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19916d118da775726c79ddb55a6b4a632d00b203
asterisk/channels/chan_sip.c
Joshua Colp 19916d118d Merged revisions 192633 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r192633 | file | 2009-05-06 10:30:51 -0300 (Wed, 06 May 2009) | 7 lines
  
  Update some old logic to stop both begin and end DTMF frames from reaching the core if rfc2833 is not enabled.
  
  (closes issue #15036)
  Reported by: dimas
  Patches:
        v1-15036.patch uploaded by dimas (license 88)
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@192634 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-06 13:34:35 +00:00

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/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 1999 - 2006, Digium, Inc.
*
* Mark Spencer <markster@digium.com>
*
* See http://www.asterisk.org for more information about
* the Asterisk project. Please do not directly contact
* any of the maintainers of this project for assistance;
* the project provides a web site, mailing lists and IRC
* channels for your use.
*
* This program is free software, distributed under the terms of
* the GNU General Public License Version 2. See the LICENSE file
* at the top of the source tree.
*/
/*!
* \file
* \brief Implementation of Session Initiation Protocol
*
* \author Mark Spencer <markster@digium.com>
*
* See Also:
* \arg \ref AstCREDITS
*
* Implementation of RFC 3261 - without S/MIME, and experimental TCP and TLS support
* Configuration file \link Config_sip sip.conf \endlink
*
* ********** IMPORTANT *
* \note TCP/TLS support is EXPERIMENTAL and WILL CHANGE. This applies to configuration
* settings, dialplan commands and dialplans apps/functions
* See \ref sip_tcp_tls
*
*
* ******** General TODO:s
* \todo Better support of forking
* \todo VIA branch tag transaction checking
* \todo Transaction support
* \todo Asterisk should send a non-100 provisional response every minute to keep proxies
* from cancelling the transaction (RFC 3261 13.3.1.1). See bug #11157.
*
* ******** Wishlist: Improvements
* - Support of SIP domains for devices, so that we match on username@domain in the From: header
* - Connect registrations with a specific device on the incoming call. It's not done
* automatically in Asterisk
*
* \ingroup channel_drivers
*
* \par Overview of the handling of SIP sessions
* The SIP channel handles several types of SIP sessions, or dialogs,
* not all of them being "telephone calls".
* - Incoming calls that will be sent to the PBX core
* - Outgoing calls, generated by the PBX
* - SIP subscriptions and notifications of states and voicemail messages
* - SIP registrations, both inbound and outbound
* - SIP peer management (peerpoke, OPTIONS)
* - SIP text messages
*
* In the SIP channel, there's a list of active SIP dialogs, which includes
* all of these when they are active. "sip show channels" in the CLI will
* show most of these, excluding subscriptions which are shown by
* "sip show subscriptions"
*
* \par incoming packets
* Incoming packets are received in the monitoring thread, then handled by
* sipsock_read() for udp only. In tcp, packets are read by the tcp_helper thread.
* sipsock_read() function parses the packet and matches an existing
* dialog or starts a new SIP dialog.
*
* sipsock_read sends the packet to handle_incoming(), that parses a bit more.
* If it is a response to an outbound request, the packet is sent to handle_response().
* If it is a request, handle_incoming() sends it to one of a list of functions
* depending on the request type - INVITE, OPTIONS, REFER, BYE, CANCEL etc
* sipsock_read locks the ast_channel if it exists (an active call) and
* unlocks it after we have processed the SIP message.
*
* A new INVITE is sent to handle_request_invite(), that will end up
* starting a new channel in the PBX, the new channel after that executing
* in a separate channel thread. This is an incoming "call".
* When the call is answered, either by a bridged channel or the PBX itself
* the sip_answer() function is called.
*
* The actual media - Video or Audio - is mostly handled by the RTP subsystem
* in rtp.c
*
* \par Outbound calls
* Outbound calls are set up by the PBX through the sip_request_call()
* function. After that, they are activated by sip_call().
*
* \par Hanging up
* The PBX issues a hangup on both incoming and outgoing calls through
* the sip_hangup() function
*/
/*!
* \page sip_tcp_tls SIP TCP and TLS support
*
* \par tcpfixes TCP implementation changes needed
* \todo Fix TCP/TLS handling in dialplan, SRV records, transfers and much more
* \todo Save TCP/TLS sessions in registry
* If someone registers a SIPS uri, this forces us to set up a TLS connection back.
* \todo Add TCP/TLS information to function SIPPEER and SIPCHANINFO
* \todo If tcpenable=yes, we must open a TCP socket on the same address as the IP for UDP.
* The tcpbindaddr config option should only be used to open ADDITIONAL ports
* So we should propably go back to
* bindaddr= the default address to bind to. If tcpenable=yes, then bind this to both udp and TCP
* if tlsenable=yes, open TLS port (provided we also have cert)
* tcpbindaddr = extra address for additional TCP connections
* tlsbindaddr = extra address for additional TCP/TLS connections
* udpbindaddr = extra address for additional UDP connections
* These three options should take multiple IP/port pairs
* Note: Since opening additional listen sockets is a *new* feature we do not have today
* the XXXbindaddr options needs to be disabled until we have support for it
*
* \todo re-evaluate the transport= setting in sip.conf. This is right now not well
* thought of. If a device in sip.conf contacts us via TCP, we should not switch transport,
* even if udp is the configured first transport.
*
* \todo Be prepared for one outbound and another incoming socket per pvt. This applies
* specially to communication with other peers (proxies).
* \todo We need to test TCP sessions with SIP proxies and in regards
* to the SIP outbound specs.
* \todo ;transport=tls was deprecated in RFC3261 and should not be used at all. See section 26.2.2.
*
* \todo If the message is smaller than the given Content-length, the request should get a 400 Bad request
* message. If it's a response, it should be dropped. (RFC 3261, Section 18.3)
* \todo Since we have had multidomain support in Asterisk for quite a while, we need to support
* multiple domains in our TLS implementation, meaning one socket and one cert per domain
* \todo Selection of transport for a request needs to be done after we've parsed all route headers,
* also considering outbound proxy options.
* First request: Outboundproxy, routes, (reg contact or URI. If URI doesn't have port: DNS naptr, srv, AAA)
* Intermediate requests: Outboundproxy(only when forced), routes, contact/uri
* DNS naptr support is crucial. A SIP uri might lead to a TLS connection.
* Also note that due to outbound proxy settings, a SIPS uri might have to be sent on UDP (not to recommend though)
* \todo Default transports are set to UDP, which cause the wrong behaviour when contacting remote
* devices directly from the dialplan. UDP is only a fallback if no other method works,
* in order to be compatible with RFC2543 (SIP/1.0) devices. For transactions that exceed the
* MTU (like INIVTE with video, audio and RTT) TCP should be preferred.
*
* When dialling unconfigured peers (with no port number) or devices in external domains
* NAPTR records MUST be consulted to find configured transport. If they are not found,
* SRV records for both TCP and UDP should be checked. If there's a record for TCP, use that.
* If there's no record for TCP, then use UDP as a last resort. If there's no SRV records,
* \note this only applies if there's no outbound proxy configured for the session. If an outbound
* proxy is configured, these procedures might apply for locating the proxy and determining
* the transport to use for communication with the proxy.
* \par Other bugs to fix ----
* __set_address_from_contact(const char *fullcontact, struct sockaddr_in *sin, int tcp)
* - sets TLS port as default for all TCP connections, unless other port is given in contact.
* parse_register_contact(struct sip_pvt *pvt, struct sip_peer *peer, struct sip_request *req)
* - assumes that the contact the UA registers is using the same transport as the REGISTER request, which is
* a bad guess.
* - Does not save any information about TCP/TLS connected devices, which is a severe BUG, as discussed on the mailing list.
* get_destination(struct sip_pvt *p, struct sip_request *oreq)
* - Doesn't store the information that we got an incoming SIPS request in the channel, so that
* we can require a secure signalling path OUT of Asterisk (on SIP or IAX2). Possibly, the call should
* fail on in-secure signalling paths if there's no override in our configuration. At least, provide a
* channel variable in the dialplan.
* get_refer_info(struct sip_pvt *transferer, struct sip_request *outgoing_req)
* - As above, if we have a SIPS: uri in the refer-to header
* - Does not check transport in refer_to uri.
*/
/*** MODULEINFO
<depend>chan_local</depend>
***/
/*! \page sip_session_timers SIP Session Timers in Asterisk Chan_sip
The SIP Session-Timers is an extension of the SIP protocol that allows end-points and proxies to
refresh a session periodically. The sessions are kept alive by sending a RE-INVITE or UPDATE
request at a negotiated interval. If a session refresh fails then all the entities that support Session-
Timers clear their internal session state. In addition, UAs generate a BYE request in order to clear
the state in the proxies and the remote UA (this is done for the benefit of SIP entities in the path
that do not support Session-Timers).
The Session-Timers can be configured on a system-wide, per-user, or per-peer basis. The peruser/
per-peer settings override the global settings. The following new parameters have been
added to the sip.conf file.
session-timers=["accept", "originate", "refuse"]
session-expires=[integer]
session-minse=[integer]
session-refresher=["uas", "uac"]
The session-timers parameter in sip.conf defines the mode of operation of SIP session-timers feature in
Asterisk. The Asterisk can be configured in one of the following three modes:
1. Accept :: In the "accept" mode, the Asterisk server honors session-timers requests
made by remote end-points. A remote end-point can request Asterisk to engage
session-timers by either sending it an INVITE request with a "Supported: timer"
header in it or by responding to Asterisk's INVITE with a 200 OK that contains
Session-Expires: header in it. In this mode, the Asterisk server does not
request session-timers from remote end-points. This is the default mode.
2. Originate :: In the "originate" mode, the Asterisk server requests the remote
end-points to activate session-timers in addition to honoring such requests
made by the remote end-pints. In order to get as much protection as possible
against hanging SIP channels due to network or end-point failures, Asterisk
resends periodic re-INVITEs even if a remote end-point does not support
the session-timers feature.
3. Refuse :: In the "refuse" mode, Asterisk acts as if it does not support session-
timers for inbound or outbound requests. If a remote end-point requests
session-timers in a dialog, then Asterisk ignores that request unless it's
noted as a requirement (Require: header), in which case the INVITE is
rejected with a 420 Bad Extension response.
*/
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include <ctype.h>
#include <sys/ioctl.h>
#include <fcntl.h>
#include <signal.h>
#include <sys/signal.h>
#include <regex.h>
#include <time.h>
#include "asterisk/network.h"
#include "asterisk/paths.h" /* need ast_config_AST_SYSTEM_NAME */
#include "asterisk/lock.h"
#include "asterisk/channel.h"
#include "asterisk/config.h"
#include "asterisk/module.h"
#include "asterisk/pbx.h"
#include "asterisk/sched.h"
#include "asterisk/io.h"
#include "asterisk/rtp_engine.h"
#include "asterisk/udptl.h"
#include "asterisk/acl.h"
#include "asterisk/manager.h"
#include "asterisk/callerid.h"
#include "asterisk/cli.h"
#include "asterisk/app.h"
#include "asterisk/musiconhold.h"
#include "asterisk/dsp.h"
#include "asterisk/features.h"
#include "asterisk/srv.h"
#include "asterisk/astdb.h"
#include "asterisk/causes.h"
#include "asterisk/utils.h"
#include "asterisk/file.h"
#include "asterisk/astobj.h"
/*
Uncomment the define below, if you are having refcount related memory leaks.
With this uncommented, this module will generate a file, /tmp/refs, which contains
a history of the ao2_ref() calls. To be useful, all calls to ao2_* functions should
be modified to ao2_t_* calls, and include a tag describing what is happening with
enough detail, to make pairing up a reference count increment with its corresponding decrement.
The refcounter program in utils/ can be invaluable in highlighting objects that are not
balanced, along with the complete history for that object.
In normal operation, the macros defined will throw away the tags, so they do not
affect the speed of the program at all. They can be considered to be documentation.
*/
/* #define REF_DEBUG 1 */
#include "asterisk/astobj2.h"
#include "asterisk/dnsmgr.h"
#include "asterisk/devicestate.h"
#include "asterisk/linkedlists.h"
#include "asterisk/stringfields.h"
#include "asterisk/monitor.h"
#include "asterisk/netsock.h"
#include "asterisk/localtime.h"
#include "asterisk/abstract_jb.h"
#include "asterisk/threadstorage.h"
#include "asterisk/translate.h"
#include "asterisk/ast_version.h"
#include "asterisk/event.h"
#include "asterisk/tcptls.h"
#include "asterisk/stun.h"
/*** DOCUMENTATION
<application name="SIPDtmfMode" language="en_US">
<synopsis>
Change the dtmfmode for a SIP call.
</synopsis>
<syntax>
<parameter name="mode" required="true">
<enumlist>
<enum name="inband" />
<enum name="info" />
<enum name="rfc2833" />
</enumlist>
</parameter>
</syntax>
<description>
<para>Changes the dtmfmode for a SIP call.</para>
</description>
</application>
<application name="SIPAddHeader" language="en_US">
<synopsis>
Add a SIP header to the outbound call.
</synopsis>
<syntax argsep=":">
<parameter name="Header" required="true" />
<parameter name="Content" required="true" />
</syntax>
<description>
<para>Adds a header to a SIP call placed with DIAL.</para>
<para>Remember to use the X-header if you are adding non-standard SIP
headers, like <literal>X-Asterisk-Accountcode:</literal>. Use this with care.
Adding the wrong headers may jeopardize the SIP dialog.</para>
<para>Always returns <literal>0</literal>.</para>
</description>
</application>
<application name="SIPRemoveHeader" language="en_US">
<synopsis>
Remove SIP headers previously added with SIPAddHeader
</synopsis>
<syntax>
<parameter name="Header" required="false" />
</syntax>
<description>
<para>SIPRemoveHeader() allows you to remove headers which were previously
added with SIPAddHeader(). If no parameter is supplied, all previously added
headers will be removed. If a parameter is supplied, only the matching headers
will be removed.</para>
<para>For example you have added these 2 headers:</para>
<para>SIPAddHeader(P-Asserted-Identity: sip:foo@bar);</para>
<para>SIPAddHeader(P-Preferred-Identity: sip:bar@foo);</para>
<para></para>
<para>// remove all headers</para>
<para>SIPRemoveHeader();</para>
<para>// remove all P- headers</para>
<para>SIPRemoveHeader(P-);</para>
<para>// remove only the PAI header (note the : at the end)</para>
<para>SIPRemoveHeader(P-Asserted-Identity:);</para>
<para></para>
<para>Always returns <literal>0</literal>.</para>
</description>
</application>
<function name="SIP_HEADER" language="en_US">
<synopsis>
Gets the specified SIP header.
</synopsis>
<syntax>
<parameter name="name" required="true" />
<parameter name="number">
<para>If not specified, defaults to <literal>1</literal>.</para>
</parameter>
</syntax>
<description>
<para>Since there are several headers (such as Via) which can occur multiple
times, SIP_HEADER takes an optional second argument to specify which header with
that name to retrieve. Headers start at offset <literal>1</literal>.</para>
</description>
</function>
<function name="SIPPEER" language="en_US">
<synopsis>
Gets SIP peer information.
</synopsis>
<syntax>
<parameter name="peername" required="true" />
<parameter name="item">
<enumlist>
<enum name="ip">
<para>(default) The ip address.</para>
</enum>
<enum name="port">
<para>The port number.</para>
</enum>
<enum name="mailbox">
<para>The configured mailbox.</para>
</enum>
<enum name="context">
<para>The configured context.</para>
</enum>
<enum name="expire">
<para>The epoch time of the next expire.</para>
</enum>
<enum name="dynamic">
<para>Is it dynamic? (yes/no).</para>
</enum>
<enum name="callerid_name">
<para>The configured Caller ID name.</para>
</enum>
<enum name="callerid_num">
<para>The configured Caller ID number.</para>
</enum>
<enum name="callgroup">
<para>The configured Callgroup.</para>
</enum>
<enum name="pickupgroup">
<para>The configured Pickupgroup.</para>
</enum>
<enum name="codecs">
<para>The configured codecs.</para>
</enum>
<enum name="status">
<para>Status (if qualify=yes).</para>
</enum>
<enum name="regexten">
<para>Registration extension.</para>
</enum>
<enum name="limit">
<para>Call limit (call-limit).</para>
</enum>
<enum name="busylevel">
<para>Configured call level for signalling busy.</para>
</enum>
<enum name="curcalls">
<para>Current amount of calls. Only available if call-limit is set.</para>
</enum>
<enum name="language">
<para>Default language for peer.</para>
</enum>
<enum name="accountcode">
<para>Account code for this peer.</para>
</enum>
<enum name="useragent">
<para>Current user agent id for peer.</para>
</enum>
<enum name="chanvar[name]">
<para>A channel variable configured with setvar for this peer.</para>
</enum>
<enum name="codec[x]">
<para>Preferred codec index number <replaceable>x</replaceable> (beginning with zero).</para>
</enum>
</enumlist>
</parameter>
</syntax>
<description />
</function>
<function name="SIPCHANINFO" language="en_US">
<synopsis>
Gets the specified SIP parameter from the current channel.
</synopsis>
<syntax>
<parameter name="item" required="true">
<enumlist>
<enum name="peerip">
<para>The IP address of the peer.</para>
</enum>
<enum name="recvip">
<para>The source IP address of the peer.</para>
</enum>
<enum name="from">
<para>The URI from the <literal>From:</literal> header.</para>
</enum>
<enum name="uri">
<para>The URI from the <literal>Contact:</literal> header.</para>
</enum>
<enum name="useragent">
<para>The useragent.</para>
</enum>
<enum name="peername">
<para>The name of the peer.</para>
</enum>
<enum name="t38passthrough">
<para><literal>1</literal> if T38 is offered or enabled in this channel,
otherwise <literal>0</literal>.</para>
</enum>
</enumlist>
</parameter>
</syntax>
<description />
</function>
<function name="CHECKSIPDOMAIN" language="en_US">
<synopsis>
Checks if domain is a local domain.
</synopsis>
<syntax>
<parameter name="domain" required="true" />
</syntax>
<description>
<para>This function checks if the <replaceable>domain</replaceable> in the argument is configured
as a local SIP domain that this Asterisk server is configured to handle.
Returns the domain name if it is locally handled, otherwise an empty string.
Check the <literal>domain=</literal> configuration in <filename>sip.conf</filename>.</para>
</description>
</function>
***/
#ifndef FALSE
#define FALSE 0
#endif
#ifndef TRUE
#define TRUE 1
#endif
#ifndef MAX
#define MAX(a,b) ((a) > (b) ? (a) : (b))
#endif
/* Arguments for find_peer */
#define FINDUSERS (1 << 0)
#define FINDPEERS (1 << 1)
#define FINDALLDEVICES (FINDUSERS | FINDPEERS)
#define SIPBUFSIZE 512 /*!< Buffer size for many operations */
#define XMIT_ERROR -2
#define SIP_RESERVED ";/?:@&=+$,# " /*!< Reserved characters in the username part of the URI */
/* #define VOCAL_DATA_HACK */
#define DEFAULT_DEFAULT_EXPIRY 120
#define DEFAULT_MIN_EXPIRY 60
#define DEFAULT_MAX_EXPIRY 3600
#define DEFAULT_MWI_EXPIRY 3600
#define DEFAULT_REGISTRATION_TIMEOUT 20
#define DEFAULT_MAX_FORWARDS "70"
/* guard limit must be larger than guard secs */
/* guard min must be < 1000, and should be >= 250 */
#define EXPIRY_GUARD_SECS 15 /*!< How long before expiry do we reregister */
#define EXPIRY_GUARD_LIMIT 30 /*!< Below here, we use EXPIRY_GUARD_PCT instead of
EXPIRY_GUARD_SECS */
#define EXPIRY_GUARD_MIN 500 /*!< This is the minimum guard time applied. If
GUARD_PCT turns out to be lower than this, it
will use this time instead.
This is in milliseconds. */
#define EXPIRY_GUARD_PCT 0.20 /*!< Percentage of expires timeout to use when
below EXPIRY_GUARD_LIMIT */
#define DEFAULT_EXPIRY 900 /*!< Expire slowly */
static int min_expiry = DEFAULT_MIN_EXPIRY; /*!< Minimum accepted registration time */
static int max_expiry = DEFAULT_MAX_EXPIRY; /*!< Maximum accepted registration time */
static int default_expiry = DEFAULT_DEFAULT_EXPIRY;
static int mwi_expiry = DEFAULT_MWI_EXPIRY;
#define DEFAULT_QUALIFY_GAP 100
#define DEFAULT_QUALIFY_PEERS 1
#define CALLERID_UNKNOWN "Unknown"
#define DEFAULT_MAXMS 2000 /*!< Qualification: Must be faster than 2 seconds by default */
#define DEFAULT_QUALIFYFREQ 60 * 1000 /*!< Qualification: How often to check for the host to be up */
#define DEFAULT_FREQ_NOTOK 10 * 1000 /*!< Qualification: How often to check, if the host is down... */
#define DEFAULT_RETRANS 1000 /*!< How frequently to retransmit Default: 2 * 500 ms in RFC 3261 */
#define MAX_RETRANS 6 /*!< Try only 6 times for retransmissions, a total of 7 transmissions */
#define DEFAULT_TIMER_T1 500 /*!< SIP timer T1 (according to RFC 3261) */
#define SIP_TRANS_TIMEOUT 64 * DEFAULT_TIMER_T1 /*!< SIP request timeout (rfc 3261) 64*T1
\todo Use known T1 for timeout (peerpoke)
*/
#define DEFAULT_TRANS_TIMEOUT -1 /*!< Use default SIP transaction timeout */
#define MAX_AUTHTRIES 3 /*!< Try authentication three times, then fail */
#define SIP_MAX_HEADERS 64 /*!< Max amount of SIP headers to read */
#define SIP_MAX_LINES 64 /*!< Max amount of lines in SIP attachment (like SDP) */
#define SIP_MIN_PACKET 4096 /*!< Initialize size of memory to allocate for packets */
#define MAX_HISTORY_ENTRIES 50 /*!< Max entires in the history list for a sip_pvt */
#define INITIAL_CSEQ 101 /*!< Our initial sip sequence number */
#define DEFAULT_MAX_SE 1800 /*!< Session-Timer Default Session-Expires period (RFC 4028) */
#define DEFAULT_MIN_SE 90 /*!< Session-Timer Default Min-SE period (RFC 4028) */
#define SDP_MAX_RTPMAP_CODECS 32 /*!< Maximum number of codecs allowed in received SDP */
/*! \brief Global jitterbuffer configuration - by default, jb is disabled */
static struct ast_jb_conf default_jbconf =
{
.flags = 0,
.max_size = -1,
.resync_threshold = -1,
.impl = ""
};
static struct ast_jb_conf global_jbconf; /*!< Global jitterbuffer configuration */
static const char config[] = "sip.conf"; /*!< Main configuration file */
static const char notify_config[] = "sip_notify.conf"; /*!< Configuration file for sending Notify with CLI commands to reconfigure or reboot phones */
#define RTP 1
#define NO_RTP 0
/*! \brief Authorization scheme for call transfers
\note Not a bitfield flag, since there are plans for other modes,
like "only allow transfers for authenticated devices" */
enum transfermodes {
TRANSFER_OPENFORALL, /*!< Allow all SIP transfers */
TRANSFER_CLOSED, /*!< Allow no SIP transfers */
};
/*! \brief The result of a lot of functions */
enum sip_result {
AST_SUCCESS = 0, /*!< FALSE means success, funny enough */
AST_FAILURE = -1, /*!< Failure code */
};
/*! \brief States for the INVITE transaction, not the dialog
\note this is for the INVITE that sets up the dialog
*/
enum invitestates {
INV_NONE = 0, /*!< No state at all, maybe not an INVITE dialog */
INV_CALLING = 1, /*!< Invite sent, no answer */
INV_PROCEEDING = 2, /*!< We got/sent 1xx message */
INV_EARLY_MEDIA = 3, /*!< We got 18x message with to-tag back */
INV_COMPLETED = 4, /*!< Got final response with error. Wait for ACK, then CONFIRMED */
INV_CONFIRMED = 5, /*!< Confirmed response - we've got an ack (Incoming calls only) */
INV_TERMINATED = 6, /*!< Transaction done - either successful (AST_STATE_UP) or failed, but done
The only way out of this is a BYE from one side */
INV_CANCELLED = 7, /*!< Transaction cancelled by client or server in non-terminated state */
};
/*! \brief Readable descriptions of device states.
\note Should be aligned to above table as index */
static const struct invstate2stringtable {
const enum invitestates state;
const char *desc;
} invitestate2string[] = {
{INV_NONE, "None" },
{INV_CALLING, "Calling (Trying)"},
{INV_PROCEEDING, "Proceeding "},
{INV_EARLY_MEDIA, "Early media"},
{INV_COMPLETED, "Completed (done)"},
{INV_CONFIRMED, "Confirmed (up)"},
{INV_TERMINATED, "Done"},
{INV_CANCELLED, "Cancelled"}
};
/*! \brief When sending a SIP message, we can send with a few options, depending on
type of SIP request. UNRELIABLE is moslty used for responses to repeated requests,
where the original response would be sent RELIABLE in an INVITE transaction */
enum xmittype {
XMIT_CRITICAL = 2, /*!< Transmit critical SIP message reliably, with re-transmits.
If it fails, it's critical and will cause a teardown of the session */
XMIT_RELIABLE = 1, /*!< Transmit SIP message reliably, with re-transmits */
XMIT_UNRELIABLE = 0, /*!< Transmit SIP message without bothering with re-transmits */
};
/*! \brief Results from the parse_register() function */
enum parse_register_result {
PARSE_REGISTER_FAILED,
PARSE_REGISTER_UPDATE,
PARSE_REGISTER_QUERY,
};
/*! \brief Type of subscription, based on the packages we do support, see \ref subscription_types */
enum subscriptiontype {
NONE = 0,
XPIDF_XML,
DIALOG_INFO_XML,
CPIM_PIDF_XML,
PIDF_XML,
MWI_NOTIFICATION
};
/*! \brief Subscription types that we support. We support
- dialoginfo updates (really device status, not dialog info as was the original intent of the standard)
- SIMPLE presence used for device status
- Voicemail notification subscriptions
*/
static const struct cfsubscription_types {
enum subscriptiontype type;
const char * const event;
const char * const mediatype;
const char * const text;
} subscription_types[] = {
{ NONE, "-", "unknown", "unknown" },
/* RFC 4235: SIP Dialog event package */
{ DIALOG_INFO_XML, "dialog", "application/dialog-info+xml", "dialog-info+xml" },
{ CPIM_PIDF_XML, "presence", "application/cpim-pidf+xml", "cpim-pidf+xml" }, /* RFC 3863 */
{ PIDF_XML, "presence", "application/pidf+xml", "pidf+xml" }, /* RFC 3863 */
{ XPIDF_XML, "presence", "application/xpidf+xml", "xpidf+xml" }, /* Pre-RFC 3863 with MS additions */
{ MWI_NOTIFICATION, "message-summary", "application/simple-message-summary", "mwi" } /* RFC 3842: Mailbox notification */
};
/*! \brief Authentication types - proxy or www authentication
\note Endpoints, like Asterisk, should always use WWW authentication to
allow multiple authentications in the same call - to the proxy and
to the end point.
*/
enum sip_auth_type {
PROXY_AUTH = 407,
WWW_AUTH = 401,
};
/*! \brief Authentication result from check_auth* functions */
enum check_auth_result {
AUTH_DONT_KNOW = -100, /*!< no result, need to check further */
/* XXX maybe this is the same as AUTH_NOT_FOUND */
AUTH_SUCCESSFUL = 0,
AUTH_CHALLENGE_SENT = 1,
AUTH_SECRET_FAILED = -1,
AUTH_USERNAME_MISMATCH = -2,
AUTH_NOT_FOUND = -3, /*!< returned by register_verify */
AUTH_FAKE_AUTH = -4,
AUTH_UNKNOWN_DOMAIN = -5,
AUTH_PEER_NOT_DYNAMIC = -6,
AUTH_ACL_FAILED = -7,
AUTH_BAD_TRANSPORT = -8,
AUTH_RTP_FAILED = 9,
};
/*! \brief States for outbound registrations (with register= lines in sip.conf */
enum sipregistrystate {
REG_STATE_UNREGISTERED = 0, /*!< We are not registered
* \note Initial state. We should have a timeout scheduled for the initial
* (or next) registration transmission, calling sip_reregister
*/
REG_STATE_REGSENT, /*!< Registration request sent
* \note sent initial request, waiting for an ack or a timeout to
* retransmit the initial request.
*/
REG_STATE_AUTHSENT, /*!< We have tried to authenticate
* \note entered after transmit_register with auth info,
* waiting for an ack.
*/
REG_STATE_REGISTERED, /*!< Registered and done */
REG_STATE_REJECTED, /*!< Registration rejected *
* \note only used when the remote party has an expire larger than
* our max-expire. This is a final state from which we do not
* recover (not sure how correctly).
*/
REG_STATE_TIMEOUT, /*!< Registration timed out *
* \note XXX unused */
REG_STATE_NOAUTH, /*!< We have no accepted credentials
* \note fatal - no chance to proceed */
REG_STATE_FAILED, /*!< Registration failed after several tries
* \note fatal - no chance to proceed */
};
/*! \brief Modes in which Asterisk can be configured to run SIP Session-Timers */
enum st_mode {
SESSION_TIMER_MODE_INVALID = 0, /*!< Invalid value */
SESSION_TIMER_MODE_ACCEPT, /*!< Honor inbound Session-Timer requests */
SESSION_TIMER_MODE_ORIGINATE, /*!< Originate outbound and honor inbound requests */
SESSION_TIMER_MODE_REFUSE /*!< Ignore inbound Session-Timers requests */
};
/*! \brief The entity playing the refresher role for Session-Timers */
enum st_refresher {
SESSION_TIMER_REFRESHER_AUTO, /*!< Negotiated */
SESSION_TIMER_REFRESHER_UAC, /*!< Session is refreshed by the UAC */
SESSION_TIMER_REFRESHER_UAS /*!< Session is refreshed by the UAS */
};
/*! \brief Define some implemented SIP transports
\note Asterisk does not support SCTP or UDP/DTLS
*/
enum sip_transport {
SIP_TRANSPORT_UDP = 1, /*!< Unreliable transport for SIP, needs retransmissions */
SIP_TRANSPORT_TCP = 1 << 1, /*!< Reliable, but unsecure */
SIP_TRANSPORT_TLS = 1 << 2, /*!< TCP/TLS - reliable and secure transport for signalling */
};
/*! \brief definition of a sip proxy server
*
* For outbound proxies, a sip_peer will contain a reference to a
* dynamically allocated instance of a sip_proxy. A sip_pvt may also
* contain a reference to a peer's outboundproxy, or it may contain
* a reference to the sip_cfg.outboundproxy.
*/
struct sip_proxy {
char name[MAXHOSTNAMELEN]; /*!< DNS name of domain/host or IP */
struct sockaddr_in ip; /*!< Currently used IP address and port */
time_t last_dnsupdate; /*!< When this was resolved */
enum sip_transport transport;
int force; /*!< If it's an outbound proxy, Force use of this outbound proxy for all outbound requests */
/* Room for a SRV record chain based on the name */
};
/*! \brief argument for the 'show channels|subscriptions' callback. */
struct __show_chan_arg {
int fd;
int subscriptions;
int numchans; /* return value */
};
/*! \brief States whether a SIP message can create a dialog in Asterisk. */
enum can_create_dialog {
CAN_NOT_CREATE_DIALOG,
CAN_CREATE_DIALOG,
CAN_CREATE_DIALOG_UNSUPPORTED_METHOD,
};
/*! \brief SIP Request methods known by Asterisk
\note Do _NOT_ make any changes to this enum, or the array following it;
if you think you are doing the right thing, you are probably
not doing the right thing. If you think there are changes
needed, get someone else to review them first _before_
submitting a patch. If these two lists do not match properly
bad things will happen.
*/
enum sipmethod {
SIP_UNKNOWN, /*!< Unknown response */
SIP_RESPONSE, /*!< Not request, response to outbound request */
SIP_REGISTER, /*!< Registration to the mothership, tell us where you are located */
SIP_OPTIONS, /*!< Check capabilities of a device, used for "ping" too */
SIP_NOTIFY, /*!< Status update, Part of the event package standard, result of a SUBSCRIBE or a REFER */
SIP_INVITE, /*!< Set up a session */
SIP_ACK, /*!< End of a three-way handshake started with INVITE. */
SIP_PRACK, /*!< Reliable pre-call signalling. Not supported in Asterisk. */
SIP_BYE, /*!< End of a session */
SIP_REFER, /*!< Refer to another URI (transfer) */
SIP_SUBSCRIBE, /*!< Subscribe for updates (voicemail, session status, device status, presence) */
SIP_MESSAGE, /*!< Text messaging */
SIP_UPDATE, /*!< Update a dialog. We can send UPDATE; but not accept it */
SIP_INFO, /*!< Information updates during a session */
SIP_CANCEL, /*!< Cancel an INVITE */
SIP_PUBLISH, /*!< Not supported in Asterisk */
SIP_PING, /*!< Not supported at all, no standard but still implemented out there */
};
/*! \brief Settings for the 'notifycid' option, see sip.conf.sample for details. */
enum notifycid_setting {
DISABLED = 0,
ENABLED = 1,
IGNORE_CONTEXT = 2,
};
/*! \brief The core structure to setup dialogs. We parse incoming messages by using
structure and then route the messages according to the type.
\note Note that sip_methods[i].id == i must hold or the code breaks */
static const struct cfsip_methods {
enum sipmethod id;
int need_rtp; /*!< when this is the 'primary' use for a pvt structure, does it need RTP? */
char * const text;
enum can_create_dialog can_create;
} sip_methods[] = {
{ SIP_UNKNOWN, RTP, "-UNKNOWN-", CAN_CREATE_DIALOG },
{ SIP_RESPONSE, NO_RTP, "SIP/2.0", CAN_NOT_CREATE_DIALOG },
{ SIP_REGISTER, NO_RTP, "REGISTER", CAN_CREATE_DIALOG },
{ SIP_OPTIONS, NO_RTP, "OPTIONS", CAN_CREATE_DIALOG },
{ SIP_NOTIFY, NO_RTP, "NOTIFY", CAN_CREATE_DIALOG },
{ SIP_INVITE, RTP, "INVITE", CAN_CREATE_DIALOG },
{ SIP_ACK, NO_RTP, "ACK", CAN_NOT_CREATE_DIALOG },
{ SIP_PRACK, NO_RTP, "PRACK", CAN_NOT_CREATE_DIALOG },
{ SIP_BYE, NO_RTP, "BYE", CAN_NOT_CREATE_DIALOG },
{ SIP_REFER, NO_RTP, "REFER", CAN_CREATE_DIALOG },
{ SIP_SUBSCRIBE, NO_RTP, "SUBSCRIBE", CAN_CREATE_DIALOG },
{ SIP_MESSAGE, NO_RTP, "MESSAGE", CAN_CREATE_DIALOG },
{ SIP_UPDATE, NO_RTP, "UPDATE", CAN_NOT_CREATE_DIALOG },
{ SIP_INFO, NO_RTP, "INFO", CAN_NOT_CREATE_DIALOG },
{ SIP_CANCEL, NO_RTP, "CANCEL", CAN_NOT_CREATE_DIALOG },
{ SIP_PUBLISH, NO_RTP, "PUBLISH", CAN_CREATE_DIALOG_UNSUPPORTED_METHOD },
{ SIP_PING, NO_RTP, "PING", CAN_CREATE_DIALOG_UNSUPPORTED_METHOD }
};
/*! Define SIP option tags, used in Require: and Supported: headers
We need to be aware of these properties in the phones to use
the replace: header. We should not do that without knowing
that the other end supports it...
This is nothing we can configure, we learn by the dialog
Supported: header on the REGISTER (peer) or the INVITE
(other devices)
We are not using many of these today, but will in the future.
This is documented in RFC 3261
*/
#define SUPPORTED 1
#define NOT_SUPPORTED 0
/* SIP options */
#define SIP_OPT_REPLACES (1 << 0)
#define SIP_OPT_100REL (1 << 1)
#define SIP_OPT_TIMER (1 << 2)
#define SIP_OPT_EARLY_SESSION (1 << 3)
#define SIP_OPT_JOIN (1 << 4)
#define SIP_OPT_PATH (1 << 5)
#define SIP_OPT_PREF (1 << 6)
#define SIP_OPT_PRECONDITION (1 << 7)
#define SIP_OPT_PRIVACY (1 << 8)
#define SIP_OPT_SDP_ANAT (1 << 9)
#define SIP_OPT_SEC_AGREE (1 << 10)
#define SIP_OPT_EVENTLIST (1 << 11)
#define SIP_OPT_GRUU (1 << 12)
#define SIP_OPT_TARGET_DIALOG (1 << 13)
#define SIP_OPT_NOREFERSUB (1 << 14)
#define SIP_OPT_HISTINFO (1 << 15)
#define SIP_OPT_RESPRIORITY (1 << 16)
#define SIP_OPT_FROMCHANGE (1 << 17)
#define SIP_OPT_RECLISTINV (1 << 18)
#define SIP_OPT_RECLISTSUB (1 << 19)
#define SIP_OPT_OUTBOUND (1 << 20)
#define SIP_OPT_UNKNOWN (1 << 21)
/*! \brief List of well-known SIP options. If we get this in a require,
we should check the list and answer accordingly. */
static const struct cfsip_options {
int id; /*!< Bitmap ID */
int supported; /*!< Supported by Asterisk ? */
char * const text; /*!< Text id, as in standard */
} sip_options[] = { /* XXX used in 3 places */
/* RFC3262: PRACK 100% reliability */
{ SIP_OPT_100REL, NOT_SUPPORTED, "100rel" },
/* RFC3959: SIP Early session support */
{ SIP_OPT_EARLY_SESSION, NOT_SUPPORTED, "early-session" },
/* SIMPLE events: RFC4662 */
{ SIP_OPT_EVENTLIST, NOT_SUPPORTED, "eventlist" },
/* RFC 4916- Connected line ID updates */
{ SIP_OPT_FROMCHANGE, NOT_SUPPORTED, "from-change" },
/* GRUU: Globally Routable User Agent URI's */
{ SIP_OPT_GRUU, NOT_SUPPORTED, "gruu" },
/* RFC4244 History info */
{ SIP_OPT_HISTINFO, NOT_SUPPORTED, "histinfo" },
/* RFC3911: SIP Join header support */
{ SIP_OPT_JOIN, NOT_SUPPORTED, "join" },
/* Disable the REFER subscription, RFC 4488 */
{ SIP_OPT_NOREFERSUB, NOT_SUPPORTED, "norefersub" },
/* SIP outbound - the final NAT battle - draft-sip-outbound */
{ SIP_OPT_OUTBOUND, NOT_SUPPORTED, "outbound" },
/* RFC3327: Path support */
{ SIP_OPT_PATH, NOT_SUPPORTED, "path" },
/* RFC3840: Callee preferences */
{ SIP_OPT_PREF, NOT_SUPPORTED, "pref" },
/* RFC3312: Precondition support */
{ SIP_OPT_PRECONDITION, NOT_SUPPORTED, "precondition" },
/* RFC3323: Privacy with proxies*/
{ SIP_OPT_PRIVACY, NOT_SUPPORTED, "privacy" },
/* RFC-ietf-sip-uri-list-conferencing-02.txt conference invite lists */
{ SIP_OPT_RECLISTINV, NOT_SUPPORTED, "recipient-list-invite" },
/* RFC-ietf-sip-uri-list-subscribe-02.txt - subscription lists */
{ SIP_OPT_RECLISTSUB, NOT_SUPPORTED, "recipient-list-subscribe" },
/* RFC3891: Replaces: header for transfer */
{ SIP_OPT_REPLACES, SUPPORTED, "replaces" },
/* One version of Polycom firmware has the wrong label */
{ SIP_OPT_REPLACES, SUPPORTED, "replace" },
/* RFC4412 Resource priorities */
{ SIP_OPT_RESPRIORITY, NOT_SUPPORTED, "resource-priority" },
/* RFC3329: Security agreement mechanism */
{ SIP_OPT_SEC_AGREE, NOT_SUPPORTED, "sec_agree" },
/* RFC4092: Usage of the SDP ANAT Semantics in the SIP */
{ SIP_OPT_SDP_ANAT, NOT_SUPPORTED, "sdp-anat" },
/* RFC4028: SIP Session-Timers */
{ SIP_OPT_TIMER, SUPPORTED, "timer" },
/* RFC4538: Target-dialog */
{ SIP_OPT_TARGET_DIALOG,NOT_SUPPORTED, "tdialog" },
};
/*! \brief Diversion header reasons
*
* The core defines a bunch of constants used to define
* redirecting reasons. This provides a translation table
* between those and the strings which may be present in
* a SIP Diversion header
*/
static const struct sip_reasons {
enum AST_REDIRECTING_REASON code;
char * const text;
} sip_reason_table[] = {
{ AST_REDIRECTING_REASON_UNKNOWN, "unknown" },
{ AST_REDIRECTING_REASON_USER_BUSY, "user-busy" },
{ AST_REDIRECTING_REASON_NO_ANSWER, "no-answer" },
{ AST_REDIRECTING_REASON_UNAVAILABLE, "unavailable" },
{ AST_REDIRECTING_REASON_UNCONDITIONAL, "unconditional" },
{ AST_REDIRECTING_REASON_TIME_OF_DAY, "time-of-day" },
{ AST_REDIRECTING_REASON_DO_NOT_DISTURB, "do-not-disturb" },
{ AST_REDIRECTING_REASON_DEFLECTION, "deflection" },
{ AST_REDIRECTING_REASON_FOLLOW_ME, "follow-me" },
{ AST_REDIRECTING_REASON_OUT_OF_ORDER, "out-of-service" },
{ AST_REDIRECTING_REASON_AWAY, "away" },
{ AST_REDIRECTING_REASON_CALL_FWD_DTE, "unknown"}
};
static enum AST_REDIRECTING_REASON sip_reason_str_to_code(const char *text)
{
enum AST_REDIRECTING_REASON ast = AST_REDIRECTING_REASON_UNKNOWN;
int i;
for (i = 0; i < ARRAY_LEN(sip_reason_table); ++i) {
if (!strcasecmp(text, sip_reason_table[i].text)) {
ast = sip_reason_table[i].code;
break;
}
}
return ast;
}
static const char *sip_reason_code_to_str(enum AST_REDIRECTING_REASON code)
{
if (code >= 0 && code < ARRAY_LEN(sip_reason_table)) {
return sip_reason_table[code].text;
}
return "unknown";
}
/*! \brief SIP Methods we support
\todo This string should be set dynamically. We only support REFER and SUBSCRIBE if we have
allowsubscribe and allowrefer on in sip.conf.
*/
#define ALLOWED_METHODS "INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY"
/*! \brief SIP Extensions we support
\note This should be generated based on the previous array
in combination with settings.
\todo We should not have "timer" if it's disabled in the configuration file.
*/
#define SUPPORTED_EXTENSIONS "replaces, timer"
/*! \brief Standard SIP unsecure port for UDP and TCP from RFC 3261. DO NOT CHANGE THIS */
#define STANDARD_SIP_PORT 5060
/*! \brief Standard SIP TLS port from RFC 3261. DO NOT CHANGE THIS */
#define STANDARD_TLS_PORT 5061
/*! \note in many SIP headers, absence of a port number implies port 5060,
* and this is why we cannot change the above constant.
* There is a limited number of places in asterisk where we could,
* in principle, use a different "default" port number, but
* we do not support this feature at the moment.
* You can run Asterisk with SIP on a different port with a configuration
* option. If you change this value, the signalling will be incorrect.
*/
/*! \name DefaultValues Default values, set and reset in reload_config before reading configuration
These are default values in the source. There are other recommended values in the
sip.conf.sample for new installations. These may differ to keep backwards compatibility,
yet encouraging new behaviour on new installations
*/
/*@{*/
#define DEFAULT_CONTEXT "default" /*!< The default context for [general] section as well as devices */
#define DEFAULT_MOHINTERPRET "default" /*!< The default music class */
#define DEFAULT_MOHSUGGEST ""
#define DEFAULT_VMEXTEN "asterisk" /*!< Default voicemail extension */
#define DEFAULT_CALLERID "asterisk" /*!< Default caller ID */
#define DEFAULT_MWI_FROM ""
#define DEFAULT_NOTIFYMIME "application/simple-message-summary"
#define DEFAULT_ALLOWGUEST TRUE
#define DEFAULT_RTPKEEPALIVE 0 /*!< Default RTPkeepalive setting */
#define DEFAULT_CALLCOUNTER FALSE
#define DEFAULT_SRVLOOKUP TRUE /*!< Recommended setting is ON */
#define DEFAULT_COMPACTHEADERS FALSE /*!< Send compact (one-character) SIP headers. Default off */
#define DEFAULT_TOS_SIP 0 /*!< Call signalling packets should be marked as DSCP CS3, but the default is 0 to be compatible with previous versions. */
#define DEFAULT_TOS_AUDIO 0 /*!< Audio packets should be marked as DSCP EF (Expedited Forwarding), but the default is 0 to be compatible with previous versions. */
#define DEFAULT_TOS_VIDEO 0 /*!< Video packets should be marked as DSCP AF41, but the default is 0 to be compatible with previous versions. */
#define DEFAULT_TOS_TEXT 0 /*!< Text packets should be marked as XXXX XXXX, but the default is 0 to be compatible with previous versions. */
#define DEFAULT_COS_SIP 4 /*!< Level 2 class of service for SIP signalling */
#define DEFAULT_COS_AUDIO 5 /*!< Level 2 class of service for audio media */
#define DEFAULT_COS_VIDEO 6 /*!< Level 2 class of service for video media */
#define DEFAULT_COS_TEXT 5 /*!< Level 2 class of service for text media (T.140) */
#define DEFAULT_ALLOW_EXT_DOM TRUE /*!< Allow external domains */
#define DEFAULT_REALM "asterisk" /*!< Realm for HTTP digest authentication */
#define DEFAULT_NOTIFYRINGING TRUE /*!< Notify devicestate system on ringing state */
#define DEFAULT_NOTIFYCID DISABLED /*!< Include CID with ringing notifications */
#define DEFAULT_PEDANTIC FALSE /*!< Avoid following SIP standards for dialog matching */
#define DEFAULT_AUTOCREATEPEER FALSE /*!< Don't create peers automagically */
#define DEFAULT_MATCHEXTERNIPLOCALLY FALSE /*!< Match extern IP locally default setting */
#define DEFAULT_QUALIFY FALSE /*!< Don't monitor devices */
#define DEFAULT_CALLEVENTS FALSE /*!< Extra manager SIP call events */
#define DEFAULT_ALWAYSAUTHREJECT FALSE /*!< Don't reject authentication requests always */
#define DEFAULT_REGEXTENONQUALIFY FALSE
#define DEFAULT_T1MIN 100 /*!< 100 MS for minimal roundtrip time */
#define DEFAULT_MAX_CALL_BITRATE (384) /*!< Max bitrate for video */
#ifndef DEFAULT_USERAGENT
#define DEFAULT_USERAGENT "Asterisk PBX" /*!< Default Useragent: header unless re-defined in sip.conf */
#define DEFAULT_SDPSESSION "Asterisk PBX" /*!< Default SDP session name, (s=) header unless re-defined in sip.conf */
#define DEFAULT_SDPOWNER "root" /*!< Default SDP username field in (o=) header unless re-defined in sip.conf */
#define DEFAULT_ENGINE "asterisk" /*!< Default RTP engine to use for sessions */
#endif
/*@}*/
/*! \name DefaultSettings
Default setttings are used as a channel setting and as a default when
configuring devices
*/
/*@{*/
static char default_language[MAX_LANGUAGE];
static char default_callerid[AST_MAX_EXTENSION];
static char default_mwi_from[80];
static char default_fromdomain[AST_MAX_EXTENSION];
static char default_notifymime[AST_MAX_EXTENSION];
static int default_qualify; /*!< Default Qualify= setting */
static char default_vmexten[AST_MAX_EXTENSION];
static char default_mohinterpret[MAX_MUSICCLASS]; /*!< Global setting for moh class to use when put on hold */
static char default_mohsuggest[MAX_MUSICCLASS]; /*!< Global setting for moh class to suggest when putting
* a bridged channel on hold */
static char default_parkinglot[AST_MAX_CONTEXT]; /*!< Parkinglot */
static char default_engine[256]; /*!< Default RTP engine */
static int default_maxcallbitrate; /*!< Maximum bitrate for call */
static struct ast_codec_pref default_prefs; /*!< Default codec prefs */
static unsigned int default_transports; /*!< Default Transports (enum sip_transport) that are acceptable */
static unsigned int default_primary_transport; /*!< Default primary Transport (enum sip_transport) for outbound connections to devices */
/*@}*/
/*! \name GlobalSettings
Global settings apply to the channel (often settings you can change in the general section
of sip.conf
*/
/*@{*/
/*! \brief a place to store all global settings for the sip channel driver
These are settings that will be possibly to apply on a group level later on.
\note Do not add settings that only apply to the channel itself and can't
be applied to devices (trunks, services, phones)
*/
struct sip_settings {
int peer_rtupdate; /*!< G: Update database with registration data for peer? */
int rtsave_sysname; /*!< G: Save system name at registration? */
int ignore_regexpire; /*!< G: Ignore expiration of peer */
int rtautoclear; /*!< Realtime ?? */
int directrtpsetup; /*!< Enable support for Direct RTP setup (no re-invites) */
int pedanticsipchecking; /*!< Extra checking ? Default off */
int autocreatepeer; /*!< Auto creation of peers at registration? Default off. */
int srvlookup; /*!< SRV Lookup on or off. Default is on */
int allowguest; /*!< allow unauthenticated peers to connect? */
int alwaysauthreject; /*!< Send 401 Unauthorized for all failing requests */
int compactheaders; /*!< send compact sip headers */
int allow_external_domains; /*!< Accept calls to external SIP domains? */
int callevents; /*!< Whether we send manager events or not */
int regextenonqualify; /*!< Whether to add/remove regexten when qualifying peers */
int matchexterniplocally; /*!< Match externip/externhost setting against localnet setting */
int notifyringing; /*!< Send notifications on ringing */
int notifyhold; /*!< Send notifications on hold */
enum notifycid_setting notifycid; /*!< Send CID with ringing notifications */
enum transfermodes allowtransfer; /*!< SIP Refer restriction scheme */
int allowsubscribe; /*!< Flag for disabling ALL subscriptions, this is FALSE only if all peers are FALSE
the global setting is in globals_flags[1] */
char realm[MAXHOSTNAMELEN]; /*!< Default realm */
struct sip_proxy outboundproxy; /*!< Outbound proxy */
char default_context[AST_MAX_CONTEXT];
char default_subscribecontext[AST_MAX_CONTEXT];
};
static struct sip_settings sip_cfg;
static int global_match_auth_username; /*!< Match auth username if available instead of From: Default off. */
static int global_relaxdtmf; /*!< Relax DTMF */
static int global_rtptimeout; /*!< Time out call if no RTP */
static int global_rtpholdtimeout; /*!< Time out call if no RTP during hold */
static int global_rtpkeepalive; /*!< Send RTP keepalives */
static int global_reg_timeout;
static int global_regattempts_max; /*!< Registration attempts before giving up */
static int global_callcounter; /*!< Enable call counters for all devices. This is currently enabled by setting the peer
call-limit to 999. When we remove the call-limit from the code, we can make it
with just a boolean flag in the device structure */
static unsigned int global_tos_sip; /*!< IP type of service for SIP packets */
static unsigned int global_tos_audio; /*!< IP type of service for audio RTP packets */
static unsigned int global_tos_video; /*!< IP type of service for video RTP packets */
static unsigned int global_tos_text; /*!< IP type of service for text RTP packets */
static unsigned int global_cos_sip; /*!< 802.1p class of service for SIP packets */
static unsigned int global_cos_audio; /*!< 802.1p class of service for audio RTP packets */
static unsigned int global_cos_video; /*!< 802.1p class of service for video RTP packets */
static unsigned int global_cos_text; /*!< 802.1p class of service for text RTP packets */
static int recordhistory; /*!< Record SIP history. Off by default */
static int dumphistory; /*!< Dump history to verbose before destroying SIP dialog */
static char global_regcontext[AST_MAX_CONTEXT]; /*!< Context for auto-extensions */
static char global_useragent[AST_MAX_EXTENSION]; /*!< Useragent for the SIP channel */
static char global_sdpsession[AST_MAX_EXTENSION]; /*!< SDP session name for the SIP channel */
static char global_sdpowner[AST_MAX_EXTENSION]; /*!< SDP owner name for the SIP channel */
static int global_authfailureevents; /*!< Whether we send authentication failure manager events or not. Default no. */
static int global_t1; /*!< T1 time */
static int global_t1min; /*!< T1 roundtrip time minimum */
static int global_timer_b; /*!< Timer B - RFC 3261 Section 17.1.1.2 */
static int global_autoframing; /*!< Turn autoframing on or off. */
static int global_qualifyfreq; /*!< Qualify frequency */
static int global_qualify_gap; /*!< Time between our group of peer pokes */
static int global_qualify_peers; /*!< Number of peers to poke at a given time */
/*! \brief Codecs that we support by default: */
static int global_capability = AST_FORMAT_ULAW | AST_FORMAT_ALAW | AST_FORMAT_GSM | AST_FORMAT_H263;
static enum st_mode global_st_mode; /*!< Mode of operation for Session-Timers */
static enum st_refresher global_st_refresher; /*!< Session-Timer refresher */
static int global_min_se; /*!< Lowest threshold for session refresh interval */
static int global_max_se; /*!< Highest threshold for session refresh interval */
/*@}*/
/*! \brief Global list of addresses dynamic peers are not allowed to use */
static struct ast_ha *global_contact_ha = NULL;
static int global_dynamic_exclude_static = 0;
/*! \name Object counters @{
* \bug These counters are not handled in a thread-safe way ast_atomic_fetchadd_int()
* should be used to modify these values. */
static int speerobjs = 0; /*!< Static peers */
static int rpeerobjs = 0; /*!< Realtime peers */
static int apeerobjs = 0; /*!< Autocreated peer objects */
static int regobjs = 0; /*!< Registry objects */
/* }@ */
static struct ast_flags global_flags[2] = {{0}}; /*!< global SIP_ flags */
static char used_context[AST_MAX_CONTEXT]; /*!< name of automatically created context for unloading */
AST_MUTEX_DEFINE_STATIC(netlock);
/*! \brief Protect the monitoring thread, so only one process can kill or start it, and not
when it's doing something critical. */
AST_MUTEX_DEFINE_STATIC(monlock);
AST_MUTEX_DEFINE_STATIC(sip_reload_lock);
/*! \brief This is the thread for the monitor which checks for input on the channels
which are not currently in use. */
static pthread_t monitor_thread = AST_PTHREADT_NULL;
static int sip_reloading = FALSE; /*!< Flag for avoiding multiple reloads at the same time */
static enum channelreloadreason sip_reloadreason; /*!< Reason for last reload/load of configuration */
static struct sched_context *sched; /*!< The scheduling context */
static struct io_context *io; /*!< The IO context */
static int *sipsock_read_id; /*!< ID of IO entry for sipsock FD */
#define DEC_CALL_LIMIT 0
#define INC_CALL_LIMIT 1
#define DEC_CALL_RINGING 2
#define INC_CALL_RINGING 3
/*! \brief The SIP socket definition */
struct sip_socket {
enum sip_transport type; /*!< UDP, TCP or TLS */
int fd; /*!< Filed descriptor, the actual socket */
uint16_t port;
struct ast_tcptls_session_instance *tcptls_session; /* If tcp or tls, a socket manager */
};
/*! \brief sip_request: The data grabbed from the UDP socket
*
* \verbatim
* Incoming messages: we first store the data from the socket in data[],
* adding a trailing \0 to make string parsing routines happy.
* Then call parse_request() and req.method = find_sip_method();
* to initialize the other fields. The \r\n at the end of each line is
* replaced by \0, so that data[] is not a conforming SIP message anymore.
* After this processing, rlPart1 is set to non-NULL to remember
* that we can run get_header() on this kind of packet.
*
* parse_request() splits the first line as follows:
* Requests have in the first line method uri SIP/2.0
* rlPart1 = method; rlPart2 = uri;
* Responses have in the first line SIP/2.0 NNN description
* rlPart1 = SIP/2.0; rlPart2 = NNN + description;
*
* For outgoing packets, we initialize the fields with init_req() or init_resp()
* (which fills the first line to "METHOD uri SIP/2.0" or "SIP/2.0 code text"),
* and then fill the rest with add_header() and add_line().
* The \r\n at the end of the line are still there, so the get_header()
* and similar functions don't work on these packets.
* \endverbatim
*/
struct sip_request {
ptrdiff_t rlPart1; /*!< Offset of the SIP Method Name or "SIP/2.0" protocol version */
ptrdiff_t rlPart2; /*!< Offset of the Request URI or Response Status */
int len; /*!< bytes used in data[], excluding trailing null terminator. Rarely used. */
int headers; /*!< # of SIP Headers */
int method; /*!< Method of this request */
int lines; /*!< Body Content */
unsigned int sdp_start; /*!< the line number where the SDP begins */
unsigned int sdp_end; /*!< the line number where the SDP ends */
char debug; /*!< print extra debugging if non zero */
char has_to_tag; /*!< non-zero if packet has To: tag */
char ignore; /*!< if non-zero This is a re-transmit, ignore it */
/* Array of offsets into the request string of each SIP header*/
ptrdiff_t header[SIP_MAX_HEADERS];
/* Array of offsets into the request string of each SDP line*/
ptrdiff_t line[SIP_MAX_LINES];
struct ast_str *data;
/* XXX Do we need to unref socket.ser when the request goes away? */
struct sip_socket socket; /*!< The socket used for this request */
AST_LIST_ENTRY(sip_request) next;
};
/* \brief given a sip_request and an offset, return the char * that resides there
*
* It used to be that rlPart1, rlPart2, and the header and line arrays were character
* pointers. They are now offsets into the ast_str portion of the sip_request structure.
* To avoid adding a bunch of redundant pointer arithmetic to the code, this macro is
* provided to retrieve the string at a particular offset within the request's buffer
*/
#define REQ_OFFSET_TO_STR(req,offset) (ast_str_buffer((req)->data) + ((req)->offset))
/*! \brief structure used in transfers */
struct sip_dual {
struct ast_channel *chan1; /*!< First channel involved */
struct ast_channel *chan2; /*!< Second channel involved */
struct sip_request req; /*!< Request that caused the transfer (REFER) */
int seqno; /*!< Sequence number */
};
struct sip_pkt;
/*! \brief Parameters to the transmit_invite function */
struct sip_invite_param {
int addsipheaders; /*!< Add extra SIP headers */
const char *uri_options; /*!< URI options to add to the URI */
const char *vxml_url; /*!< VXML url for Cisco phones */
char *auth; /*!< Authentication */
char *authheader; /*!< Auth header */
enum sip_auth_type auth_type; /*!< Authentication type */
const char *replaces; /*!< Replaces header for call transfers */
int transfer; /*!< Flag - is this Invite part of a SIP transfer? (invite/replaces) */
};
/*! \brief Structure to save routing information for a SIP session */
struct sip_route {
struct sip_route *next;
char hop[0];
};
/*! \brief Modes for SIP domain handling in the PBX */
enum domain_mode {
SIP_DOMAIN_AUTO, /*!< This domain is auto-configured */
SIP_DOMAIN_CONFIG, /*!< This domain is from configuration */
};
/*! \brief Domain data structure.
\note In the future, we will connect this to a configuration tree specific
for this domain
*/
struct domain {
char domain[MAXHOSTNAMELEN]; /*!< SIP domain we are responsible for */
char context[AST_MAX_EXTENSION]; /*!< Incoming context for this domain */
enum domain_mode mode; /*!< How did we find this domain? */
AST_LIST_ENTRY(domain) list; /*!< List mechanics */
};
static AST_LIST_HEAD_STATIC(domain_list, domain); /*!< The SIP domain list */
/*! \brief sip_history: Structure for saving transactions within a SIP dialog */
struct sip_history {
AST_LIST_ENTRY(sip_history) list;
char event[0]; /* actually more, depending on needs */
};
AST_LIST_HEAD_NOLOCK(sip_history_head, sip_history); /*!< history list, entry in sip_pvt */
/*! \brief sip_auth: Credentials for authentication to other SIP services */
struct sip_auth {
char realm[AST_MAX_EXTENSION]; /*!< Realm in which these credentials are valid */
char username[256]; /*!< Username */
char secret[256]; /*!< Secret */
char md5secret[256]; /*!< MD5Secret */
struct sip_auth *next; /*!< Next auth structure in list */
};
/*! \name SIPflags
Various flags for the flags field in the pvt structure
Trying to sort these up (one or more of the following):
D: Dialog
P: Peer/user
G: Global flag
When flags are used by multiple structures, it is important that
they have a common layout so it is easy to copy them.
*/
/*@{*/
#define SIP_OUTGOING (1 << 0) /*!< D: Direction of the last transaction in this dialog */
#define SIP_RINGING (1 << 2) /*!< D: Have sent 180 ringing */
#define SIP_PROGRESS_SENT (1 << 3) /*!< D: Have sent 183 message progress */
#define SIP_NEEDREINVITE (1 << 4) /*!< D: Do we need to send another reinvite? */
#define SIP_PENDINGBYE (1 << 5) /*!< D: Need to send bye after we ack? */
#define SIP_GOTREFER (1 << 6) /*!< D: Got a refer? */
#define SIP_CALL_LIMIT (1 << 7) /*!< D: Call limit enforced for this call */
#define SIP_INC_COUNT (1 << 8) /*!< D: Did this dialog increment the counter of in-use calls? */
#define SIP_INC_RINGING (1 << 9) /*!< D: Did this connection increment the counter of in-use calls? */
#define SIP_DEFER_BYE_ON_TRANSFER (1 << 10) /*!< D: Do not hangup at first ast_hangup */
#define SIP_PROMISCREDIR (1 << 11) /*!< DP: Promiscuous redirection */
#define SIP_TRUSTRPID (1 << 12) /*!< DP: Trust RPID headers? */
#define SIP_USEREQPHONE (1 << 13) /*!< DP: Add user=phone to numeric URI. Default off */
#define SIP_USECLIENTCODE (1 << 14) /*!< DP: Trust X-ClientCode info message */
/* DTMF flags - see str2dtmfmode() and dtmfmode2str() */
#define SIP_DTMF (7 << 15) /*!< DP: DTMF Support: five settings, uses three bits */
#define SIP_DTMF_RFC2833 (0 << 15) /*!< DP: DTMF Support: RTP DTMF - "rfc2833" */
#define SIP_DTMF_INBAND (1 << 15) /*!< DP: DTMF Support: Inband audio, only for ULAW/ALAW - "inband" */
#define SIP_DTMF_INFO (2 << 15) /*!< DP: DTMF Support: SIP Info messages - "info" */
#define SIP_DTMF_AUTO (3 << 15) /*!< DP: DTMF Support: AUTO switch between rfc2833 and in-band DTMF */
#define SIP_DTMF_SHORTINFO (4 << 15) /*!< DP: DTMF Support: SIP Info messages - "info" - short variant */
/* NAT settings - see nat2str() */
#define SIP_NAT (3 << 18) /*!< DP: four settings, uses two bits */
#define SIP_NAT_NEVER (0 << 18) /*!< DP: No nat support */
#define SIP_NAT_RFC3581 (1 << 18) /*!< DP: NAT RFC3581 */
#define SIP_NAT_ROUTE (2 << 18) /*!< DP: NAT Only ROUTE */
#define SIP_NAT_ALWAYS (3 << 18) /*!< DP: NAT Both ROUTE and RFC3581 */
/* re-INVITE related settings */
#define SIP_REINVITE (7 << 20) /*!< DP: four settings, uses three bits */
#define SIP_REINVITE_NONE (0 << 20) /*!< DP: no reinvite allowed */
#define SIP_CAN_REINVITE (1 << 20) /*!< DP: allow peers to be reinvited to send media directly p2p */
#define SIP_CAN_REINVITE_NAT (2 << 20) /*!< DP: allow media reinvite when new peer is behind NAT */
#define SIP_REINVITE_UPDATE (4 << 20) /*!< DP: use UPDATE (RFC3311) when reinviting this peer */
/* "insecure" settings - see insecure2str() */
#define SIP_INSECURE (3 << 23) /*!< DP: three settings, uses two bits */
#define SIP_INSECURE_NONE (0 << 23) /*!< DP: secure mode */
#define SIP_INSECURE_PORT (1 << 23) /*!< DP: don't require matching port for incoming requests */
#define SIP_INSECURE_INVITE (1 << 24) /*!< DP: don't require authentication for incoming INVITEs */
/* Sending PROGRESS in-band settings */
#define SIP_PROG_INBAND (3 << 25) /*!< DP: three settings, uses two bits */
#define SIP_PROG_INBAND_NEVER (0 << 25)
#define SIP_PROG_INBAND_NO (1 << 25)
#define SIP_PROG_INBAND_YES (2 << 25)
#define SIP_SENDRPID (3 << 29) /*!< DP: Remote Party-ID Support */
#define SIP_SENDRPID_NO (0 << 29)
#define SIP_SENDRPID_PAI (1 << 29) /*!< Use "P-Asserted-Identity" for rpid */
#define SIP_SENDRPID_RPID (2 << 29) /*!< Use "Remote-Party-ID" for rpid */
#define SIP_G726_NONSTANDARD (1 << 31) /*!< DP: Use non-standard packing for G726-32 data */
/*! \brief Flags to copy from peer/user to dialog */
#define SIP_FLAGS_TO_COPY \
(SIP_PROMISCREDIR | SIP_TRUSTRPID | SIP_SENDRPID | SIP_DTMF | SIP_REINVITE | \
SIP_PROG_INBAND | SIP_USECLIENTCODE | SIP_NAT | SIP_G726_NONSTANDARD | \
SIP_USEREQPHONE | SIP_INSECURE)
/*@}*/
/*! \name SIPflags2
a second page of flags (for flags[1] */
/*@{*/
/* realtime flags */
#define SIP_PAGE2_RTCACHEFRIENDS (1 << 0) /*!< GP: Should we keep RT objects in memory for extended time? */
#define SIP_PAGE2_RTAUTOCLEAR (1 << 2) /*!< GP: Should we clean memory from peers after expiry? */
/* Space for addition of other realtime flags in the future */
#define SIP_PAGE2_STATECHANGEQUEUE (1 << 9) /*!< D: Unsent state pending change exists */
#define SIP_PAGE2_CONNECTLINEUPDATE_PEND (1 << 10)
#define SIP_PAGE2_RPID_IMMEDIATE (1 << 11)
#define SIP_PAGE2_PREFERRED_CODEC (1 << 13) /*!< GDP: Only respond with single most preferred joint codec */
#define SIP_PAGE2_VIDEOSUPPORT (1 << 14) /*!< DP: Video supported if offered? */
#define SIP_PAGE2_TEXTSUPPORT (1 << 15) /*!< GDP: Global text enable */
#define SIP_PAGE2_ALLOWSUBSCRIBE (1 << 16) /*!< GP: Allow subscriptions from this peer? */
#define SIP_PAGE2_ALLOWOVERLAP (1 << 17) /*!< DP: Allow overlap dialing ? */
#define SIP_PAGE2_SUBSCRIBEMWIONLY (1 << 18) /*!< GP: Only issue MWI notification if subscribed to */
#define SIP_PAGE2_IGNORESDPVERSION (1 << 19) /*!< GDP: Ignore the SDP session version number we receive and treat all sessions as new */
#define SIP_PAGE2_T38SUPPORT (7 << 20) /*!< GDP: T38 Fax Passthrough Support */
#define SIP_PAGE2_T38SUPPORT_UDPTL (1 << 20) /*!< GDP: T38 Fax Passthrough Support */
#define SIP_PAGE2_T38SUPPORT_RTP (2 << 20) /*!< GDP: T38 Fax Passthrough Support (not implemented) */
#define SIP_PAGE2_T38SUPPORT_TCP (4 << 20) /*!< GDP: T38 Fax Passthrough Support (not implemented) */
#define SIP_PAGE2_CALL_ONHOLD (3 << 23) /*!< D: Call hold states: */
#define SIP_PAGE2_CALL_ONHOLD_ACTIVE (1 << 23) /*!< D: Active hold */
#define SIP_PAGE2_CALL_ONHOLD_ONEDIR (2 << 23) /*!< D: One directional hold */
#define SIP_PAGE2_CALL_ONHOLD_INACTIVE (3 << 23) /*!< D: Inactive hold */
#define SIP_PAGE2_RFC2833_COMPENSATE (1 << 25) /*!< DP: Compensate for buggy RFC2833 implementations */
#define SIP_PAGE2_BUGGY_MWI (1 << 26) /*!< DP: Buggy CISCO MWI fix */
#define SIP_PAGE2_DIALOG_ESTABLISHED (1 << 27) /*!< 29: Has a dialog been established? */
#define SIP_PAGE2_FAX_DETECT (1 << 28) /*!< DP: Fax Detection support */
#define SIP_PAGE2_REGISTERTRYING (1 << 29) /*!< DP: Send 100 Trying on REGISTER attempts */
#define SIP_PAGE2_UDPTL_DESTINATION (1 << 30) /*!< DP: Use source IP of RTP as destination if NAT is enabled */
#define SIP_PAGE2_VIDEOSUPPORT_ALWAYS (1 << 31) /*!< DP: Always set up video, even if endpoints don't support it */
#define SIP_PAGE2_FLAGS_TO_COPY \
(SIP_PAGE2_ALLOWSUBSCRIBE | SIP_PAGE2_ALLOWOVERLAP | SIP_PAGE2_IGNORESDPVERSION | \
SIP_PAGE2_VIDEOSUPPORT | SIP_PAGE2_T38SUPPORT | SIP_PAGE2_RFC2833_COMPENSATE | \
SIP_PAGE2_BUGGY_MWI | SIP_PAGE2_TEXTSUPPORT | SIP_PAGE2_FAX_DETECT | \
SIP_PAGE2_UDPTL_DESTINATION | SIP_PAGE2_VIDEOSUPPORT_ALWAYS | SIP_PAGE2_PREFERRED_CODEC | \
SIP_PAGE2_RPID_IMMEDIATE)
/*@}*/
/*! \name SIPflagsT38
T.38 set of flags */
/*@{*/
#define T38FAX_FILL_BIT_REMOVAL (1 << 0) /*!< Default: 0 (unset)*/
#define T38FAX_TRANSCODING_MMR (1 << 1) /*!< Default: 0 (unset)*/
#define T38FAX_TRANSCODING_JBIG (1 << 2) /*!< Default: 0 (unset)*/
/* Rate management */
#define T38FAX_RATE_MANAGEMENT_TRANSFERED_TCF (0 << 3)
#define T38FAX_RATE_MANAGEMENT_LOCAL_TCF (1 << 3) /*!< Unset for transferredTCF (UDPTL), set for localTCF (TPKT) */
/* UDP Error correction */
#define T38FAX_UDP_EC_NONE (0 << 4) /*!< two bits, if unset NO t38UDPEC field in T38 SDP*/
#define T38FAX_UDP_EC_FEC (1 << 4) /*!< Set for t38UDPFEC */
#define T38FAX_UDP_EC_REDUNDANCY (2 << 4) /*!< Set for t38UDPRedundancy */
/* T38 Spec version */
#define T38FAX_VERSION (3 << 6) /*!< two bits, 2 values so far, up to 4 values max */
#define T38FAX_VERSION_0 (0 << 6) /*!< Version 0 */
#define T38FAX_VERSION_1 (1 << 6) /*!< Version 1 */
/* Maximum Fax Rate */
#define T38FAX_RATE_2400 (1 << 8) /*!< 2400 bps t38FaxRate */
#define T38FAX_RATE_4800 (1 << 9) /*!< 4800 bps t38FaxRate */
#define T38FAX_RATE_7200 (1 << 10) /*!< 7200 bps t38FaxRate */
#define T38FAX_RATE_9600 (1 << 11) /*!< 9600 bps t38FaxRate */
#define T38FAX_RATE_12000 (1 << 12) /*!< 12000 bps t38FaxRate */
#define T38FAX_RATE_14400 (1 << 13) /*!< 14400 bps t38FaxRate */
/*!< This is default: NO MMR and JBIG transcoding, NO fill bit removal, transferredTCF TCF, UDP FEC, Version 0 and 9600 max fax rate */
static int global_t38_capability = T38FAX_VERSION_0 | T38FAX_RATE_2400 | T38FAX_RATE_4800 | T38FAX_RATE_7200 | T38FAX_RATE_9600;
/*@}*/
/*! \brief debugging state
* We store separately the debugging requests from the config file
* and requests from the CLI. Debugging is enabled if either is set
* (which means that if sipdebug is set in the config file, we can
* only turn it off by reloading the config).
*/
enum sip_debug_e {
sip_debug_none = 0,
sip_debug_config = 1,
sip_debug_console = 2,
};
static enum sip_debug_e sipdebug;
/*! \brief extra debugging for 'text' related events.
* At the moment this is set together with sip_debug_console.
* \note It should either go away or be implemented properly.
*/
static int sipdebug_text;
/*! \brief T38 States for a call */
enum t38state {
T38_DISABLED = 0, /*!< Not enabled */
T38_LOCAL_REINVITE, /*!< Offered from local - REINVITE */
T38_PEER_DIRECT, /*!< Offered from peer */
T38_PEER_REINVITE, /*!< Offered from peer - REINVITE */
T38_ENABLED /*!< Negotiated (enabled) */
};
/*! \brief T.38 channel settings (at some point we need to make this alloc'ed */
struct t38properties {
struct ast_flags t38support; /*!< Flag for udptl, rtp or tcp support for this session */
int capability; /*!< Our T38 capability */
int peercapability; /*!< Peers T38 capability */
int jointcapability; /*!< Supported T38 capability at both ends */
enum t38state state; /*!< T.38 state */
unsigned int direct:1; /*!< Whether the T38 came from the initial invite or not */
};
/*! \brief Parameters to know status of transfer */
enum referstatus {
REFER_IDLE, /*!< No REFER is in progress */
REFER_SENT, /*!< Sent REFER to transferee */
REFER_RECEIVED, /*!< Received REFER from transferrer */
REFER_CONFIRMED, /*!< Refer confirmed with a 100 TRYING (unused) */
REFER_ACCEPTED, /*!< Accepted by transferee */
REFER_RINGING, /*!< Target Ringing */
REFER_200OK, /*!< Answered by transfer target */
REFER_FAILED, /*!< REFER declined - go on */
REFER_NOAUTH /*!< We had no auth for REFER */
};
/*! \brief generic struct to map between strings and integers.
* Fill it with x-s pairs, terminate with an entry with s = NULL;
* Then you can call map_x_s(...) to map an integer to a string,
* and map_s_x() for the string -> integer mapping.
*/
struct _map_x_s {
int x;
const char *s;
};
static const struct _map_x_s referstatusstrings[] = {
{ REFER_IDLE, "<none>" },
{ REFER_SENT, "Request sent" },
{ REFER_RECEIVED, "Request received" },
{ REFER_CONFIRMED, "Confirmed" },
{ REFER_ACCEPTED, "Accepted" },
{ REFER_RINGING, "Target ringing" },
{ REFER_200OK, "Done" },
{ REFER_FAILED, "Failed" },
{ REFER_NOAUTH, "Failed - auth failure" },
{ -1, NULL} /* terminator */
};
/*! \brief Structure to handle SIP transfers. Dynamically allocated when needed
\note OEJ: Should be moved to string fields */
struct sip_refer {
char refer_to[AST_MAX_EXTENSION]; /*!< Place to store REFER-TO extension */
char refer_to_domain[AST_MAX_EXTENSION]; /*!< Place to store REFER-TO domain */
char refer_to_urioption[AST_MAX_EXTENSION]; /*!< Place to store REFER-TO uri options */
char refer_to_context[AST_MAX_EXTENSION]; /*!< Place to store REFER-TO context */
char referred_by[AST_MAX_EXTENSION]; /*!< Place to store REFERRED-BY extension */
char referred_by_name[AST_MAX_EXTENSION]; /*!< Place to store REFERRED-BY extension */
char refer_contact[AST_MAX_EXTENSION]; /*!< Place to store Contact info from a REFER extension */
char replaces_callid[SIPBUFSIZE]; /*!< Replace info: callid */
char replaces_callid_totag[SIPBUFSIZE/2]; /*!< Replace info: to-tag */
char replaces_callid_fromtag[SIPBUFSIZE/2]; /*!< Replace info: from-tag */
struct sip_pvt *refer_call; /*!< Call we are referring. This is just a reference to a
* dialog owned by someone else, so we should not destroy
* it when the sip_refer object goes.
*/
int attendedtransfer; /*!< Attended or blind transfer? */
int localtransfer; /*!< Transfer to local domain? */
enum referstatus status; /*!< REFER status */
};
/*! \brief Structure that encapsulates all attributes related to running
* SIP Session-Timers feature on a per dialog basis.
*/
struct sip_st_dlg {
int st_active; /*!< Session-Timers on/off */
int st_interval; /*!< Session-Timers negotiated session refresh interval */
int st_schedid; /*!< Session-Timers ast_sched scheduler id */
enum st_refresher st_ref; /*!< Session-Timers session refresher */
int st_expirys; /*!< Session-Timers number of expirys */
int st_active_peer_ua; /*!< Session-Timers on/off in peer UA */
int st_cached_min_se; /*!< Session-Timers cached Min-SE */
int st_cached_max_se; /*!< Session-Timers cached Session-Expires */
enum st_mode st_cached_mode; /*!< Session-Timers cached M.O. */
enum st_refresher st_cached_ref; /*!< Session-Timers cached refresher */
};
/*! \brief Structure that encapsulates all attributes related to configuration
* of SIP Session-Timers feature on a per user/peer basis.
*/
struct sip_st_cfg {
enum st_mode st_mode_oper; /*!< Mode of operation for Session-Timers */
enum st_refresher st_ref; /*!< Session-Timer refresher */
int st_min_se; /*!< Lowest threshold for session refresh interval */
int st_max_se; /*!< Highest threshold for session refresh interval */
};
/*! \brief Structure used for each SIP dialog, ie. a call, a registration, a subscribe.
* Created and initialized by sip_alloc(), the descriptor goes into the list of
* descriptors (dialoglist).
*/
struct sip_pvt {
struct sip_pvt *next; /*!< Next dialog in chain */
enum invitestates invitestate; /*!< Track state of SIP_INVITEs */
int method; /*!< SIP method that opened this dialog */
AST_DECLARE_STRING_FIELDS(
AST_STRING_FIELD(callid); /*!< Global CallID */
AST_STRING_FIELD(randdata); /*!< Random data */
AST_STRING_FIELD(accountcode); /*!< Account code */
AST_STRING_FIELD(realm); /*!< Authorization realm */
AST_STRING_FIELD(nonce); /*!< Authorization nonce */
AST_STRING_FIELD(opaque); /*!< Opaque nonsense */
AST_STRING_FIELD(qop); /*!< Quality of Protection, since SIP wasn't complicated enough yet. */
AST_STRING_FIELD(domain); /*!< Authorization domain */
AST_STRING_FIELD(from); /*!< The From: header */
AST_STRING_FIELD(useragent); /*!< User agent in SIP request */
AST_STRING_FIELD(exten); /*!< Extension where to start */
AST_STRING_FIELD(context); /*!< Context for this call */
AST_STRING_FIELD(subscribecontext); /*!< Subscribecontext */
AST_STRING_FIELD(subscribeuri); /*!< Subscribecontext */
AST_STRING_FIELD(fromdomain); /*!< Domain to show in the from field */
AST_STRING_FIELD(fromuser); /*!< User to show in the user field */
AST_STRING_FIELD(fromname); /*!< Name to show in the user field */
AST_STRING_FIELD(tohost); /*!< Host we should put in the "to" field */
AST_STRING_FIELD(todnid); /*!< DNID of this call (overrides host) */
AST_STRING_FIELD(language); /*!< Default language for this call */
AST_STRING_FIELD(mohinterpret); /*!< MOH class to use when put on hold */
AST_STRING_FIELD(mohsuggest); /*!< MOH class to suggest when putting a peer on hold */
AST_STRING_FIELD(rdnis); /*!< Referring DNIS */
AST_STRING_FIELD(redircause); /*!< Referring cause */
AST_STRING_FIELD(theirtag); /*!< Their tag */
AST_STRING_FIELD(username); /*!< [user] name */
AST_STRING_FIELD(peername); /*!< [peer] name, not set if [user] */
AST_STRING_FIELD(authname); /*!< Who we use for authentication */
AST_STRING_FIELD(uri); /*!< Original requested URI */
AST_STRING_FIELD(okcontacturi); /*!< URI from the 200 OK on INVITE */
AST_STRING_FIELD(peersecret); /*!< Password */
AST_STRING_FIELD(peermd5secret);
AST_STRING_FIELD(cid_num); /*!< Caller*ID number */
AST_STRING_FIELD(cid_name); /*!< Caller*ID name */
AST_STRING_FIELD(mwi_from); /*!< Name to place in the From header in outgoing NOTIFY requests */
AST_STRING_FIELD(fullcontact); /*!< The Contact: that the UA registers with us */
/* we only store the part in <brackets> in this field. */
AST_STRING_FIELD(our_contact); /*!< Our contact header */
AST_STRING_FIELD(url); /*!< URL to be sent with next message to peer */
AST_STRING_FIELD(parkinglot); /*!< Parkinglot */
AST_STRING_FIELD(engine); /*!< RTP engine to use */
);
char via[128]; /*!< Via: header */
struct sip_socket socket; /*!< The socket used for this dialog */
unsigned int ocseq; /*!< Current outgoing seqno */
unsigned int icseq; /*!< Current incoming seqno */
ast_group_t callgroup; /*!< Call group */
ast_group_t pickupgroup; /*!< Pickup group */
int lastinvite; /*!< Last Cseq of invite */
int lastnoninvite; /*!< Last Cseq of non-invite */
struct ast_flags flags[2]; /*!< SIP_ flags */
/* boolean or small integers that don't belong in flags */
char do_history; /*!< Set if we want to record history */
char alreadygone; /*!< already destroyed by our peer */
char needdestroy; /*!< need to be destroyed by the monitor thread */
char outgoing_call; /*!< this is an outgoing call */
char answered_elsewhere; /*!< This call is cancelled due to answer on another channel */
char novideo; /*!< Didn't get video in invite, don't offer */
char notext; /*!< Text not supported (?) */
int timer_t1; /*!< SIP timer T1, ms rtt */
int timer_b; /*!< SIP timer B, ms */
unsigned int sipoptions; /*!< Supported SIP options on the other end */
unsigned int reqsipoptions; /*!< Required SIP options on the other end */
struct ast_codec_pref prefs; /*!< codec prefs */
int capability; /*!< Special capability (codec) */
int jointcapability; /*!< Supported capability at both ends (codecs) */
int peercapability; /*!< Supported peer capability */
int prefcodec; /*!< Preferred codec (outbound only) */
int noncodeccapability; /*!< DTMF RFC2833 telephony-event */
int jointnoncodeccapability; /*!< Joint Non codec capability */
int redircodecs; /*!< Redirect codecs */
int maxcallbitrate; /*!< Maximum Call Bitrate for Video Calls */
struct sip_proxy *outboundproxy; /*!< Outbound proxy for this dialog. Use ref_proxy to set this instead of setting it directly*/
struct t38properties t38; /*!< T38 settings */
struct sockaddr_in udptlredirip; /*!< Where our T.38 UDPTL should be going if not to us */
struct ast_udptl *udptl; /*!< T.38 UDPTL session */
int callingpres; /*!< Calling presentation */
int authtries; /*!< Times we've tried to authenticate */
int expiry; /*!< How long we take to expire */
long branch; /*!< The branch identifier of this session */
long invite_branch; /*!< The branch used when we sent the initial INVITE */
char tag[11]; /*!< Our tag for this session */
int sessionid; /*!< SDP Session ID */
int sessionversion; /*!< SDP Session Version */
uint64_t sessionversion_remote; /*!< Remote UA's SDP Session Version */
int session_modify; /*!< Session modification request true/false */
struct sockaddr_in sa; /*!< Our peer */
struct sockaddr_in redirip; /*!< Where our RTP should be going if not to us */
struct sockaddr_in vredirip; /*!< Where our Video RTP should be going if not to us */
struct sockaddr_in tredirip; /*!< Where our Text RTP should be going if not to us */
time_t lastrtprx; /*!< Last RTP received */
time_t lastrtptx; /*!< Last RTP sent */
int rtptimeout; /*!< RTP timeout time */
struct sockaddr_in recv; /*!< Received as */
struct sockaddr_in ourip; /*!< Our IP (as seen from the outside) */
struct ast_channel *owner; /*!< Who owns us (if we have an owner) */
struct sip_route *route; /*!< Head of linked list of routing steps (fm Record-Route) */
int route_persistant; /*!< Is this the "real" route? */
struct ast_variable *notify_headers; /*!< Custom notify type */
struct sip_auth *peerauth; /*!< Realm authentication */
int noncecount; /*!< Nonce-count */
char lastmsg[256]; /*!< Last Message sent/received */
int amaflags; /*!< AMA Flags */
int pendinginvite; /*!< Any pending INVITE or state NOTIFY (in subscribe pvt's) ? (seqno of this) */
int glareinvite; /*!< A invite received while a pending invite is already present is stored here. Its seqno is the
value. Since this glare invite's seqno is not the same as the pending invite's, it must be
held in order to properly process acknowledgements for our 491 response. */
struct sip_request initreq; /*!< Latest request that opened a new transaction
within this dialog.
NOT the request that opened the dialog */
int initid; /*!< Auto-congest ID if appropriate (scheduler) */
int waitid; /*!< Wait ID for scheduler after 491 or other delays */
int autokillid; /*!< Auto-kill ID (scheduler) */
int t38id; /*!< T.38 Response ID */
enum transfermodes allowtransfer; /*!< REFER: restriction scheme */
struct sip_refer *refer; /*!< REFER: SIP transfer data structure */
enum subscriptiontype subscribed; /*!< SUBSCRIBE: Is this dialog a subscription? */
int stateid; /*!< SUBSCRIBE: ID for devicestate subscriptions */
int laststate; /*!< SUBSCRIBE: Last known extension state */
int dialogver; /*!< SUBSCRIBE: Version for subscription dialog-info */
struct ast_dsp *vad; /*!< Inband DTMF Detection dsp */
struct sip_peer *relatedpeer; /*!< If this dialog is related to a peer, which one
Used in peerpoke, mwi subscriptions */
struct sip_registry *registry; /*!< If this is a REGISTER dialog, to which registry */
struct ast_rtp_instance *rtp; /*!< RTP Session */
struct ast_rtp_instance *vrtp; /*!< Video RTP session */
struct ast_rtp_instance *trtp; /*!< Text RTP session */
struct sip_pkt *packets; /*!< Packets scheduled for re-transmission */
struct sip_history_head *history; /*!< History of this SIP dialog */
size_t history_entries; /*!< Number of entires in the history */
struct ast_variable *chanvars; /*!< Channel variables to set for inbound call */
AST_LIST_HEAD_NOLOCK(request_queue, sip_request) request_queue; /*!< Requests that arrived but could not be processed immediately */
int request_queue_sched_id; /*!< Scheduler ID of any scheduled action to process queued requests */
struct sip_invite_param *options; /*!< Options for INVITE */
int autoframing; /*!< The number of Asters we group in a Pyroflax
before strolling to the Grokyzpå
(A bit unsure of this, please correct if
you know more) */
struct sip_st_dlg *stimer; /*!< SIP Session-Timers */
int red; /*!< T.140 RTP Redundancy */
int hangupcause; /*!< Storage of hangupcause copied from our owner before we disconnect from the AST channel (only used at hangup) */
struct sip_subscription_mwi *mwi; /*!< If this is a subscription MWI dialog, to which subscription */
};
/*! \brief
* Here we implement the container for dialogs (sip_pvt), defining
* generic wrapper functions to ease the transition from the current
* implementation (a single linked list) to a different container.
* In addition to a reference to the container, we need functions to lock/unlock
* the container and individual items, and functions to add/remove
* references to the individual items.
*/
struct ao2_container *dialogs;
#define sip_pvt_lock(x) ao2_lock(x)
#define sip_pvt_trylock(x) ao2_trylock(x)
#define sip_pvt_unlock(x) ao2_unlock(x)
/*! \brief
* when we create or delete references, make sure to use these
* functions so we keep track of the refcounts.
* To simplify the code, we allow a NULL to be passed to dialog_unref().
*/
#ifdef REF_DEBUG
#define dialog_ref(arg1,arg2) dialog_ref_debug((arg1),(arg2), __FILE__, __LINE__, __PRETTY_FUNCTION__)
#define dialog_unref(arg1,arg2) dialog_unref_debug((arg1),(arg2), __FILE__, __LINE__, __PRETTY_FUNCTION__)
static struct sip_pvt *dialog_ref_debug(struct sip_pvt *p, char *tag, char *file, int line, const char *func)
{
if (p)
_ao2_ref_debug(p, 1, tag, file, line, func);
else
ast_log(LOG_ERROR, "Attempt to Ref a null pointer\n");
return p;
}
static struct sip_pvt *dialog_unref_debug(struct sip_pvt *p, char *tag, char *file, int line, const char *func)
{
if (p)
_ao2_ref_debug(p, -1, tag, file, line, func);
return NULL;
}
#else
static struct sip_pvt *dialog_ref(struct sip_pvt *p, char *tag)
{
if (p)
ao2_ref(p, 1);
else
ast_log(LOG_ERROR, "Attempt to Ref a null pointer\n");
return p;
}
static struct sip_pvt *dialog_unref(struct sip_pvt *p, char *tag)
{
if (p)
ao2_ref(p, -1);
return NULL;
}
#endif
/*! \brief sip packet - raw format for outbound packets that are sent or scheduled for transmission
* Packets are linked in a list, whose head is in the struct sip_pvt they belong to.
* Each packet holds a reference to the parent struct sip_pvt.
* This structure is allocated in __sip_reliable_xmit() and only for packets that
* require retransmissions.
*/
struct sip_pkt {
struct sip_pkt *next; /*!< Next packet in linked list */
int retrans; /*!< Retransmission number */
int method; /*!< SIP method for this packet */
int seqno; /*!< Sequence number */
char is_resp; /*!< 1 if this is a response packet (e.g. 200 OK), 0 if it is a request */
char is_fatal; /*!< non-zero if there is a fatal error */
struct sip_pvt *owner; /*!< Owner AST call */
int retransid; /*!< Retransmission ID */
int timer_a; /*!< SIP timer A, retransmission timer */
int timer_t1; /*!< SIP Timer T1, estimated RTT or 500 ms */
int packetlen; /*!< Length of packet */
struct ast_str *data;
};
/*!
* \brief A peer's mailbox
*
* We could use STRINGFIELDS here, but for only two strings, it seems like
* too much effort ...
*/
struct sip_mailbox {
char *mailbox;
char *context;
/*! Associated MWI subscription */
struct ast_event_sub *event_sub;
AST_LIST_ENTRY(sip_mailbox) entry;
};
enum sip_peer_type {
SIP_TYPE_PEER = (1 << 0),
SIP_TYPE_USER = (1 << 1),
};
/*! \brief Structure for SIP peer data, we place calls to peers if registered or fixed IP address (host)
*/
/* XXX field 'name' must be first otherwise sip_addrcmp() will fail, as will astobj2 hashing of the structure */
struct sip_peer {
char name[80]; /*!< the unique name of this object */
AST_DECLARE_STRING_FIELDS(
AST_STRING_FIELD(secret); /*!< Password for inbound auth */
AST_STRING_FIELD(md5secret); /*!< Password in MD5 */
AST_STRING_FIELD(remotesecret); /*!< Remote secret (trunks, remote devices) */
AST_STRING_FIELD(context); /*!< Default context for incoming calls */
AST_STRING_FIELD(subscribecontext); /*!< Default context for subscriptions */
AST_STRING_FIELD(username); /*!< Temporary username until registration */
AST_STRING_FIELD(accountcode); /*!< Account code */
AST_STRING_FIELD(tohost); /*!< If not dynamic, IP address */
AST_STRING_FIELD(regexten); /*!< Extension to register (if regcontext is used) */
AST_STRING_FIELD(fromuser); /*!< From: user when calling this peer */
AST_STRING_FIELD(fromdomain); /*!< From: domain when calling this peer */
AST_STRING_FIELD(fullcontact); /*!< Contact registered with us (not in sip.conf) */
AST_STRING_FIELD(cid_num); /*!< Caller ID num */
AST_STRING_FIELD(cid_name); /*!< Caller ID name */
AST_STRING_FIELD(vmexten); /*!< Dialplan extension for MWI notify message*/
AST_STRING_FIELD(language); /*!< Default language for prompts */
AST_STRING_FIELD(mohinterpret); /*!< Music on Hold class */
AST_STRING_FIELD(mohsuggest); /*!< Music on Hold class */
AST_STRING_FIELD(parkinglot); /*!< Parkinglot */
AST_STRING_FIELD(useragent); /*!< User agent in SIP request (saved from registration) */
AST_STRING_FIELD(mwi_from); /*!< Name to place in From header for outgoing NOTIFY requests */
AST_STRING_FIELD(engine); /*!< RTP Engine to use */
);
struct sip_socket socket; /*!< Socket used for this peer */
unsigned int transports:3; /*!< Transports (enum sip_transport) that are acceptable for this peer */
struct sip_auth *auth; /*!< Realm authentication list */
int amaflags; /*!< AMA Flags (for billing) */
int callingpres; /*!< Calling id presentation */
int inUse; /*!< Number of calls in use */
int inRinging; /*!< Number of calls ringing */
int onHold; /*!< Peer has someone on hold */
int call_limit; /*!< Limit of concurrent calls */
int busy_level; /*!< Level of active channels where we signal busy */
enum transfermodes allowtransfer; /*! SIP Refer restriction scheme */
struct ast_codec_pref prefs; /*!< codec prefs */
int lastmsgssent;
unsigned int sipoptions; /*!< Supported SIP options */
struct ast_flags flags[2]; /*!< SIP_ flags */
/*! Mailboxes that this peer cares about */
AST_LIST_HEAD_NOLOCK(, sip_mailbox) mailboxes;
/* things that don't belong in flags */
char is_realtime; /*!< this is a 'realtime' peer */
char rt_fromcontact; /*!< copy fromcontact from realtime */
char host_dynamic; /*!< Dynamic Peers register with Asterisk */
char selfdestruct; /*!< Automatic peers need to destruct themselves */
char the_mark; /*!< moved out of ASTOBJ into struct proper; That which bears the_mark should be deleted! */
int expire; /*!< When to expire this peer registration */
int capability; /*!< Codec capability */
int rtptimeout; /*!< RTP timeout */
int rtpholdtimeout; /*!< RTP Hold Timeout */
int rtpkeepalive; /*!< Send RTP packets for keepalive */
ast_group_t callgroup; /*!< Call group */
ast_group_t pickupgroup; /*!< Pickup group */
struct sip_proxy *outboundproxy; /*!< Outbound proxy for this peer */
struct ast_dnsmgr_entry *dnsmgr;/*!< DNS refresh manager for peer */
struct sockaddr_in addr; /*!< IP address of peer */
int maxcallbitrate; /*!< Maximum Bitrate for a video call */
/* Qualification */
struct sip_pvt *call; /*!< Call pointer */
int pokeexpire; /*!< When to expire poke (qualify= checking) */
int lastms; /*!< How long last response took (in ms), or -1 for no response */
int maxms; /*!< Max ms we will accept for the host to be up, 0 to not monitor */
int qualifyfreq; /*!< Qualification: How often to check for the host to be up */
struct timeval ps; /*!< Time for sending SIP OPTION in sip_pke_peer() */
struct sockaddr_in defaddr; /*!< Default IP address, used until registration */
struct ast_ha *ha; /*!< Access control list */
struct ast_ha *contactha; /*!< Restrict what IPs are allowed in the Contact header (for registration) */
struct ast_variable *chanvars; /*!< Variables to set for channel created by user */
struct sip_pvt *mwipvt; /*!< Subscription for MWI */
int autoframing;
struct sip_st_cfg stimer; /*!< SIP Session-Timers */
int timer_t1; /*!< The maximum T1 value for the peer */
int timer_b; /*!< The maximum timer B (transaction timeouts) */
int deprecated_username; /*!< If it's a realtime peer, are they using the deprecated "username" instead of "defaultuser" */
/*XXX Seems like we suddenly have two flags with the same content. Why? To be continued... */
enum sip_peer_type type; /*!< Distinguish between "user" and "peer" types. This is used solely for CLI and manager commands */
};
/*!
* \brief Registrations with other SIP proxies
*
* Created by sip_register(), the entry is linked in the 'regl' list,
* and never deleted (other than at 'sip reload' or module unload times).
* The entry always has a pending timeout, either waiting for an ACK to
* the REGISTER message (in which case we have to retransmit the request),
* or waiting for the next REGISTER message to be sent (either the initial one,
* or once the previously completed registration one expires).
* The registration can be in one of many states, though at the moment
* the handling is a bit mixed.
*
* XXX \todo Reference count handling for this object has some problems with
* respect to scheduler entries. The ref count is handled in some places,
* but not all of them. There are some places where references get leaked
* when this scheduler entry gets cancelled. At worst, this would cause
* memory leaks on reloads if registrations get removed from configuration.
*/
struct sip_registry {
ASTOBJ_COMPONENTS_FULL(struct sip_registry,1,1);
AST_DECLARE_STRING_FIELDS(
AST_STRING_FIELD(callid); /*!< Global Call-ID */
AST_STRING_FIELD(realm); /*!< Authorization realm */
AST_STRING_FIELD(nonce); /*!< Authorization nonce */
AST_STRING_FIELD(opaque); /*!< Opaque nonsense */
AST_STRING_FIELD(qop); /*!< Quality of Protection, since SIP wasn't complicated enough yet. */
AST_STRING_FIELD(domain); /*!< Authorization domain */
AST_STRING_FIELD(username); /*!< Who we are registering as */
AST_STRING_FIELD(authuser); /*!< Who we *authenticate* as */
AST_STRING_FIELD(hostname); /*!< Domain or host we register to */
AST_STRING_FIELD(secret); /*!< Password in clear text */
AST_STRING_FIELD(md5secret); /*!< Password in md5 */
AST_STRING_FIELD(callback); /*!< Contact extension */
AST_STRING_FIELD(random);
);
enum sip_transport transport; /*!< Transport for this registration UDP, TCP or TLS */
int portno; /*!< Optional port override */
int expire; /*!< Sched ID of expiration */
int expiry; /*!< Value to use for the Expires header */
int regattempts; /*!< Number of attempts (since the last success) */
int timeout; /*!< sched id of sip_reg_timeout */
int refresh; /*!< How often to refresh */
struct sip_pvt *call; /*!< create a sip_pvt structure for each outbound "registration dialog" in progress */
enum sipregistrystate regstate; /*!< Registration state (see above) */
struct timeval regtime; /*!< Last successful registration time */
int callid_valid; /*!< 0 means we haven't chosen callid for this registry yet. */
unsigned int ocseq; /*!< Sequence number we got to for REGISTERs for this registry */
struct ast_dnsmgr_entry *dnsmgr; /*!< DNS refresh manager for register */
struct sockaddr_in us; /*!< Who the server thinks we are */
int noncecount; /*!< Nonce-count */
char lastmsg[256]; /*!< Last Message sent/received */
};
/*! \brief Definition of a thread that handles a socket */
struct sip_threadinfo {
int stop;
pthread_t threadid;
struct ast_tcptls_session_instance *tcptls_session;
enum sip_transport type; /*!< We keep a copy of the type here so we can display it in the connection list */
AST_LIST_ENTRY(sip_threadinfo) list;
};
/*! \brief Definition of an MWI subscription to another server */
struct sip_subscription_mwi {
ASTOBJ_COMPONENTS_FULL(struct sip_subscription_mwi,1,1);
AST_DECLARE_STRING_FIELDS(
AST_STRING_FIELD(username); /*!< Who we are sending the subscription as */
AST_STRING_FIELD(authuser); /*!< Who we *authenticate* as */
AST_STRING_FIELD(hostname); /*!< Domain or host we subscribe to */
AST_STRING_FIELD(secret); /*!< Password in clear text */
AST_STRING_FIELD(mailbox); /*!< Mailbox store to put MWI into */
);
enum sip_transport transport; /*!< Transport to use */
int portno; /*!< Optional port override */
int resub; /*!< Sched ID of resubscription */
unsigned int subscribed:1; /*!< Whether we are currently subscribed or not */
struct sip_pvt *call; /*!< Outbound subscription dialog */
struct ast_dnsmgr_entry *dnsmgr; /*!< DNS refresh manager for subscription */
struct sockaddr_in us; /*!< Who the server thinks we are */
};
/* --- Hash tables of various objects --------*/
#ifdef LOW_MEMORY
static int hash_peer_size = 17;
static int hash_dialog_size = 17;
static int hash_user_size = 17;
#else
static int hash_peer_size = 563; /*!< Size of peer hash table, prime number preferred! */
static int hash_dialog_size = 563;
static int hash_user_size = 563;
#endif
/*! \brief The thread list of TCP threads */
static AST_LIST_HEAD_STATIC(threadl, sip_threadinfo);
/*! \brief The peer list: Users, Peers and Friends */
struct ao2_container *peers;
struct ao2_container *peers_by_ip;
/*! \brief The register list: Other SIP proxies we register with and place calls to */
static struct ast_register_list {
ASTOBJ_CONTAINER_COMPONENTS(struct sip_registry);
int recheck;
} regl;
/*! \brief The MWI subscription list */
static struct ast_subscription_mwi_list {
ASTOBJ_CONTAINER_COMPONENTS(struct sip_subscription_mwi);
} submwil;
/*! \brief
* \note The only member of the peer used here is the name field
*/
static int peer_hash_cb(const void *obj, const int flags)
{
const struct sip_peer *peer = obj;
return ast_str_case_hash(peer->name);
}
/*!
* \note The only member of the peer used here is the name field
*/
static int peer_cmp_cb(void *obj, void *arg, int flags)
{
struct sip_peer *peer = obj, *peer2 = arg;
return !strcasecmp(peer->name, peer2->name) ? CMP_MATCH | CMP_STOP : 0;
}
/*!
* \note the peer's addr struct provides to fields combined to make a key: the sin_addr.s_addr and sin_port fields.
*/
static int peer_iphash_cb(const void *obj, const int flags)
{
const struct sip_peer *peer = obj;
int ret1 = peer->addr.sin_addr.s_addr;
if (ret1 < 0)
ret1 = -ret1;
if (ast_test_flag(&peer->flags[0], SIP_INSECURE_PORT)) {
return ret1;
} else {
return ret1 + peer->addr.sin_port;
}
}
/*!
* \note the peer's addr struct provides to fields combined to make a key: the sin_addr.s_addr and sin_port fields.
*/
static int peer_ipcmp_cb(void *obj, void *arg, int flags)
{
struct sip_peer *peer = obj, *peer2 = arg;
if (peer->addr.sin_addr.s_addr != peer2->addr.sin_addr.s_addr)
return 0;
if (!ast_test_flag(&peer->flags[0], SIP_INSECURE_PORT) && !ast_test_flag(&peer2->flags[0], SIP_INSECURE_PORT)) {
if (peer->addr.sin_port == peer2->addr.sin_port)
return CMP_MATCH | CMP_STOP;
else
return 0;
}
return CMP_MATCH | CMP_STOP;
}
/*!
* \note The only member of the dialog used here callid string
*/
static int dialog_hash_cb(const void *obj, const int flags)
{
const struct sip_pvt *pvt = obj;
return ast_str_case_hash(pvt->callid);
}
/*!
* \note The only member of the dialog used here callid string
*/
static int dialog_cmp_cb(void *obj, void *arg, int flags)
{
struct sip_pvt *pvt = obj, *pvt2 = arg;
return !strcasecmp(pvt->callid, pvt2->callid) ? CMP_MATCH | CMP_STOP : 0;
}
static int temp_pvt_init(void *);
static void temp_pvt_cleanup(void *);
/*! \brief A per-thread temporary pvt structure */
AST_THREADSTORAGE_CUSTOM(ts_temp_pvt, temp_pvt_init, temp_pvt_cleanup);
#ifdef LOW_MEMORY
static void ts_ast_rtp_destroy(void *);
AST_THREADSTORAGE_CUSTOM(ts_audio_rtp, NULL, ts_ast_rtp_destroy);
AST_THREADSTORAGE_CUSTOM(ts_video_rtp, NULL, ts_ast_rtp_destroy);
AST_THREADSTORAGE_CUSTOM(ts_text_rtp, NULL, ts_ast_rtp_destroy);
#endif
/*! \brief Authentication list for realm authentication
* \todo Move the sip_auth list to AST_LIST */
static struct sip_auth *authl = NULL;
/* --- Sockets and networking --------------*/
/*! \brief Main socket for UDP SIP communication.
*
* sipsock is shared between the SIP manager thread (which handles reload
* requests), the udp io handler (sipsock_read()) and the user routines that
* issue udp writes (using __sip_xmit()).
* The socket is -1 only when opening fails (this is a permanent condition),
* or when we are handling a reload() that changes its address (this is
* a transient situation during which we might have a harmless race, see
* below). Because the conditions for the race to be possible are extremely
* rare, we don't want to pay the cost of locking on every I/O.
* Rather, we remember that when the race may occur, communication is
* bound to fail anyways, so we just live with this event and let
* the protocol handle this above us.
*/
static int sipsock = -1;
static struct sockaddr_in bindaddr; /*!< UDP: The address we bind to */
/*! \brief our (internal) default address/port to put in SIP/SDP messages
* internip is initialized picking a suitable address from one of the
* interfaces, and the same port number we bind to. It is used as the
* default address/port in SIP messages, and as the default address
* (but not port) in SDP messages.
*/
static struct sockaddr_in internip;
/*! \brief our external IP address/port for SIP sessions.
* externip.sin_addr is only set when we know we might be behind
* a NAT, and this is done using a variety of (mutually exclusive)
* ways from the config file:
*
* + with "externip = host[:port]" we specify the address/port explicitly.
* The address is looked up only once when (re)loading the config file;
*
* + with "externhost = host[:port]" we do a similar thing, but the
* hostname is stored in externhost, and the hostname->IP mapping
* is refreshed every 'externrefresh' seconds;
*
* + with "stunaddr = host[:port]" we run queries every externrefresh seconds
* to the specified server, and store the result in externip.
*
* Other variables (externhost, externexpire, externrefresh) are used
* to support the above functions.
*/
static struct sockaddr_in externip; /*!< External IP address if we are behind NAT */
static char externhost[MAXHOSTNAMELEN]; /*!< External host name */
static time_t externexpire; /*!< Expiration counter for re-resolving external host name in dynamic DNS */
static int externrefresh = 10;
static struct sockaddr_in stunaddr; /*!< stun server address */
/*! \brief List of local networks
* We store "localnet" addresses from the config file into an access list,
* marked as 'DENY', so the call to ast_apply_ha() will return
* AST_SENSE_DENY for 'local' addresses, and AST_SENSE_ALLOW for 'non local'
* (i.e. presumably public) addresses.
*/
static struct ast_ha *localaddr; /*!< List of local networks, on the same side of NAT as this Asterisk */
static int ourport_tcp; /*!< The port used for TCP connections */
static int ourport_tls; /*!< The port used for TCP/TLS connections */
static struct sockaddr_in debugaddr;
static struct ast_config *notify_types; /*!< The list of manual NOTIFY types we know how to send */
/*! some list management macros. */
#define UNLINK(element, head, prev) do { \
if (prev) \
(prev)->next = (element)->next; \
else \
(head) = (element)->next; \
} while (0)
enum t38_action_flag {
SDP_T38_NONE = 0, /*!< Do not modify T38 information at all */
SDP_T38_INITIATE, /*!< Remote side has requested T38 with us */
SDP_T38_ACCEPT, /*!< Remote side accepted our T38 request */
};
/*---------------------------- Forward declarations of functions in chan_sip.c */
/* Note: This is added to help splitting up chan_sip.c into several files
in coming releases. */
/*--- PBX interface functions */
static struct ast_channel *sip_request_call(const char *type, int format, void *data, int *cause);
static int sip_devicestate(void *data);
static int sip_sendtext(struct ast_channel *ast, const char *text);
static int sip_call(struct ast_channel *ast, char *dest, int timeout);
static int sip_sendhtml(struct ast_channel *chan, int subclass, const char *data, int datalen);
static int sip_hangup(struct ast_channel *ast);
static int sip_answer(struct ast_channel *ast);
static struct ast_frame *sip_read(struct ast_channel *ast);
static int sip_write(struct ast_channel *ast, struct ast_frame *frame);
static int sip_indicate(struct ast_channel *ast, int condition, const void *data, size_t datalen);
static int sip_transfer(struct ast_channel *ast, const char *dest);
static int sip_fixup(struct ast_channel *oldchan, struct ast_channel *newchan);
static int sip_senddigit_begin(struct ast_channel *ast, char digit);
static int sip_senddigit_end(struct ast_channel *ast, char digit, unsigned int duration);
static int sip_setoption(struct ast_channel *chan, int option, void *data, int datalen);
static int sip_queryoption(struct ast_channel *chan, int option, void *data, int *datalen);
static const char *sip_get_callid(struct ast_channel *chan);
static int handle_request_do(struct sip_request *req, struct sockaddr_in *sin);
static int sip_standard_port(enum sip_transport type, int port);
static int sip_prepare_socket(struct sip_pvt *p);
static int sip_parse_host(char *line, int lineno, char **hostname, int *portnum, enum sip_transport *transport);
/*--- Transmitting responses and requests */
static int sipsock_read(int *id, int fd, short events, void *ignore);
static int __sip_xmit(struct sip_pvt *p, struct ast_str *data, int len);
static int __sip_reliable_xmit(struct sip_pvt *p, int seqno, int resp, struct ast_str *data, int len, int fatal, int sipmethod);
static int __transmit_response(struct sip_pvt *p, const char *msg, const struct sip_request *req, enum xmittype reliable);
static int retrans_pkt(const void *data);
static int transmit_response_using_temp(ast_string_field callid, struct sockaddr_in *sin, int useglobal_nat, const int intended_method, const struct sip_request *req, const char *msg);
static int transmit_response(struct sip_pvt *p, const char *msg, const struct sip_request *req);
static int transmit_response_reliable(struct sip_pvt *p, const char *msg, const struct sip_request *req);
static int transmit_response_with_date(struct sip_pvt *p, const char *msg, const struct sip_request *req);
static int transmit_response_with_sdp(struct sip_pvt *p, const char *msg, const struct sip_request *req, enum xmittype reliable, int oldsdp, int rpid);
static int transmit_response_with_unsupported(struct sip_pvt *p, const char *msg, const struct sip_request *req, const char *unsupported);
static int transmit_response_with_auth(struct sip_pvt *p, const char *msg, const struct sip_request *req, const char *rand, enum xmittype reliable, const char *header, int stale);
static int transmit_response_with_allow(struct sip_pvt *p, const char *msg, const struct sip_request *req, enum xmittype reliable);
static void transmit_fake_auth_response(struct sip_pvt *p, int sipmethod, struct sip_request *req, enum xmittype reliable);
static int transmit_request(struct sip_pvt *p, int sipmethod, int inc, enum xmittype reliable, int newbranch);
static int transmit_request_with_auth(struct sip_pvt *p, int sipmethod, int seqno, enum xmittype reliable, int newbranch);
static int transmit_invite(struct sip_pvt *p, int sipmethod, int sdp, int init);
static int transmit_reinvite_with_sdp(struct sip_pvt *p, int t38version, int oldsdp);
static int transmit_info_with_digit(struct sip_pvt *p, const char digit, unsigned int duration);
static int transmit_info_with_vidupdate(struct sip_pvt *p);
static int transmit_message_with_text(struct sip_pvt *p, const char *text);
static int transmit_refer(struct sip_pvt *p, const char *dest);
static int transmit_notify_with_mwi(struct sip_pvt *p, int newmsgs, int oldmsgs, const char *vmexten);
static int transmit_notify_with_sipfrag(struct sip_pvt *p, int cseq, char *message, int terminate);
static int transmit_notify_custom(struct sip_pvt *p, struct ast_variable *vars);
static int transmit_register(struct sip_registry *r, int sipmethod, const char *auth, const char *authheader);
static int send_response(struct sip_pvt *p, struct sip_request *req, enum xmittype reliable, int seqno);
static int send_request(struct sip_pvt *p, struct sip_request *req, enum xmittype reliable, int seqno);
static void copy_request(struct sip_request *dst, const struct sip_request *src);
static void receive_message(struct sip_pvt *p, struct sip_request *req);
static void parse_moved_contact(struct sip_pvt *p, struct sip_request *req, char **name, char **number, int set_call_forward);
static int sip_send_mwi_to_peer(struct sip_peer *peer, const struct ast_event *event, int cache_only);
/*--- Dialog management */
static struct sip_pvt *sip_alloc(ast_string_field callid, struct sockaddr_in *sin,
int useglobal_nat, const int intended_method);
static int __sip_autodestruct(const void *data);
static void sip_scheddestroy(struct sip_pvt *p, int ms);
static int sip_cancel_destroy(struct sip_pvt *p);
static struct sip_pvt *sip_destroy(struct sip_pvt *p);
static void *dialog_unlink_all(struct sip_pvt *dialog, int lockowner, int lockdialoglist);
static void *registry_unref(struct sip_registry *reg, char *tag);
static void __sip_destroy(struct sip_pvt *p, int lockowner, int lockdialoglist);
static void __sip_ack(struct sip_pvt *p, int seqno, int resp, int sipmethod);
static void __sip_pretend_ack(struct sip_pvt *p);
static int __sip_semi_ack(struct sip_pvt *p, int seqno, int resp, int sipmethod);
static int auto_congest(const void *arg);
static int update_call_counter(struct sip_pvt *fup, int event);
static int hangup_sip2cause(int cause);
static const char *hangup_cause2sip(int cause);
static struct sip_pvt *find_call(struct sip_request *req, struct sockaddr_in *sin, const int intended_method);
static void free_old_route(struct sip_route *route);
static void list_route(struct sip_route *route);
static void build_route(struct sip_pvt *p, struct sip_request *req, int backwards);
static enum check_auth_result register_verify(struct sip_pvt *p, struct sockaddr_in *sin,
struct sip_request *req, char *uri);
static struct sip_pvt *get_sip_pvt_byid_locked(const char *callid, const char *totag, const char *fromtag);
static void check_pendings(struct sip_pvt *p);
static void *sip_park_thread(void *stuff);
static int sip_park(struct ast_channel *chan1, struct ast_channel *chan2, struct sip_request *req, int seqno);
static int sip_sipredirect(struct sip_pvt *p, const char *dest);
/*--- Codec handling / SDP */
static void try_suggested_sip_codec(struct sip_pvt *p);
static const char* get_sdp_iterate(int* start, struct sip_request *req, const char *name);
static const char *get_sdp(struct sip_request *req, const char *name);
static int find_sdp(struct sip_request *req);
static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action);
static void add_codec_to_sdp(const struct sip_pvt *p, int codec,
struct ast_str **m_buf, struct ast_str **a_buf,
int debug, int *min_packet_size);
static void add_noncodec_to_sdp(const struct sip_pvt *p, int format,
struct ast_str **m_buf, struct ast_str **a_buf,
int debug);
static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int oldsdp, int add_audio, int add_t38);
static void do_setnat(struct sip_pvt *p, int natflags);
static void stop_media_flows(struct sip_pvt *p);
/*--- Authentication stuff */
static int reply_digest(struct sip_pvt *p, struct sip_request *req, char *header, int sipmethod, char *digest, int digest_len);
static int build_reply_digest(struct sip_pvt *p, int method, char *digest, int digest_len);
static enum check_auth_result check_auth(struct sip_pvt *p, struct sip_request *req, const char *username,
const char *secret, const char *md5secret, int sipmethod,
char *uri, enum xmittype reliable, int ignore);
static enum check_auth_result check_user_full(struct sip_pvt *p, struct sip_request *req,
int sipmethod, char *uri, enum xmittype reliable,
struct sockaddr_in *sin, struct sip_peer **authpeer);
static int check_user(struct sip_pvt *p, struct sip_request *req, int sipmethod, char *uri, enum xmittype reliable, struct sockaddr_in *sin);
/*--- Domain handling */
static int check_sip_domain(const char *domain, char *context, size_t len); /* Check if domain is one of our local domains */
static int add_sip_domain(const char *domain, const enum domain_mode mode, const char *context);
static void clear_sip_domains(void);
/*--- SIP realm authentication */
static struct sip_auth *add_realm_authentication(struct sip_auth *authlist, const char *configuration, int lineno);
static int clear_realm_authentication(struct sip_auth *authlist); /* Clear realm authentication list (at reload) */
static struct sip_auth *find_realm_authentication(struct sip_auth *authlist, const char *realm);
/*--- Misc functions */
static void check_rtp_timeout(struct sip_pvt *dialog, time_t t);
static int sip_do_reload(enum channelreloadreason reason);
static int reload_config(enum channelreloadreason reason);
static int expire_register(const void *data);
static void *do_monitor(void *data);
static int restart_monitor(void);
static void peer_mailboxes_to_str(struct ast_str **mailbox_str, struct sip_peer *peer);
static struct ast_variable *copy_vars(struct ast_variable *src);
/* static int sip_addrcmp(char *name, struct sockaddr_in *sin); Support for peer matching */
static int sip_refer_allocate(struct sip_pvt *p);
static void ast_quiet_chan(struct ast_channel *chan);
static int attempt_transfer(struct sip_dual *transferer, struct sip_dual *target);
static int do_magic_pickup(struct ast_channel *channel, const char *extension, const char *context);
/*!
* \brief generic function for determining if a correct transport is being
* used to contact a peer
*
* this is done as a macro so that the "tmpl" var can be passed either a
* sip_request or a sip_peer
*/
#define check_request_transport(peer, tmpl) ({ \
int ret = 0; \
if (peer->socket.type == tmpl->socket.type) \
; \
else if (!(peer->transports & tmpl->socket.type)) {\
ast_log(LOG_ERROR, \
"'%s' is not a valid transport for '%s'. we only use '%s'! ending call.\n", \
get_transport(tmpl->socket.type), peer->name, get_transport_list(peer->transports) \
); \
ret = 1; \
} else if (peer->socket.type & SIP_TRANSPORT_TLS) { \
ast_log(LOG_WARNING, \
"peer '%s' HAS NOT USED (OR SWITCHED TO) TLS in favor of '%s' (but this was allowed in sip.conf)!\n", \
peer->name, get_transport(tmpl->socket.type) \
); \
} else { \
ast_debug(1, \
"peer '%s' has contacted us over %s even though we prefer %s.\n", \
peer->name, get_transport(tmpl->socket.type), get_transport(peer->socket.type) \
); \
}\
(ret); \
})
/*--- Device monitoring and Device/extension state/event handling */
static int cb_extensionstate(char *context, char* exten, int state, void *data);
static int sip_devicestate(void *data);
static int sip_poke_noanswer(const void *data);
static int sip_poke_peer(struct sip_peer *peer, int force);
static void sip_poke_all_peers(void);
static void sip_peer_hold(struct sip_pvt *p, int hold);
static void mwi_event_cb(const struct ast_event *, void *);
/*--- Applications, functions, CLI and manager command helpers */
static const char *sip_nat_mode(const struct sip_pvt *p);
static char *sip_show_inuse(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
static char *transfermode2str(enum transfermodes mode) attribute_const;
static const char *nat2str(int nat) attribute_const;
static int peer_status(struct sip_peer *peer, char *status, int statuslen);
static char *sip_show_sched(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
static char * _sip_show_peers(int fd, int *total, struct mansession *s, const struct message *m, int argc, const char *argv[]);
static char *sip_show_peers(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
static char *sip_show_objects(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
static void print_group(int fd, ast_group_t group, int crlf);
static const char *dtmfmode2str(int mode) attribute_const;
static int str2dtmfmode(const char *str) attribute_unused;
static const char *insecure2str(int mode) attribute_const;
static void cleanup_stale_contexts(char *new, char *old);
static void print_codec_to_cli(int fd, struct ast_codec_pref *pref);
static const char *domain_mode_to_text(const enum domain_mode mode);
static char *sip_show_domains(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
static char *_sip_show_peer(int type, int fd, struct mansession *s, const struct message *m, int argc, const char *argv[]);
static char *sip_show_peer(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
static char *_sip_qualify_peer(int type, int fd, struct mansession *s, const struct message *m, int argc, const char *argv[]);
static char *sip_qualify_peer(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
static char *sip_show_registry(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
static char *sip_unregister(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
static char *sip_show_settings(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
static char *sip_show_mwi(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
static const char *subscription_type2str(enum subscriptiontype subtype) attribute_pure;
static const struct cfsubscription_types *find_subscription_type(enum subscriptiontype subtype);
static char *complete_sip_peer(const char *word, int state, int flags2);
static char *complete_sip_registered_peer(const char *word, int state, int flags2);
static char *complete_sip_show_history(const char *line, const char *word, int pos, int state);
static char *complete_sip_show_peer(const char *line, const char *word, int pos, int state);
static char *complete_sip_unregister(const char *line, const char *word, int pos, int state);
static char *complete_sipnotify(const char *line, const char *word, int pos, int state);
static char *sip_show_channel(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
static char *sip_show_channelstats(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
static char *sip_show_history(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
static char *sip_do_debug_ip(int fd, char *arg);
static char *sip_do_debug_peer(int fd, char *arg);
static char *sip_do_debug(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
static char *sip_cli_notify(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
static char *sip_set_history(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
static int sip_dtmfmode(struct ast_channel *chan, void *data);
static int sip_addheader(struct ast_channel *chan, void *data);
static int sip_do_reload(enum channelreloadreason reason);
static char *sip_reload(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
static int acf_channel_read(struct ast_channel *chan, const char *funcname, char *preparse, char *buf, size_t buflen);
/*--- Debugging
Functions for enabling debug per IP or fully, or enabling history logging for
a SIP dialog
*/
static void sip_dump_history(struct sip_pvt *dialog); /* Dump history to debuglog at end of dialog, before destroying data */
static inline int sip_debug_test_addr(const struct sockaddr_in *addr);
static inline int sip_debug_test_pvt(struct sip_pvt *p);
/*! \brief Append to SIP dialog history
\return Always returns 0 */
#define append_history(p, event, fmt , args... ) append_history_full(p, "%-15s " fmt, event, ## args)
static void append_history_full(struct sip_pvt *p, const char *fmt, ...);
static void sip_dump_history(struct sip_pvt *dialog);
/*--- Device object handling */
static struct sip_peer *build_peer(const char *name, struct ast_variable *v, struct ast_variable *alt, int realtime);
static int update_call_counter(struct sip_pvt *fup, int event);
static void sip_destroy_peer(struct sip_peer *peer);
static void sip_destroy_peer_fn(void *peer);
static void set_peer_defaults(struct sip_peer *peer);
static struct sip_peer *temp_peer(const char *name);
static void register_peer_exten(struct sip_peer *peer, int onoff);
static struct sip_peer *find_peer(const char *peer, struct sockaddr_in *sin, int realtime, int forcenamematch, int devstate_only);
static int sip_poke_peer_s(const void *data);
static enum parse_register_result parse_register_contact(struct sip_pvt *pvt, struct sip_peer *p, struct sip_request *req);
static void reg_source_db(struct sip_peer *peer);
static void destroy_association(struct sip_peer *peer);
static void set_insecure_flags(struct ast_flags *flags, const char *value, int lineno);
static int handle_common_options(struct ast_flags *flags, struct ast_flags *mask, struct ast_variable *v);
/* Realtime device support */
static void realtime_update_peer(const char *peername, struct sockaddr_in *sin, const char *username, const char *fullcontact, const char *useragent, int expirey, int deprecated_username, int lastms);
static void update_peer(struct sip_peer *p, int expire);
static struct ast_variable *get_insecure_variable_from_config(struct ast_config *config);
static const char *get_name_from_variable(struct ast_variable *var, const char *newpeername);
static struct sip_peer *realtime_peer(const char *peername, struct sockaddr_in *sin, int devstate_only);
static char *sip_prune_realtime(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
/*--- Internal UA client handling (outbound registrations) */
static void ast_sip_ouraddrfor(struct in_addr *them, struct sockaddr_in *us);
static void sip_registry_destroy(struct sip_registry *reg);
static int sip_register(const char *value, int lineno);
static const char *regstate2str(enum sipregistrystate regstate) attribute_const;
static int sip_reregister(const void *data);
static int __sip_do_register(struct sip_registry *r);
static int sip_reg_timeout(const void *data);
static void sip_send_all_registers(void);
static int sip_reinvite_retry(const void *data);
/*--- Parsing SIP requests and responses */
static void append_date(struct sip_request *req); /* Append date to SIP packet */
static int determine_firstline_parts(struct sip_request *req);
static const struct cfsubscription_types *find_subscription_type(enum subscriptiontype subtype);
static const char *gettag(const struct sip_request *req, const char *header, char *tagbuf, int tagbufsize);
static int find_sip_method(const char *msg);
static unsigned int parse_sip_options(struct sip_pvt *pvt, const char *supported);
static int parse_request(struct sip_request *req);
static const char *get_header(const struct sip_request *req, const char *name);
static const char *referstatus2str(enum referstatus rstatus) attribute_pure;
static int method_match(enum sipmethod id, const char *name);
static void parse_copy(struct sip_request *dst, const struct sip_request *src);
static char *get_in_brackets(char *tmp);
static const char *find_alias(const char *name, const char *_default);
static const char *__get_header(const struct sip_request *req, const char *name, int *start);
static int lws2sws(char *msgbuf, int len);
static void extract_uri(struct sip_pvt *p, struct sip_request *req);
static char *remove_uri_parameters(char *uri);
static int get_refer_info(struct sip_pvt *transferer, struct sip_request *outgoing_req);
static int get_also_info(struct sip_pvt *p, struct sip_request *oreq);
static int parse_ok_contact(struct sip_pvt *pvt, struct sip_request *req);
static int set_address_from_contact(struct sip_pvt *pvt);
static void check_via(struct sip_pvt *p, struct sip_request *req);
static char *get_calleridname(const char *input, char *output, size_t outputsize);
static int get_rpid(struct sip_pvt *p, struct sip_request *oreq);
static int get_rdnis(struct sip_pvt *p, struct sip_request *oreq, char **name, char **number, int *reason);
static int get_destination(struct sip_pvt *p, struct sip_request *oreq);
static int get_msg_text(char *buf, int len, struct sip_request *req, int addnewline);
static int transmit_state_notify(struct sip_pvt *p, int state, int full, int timeout);
static void update_connectedline(struct sip_pvt *p, const void *data, size_t datalen);
static void update_redirecting(struct sip_pvt *p, const void *data, size_t datalen);
static void change_redirecting_information(struct sip_pvt *p, struct sip_request *req, struct ast_party_redirecting *redirecting, int set_call_forward);
/*-- TCP connection handling ---*/
static void *_sip_tcp_helper_thread(struct sip_pvt *pvt, struct ast_tcptls_session_instance *tcptls_session);
static void *sip_tcp_worker_fn(void *);
/*--- Constructing requests and responses */
static void initialize_initreq(struct sip_pvt *p, struct sip_request *req);
static int init_req(struct sip_request *req, int sipmethod, const char *recip);
static int reqprep(struct sip_request *req, struct sip_pvt *p, int sipmethod, int seqno, int newbranch);
static void initreqprep(struct sip_request *req, struct sip_pvt *p, int sipmethod);
static int init_resp(struct sip_request *resp, const char *msg);
static inline int resp_needs_contact(const char *msg, enum sipmethod method);
static int respprep(struct sip_request *resp, struct sip_pvt *p, const char *msg, const struct sip_request *req);
static const struct sockaddr_in *sip_real_dst(const struct sip_pvt *p);
static void build_via(struct sip_pvt *p);
static int create_addr_from_peer(struct sip_pvt *r, struct sip_peer *peer);
static int create_addr(struct sip_pvt *dialog, const char *opeer, struct sockaddr_in *sin, int newdialog);
static char *generate_random_string(char *buf, size_t size);
static void build_callid_pvt(struct sip_pvt *pvt);
static void build_callid_registry(struct sip_registry *reg, struct in_addr ourip, const char *fromdomain);
static void make_our_tag(char *tagbuf, size_t len);
static int add_header(struct sip_request *req, const char *var, const char *value);
static int add_header_contentLength(struct sip_request *req, int len);
static int add_line(struct sip_request *req, const char *line);
static int add_text(struct sip_request *req, const char *text);
static int add_digit(struct sip_request *req, char digit, unsigned int duration, int mode);
static int add_rpid(struct sip_request *req, struct sip_pvt *p);
static int add_vidupdate(struct sip_request *req);
static void add_route(struct sip_request *req, struct sip_route *route);
static int copy_header(struct sip_request *req, const struct sip_request *orig, const char *field);
static int copy_all_header(struct sip_request *req, const struct sip_request *orig, const char *field);
static int copy_via_headers(struct sip_pvt *p, struct sip_request *req, const struct sip_request *orig, const char *field);
static void set_destination(struct sip_pvt *p, char *uri);
static void append_date(struct sip_request *req);
static void build_contact(struct sip_pvt *p);
/*------Request handling functions */
static int handle_incoming(struct sip_pvt *p, struct sip_request *req, struct sockaddr_in *sin, int *recount, int *nounlock);
static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int debug, int seqno, struct sockaddr_in *sin, int *recount, char *e, int *nounlock);
static int handle_request_refer(struct sip_pvt *p, struct sip_request *req, int debug, int seqno, int *nounlock);
static int handle_request_bye(struct sip_pvt *p, struct sip_request *req);
static int handle_request_register(struct sip_pvt *p, struct sip_request *req, struct sockaddr_in *sin, char *e);
static int handle_request_cancel(struct sip_pvt *p, struct sip_request *req);
static int handle_request_message(struct sip_pvt *p, struct sip_request *req);
static int handle_request_subscribe(struct sip_pvt *p, struct sip_request *req, struct sockaddr_in *sin, int seqno, char *e);
static void handle_request_info(struct sip_pvt *p, struct sip_request *req);
static int handle_request_options(struct sip_pvt *p, struct sip_request *req);
static int handle_invite_replaces(struct sip_pvt *p, struct sip_request *req, int debug, int seqno, struct sockaddr_in *sin);
static int handle_request_notify(struct sip_pvt *p, struct sip_request *req, struct sockaddr_in *sin, int seqno, char *e);
static int local_attended_transfer(struct sip_pvt *transferer, struct sip_dual *current, struct sip_request *req, int seqno);
/*------Response handling functions */
static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int seqno);
static void handle_response_notify(struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int seqno);
static void handle_response_refer(struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int seqno);
static void handle_response_subscribe(struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int seqno);
static int handle_response_register(struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int seqno);
static void handle_response(struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int seqno);
/*------ T38 Support --------- */
static int transmit_response_with_t38_sdp(struct sip_pvt *p, char *msg, struct sip_request *req, int retrans);
static struct ast_udptl *sip_get_udptl_peer(struct ast_channel *chan);
static int sip_set_udptl_peer(struct ast_channel *chan, struct ast_udptl *udptl);
static void change_t38_state(struct sip_pvt *p, int state);
/*------ Session-Timers functions --------- */
static void proc_422_rsp(struct sip_pvt *p, struct sip_request *rsp);
static int proc_session_timer(const void *vp);
static void stop_session_timer(struct sip_pvt *p);
static void start_session_timer(struct sip_pvt *p);
static void restart_session_timer(struct sip_pvt *p);
static const char *strefresher2str(enum st_refresher r);
static int parse_session_expires(const char *p_hdrval, int *const p_interval, enum st_refresher *const p_ref);
static int parse_minse(const char *p_hdrval, int *const p_interval);
static int st_get_se(struct sip_pvt *, int max);
static enum st_refresher st_get_refresher(struct sip_pvt *);
static enum st_mode st_get_mode(struct sip_pvt *);
static struct sip_st_dlg* sip_st_alloc(struct sip_pvt *const p);
/*------- RTP Glue functions -------- */
static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance *instance, struct ast_rtp_instance *vinstance, struct ast_rtp_instance *tinstance, int codecs, int nat_active);
/*!--- SIP MWI Subscription support */
static int sip_subscribe_mwi(const char *value, int lineno);
static void sip_subscribe_mwi_destroy(struct sip_subscription_mwi *mwi);
static void sip_send_all_mwi_subscriptions(void);
static int sip_subscribe_mwi_do(const void *data);
static int __sip_subscribe_mwi_do(struct sip_subscription_mwi *mwi);
/*! \brief Definition of this channel for PBX channel registration */
static const struct ast_channel_tech sip_tech = {
.type = "SIP",
.description = "Session Initiation Protocol (SIP)",
.capabilities = AST_FORMAT_AUDIO_MASK, /* all audio formats */
.properties = AST_CHAN_TP_WANTSJITTER | AST_CHAN_TP_CREATESJITTER,
.requester = sip_request_call, /* called with chan unlocked */
.devicestate = sip_devicestate, /* called with chan unlocked (not chan-specific) */
.call = sip_call, /* called with chan locked */
.send_html = sip_sendhtml,
.hangup = sip_hangup, /* called with chan locked */
.answer = sip_answer, /* called with chan locked */
.read = sip_read, /* called with chan locked */
.write = sip_write, /* called with chan locked */
.write_video = sip_write, /* called with chan locked */
.write_text = sip_write,
.indicate = sip_indicate, /* called with chan locked */
.transfer = sip_transfer, /* called with chan locked */
.fixup = sip_fixup, /* called with chan locked */
.send_digit_begin = sip_senddigit_begin, /* called with chan unlocked */
.send_digit_end = sip_senddigit_end,
.bridge = ast_rtp_instance_bridge, /* XXX chan unlocked ? */
.early_bridge = ast_rtp_instance_early_bridge,
.send_text = sip_sendtext, /* called with chan locked */
.func_channel_read = acf_channel_read,
.setoption = sip_setoption,
.queryoption = sip_queryoption,
.get_pvt_uniqueid = sip_get_callid,
};
/*! \brief This version of the sip channel tech has no send_digit_begin
* callback so that the core knows that the channel does not want
* DTMF BEGIN frames.
* The struct is initialized just before registering the channel driver,
* and is for use with channels using SIP INFO DTMF.
*/
static struct ast_channel_tech sip_tech_info;
/*! \brief Working TLS connection configuration */
static struct ast_tls_config sip_tls_cfg;
/*! \brief Default TLS connection configuration */
static struct ast_tls_config default_tls_cfg;
/*! \brief The TCP server definition */
static struct ast_tcptls_session_args sip_tcp_desc = {
.accept_fd = -1,
.master = AST_PTHREADT_NULL,
.tls_cfg = NULL,
.poll_timeout = -1,
.name = "SIP TCP server",
.accept_fn = ast_tcptls_server_root,
.worker_fn = sip_tcp_worker_fn,
};
/*! \brief The TCP/TLS server definition */
static struct ast_tcptls_session_args sip_tls_desc = {
.accept_fd = -1,
.master = AST_PTHREADT_NULL,
.tls_cfg = &sip_tls_cfg,
.poll_timeout = -1,
.name = "SIP TLS server",
.accept_fn = ast_tcptls_server_root,
.worker_fn = sip_tcp_worker_fn,
};
/* wrapper macro to tell whether t points to one of the sip_tech descriptors */
#define IS_SIP_TECH(t) ((t) == &sip_tech || (t) == &sip_tech_info)
/*! \brief map from an integer value to a string.
* If no match is found, return errorstring
*/
static const char *map_x_s(const struct _map_x_s *table, int x, const char *errorstring)
{
const struct _map_x_s *cur;
for (cur = table; cur->s; cur++)
if (cur->x == x)
return cur->s;
return errorstring;
}
/*! \brief map from a string to an integer value, case insensitive.
* If no match is found, return errorvalue.
*/
static int map_s_x(const struct _map_x_s *table, const char *s, int errorvalue)
{
const struct _map_x_s *cur;
for (cur = table; cur->s; cur++)
if (!strcasecmp(cur->s, s))
return cur->x;
return errorvalue;
}
/*!
* duplicate a list of channel variables, \return the copy.
*/
static struct ast_variable *copy_vars(struct ast_variable *src)
{
struct ast_variable *res = NULL, *tmp, *v = NULL;
for (v = src ; v ; v = v->next) {
if ((tmp = ast_variable_new(v->name, v->value, v->file))) {
tmp->next = res;
res = tmp;
}
}
return res;
}
/*! \brief SIP TCP connection handler */
static void *sip_tcp_worker_fn(void *data)
{
struct ast_tcptls_session_instance *tcptls_session = data;
return _sip_tcp_helper_thread(NULL, tcptls_session);
}
/*! \brief SIP TCP thread management function
This function reads from the socket, parses the packet into a request
*/
static void *_sip_tcp_helper_thread(struct sip_pvt *pvt, struct ast_tcptls_session_instance *tcptls_session)
{
int res, cl;
struct sip_request req = { 0, } , reqcpy = { 0, };
struct sip_threadinfo *me;
char buf[1024] = "";
me = ast_calloc(1, sizeof(*me));
if (!me)
goto cleanup2;
me->threadid = pthread_self();
me->tcptls_session = tcptls_session;
if (tcptls_session->ssl)
me->type = SIP_TRANSPORT_TLS;
else
me->type = SIP_TRANSPORT_TCP;
ast_debug(2, "Starting thread for %s server\n", tcptls_session->ssl ? "SSL" : "TCP");
AST_LIST_LOCK(&threadl);
AST_LIST_INSERT_TAIL(&threadl, me, list);
AST_LIST_UNLOCK(&threadl);
if (!(req.data = ast_str_create(SIP_MIN_PACKET)))
goto cleanup;
if (!(reqcpy.data = ast_str_create(SIP_MIN_PACKET)))
goto cleanup;
for (;;) {
struct ast_str *str_save;
str_save = req.data;
memset(&req, 0, sizeof(req));
req.data = str_save;
ast_str_reset(req.data);
str_save = reqcpy.data;
memset(&reqcpy, 0, sizeof(reqcpy));
reqcpy.data = str_save;
ast_str_reset(reqcpy.data);
req.socket.fd = tcptls_session->fd;
if (tcptls_session->ssl) {
req.socket.type = SIP_TRANSPORT_TLS;
req.socket.port = htons(ourport_tls);
} else {
req.socket.type = SIP_TRANSPORT_TCP;
req.socket.port = htons(ourport_tcp);
}
res = ast_wait_for_input(tcptls_session->fd, -1);
if (res < 0) {
ast_debug(2, "SIP %s server :: ast_wait_for_input returned %d\n", tcptls_session->ssl ? "SSL": "TCP", res);
goto cleanup;
}
/* Read in headers one line at a time */
while (req.len < 4 || strncmp(REQ_OFFSET_TO_STR(&req, len - 4), "\r\n\r\n", 4)) {
ast_mutex_lock(&tcptls_session->lock);
if (!fgets(buf, sizeof(buf), tcptls_session->f)) {
ast_mutex_unlock(&tcptls_session->lock);
goto cleanup;
}
ast_mutex_unlock(&tcptls_session->lock);
if (me->stop)
goto cleanup;
ast_str_append(&req.data, 0, "%s", buf);
req.len = req.data->used;
}
copy_request(&reqcpy, &req);
parse_request(&reqcpy);
/* In order to know how much to read, we need the content-length header */
if (sscanf(get_header(&reqcpy, "Content-Length"), "%d", &cl)) {
while (cl > 0) {
ast_mutex_lock(&tcptls_session->lock);
if (!fread(buf, (cl < sizeof(buf)) ? cl : sizeof(buf), 1, tcptls_session->f)) {
ast_mutex_unlock(&tcptls_session->lock);
goto cleanup;
}
ast_mutex_unlock(&tcptls_session->lock);
if (me->stop)
goto cleanup;
cl -= strlen(buf);
ast_str_append(&req.data, 0, "%s", buf);
req.len = req.data->used;
}
}
/*! \todo XXX If there's no Content-Length or if the content-length and what
we receive is not the same - we should generate an error */
req.socket.tcptls_session = tcptls_session;
handle_request_do(&req, &tcptls_session->remote_address);
}
cleanup:
AST_LIST_LOCK(&threadl);
AST_LIST_REMOVE(&threadl, me, list);
AST_LIST_UNLOCK(&threadl);
ast_free(me);
cleanup2:
fclose(tcptls_session->f);
tcptls_session->f = NULL;
tcptls_session->fd = -1;
if (reqcpy.data) {
ast_free(reqcpy.data);
}
if (req.data) {
ast_free(req.data);
req.data = NULL;
}
ast_debug(2, "Shutting down thread for %s server\n", tcptls_session->ssl ? "SSL" : "TCP");
ao2_ref(tcptls_session, -1);
tcptls_session = NULL;
return NULL;
}
/*!
* helper functions to unreference various types of objects.
* By handling them this way, we don't have to declare the
* destructor on each call, which removes the chance of errors.
*/
static void *unref_peer(struct sip_peer *peer, char *tag)
{
ao2_t_ref(peer, -1, tag);
return NULL;
}
static struct sip_peer *ref_peer(struct sip_peer *peer, char *tag)
{
ao2_t_ref(peer, 1, tag);
return peer;
}
/*! \brief maintain proper refcounts for a sip_pvt's outboundproxy
*
* This function sets pvt's outboundproxy pointer to the one referenced
* by the proxy parameter. Because proxy may be a refcounted object, and
* because pvt's old outboundproxy may also be a refcounted object, we need
* to maintain the proper refcounts.
*
* \param pvt The sip_pvt for which we wish to set the outboundproxy
* \param proxy The sip_proxy which we will point pvt towards.
* \return Returns void
*/
static void ref_proxy(struct sip_pvt *pvt, struct sip_proxy *proxy)
{
struct sip_proxy *old_obproxy = pvt->outboundproxy;
/* The sip_cfg.outboundproxy is statically allocated, and so
* we don't ever need to adjust refcounts for it
*/
if (proxy && proxy != &sip_cfg.outboundproxy) {
ao2_ref(proxy, +1);
}
pvt->outboundproxy = proxy;
if (old_obproxy && old_obproxy != &sip_cfg.outboundproxy) {
ao2_ref(old_obproxy, -1);
}
}
/*!
* \brief Unlink a dialog from the dialogs container, as well as any other places
* that it may be currently stored.
*
* \note A reference to the dialog must be held before calling this function, and this
* function does not release that reference.
*/
static void *dialog_unlink_all(struct sip_pvt *dialog, int lockowner, int lockdialoglist)
{
struct sip_pkt *cp;
dialog_ref(dialog, "Let's bump the count in the unlink so it doesn't accidentally become dead before we are done");
ao2_t_unlink(dialogs, dialog, "unlinking dialog via ao2_unlink");
/* Unlink us from the owner (channel) if we have one */
if (dialog->owner) {
if (lockowner)
ast_channel_lock(dialog->owner);
ast_debug(1, "Detaching from channel %s\n", dialog->owner->name);
dialog->owner->tech_pvt = dialog_unref(dialog->owner->tech_pvt, "resetting channel dialog ptr in unlink_all");
if (lockowner)
ast_channel_unlock(dialog->owner);
}
if (dialog->registry) {
if (dialog->registry->call == dialog)
dialog->registry->call = dialog_unref(dialog->registry->call, "nulling out the registry's call dialog field in unlink_all");
dialog->registry = registry_unref(dialog->registry, "delete dialog->registry");
}
if (dialog->stateid > -1) {
ast_extension_state_del(dialog->stateid, NULL);
dialog_unref(dialog, "removing extension_state, should unref the associated dialog ptr that was stored there.");
dialog->stateid = -1; /* shouldn't we 'zero' this out? */
}
/* Remove link from peer to subscription of MWI */
if (dialog->relatedpeer && dialog->relatedpeer->mwipvt == dialog)
dialog->relatedpeer->mwipvt = dialog_unref(dialog->relatedpeer->mwipvt, "delete ->relatedpeer->mwipvt");
if (dialog->relatedpeer && dialog->relatedpeer->call == dialog)
dialog->relatedpeer->call = dialog_unref(dialog->relatedpeer->call, "unset the relatedpeer->call field in tandem with relatedpeer field itself");
/* remove all current packets in this dialog */
while((cp = dialog->packets)) {
dialog->packets = dialog->packets->next;
AST_SCHED_DEL(sched, cp->retransid);
dialog_unref(cp->owner, "remove all current packets in this dialog, and the pointer to the dialog too as part of __sip_destroy");
ast_free(cp);
}
AST_SCHED_DEL_UNREF(sched, dialog->waitid, dialog_unref(dialog, "when you delete the waitid sched, you should dec the refcount for the stored dialog ptr"));
AST_SCHED_DEL_UNREF(sched, dialog->initid, dialog_unref(dialog, "when you delete the initid sched, you should dec the refcount for the stored dialog ptr"));
if (dialog->autokillid > -1)
AST_SCHED_DEL_UNREF(sched, dialog->autokillid, dialog_unref(dialog, "when you delete the autokillid sched, you should dec the refcount for the stored dialog ptr"));
if (dialog->request_queue_sched_id > -1) {
AST_SCHED_DEL_UNREF(sched, dialog->request_queue_sched_id, dialog_unref(dialog, "when you delete the request_queue_sched_id sched, you should dec the refcount for the stored dialog ptr"));
}
if (dialog->t38id > -1) {
AST_SCHED_DEL_UNREF(sched, dialog->t38id, dialog_unref(dialog, "when you delete the t38id sched, you should dec the refcount for the stored dialog ptr"));
}
dialog_unref(dialog, "Let's unbump the count in the unlink so the poor pvt can disappear if it is time");
return NULL;
}
static void *registry_unref(struct sip_registry *reg, char *tag)
{
ast_debug(3, "SIP Registry %s: refcount now %d\n", reg->hostname, reg->refcount - 1);
ASTOBJ_UNREF(reg, sip_registry_destroy);
return NULL;
}
/*! \brief Add object reference to SIP registry */
static struct sip_registry *registry_addref(struct sip_registry *reg, char *tag)
{
ast_debug(3, "SIP Registry %s: refcount now %d\n", reg->hostname, reg->refcount + 1);
return ASTOBJ_REF(reg); /* Add pointer to registry in packet */
}
/*! \brief Interface structure with callbacks used to connect to UDPTL module*/
static struct ast_udptl_protocol sip_udptl = {
type: "SIP",
get_udptl_info: sip_get_udptl_peer,
set_udptl_peer: sip_set_udptl_peer,
};
static void append_history_full(struct sip_pvt *p, const char *fmt, ...)
__attribute__((format(printf, 2, 3)));
/*! \brief Convert transfer status to string */
static const char *referstatus2str(enum referstatus rstatus)
{
return map_x_s(referstatusstrings, rstatus, "");
}
static inline void pvt_set_needdestroy(struct sip_pvt *pvt, const char *reason)
{
append_history(pvt, "NeedDestroy", "Setting needdestroy because %s", reason);
pvt->needdestroy = 1;
}
/*! \brief Initialize the initital request packet in the pvt structure.
This packet is used for creating replies and future requests in
a dialog */
static void initialize_initreq(struct sip_pvt *p, struct sip_request *req)
{
if (p->initreq.headers)
ast_debug(1, "Initializing already initialized SIP dialog %s (presumably reinvite)\n", p->callid);
else
ast_debug(1, "Initializing initreq for method %s - callid %s\n", sip_methods[req->method].text, p->callid);
/* Use this as the basis */
copy_request(&p->initreq, req);
parse_request(&p->initreq);
if (req->debug)
ast_verbose("Initreq: %d headers, %d lines\n", p->initreq.headers, p->initreq.lines);
}
/*! \brief Encapsulate setting of SIP_ALREADYGONE to be able to trace it with debugging */
static void sip_alreadygone(struct sip_pvt *dialog)
{
ast_debug(3, "Setting SIP_ALREADYGONE on dialog %s\n", dialog->callid);
dialog->alreadygone = 1;
}
/*! Resolve DNS srv name or host name in a sip_proxy structure */
static int proxy_update(struct sip_proxy *proxy)
{
/* if it's actually an IP address and not a name,
there's no need for a managed lookup */
if (!inet_aton(proxy->name, &proxy->ip.sin_addr)) {
/* Ok, not an IP address, then let's check if it's a domain or host */
/* XXX Todo - if we have proxy port, don't do SRV */
if (ast_get_ip_or_srv(&proxy->ip, proxy->name, sip_cfg.srvlookup ? "_sip._udp" : NULL) < 0) {
ast_log(LOG_WARNING, "Unable to locate host '%s'\n", proxy->name);
return FALSE;
}
}
proxy->last_dnsupdate = time(NULL);
return TRUE;
}
/*! \brief Allocate and initialize sip proxy */
static struct sip_proxy *proxy_allocate(char *name, char *port, int force)
{
struct sip_proxy *proxy;
proxy = ao2_alloc(sizeof(*proxy), NULL);
if (!proxy)
return NULL;
proxy->force = force;
ast_copy_string(proxy->name, name, sizeof(proxy->name));
proxy->ip.sin_port = htons((!ast_strlen_zero(port) ? atoi(port) : STANDARD_SIP_PORT));
proxy_update(proxy);
return proxy;
}
/*! \brief Get default outbound proxy or global proxy */
static struct sip_proxy *obproxy_get(struct sip_pvt *dialog, struct sip_peer *peer)
{
if (peer && peer->outboundproxy) {
if (sipdebug)
ast_debug(1, "OBPROXY: Applying peer OBproxy to this call\n");
append_history(dialog, "OBproxy", "Using peer obproxy %s", peer->outboundproxy->name);
return peer->outboundproxy;
}
if (sip_cfg.outboundproxy.name[0]) {
if (sipdebug)
ast_debug(1, "OBPROXY: Applying global OBproxy to this call\n");
append_history(dialog, "OBproxy", "Using global obproxy %s", sip_cfg.outboundproxy.name);
return &sip_cfg.outboundproxy;
}
if (sipdebug)
ast_debug(1, "OBPROXY: Not applying OBproxy to this call\n");
return NULL;
}
/*! \brief returns true if 'name' (with optional trailing whitespace)
* matches the sip method 'id'.
* Strictly speaking, SIP methods are case SENSITIVE, but we do
* a case-insensitive comparison to be more tolerant.
* following Jon Postel's rule: Be gentle in what you accept, strict with what you send
*/
static int method_match(enum sipmethod id, const char *name)
{
int len = strlen(sip_methods[id].text);
int l_name = name ? strlen(name) : 0;
/* true if the string is long enough, and ends with whitespace, and matches */
return (l_name >= len && name[len] < 33 &&
!strncasecmp(sip_methods[id].text, name, len));
}
/*! \brief find_sip_method: Find SIP method from header */
static int find_sip_method(const char *msg)
{
int i, res = 0;
if (ast_strlen_zero(msg))
return 0;
for (i = 1; i < ARRAY_LEN(sip_methods) && !res; i++) {
if (method_match(i, msg))
res = sip_methods[i].id;
}
return res;
}
/*! \brief Parse supported header in incoming packet */
static unsigned int parse_sip_options(struct sip_pvt *pvt, const char *supported)
{
char *next, *sep;
char *temp;
unsigned int profile = 0;
int i, found;
if (ast_strlen_zero(supported) )
return 0;
temp = ast_strdupa(supported);
if (sipdebug)
ast_debug(3, "Begin: parsing SIP \"Supported: %s\"\n", supported);
for (next = temp; next; next = sep) {
found = FALSE;
if ( (sep = strchr(next, ',')) != NULL)
*sep++ = '\0';
next = ast_skip_blanks(next);
if (sipdebug)
ast_debug(3, "Found SIP option: -%s-\n", next);
for (i = 0; i < ARRAY_LEN(sip_options); i++) {
if (!strcasecmp(next, sip_options[i].text)) {
profile |= sip_options[i].id;
found = TRUE;
if (sipdebug)
ast_debug(3, "Matched SIP option: %s\n", next);
break;
}
}
/* This function is used to parse both Suported: and Require: headers.
Let the caller of this function know that an unknown option tag was
encountered, so that if the UAC requires it then the request can be
rejected with a 420 response. */
if (!found)
profile |= SIP_OPT_UNKNOWN;
if (!found && sipdebug) {
if (!strncasecmp(next, "x-", 2))
ast_debug(3, "Found private SIP option, not supported: %s\n", next);
else
ast_debug(3, "Found no match for SIP option: %s (Please file bug report!)\n", next);
}
}
if (pvt)
pvt->sipoptions = profile;
return profile;
}
/*! \brief See if we pass debug IP filter */
static inline int sip_debug_test_addr(const struct sockaddr_in *addr)
{
if (!sipdebug)
return 0;
if (debugaddr.sin_addr.s_addr) {
if (((ntohs(debugaddr.sin_port) != 0)
&& (debugaddr.sin_port != addr->sin_port))
|| (debugaddr.sin_addr.s_addr != addr->sin_addr.s_addr))
return 0;
}
return 1;
}
/*! \brief The real destination address for a write */
static const struct sockaddr_in *sip_real_dst(const struct sip_pvt *p)
{
if (p->outboundproxy)
return &p->outboundproxy->ip;
return ast_test_flag(&p->flags[0], SIP_NAT) & SIP_NAT_ROUTE ? &p->recv : &p->sa;
}
/*! \brief Display SIP nat mode */
static const char *sip_nat_mode(const struct sip_pvt *p)
{
return ast_test_flag(&p->flags[0], SIP_NAT) & SIP_NAT_ROUTE ? "NAT" : "no NAT";
}
/*! \brief Test PVT for debugging output */
static inline int sip_debug_test_pvt(struct sip_pvt *p)
{
if (!sipdebug)
return 0;
return sip_debug_test_addr(sip_real_dst(p));
}
/*! \brief Return configuration of transports for a device */
static inline const char *get_transport_list(unsigned int transports) {
switch (transports) {
case SIP_TRANSPORT_UDP:
return "UDP";
case SIP_TRANSPORT_TCP:
return "TCP";
case SIP_TRANSPORT_TLS:
return "TLS";
case SIP_TRANSPORT_UDP | SIP_TRANSPORT_TCP:
return "TCP,UDP";
case SIP_TRANSPORT_UDP | SIP_TRANSPORT_TLS:
return "TLS,UDP";
case SIP_TRANSPORT_TCP | SIP_TRANSPORT_TLS:
return "TLS,TCP";
default:
return transports ?
"TLS,TCP,UDP" : "UNKNOWN";
}
}
/*! \brief Return transport as string */
static inline const char *get_transport(enum sip_transport t)
{
switch (t) {
case SIP_TRANSPORT_UDP:
return "UDP";
case SIP_TRANSPORT_TCP:
return "TCP";
case SIP_TRANSPORT_TLS:
return "TLS";
}
return "UNKNOWN";
}
/*! \brief Return transport of dialog.
\note this is based on a false assumption. We don't always use the
outbound proxy for all requests in a dialog. It depends on the
"force" parameter. The FIRST request is always sent to the ob proxy.
\todo Fix this function to work correctly
*/
static inline const char *get_transport_pvt(struct sip_pvt *p)
{
if (p->outboundproxy && p->outboundproxy->transport)
p->socket.type = p->outboundproxy->transport;
return get_transport(p->socket.type);
}
/*! \brief Transmit SIP message
Sends a SIP request or response on a given socket (in the pvt)
Called by retrans_pkt, send_request, send_response and
__sip_reliable_xmit
*/
static int __sip_xmit(struct sip_pvt *p, struct ast_str *data, int len)
{
int res = 0;
const struct sockaddr_in *dst = sip_real_dst(p);
ast_debug(2, "Trying to put '%.10s' onto %s socket destined for %s:%d\n", data->str, get_transport_pvt(p), ast_inet_ntoa(dst->sin_addr), htons(dst->sin_port));
if (sip_prepare_socket(p) < 0)
return XMIT_ERROR;
if (p->socket.tcptls_session)
ast_mutex_lock(&p->socket.tcptls_session->lock);
if (p->socket.type & SIP_TRANSPORT_UDP) {
res = sendto(p->socket.fd, data->str, len, 0, (const struct sockaddr *)dst, sizeof(struct sockaddr_in));
} else if (p->socket.tcptls_session) {
if (p->socket.tcptls_session->f) {
res = ast_tcptls_server_write(p->socket.tcptls_session, data->str, len);
} else {
ast_debug(2, "No p->socket.tcptls_session->f len=%d\n", len);
}
} else {
ast_debug(2, "Socket type is TCP but no tcptls_session is present to write to\n");
return XMIT_ERROR;
}
if (p->socket.tcptls_session)
ast_mutex_unlock(&p->socket.tcptls_session->lock);
if (res == -1) {
switch (errno) {
case EBADF: /* Bad file descriptor - seems like this is generated when the host exist, but doesn't accept the UDP packet */
case EHOSTUNREACH: /* Host can't be reached */
case ENETDOWN: /* Inteface down */
case ENETUNREACH: /* Network failure */
case ECONNREFUSED: /* ICMP port unreachable */
res = XMIT_ERROR; /* Don't bother with trying to transmit again */
}
}
if (res != len)
ast_log(LOG_WARNING, "sip_xmit of %p (len %d) to %s:%d returned %d: %s\n", data, len, ast_inet_ntoa(dst->sin_addr), ntohs(dst->sin_port), res, strerror(errno));
return res;
}
/*! \brief Build a Via header for a request */
static void build_via(struct sip_pvt *p)
{
/* Work around buggy UNIDEN UIP200 firmware */
const char *rport = ast_test_flag(&p->flags[0], SIP_NAT) & SIP_NAT_RFC3581 ? ";rport" : "";
/* z9hG4bK is a magic cookie. See RFC 3261 section 8.1.1.7 */
snprintf(p->via, sizeof(p->via), "SIP/2.0/%s %s:%d;branch=z9hG4bK%08x%s",
get_transport_pvt(p),
ast_inet_ntoa(p->ourip.sin_addr),
ntohs(p->ourip.sin_port), (int) p->branch, rport);
}
/*! \brief NAT fix - decide which IP address to use for Asterisk server?
*
* Using the localaddr structure built up with localnet statements in sip.conf
* apply it to their address to see if we need to substitute our
* externip or can get away with our internal bindaddr
* 'us' is always overwritten.
*/
static void ast_sip_ouraddrfor(struct in_addr *them, struct sockaddr_in *us)
{
struct sockaddr_in theirs;
/* Set want_remap to non-zero if we want to remap 'us' to an externally
* reachable IP address and port. This is done if:
* 1. we have a localaddr list (containing 'internal' addresses marked
* as 'deny', so ast_apply_ha() will return AST_SENSE_DENY on them,
* and AST_SENSE_ALLOW on 'external' ones);
* 2. either stunaddr or externip is set, so we know what to use as the
* externally visible address;
* 3. the remote address, 'them', is external;
* 4. the address returned by ast_ouraddrfor() is 'internal' (AST_SENSE_DENY
* when passed to ast_apply_ha() so it does need to be remapped.
* This fourth condition is checked later.
*/
int want_remap;
*us = internip; /* starting guess for the internal address */
/* now ask the system what would it use to talk to 'them' */
ast_ouraddrfor(them, &us->sin_addr);
theirs.sin_addr = *them;
want_remap = localaddr &&
(externip.sin_addr.s_addr || stunaddr.sin_addr.s_addr) &&
ast_apply_ha(localaddr, &theirs) == AST_SENSE_ALLOW ;
if (want_remap &&
(!sip_cfg.matchexterniplocally || !ast_apply_ha(localaddr, us)) ) {
/* if we used externhost or stun, see if it is time to refresh the info */
if (externexpire && time(NULL) >= externexpire) {
if (stunaddr.sin_addr.s_addr) {
ast_stun_request(sipsock, &stunaddr, NULL, &externip);
} else {
if (ast_parse_arg(externhost, PARSE_INADDR, &externip))
ast_log(LOG_NOTICE, "Warning: Re-lookup of '%s' failed!\n", externhost);
}
externexpire = time(NULL) + externrefresh;
}
if (externip.sin_addr.s_addr)
*us = externip;
else
ast_log(LOG_WARNING, "stun failed\n");
ast_debug(1, "Target address %s is not local, substituting externip\n",
ast_inet_ntoa(*(struct in_addr *)&them->s_addr));
} else if (bindaddr.sin_addr.s_addr) {
/* no remapping, but we bind to a specific address, so use it. */
*us = bindaddr;
}
}
/*! \brief Append to SIP dialog history with arg list */
static __attribute__((format(printf, 2, 0))) void append_history_va(struct sip_pvt *p, const char *fmt, va_list ap)
{
char buf[80], *c = buf; /* max history length */
struct sip_history *hist;
int l;
vsnprintf(buf, sizeof(buf), fmt, ap);
strsep(&c, "\r\n"); /* Trim up everything after \r or \n */
l = strlen(buf) + 1;
if (!(hist = ast_calloc(1, sizeof(*hist) + l)))
return;
if (!p->history && !(p->history = ast_calloc(1, sizeof(*p->history)))) {
ast_free(hist);
return;
}
memcpy(hist->event, buf, l);
if (p->history_entries == MAX_HISTORY_ENTRIES) {
struct sip_history *oldest;
oldest = AST_LIST_REMOVE_HEAD(p->history, list);
p->history_entries--;
ast_free(oldest);
}
AST_LIST_INSERT_TAIL(p->history, hist, list);
p->history_entries++;
}
/*! \brief Append to SIP dialog history with arg list */
static void append_history_full(struct sip_pvt *p, const char *fmt, ...)
{
va_list ap;
if (!p)
return;
if (!p->do_history && !recordhistory && !dumphistory)
return;
va_start(ap, fmt);
append_history_va(p, fmt, ap);
va_end(ap);
return;
}
/*! \brief Retransmit SIP message if no answer (Called from scheduler) */
static int retrans_pkt(const void *data)
{
struct sip_pkt *pkt = (struct sip_pkt *)data, *prev, *cur = NULL;
int reschedule = DEFAULT_RETRANS;
int xmitres = 0;
/* Lock channel PVT */
sip_pvt_lock(pkt->owner);
if (pkt->retrans < MAX_RETRANS) {
pkt->retrans++;
if (!pkt->timer_t1) { /* Re-schedule using timer_a and timer_t1 */
if (sipdebug)
ast_debug(4, "SIP TIMER: Not rescheduling id #%d:%s (Method %d) (No timer T1)\n", pkt->retransid, sip_methods[pkt->method].text, pkt->method);
} else {
int siptimer_a;
if (sipdebug)
ast_debug(4, "SIP TIMER: Rescheduling retransmission #%d (%d) %s - %d\n", pkt->retransid, pkt->retrans, sip_methods[pkt->method].text, pkt->method);
if (!pkt->timer_a)
pkt->timer_a = 2 ;
else
pkt->timer_a = 2 * pkt->timer_a;
/* For non-invites, a maximum of 4 secs */
siptimer_a = pkt->timer_t1 * pkt->timer_a; /* Double each time */
if (pkt->method != SIP_INVITE && siptimer_a > 4000)
siptimer_a = 4000;
/* Reschedule re-transmit */
reschedule = siptimer_a;
ast_debug(4, "** SIP timers: Rescheduling retransmission %d to %d ms (t1 %d ms (Retrans id #%d)) \n", pkt->retrans +1, siptimer_a, pkt->timer_t1, pkt->retransid);
}
if (sip_debug_test_pvt(pkt->owner)) {
const struct sockaddr_in *dst = sip_real_dst(pkt->owner);
ast_verbose("Retransmitting #%d (%s) to %s:%d:\n%s\n---\n",
pkt->retrans, sip_nat_mode(pkt->owner),
ast_inet_ntoa(dst->sin_addr),
ntohs(dst->sin_port), pkt->data->str);
}
append_history(pkt->owner, "ReTx", "%d %s", reschedule, pkt->data->str);
xmitres = __sip_xmit(pkt->owner, pkt->data, pkt->packetlen);
sip_pvt_unlock(pkt->owner);
if (xmitres == XMIT_ERROR)
ast_log(LOG_WARNING, "Network error on retransmit in dialog %s\n", pkt->owner->callid);
else
return reschedule;
}
/* Too many retries */
if (pkt->owner && pkt->method != SIP_OPTIONS && xmitres == 0) {
if (pkt->is_fatal || sipdebug) /* Tell us if it's critical or if we're debugging */
ast_log(LOG_WARNING, "Maximum retries exceeded on transmission %s for seqno %d (%s %s) -- See doc/sip-retransmit.txt.\n",
pkt->owner->callid, pkt->seqno,
pkt->is_fatal ? "Critical" : "Non-critical", pkt->is_resp ? "Response" : "Request");
} else if (pkt->method == SIP_OPTIONS && sipdebug) {
ast_log(LOG_WARNING, "Cancelling retransmit of OPTIONs (call id %s) -- See doc/sip-retransmit.txt.\n", pkt->owner->callid);
}
if (xmitres == XMIT_ERROR) {
ast_log(LOG_WARNING, "Transmit error :: Cancelling transmission on Call ID %s\n", pkt->owner->callid);
append_history(pkt->owner, "XmitErr", "%s", pkt->is_fatal ? "(Critical)" : "(Non-critical)");
} else
append_history(pkt->owner, "MaxRetries", "%s", pkt->is_fatal ? "(Critical)" : "(Non-critical)");
pkt->retransid = -1;
if (pkt->is_fatal) {
while(pkt->owner->owner && ast_channel_trylock(pkt->owner->owner)) {
sip_pvt_unlock(pkt->owner); /* SIP_PVT, not channel */
usleep(1);
sip_pvt_lock(pkt->owner);
}
if (pkt->owner->owner && !pkt->owner->owner->hangupcause)
pkt->owner->owner->hangupcause = AST_CAUSE_NO_USER_RESPONSE;
if (pkt->owner->owner) {
sip_alreadygone(pkt->owner);
ast_log(LOG_WARNING, "Hanging up call %s - no reply to our critical packet (see doc/sip-retransmit.txt).\n", pkt->owner->callid);
ast_queue_hangup_with_cause(pkt->owner->owner, AST_CAUSE_PROTOCOL_ERROR);
ast_channel_unlock(pkt->owner->owner);
} else {
/* If no channel owner, destroy now */
/* Let the peerpoke system expire packets when the timer expires for poke_noanswer */
if (pkt->method != SIP_OPTIONS && pkt->method != SIP_REGISTER) {
pvt_set_needdestroy(pkt->owner, "no response to critical packet");
sip_alreadygone(pkt->owner);
append_history(pkt->owner, "DialogKill", "Killing this failed dialog immediately");
}
}
}
if (pkt->method == SIP_BYE) {
/* We're not getting answers on SIP BYE's. Tear down the call anyway. */
if (pkt->owner->owner)
ast_channel_unlock(pkt->owner->owner);
append_history(pkt->owner, "ByeFailure", "Remote peer doesn't respond to bye. Destroying call anyway.");
pvt_set_needdestroy(pkt->owner, "no response to BYE");
}
/* Remove the packet */
for (prev = NULL, cur = pkt->owner->packets; cur; prev = cur, cur = cur->next) {
if (cur == pkt) {
UNLINK(cur, pkt->owner->packets, prev);
sip_pvt_unlock(pkt->owner);
if (pkt->owner)
pkt->owner = dialog_unref(pkt->owner,"pkt is being freed, its dialog ref is dead now");
if (pkt->data)
ast_free(pkt->data);
pkt->data = NULL;
ast_free(pkt);
return 0;
}
}
/* error case */
ast_log(LOG_WARNING, "Weird, couldn't find packet owner!\n");
sip_pvt_unlock(pkt->owner);
return 0;
}
/*! \brief Transmit packet with retransmits
\return 0 on success, -1 on failure to allocate packet
*/
static enum sip_result __sip_reliable_xmit(struct sip_pvt *p, int seqno, int resp, struct ast_str *data, int len, int fatal, int sipmethod)
{
struct sip_pkt *pkt = NULL;
int siptimer_a = DEFAULT_RETRANS;
int xmitres = 0;
if (sipmethod == SIP_INVITE) {
/* Note this is a pending invite */
p->pendinginvite = seqno;
}
/* If the transport is something reliable (TCP or TLS) then don't really send this reliably */
/* I removed the code from retrans_pkt that does the same thing so it doesn't get loaded into the scheduler */
/*! \todo According to the RFC some packets need to be retransmitted even if its TCP, so this needs to get revisited */
if (!(p->socket.type & SIP_TRANSPORT_UDP)) {
xmitres = __sip_xmit(p, data, len); /* Send packet */
if (xmitres == XMIT_ERROR) { /* Serious network trouble, no need to try again */
append_history(p, "XmitErr", "%s", fatal ? "(Critical)" : "(Non-critical)");
return AST_FAILURE;
} else {
return AST_SUCCESS;
}
}
if (!(pkt = ast_calloc(1, sizeof(*pkt) + len + 1)))
return AST_FAILURE;
/* copy data, add a terminator and save length */
if (!(pkt->data = ast_str_create(len))) {
ast_free(pkt);
return AST_FAILURE;
}
ast_str_set(&pkt->data, 0, "%s%s", data->str, "\0");
pkt->packetlen = len;
/* copy other parameters from the caller */
pkt->method = sipmethod;
pkt->seqno = seqno;
pkt->is_resp = resp;
pkt->is_fatal = fatal;
pkt->owner = dialog_ref(p, "__sip_reliable_xmit: setting pkt->owner");
pkt->next = p->packets;
p->packets = pkt; /* Add it to the queue */
pkt->timer_t1 = p->timer_t1; /* Set SIP timer T1 */
pkt->retransid = -1;
if (pkt->timer_t1)
siptimer_a = pkt->timer_t1 * 2;
/* Schedule retransmission */
AST_SCHED_REPLACE_VARIABLE(pkt->retransid, sched, siptimer_a, retrans_pkt, pkt, 1);
if (sipdebug)
ast_debug(4, "*** SIP TIMER: Initializing retransmit timer on packet: Id #%d\n", pkt->retransid);
xmitres = __sip_xmit(pkt->owner, pkt->data, pkt->packetlen); /* Send packet */
if (xmitres == XMIT_ERROR) { /* Serious network trouble, no need to try again */
append_history(pkt->owner, "XmitErr", "%s", pkt->is_fatal ? "(Critical)" : "(Non-critical)");
ast_log(LOG_ERROR, "Serious Network Trouble; __sip_xmit returns error for pkt data\n");
AST_SCHED_DEL(sched, pkt->retransid);
p->packets = pkt->next;
pkt->owner = dialog_unref(pkt->owner,"pkt is being freed, its dialog ref is dead now");
ast_free(pkt->data);
ast_free(pkt);
return AST_FAILURE;
} else {
return AST_SUCCESS;
}
}
/*! \brief Kill a SIP dialog (called only by the scheduler)
* The scheduler has a reference to this dialog when p->autokillid != -1,
* and we are called using that reference. So if the event is not
* rescheduled, we need to call dialog_unref().
*/
static int __sip_autodestruct(const void *data)
{
struct sip_pvt *p = (struct sip_pvt *)data;
/* If this is a subscription, tell the phone that we got a timeout */
if (p->subscribed) {
transmit_state_notify(p, AST_EXTENSION_DEACTIVATED, 1, TRUE); /* Send last notification */
p->subscribed = NONE;
append_history(p, "Subscribestatus", "timeout");
ast_debug(3, "Re-scheduled destruction of SIP subscription %s\n", p->callid ? p->callid : "<unknown>");
return 10000; /* Reschedule this destruction so that we know that it's gone */
}
/* If there are packets still waiting for delivery, delay the destruction */
if (p->packets) {
ast_debug(3, "Re-scheduled destruction of SIP call %s\n", p->callid ? p->callid : "<unknown>");
append_history(p, "ReliableXmit", "timeout");
return 10000;
}
if (p->subscribed == MWI_NOTIFICATION)
if (p->relatedpeer)
p->relatedpeer = unref_peer(p->relatedpeer, "__sip_autodestruct: unref peer p->relatedpeer"); /* Remove link to peer. If it's realtime, make sure it's gone from memory) */
/* Reset schedule ID */
p->autokillid = -1;
if (p->owner) {
ast_log(LOG_WARNING, "Autodestruct on dialog '%s' with owner in place (Method: %s)\n", p->callid, sip_methods[p->method].text);
ast_queue_hangup_with_cause(p->owner, AST_CAUSE_PROTOCOL_ERROR);
} else if (p->refer && !p->alreadygone) {
ast_debug(3, "Finally hanging up channel after transfer: %s\n", p->callid);
transmit_request_with_auth(p, SIP_BYE, 0, XMIT_RELIABLE, 1);
append_history(p, "ReferBYE", "Sending BYE on transferer call leg %s", p->callid);
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
} else {
append_history(p, "AutoDestroy", "%s", p->callid);
ast_debug(3, "Auto destroying SIP dialog '%s'\n", p->callid);
dialog_unlink_all(p, TRUE, TRUE); /* once it's unlinked and unrefd everywhere, it'll be freed automagically */
/* dialog_unref(p, "unref dialog-- no other matching conditions"); -- unlink all now should finish off the dialog's references and free it. */
/* sip_destroy(p); */ /* Go ahead and destroy dialog. All attempts to recover is done */
/* sip_destroy also absorbs the reference */
}
dialog_unref(p, "The ref to a dialog passed to this sched callback is going out of scope; unref it.");
return 0;
}
/*! \brief Schedule destruction of SIP dialog */
static void sip_scheddestroy(struct sip_pvt *p, int ms)
{
if (ms < 0) {
if (p->timer_t1 == 0) {
p->timer_t1 = global_t1; /* Set timer T1 if not set (RFC 3261) */
p->timer_b = global_timer_b; /* Set timer B if not set (RFC 3261) */
}
ms = p->timer_t1 * 64;
}
if (sip_debug_test_pvt(p))
ast_verbose("Scheduling destruction of SIP dialog '%s' in %d ms (Method: %s)\n", p->callid, ms, sip_methods[p->method].text);
if (sip_cancel_destroy(p))
ast_log(LOG_WARNING, "Unable to cancel SIP destruction. Expect bad things.\n");
if (p->do_history)
append_history(p, "SchedDestroy", "%d ms", ms);
p->autokillid = ast_sched_add(sched, ms, __sip_autodestruct, dialog_ref(p, "setting ref as passing into ast_sched_add for __sip_autodestruct"));
if (p->stimer && p->stimer->st_active == TRUE && p->stimer->st_schedid > 0)
stop_session_timer(p);
}
/*! \brief Cancel destruction of SIP dialog.
* Be careful as this also absorbs the reference - if you call it
* from within the scheduler, this might be the last reference.
*/
static int sip_cancel_destroy(struct sip_pvt *p)
{
int res = 0;
if (p->autokillid > -1) {
int res3;
if (!(res3 = ast_sched_del(sched, p->autokillid))) {
append_history(p, "CancelDestroy", "");
p->autokillid = -1;
dialog_unref(p, "dialog unrefd because autokillid is de-sched'd");
}
}
return res;
}
/*! \brief Acknowledges receipt of a packet and stops retransmission
* called with p locked*/
static void __sip_ack(struct sip_pvt *p, int seqno, int resp, int sipmethod)
{
struct sip_pkt *cur, *prev = NULL;
const char *msg = "Not Found"; /* used only for debugging */
/* If we have an outbound proxy for this dialog, then delete it now since
the rest of the requests in this dialog needs to follow the routing.
If obforcing is set, we will keep the outbound proxy during the whole
dialog, regardless of what the SIP rfc says
*/
if (p->outboundproxy && !p->outboundproxy->force){
ref_proxy(p, NULL);
}
for (cur = p->packets; cur; prev = cur, cur = cur->next) {
if (cur->seqno != seqno || cur->is_resp != resp)
continue;
if (cur->is_resp || cur->method == sipmethod) {
msg = "Found";
if (!resp && (seqno == p->pendinginvite)) {
ast_debug(1, "Acked pending invite %d\n", p->pendinginvite);
p->pendinginvite = 0;
}
if (cur->retransid > -1) {
if (sipdebug)
ast_debug(4, "** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #%d\n", cur->retransid);
}
/* This odd section is designed to thwart a
* race condition in the packet scheduler. There are
* two conditions under which deleting the packet from the
* scheduler can fail.
*
* 1. The packet has been removed from the scheduler because retransmission
* is being attempted. The problem is that if the packet is currently attempting
* retransmission and we are at this point in the code, then that MUST mean
* that retrans_pkt is waiting on p's lock. Therefore we will relinquish the
* lock temporarily to allow retransmission.
*
* 2. The packet has reached its maximum number of retransmissions and has
* been permanently removed from the packet scheduler. If this is the case, then
* the packet's retransid will be set to -1. The atomicity of the setting and checking
* of the retransid to -1 is ensured since in both cases p's lock is held.
*/
while (cur->retransid > -1 && ast_sched_del(sched, cur->retransid)) {
sip_pvt_unlock(p);
usleep(1);
sip_pvt_lock(p);
}
UNLINK(cur, p->packets, prev);
dialog_unref(cur->owner, "unref pkt cur->owner dialog from sip ack before freeing pkt");
if (cur->data)
ast_free(cur->data);
ast_free(cur);
break;
}
}
ast_debug(1, "Stopping retransmission on '%s' of %s %d: Match %s\n",
p->callid, resp ? "Response" : "Request", seqno, msg);
}
/*! \brief Pretend to ack all packets
* called with p locked */
static void __sip_pretend_ack(struct sip_pvt *p)
{
struct sip_pkt *cur = NULL;
while (p->packets) {
int method;
if (cur == p->packets) {
ast_log(LOG_WARNING, "Have a packet that doesn't want to give up! %s\n", sip_methods[cur->method].text);
return;
}
cur = p->packets;
method = (cur->method) ? cur->method : find_sip_method(cur->data->str);
__sip_ack(p, cur->seqno, cur->is_resp, method);
}
}
/*! \brief Acks receipt of packet, keep it around (used for provisional responses) */
static int __sip_semi_ack(struct sip_pvt *p, int seqno, int resp, int sipmethod)
{
struct sip_pkt *cur;
int res = -1;
for (cur = p->packets; cur; cur = cur->next) {
if (cur->seqno == seqno && cur->is_resp == resp &&
(cur->is_resp || method_match(sipmethod, cur->data->str))) {
/* this is our baby */
if (cur->retransid > -1) {
if (sipdebug)
ast_debug(4, "*** SIP TIMER: Cancelling retransmission #%d - %s (got response)\n", cur->retransid, sip_methods[sipmethod].text);
}
AST_SCHED_DEL(sched, cur->retransid);
res = 0;
break;
}
}
ast_debug(1, "(Provisional) Stopping retransmission (but retaining packet) on '%s' %s %d: %s\n", p->callid, resp ? "Response" : "Request", seqno, res == -1 ? "Not Found" : "Found");
return res;
}
/*! \brief Copy SIP request, parse it */
static void parse_copy(struct sip_request *dst, const struct sip_request *src)
{
copy_request(dst, src);
parse_request(dst);
}
/*! \brief add a blank line if no body */
static void add_blank(struct sip_request *req)
{
if (!req->lines) {
/* Add extra empty return. add_header() reserves 4 bytes so cannot be truncated */
ast_str_append(&req->data, 0, "\r\n");
req->len = ast_str_strlen(req->data);
}
}
/*! \brief Transmit response on SIP request*/
static int send_response(struct sip_pvt *p, struct sip_request *req, enum xmittype reliable, int seqno)
{
int res;
add_blank(req);
if (sip_debug_test_pvt(p)) {
const struct sockaddr_in *dst = sip_real_dst(p);
ast_verbose("\n<--- %sTransmitting (%s) to %s:%d --->\n%s\n<------------>\n",
reliable ? "Reliably " : "", sip_nat_mode(p),
ast_inet_ntoa(dst->sin_addr),
ntohs(dst->sin_port), req->data->str);
}
if (p->do_history) {
struct sip_request tmp = { .rlPart1 = 0, };
parse_copy(&tmp, req);
append_history(p, reliable ? "TxRespRel" : "TxResp", "%s / %s - %s", tmp.data->str, get_header(&tmp, "CSeq"),
(tmp.method == SIP_RESPONSE || tmp.method == SIP_UNKNOWN) ? REQ_OFFSET_TO_STR(&tmp, rlPart2) : sip_methods[tmp.method].text);
ast_free(tmp.data);
}
res = (reliable) ?
__sip_reliable_xmit(p, seqno, 1, req->data, req->len, (reliable == XMIT_CRITICAL), req->method) :
__sip_xmit(p, req->data, req->len);
ast_free(req->data);
req->data = NULL;
if (res > 0)
return 0;
return res;
}
/*! \brief Send SIP Request to the other part of the dialogue */
static int send_request(struct sip_pvt *p, struct sip_request *req, enum xmittype reliable, int seqno)
{
int res;
/* If we have an outbound proxy, reset peer address
Only do this once.
*/
if (p->outboundproxy) {
p->sa = p->outboundproxy->ip;
}
add_blank(req);
if (sip_debug_test_pvt(p)) {
if (ast_test_flag(&p->flags[0], SIP_NAT_ROUTE))
ast_verbose("%sTransmitting (NAT) to %s:%d:\n%s\n---\n", reliable ? "Reliably " : "", ast_inet_ntoa(p->recv.sin_addr), ntohs(p->recv.sin_port), req->data->str);
else
ast_verbose("%sTransmitting (no NAT) to %s:%d:\n%s\n---\n", reliable ? "Reliably " : "", ast_inet_ntoa(p->sa.sin_addr), ntohs(p->sa.sin_port), req->data->str);
}
if (p->do_history) {
struct sip_request tmp = { .rlPart1 = 0, };
parse_copy(&tmp, req);
append_history(p, reliable ? "TxReqRel" : "TxReq", "%s / %s - %s", tmp.data->str, get_header(&tmp, "CSeq"), sip_methods[tmp.method].text);
ast_free(tmp.data);
}
res = (reliable) ?
__sip_reliable_xmit(p, seqno, 0, req->data, req->len, (reliable == XMIT_CRITICAL), req->method) :
__sip_xmit(p, req->data, req->len);
if (req->data) {
ast_free(req->data);
req->data = NULL;
}
return res;
}
/*! \brief Set an option on a SIP dialog */
static int sip_setoption(struct ast_channel *chan, int option, void *data, int datalen)
{
int res = -1;
struct sip_pvt *p = chan->tech_pvt;
if (option == AST_OPTION_FORMAT_READ) {
int format = *(int *)data;
res = ast_rtp_instance_set_read_format(p->rtp, format);
} else if (option == AST_OPTION_FORMAT_WRITE) {
int format = *(int *)data;
res = ast_rtp_instance_set_write_format(p->rtp, format);
} else if (option == AST_OPTION_MAKE_COMPATIBLE) {
struct ast_channel *peer = data;
res = ast_rtp_instance_make_compatible(chan, p->rtp, peer);
}
return res;
}
/*! \brief Query an option on a SIP dialog */
static int sip_queryoption(struct ast_channel *chan, int option, void *data, int *datalen)
{
int res = -1;
enum ast_t38_state state = T38_STATE_UNAVAILABLE;
struct sip_pvt *p = (struct sip_pvt *) chan->tech_pvt;
switch (option) {
case AST_OPTION_T38_STATE:
/* Make sure we got an ast_t38_state enum passed in */
if (*datalen != sizeof(enum ast_t38_state)) {
ast_log(LOG_ERROR, "Invalid datalen for AST_OPTION_T38_STATE option. Expected %d, got %d\n", (int)sizeof(enum ast_t38_state), *datalen);
return -1;
}
sip_pvt_lock(p);
/* Now if T38 support is enabled we need to look and see what the current state is to get what we want to report back */
if (ast_test_flag(&p->t38.t38support, SIP_PAGE2_T38SUPPORT)) {
switch (p->t38.state) {
case T38_LOCAL_REINVITE:
case T38_PEER_DIRECT:
case T38_PEER_REINVITE:
state = T38_STATE_NEGOTIATING;
break;
case T38_ENABLED:
state = T38_STATE_NEGOTIATED;
break;
default:
state = T38_STATE_UNKNOWN;
}
}
sip_pvt_unlock(p);
*((enum ast_t38_state *) data) = state;
res = 0;
break;
default:
break;
}
return res;
}
/*! \brief Locate closing quote in a string, skipping escaped quotes.
* optionally with a limit on the search.
* start must be past the first quote.
*/
static const char *find_closing_quote(const char *start, const char *lim)
{
char last_char = '\0';
const char *s;
for (s = start; *s && s != lim; last_char = *s++) {
if (*s == '"' && last_char != '\\')
break;
}
return s;
}
/*! \brief Pick out text in brackets from character string
\return pointer to terminated stripped string
\param tmp input string that will be modified
Examples:
\verbatim
"foo" <bar> valid input, returns bar
foo returns the whole string
< "foo ... > returns the string between brackets
< "foo... bogus (missing closing bracket), returns the whole string
XXX maybe should still skip the opening bracket
\endverbatim
*/
static char *get_in_brackets(char *tmp)
{
const char *parse = tmp;
char *first_bracket;
/*
* Skip any quoted text until we find the part in brackets.
* On any error give up and return the full string.
*/
while ( (first_bracket = strchr(parse, '<')) ) {
char *first_quote = strchr(parse, '"');
if (!first_quote || first_quote > first_bracket)
break; /* no need to look at quoted part */
/* the bracket is within quotes, so ignore it */
parse = find_closing_quote(first_quote + 1, NULL);
if (!*parse) { /* not found, return full string ? */
/* XXX or be robust and return in-bracket part ? */
ast_log(LOG_WARNING, "No closing quote found in '%s'\n", tmp);
break;
}
parse++;
}
if (first_bracket) {
char *second_bracket = strchr(first_bracket + 1, '>');
if (second_bracket) {
*second_bracket = '\0';
tmp = first_bracket + 1;
} else {
ast_log(LOG_WARNING, "No closing bracket found in '%s'\n", tmp);
}
}
return tmp;
}
/*! \brief * parses a URI in its components.
*
* \note
* - If scheme is specified, drop it from the top.
* - If a component is not requested, do not split around it.
*
* This means that if we don't have domain, we cannot split
* name:pass and domain:port.
* It is safe to call with ret_name, pass, domain, port
* pointing all to the same place.
* Init pointers to empty string so we never get NULL dereferencing.
* Overwrites the string.
* return 0 on success, other values on error.
* \verbatim
* general form we are expecting is sip[s]:username[:password][;parameter]@host[:port][;...]
* \endverbatim
*
* \todo This function needs to look for ;transport= too
*/
static int parse_uri(char *uri, char *scheme,
char **ret_name, char **pass, char **domain, char **port, char **options)
{
char *name = NULL;
int error = 0;
/* init field as required */
if (pass)
*pass = "";
if (port)
*port = "";
if (scheme) {
int l = strlen(scheme);
if (!strncasecmp(uri, scheme, l))
uri += l;
else {
ast_debug(1, "Missing scheme '%s' in '%s'\n", scheme, uri);
error = -1;
}
}
if (!domain) {
/* if we don't want to split around domain, keep everything as a name,
* so we need to do nothing here, except remember why.
*/
} else {
/* store the result in a temp. variable to avoid it being
* overwritten if arguments point to the same place.
*/
char *c, *dom = "";
if ((c = strchr(uri, '@')) == NULL) {
/* domain-only URI, according to the SIP RFC. */
dom = uri;
name = "";
} else {
*c++ = '\0';
dom = c;
name = uri;
}
/* Remove options in domain and name */
dom = strsep(&dom, ";");
name = strsep(&name, ";");
if (port && (c = strchr(dom, ':'))) { /* Remove :port */
*c++ = '\0';
*port = c;
}
if (pass && (c = strchr(name, ':'))) { /* user:password */
*c++ = '\0';
*pass = c;
}
*domain = dom;
}
if (ret_name) /* same as for domain, store the result only at the end */
*ret_name = name;
if (options)
*options = uri ? uri : "";
return error;
}
/*! \brief Send message with Access-URL header, if this is an HTML URL only! */
static int sip_sendhtml(struct ast_channel *chan, int subclass, const char *data, int datalen)
{
struct sip_pvt *p = chan->tech_pvt;
if (subclass != AST_HTML_URL)
return -1;
ast_string_field_build(p, url, "<%s>;mode=active", data);
if (sip_debug_test_pvt(p))
ast_debug(1, "Send URL %s, state = %d!\n", data, chan->_state);
switch (chan->_state) {
case AST_STATE_RING:
transmit_response(p, "100 Trying", &p->initreq);
break;
case AST_STATE_RINGING:
transmit_response(p, "180 Ringing", &p->initreq);
break;
case AST_STATE_UP:
if (!p->pendinginvite) { /* We are up, and have no outstanding invite */
transmit_reinvite_with_sdp(p, FALSE, FALSE);
} else if (!ast_test_flag(&p->flags[0], SIP_PENDINGBYE)) {
ast_set_flag(&p->flags[0], SIP_NEEDREINVITE);
}
break;
default:
ast_log(LOG_WARNING, "Don't know how to send URI when state is %d!\n", chan->_state);
}
return 0;
}
/*! \brief Deliver SIP call ID for the call */
static const char *sip_get_callid(struct ast_channel *chan)
{
return chan->tech_pvt ? ((struct sip_pvt *) chan->tech_pvt)->callid : "";
}
/*! \brief Send SIP MESSAGE text within a call
Called from PBX core sendtext() application */
static int sip_sendtext(struct ast_channel *ast, const char *text)
{
struct sip_pvt *p = ast->tech_pvt;
int debug = sip_debug_test_pvt(p);
if (debug)
ast_verbose("Sending text %s on %s\n", text, ast->name);
if (!p)
return -1;
/* NOT ast_strlen_zero, because a zero-length message is specifically
* allowed by RFC 3428 (See section 10, Examples) */
if (!text)
return 0;
if (debug)
ast_verbose("Really sending text %s on %s\n", text, ast->name);
transmit_message_with_text(p, text);
return 0;
}
/*! \brief Update peer object in realtime storage
If the Asterisk system name is set in asterisk.conf, we will use
that name and store that in the "regserver" field in the sippeers
table to facilitate multi-server setups.
*/
static void realtime_update_peer(const char *peername, struct sockaddr_in *sin, const char *defaultuser, const char *fullcontact, const char *useragent, int expirey, int deprecated_username, int lastms)
{
char port[10];
char ipaddr[INET_ADDRSTRLEN];
char regseconds[20];
char *tablename = NULL;
char str_lastms[20];
const char *sysname = ast_config_AST_SYSTEM_NAME;
char *syslabel = NULL;
time_t nowtime = time(NULL) + expirey;
const char *fc = fullcontact ? "fullcontact" : NULL;
int realtimeregs = ast_check_realtime("sipregs");
tablename = realtimeregs ? "sipregs" : "sippeers";
snprintf(str_lastms, sizeof(str_lastms), "%d", lastms);
snprintf(regseconds, sizeof(regseconds), "%d", (int)nowtime); /* Expiration time */
ast_copy_string(ipaddr, ast_inet_ntoa(sin->sin_addr), sizeof(ipaddr));
snprintf(port, sizeof(port), "%d", ntohs(sin->sin_port));
if (ast_strlen_zero(sysname)) /* No system name, disable this */
sysname = NULL;
else if (sip_cfg.rtsave_sysname)
syslabel = "regserver";
if (fc) {
ast_update_realtime(tablename, "name", peername, "ipaddr", ipaddr,
"port", port, "regseconds", regseconds,
deprecated_username ? "username" : "defaultuser", defaultuser,
"useragent", useragent, "lastms", str_lastms,
fc, fullcontact, syslabel, sysname, SENTINEL); /* note fc and syslabel _can_ be NULL */
} else {
ast_update_realtime(tablename, "name", peername, "ipaddr", ipaddr,
"port", port, "regseconds", regseconds,
"useragent", useragent, "lastms", str_lastms,
deprecated_username ? "username" : "defaultuser", defaultuser,
syslabel, sysname, SENTINEL); /* note syslabel _can_ be NULL */
}
}
/*! \brief Automatically add peer extension to dial plan */
static void register_peer_exten(struct sip_peer *peer, int onoff)
{
char multi[256];
char *stringp, *ext, *context;
struct pbx_find_info q = { .stacklen = 0 };
/* XXX note that global_regcontext is both a global 'enable' flag and
* the name of the global regexten context, if not specified
* individually.
*/
if (ast_strlen_zero(global_regcontext))
return;
ast_copy_string(multi, S_OR(peer->regexten, peer->name), sizeof(multi));
stringp = multi;
while ((ext = strsep(&stringp, "&"))) {
if ((context = strchr(ext, '@'))) {
*context++ = '\0'; /* split ext@context */
if (!ast_context_find(context)) {
ast_log(LOG_WARNING, "Context %s must exist in regcontext= in sip.conf!\n", context);
continue;
}
} else {
context = global_regcontext;
}
if (onoff) {
if (!ast_exists_extension(NULL, context, ext, 1, NULL)) {
ast_add_extension(context, 1, ext, 1, NULL, NULL, "Noop",
ast_strdup(peer->name), ast_free_ptr, "SIP");
}
} else if (pbx_find_extension(NULL, NULL, &q, context, ext, 1, NULL, "", E_MATCH)) {
ast_context_remove_extension(context, ext, 1, NULL);
}
}
}
/*! Destroy mailbox subscriptions */
static void destroy_mailbox(struct sip_mailbox *mailbox)
{
if (mailbox->mailbox)
ast_free(mailbox->mailbox);
if (mailbox->context)
ast_free(mailbox->context);
if (mailbox->event_sub)
ast_event_unsubscribe(mailbox->event_sub);
ast_free(mailbox);
}
/*! Destroy all peer-related mailbox subscriptions */
static void clear_peer_mailboxes(struct sip_peer *peer)
{
struct sip_mailbox *mailbox;
while ((mailbox = AST_LIST_REMOVE_HEAD(&peer->mailboxes, entry)))
destroy_mailbox(mailbox);
}
static void sip_destroy_peer_fn(void *peer)
{
sip_destroy_peer(peer);
}
/*! \brief Destroy peer object from memory */
static void sip_destroy_peer(struct sip_peer *peer)
{
ast_debug(3, "Destroying SIP peer %s\n", peer->name);
if (peer->outboundproxy)
ao2_ref(peer->outboundproxy, -1);
peer->outboundproxy = NULL;
/* Delete it, it needs to disappear */
if (peer->call) {
dialog_unlink_all(peer->call, TRUE, TRUE);
peer->call = dialog_unref(peer->call, "peer->call is being unset");
}
if (peer->mwipvt) { /* We have an active subscription, delete it */
dialog_unlink_all(peer->mwipvt, TRUE, TRUE);
peer->mwipvt = dialog_unref(peer->mwipvt, "unreffing peer->mwipvt");
}
if (peer->chanvars) {
ast_variables_destroy(peer->chanvars);
peer->chanvars = NULL;
}
register_peer_exten(peer, FALSE);
ast_free_ha(peer->ha);
if (peer->selfdestruct)
ast_atomic_fetchadd_int(&apeerobjs, -1);
else if (peer->is_realtime) {
ast_atomic_fetchadd_int(&rpeerobjs, -1);
ast_debug(3, "-REALTIME- peer Destroyed. Name: %s. Realtime Peer objects: %d\n", peer->name, rpeerobjs);
} else
ast_atomic_fetchadd_int(&speerobjs, -1);
clear_realm_authentication(peer->auth);
peer->auth = NULL;
if (peer->dnsmgr)
ast_dnsmgr_release(peer->dnsmgr);
clear_peer_mailboxes(peer);
if (peer->socket.tcptls_session) {
ao2_ref(peer->socket.tcptls_session, -1);
peer->socket.tcptls_session = NULL;
}
ast_string_field_free_memory(peer);
}
/*! \brief Update peer data in database (if used) */
static void update_peer(struct sip_peer *p, int expire)
{
int rtcachefriends = ast_test_flag(&p->flags[1], SIP_PAGE2_RTCACHEFRIENDS);
if (sip_cfg.peer_rtupdate &&
(p->is_realtime || rtcachefriends)) {
realtime_update_peer(p->name, &p->addr, p->username, rtcachefriends ? p->fullcontact : NULL, p->useragent, expire, p->deprecated_username, p->lastms);
}
}
static struct ast_variable *get_insecure_variable_from_config(struct ast_config *cfg)
{
struct ast_variable *var = NULL;
struct ast_flags flags = {0};
char *cat = NULL;
const char *insecure;
while ((cat = ast_category_browse(cfg, cat))) {
insecure = ast_variable_retrieve(cfg, cat, "insecure");
set_insecure_flags(&flags, insecure, -1);
if (ast_test_flag(&flags, SIP_INSECURE_PORT)) {
var = ast_category_root(cfg, cat);
break;
}
}
return var;
}
static const char *get_name_from_variable(struct ast_variable *var, const char *newpeername)
{
struct ast_variable *tmp;
for (tmp = var; tmp; tmp = tmp->next) {
if (!newpeername && !strcasecmp(tmp->name, "name"))
newpeername = tmp->value;
}
return newpeername;
}
/*! \brief realtime_peer: Get peer from realtime storage
* Checks the "sippeers" realtime family from extconfig.conf
* Checks the "sipregs" realtime family from extconfig.conf if it's configured.
* This returns a pointer to a peer and because we use build_peer, we can rest
* assured that the refcount is bumped.
*/
static struct sip_peer *realtime_peer(const char *newpeername, struct sockaddr_in *sin, int devstate_only)
{
struct sip_peer *peer;
struct ast_variable *var = NULL;
struct ast_variable *varregs = NULL;
struct ast_variable *tmp;
struct ast_config *peerlist = NULL;
char ipaddr[INET_ADDRSTRLEN];
char portstring[6]; /*up to 5 digits plus null terminator*/
char *cat = NULL;
unsigned short portnum;
int realtimeregs = ast_check_realtime("sipregs");
/* First check on peer name */
if (newpeername) {
if (realtimeregs)
varregs = ast_load_realtime("sipregs", "name", newpeername, SENTINEL);
var = ast_load_realtime("sippeers", "name", newpeername, "host", "dynamic", SENTINEL);
if (!var && sin)
var = ast_load_realtime("sippeers", "name", newpeername, "host", ast_inet_ntoa(sin->sin_addr), SENTINEL);
if (!var) {
var = ast_load_realtime("sippeers", "name", newpeername, SENTINEL);
/*!\note
* If this one loaded something, then we need to ensure that the host
* field matched. The only reason why we can't have this as a criteria
* is because we only have the IP address and the host field might be
* set as a name (and the reverse PTR might not match).
*/
if (var && sin) {
for (tmp = var; tmp; tmp = tmp->next) {
if (!strcasecmp(tmp->name, "host")) {
struct hostent *hp;
struct ast_hostent ahp;
if (!(hp = ast_gethostbyname(tmp->value, &ahp)) || (memcmp(&hp->h_addr, &sin->sin_addr, sizeof(hp->h_addr)))) {
/* No match */
ast_variables_destroy(var);
var = NULL;
}
break;
}
}
}
}
}
if (!var && sin) { /* Then check on IP address for dynamic peers */
ast_copy_string(ipaddr, ast_inet_ntoa(sin->sin_addr), sizeof(ipaddr));
portnum = ntohs(sin->sin_port);
sprintf(portstring, "%u", portnum);
var = ast_load_realtime("sippeers", "host", ipaddr, "port", portstring, SENTINEL); /* First check for fixed IP hosts */
if (var) {
if (realtimeregs) {
newpeername = get_name_from_variable(var, newpeername);
varregs = ast_load_realtime("sipregs", "name", newpeername, SENTINEL);
}
} else {
if (realtimeregs)
varregs = ast_load_realtime("sipregs", "ipaddr", ipaddr, "port", portstring, SENTINEL); /* Then check for registered hosts */
else
var = ast_load_realtime("sippeers", "ipaddr", ipaddr, "port", portstring, SENTINEL); /* Then check for registered hosts */
if (varregs) {
newpeername = get_name_from_variable(varregs, newpeername);
var = ast_load_realtime("sippeers", "name", newpeername, SENTINEL);
}
}
if (!var) { /*We couldn't match on ipaddress and port, so we need to check if port is insecure*/
peerlist = ast_load_realtime_multientry("sippeers", "host", ipaddr, SENTINEL);
if (peerlist) {
var = get_insecure_variable_from_config(peerlist);
if(var) {
if (realtimeregs) {
newpeername = get_name_from_variable(var, newpeername);
varregs = ast_load_realtime("sipregs", "name", newpeername, SENTINEL);
}
} else { /*var wasn't found in the list of "hosts", so try "ipaddr"*/
peerlist = NULL;
cat = NULL;
peerlist = ast_load_realtime_multientry("sippeers", "ipaddr", ipaddr, SENTINEL);
if(peerlist) {
var = get_insecure_variable_from_config(peerlist);
if(var) {
if (realtimeregs) {
newpeername = get_name_from_variable(var, newpeername);
varregs = ast_load_realtime("sipregs", "name", newpeername, SENTINEL);
}
}
}
}
} else {
if (realtimeregs) {
peerlist = ast_load_realtime_multientry("sipregs", "ipaddr", ipaddr, SENTINEL);
if (peerlist) {
varregs = get_insecure_variable_from_config(peerlist);
if (varregs) {
newpeername = get_name_from_variable(varregs, newpeername);
var = ast_load_realtime("sippeers", "name", newpeername, SENTINEL);
}
}
} else {
peerlist = ast_load_realtime_multientry("sippeers", "ipaddr", ipaddr, SENTINEL);
if (peerlist) {
var = get_insecure_variable_from_config(peerlist);
if (var) {
newpeername = get_name_from_variable(var, newpeername);
varregs = ast_load_realtime("sipregs", "name", newpeername, SENTINEL);
}
}
}
}
}
}
if (!var) {
if (peerlist)
ast_config_destroy(peerlist);
return NULL;
}
for (tmp = var; tmp; tmp = tmp->next) {
/* If this is type=user, then skip this object. */
if (!strcasecmp(tmp->name, "type") &&
!strcasecmp(tmp->value, "user")) {
if(peerlist)
ast_config_destroy(peerlist);
else {
ast_variables_destroy(var);
ast_variables_destroy(varregs);
}
return NULL;
} else if (!newpeername && !strcasecmp(tmp->name, "name")) {
newpeername = tmp->value;
}
}
if (!newpeername) { /* Did not find peer in realtime */
ast_log(LOG_WARNING, "Cannot Determine peer name ip=%s\n", ipaddr);
if(peerlist)
ast_config_destroy(peerlist);
else
ast_variables_destroy(var);
return NULL;
}
/* Peer found in realtime, now build it in memory */
peer = build_peer(newpeername, var, varregs, TRUE);
if (!peer) {
if(peerlist)
ast_config_destroy(peerlist);
else {
ast_variables_destroy(var);
ast_variables_destroy(varregs);
}
return NULL;
}
ast_debug(3, "-REALTIME- loading peer from database to memory. Name: %s. Peer objects: %d\n", peer->name, rpeerobjs);
if (ast_test_flag(&global_flags[1], SIP_PAGE2_RTCACHEFRIENDS) && !devstate_only) {
/* Cache peer */
ast_copy_flags(&peer->flags[1], &global_flags[1], SIP_PAGE2_RTAUTOCLEAR|SIP_PAGE2_RTCACHEFRIENDS);
if (ast_test_flag(&global_flags[1], SIP_PAGE2_RTAUTOCLEAR)) {
AST_SCHED_REPLACE_UNREF(peer->expire, sched, sip_cfg.rtautoclear * 1000, expire_register, peer,
unref_peer(_data, "remove registration ref"),
unref_peer(peer, "remove registration ref"),
ref_peer(peer, "add registration ref"));
}
ao2_t_link(peers, peer, "link peer into peers table");
if (peer->addr.sin_addr.s_addr) {
ao2_t_link(peers_by_ip, peer, "link peer into peers_by_ip table");
}
}
peer->is_realtime = 1;
if (peerlist)
ast_config_destroy(peerlist);
else {
ast_variables_destroy(var);
ast_variables_destroy(varregs);
}
return peer;
}
/* Function to assist finding peers by name only */
static int find_by_name(void *obj, void *arg, void *data, int flags)
{
struct sip_peer *search = obj, *match = arg;
int *which_objects = data;
/* Usernames in SIP uri's are case sensitive. Domains are not */
if (strcmp(search->name, match->name)) {
return 0;
}
switch (*which_objects) {
case FINDUSERS:
if (!(search->type & SIP_TYPE_USER)) {
return 0;
}
break;
case FINDPEERS:
if (!(search->type & SIP_TYPE_PEER)) {
return 0;
}
break;
case FINDALLDEVICES:
break;
}
return CMP_MATCH | CMP_STOP;
}
/*!
* \brief Locate device by name or ip address
*
* \param which_objects Define which objects should be matched when doing a lookup
* by name. Valid options are FINDUSERS, FINDPEERS, or FINDALLDEVICES.
* Note that this option is not used at all when doing a lookup by IP.
*
* This is used on find matching device on name or ip/port.
* If the device was declared as type=peer, we don't match on peer name on incoming INVITEs.
*
* \note Avoid using this function in new functions if there is a way to avoid it,
* since it might cause a database lookup.
*/
static struct sip_peer *find_peer(const char *peer, struct sockaddr_in *sin, int realtime, int which_objects, int devstate_only)
{
struct sip_peer *p = NULL;
struct sip_peer tmp_peer;
if (peer) {
ast_copy_string(tmp_peer.name, peer, sizeof(tmp_peer.name));
p = ao2_t_callback_data(peers, OBJ_POINTER, find_by_name, &tmp_peer, &which_objects, "ao2_find in peers table");
} else if (sin) { /* search by addr? */
tmp_peer.addr.sin_addr.s_addr = sin->sin_addr.s_addr;
tmp_peer.addr.sin_port = sin->sin_port;
tmp_peer.flags[0].flags = 0;
p = ao2_t_find(peers_by_ip, &tmp_peer, OBJ_POINTER, "ao2_find in peers_by_ip table"); /* WAS: p = ASTOBJ_CONTAINER_FIND_FULL(&peerl, sin, name, sip_addr_hashfunc, 1, sip_addrcmp); */
if (!p) {
ast_set_flag(&tmp_peer.flags[0], SIP_INSECURE_PORT);
p = ao2_t_find(peers_by_ip, &tmp_peer, OBJ_POINTER, "ao2_find in peers_by_ip table 2"); /* WAS: p = ASTOBJ_CONTAINER_FIND_FULL(&peerl, sin, name, sip_addr_hashfunc, 1, sip_addrcmp); */
if (p) {
return p;
}
}
}
if (!p && (realtime || devstate_only)) {
p = realtime_peer(peer, sin, devstate_only);
}
return p;
}
/*! \brief Set nat mode on the various data sockets */
static void do_setnat(struct sip_pvt *p, int natflags)
{
const char *mode = natflags ? "On" : "Off";
if (p->rtp) {
ast_debug(1, "Setting NAT on RTP to %s\n", mode);
ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_NAT, natflags);
}
if (p->vrtp) {
ast_debug(1, "Setting NAT on VRTP to %s\n", mode);
ast_rtp_instance_set_prop(p->vrtp, AST_RTP_PROPERTY_NAT, natflags);
}
if (p->udptl) {
ast_debug(1, "Setting NAT on UDPTL to %s\n", mode);
ast_udptl_setnat(p->udptl, natflags);
}
if (p->trtp) {
ast_debug(1, "Setting NAT on TRTP to %s\n", mode);
ast_rtp_instance_set_prop(p->trtp, AST_RTP_PROPERTY_NAT, natflags);
}
}
/*! \brief Change the T38 state on a SIP dialog */
static void change_t38_state(struct sip_pvt *p, int state)
{
int old = p->t38.state;
struct ast_channel *chan = p->owner;
enum ast_control_t38 message = 0;
/* Don't bother changing if we are already in the state wanted */
if (old == state)
return;
if (state == T38_PEER_DIRECT) {
p->t38.direct = 1;
}
p->t38.state = state;
ast_debug(2, "T38 state changed to %d on channel %s\n", p->t38.state, chan ? chan->name : "<none>");
/* If no channel was provided we can't send off a control frame */
if (!chan)
return;
/* Given the state requested and old state determine what control frame we want to queue up */
if (state == T38_PEER_REINVITE)
message = AST_T38_REQUEST_NEGOTIATE;
else if (state == T38_ENABLED)
message = AST_T38_NEGOTIATED;
else if (state == T38_DISABLED && old == T38_ENABLED)
message = AST_T38_TERMINATED;
else if (state == T38_DISABLED && old == T38_LOCAL_REINVITE)
message = AST_T38_REFUSED;
/* Woot we got a message, create a control frame and send it on! */
if (message)
ast_queue_control_data(chan, AST_CONTROL_T38, &message, sizeof(message));
if (ast_test_flag(&p->flags[1], SIP_PAGE2_FAX_DETECT) && !p->outgoing_call) {
/* fax detection is enabled and this is an incoming call */
ast_channel_lock(chan);
if (strcmp(chan->exten, "fax") && state == T38_ENABLED) {
const char *target_context = S_OR(chan->macrocontext, chan->context);
ast_channel_unlock(chan);
if (ast_exists_extension(chan, target_context, "fax", 1, chan->cid.cid_num)) {
/* save the DID/DNIS when we transfer the fax call to a 'fax' extension */
ast_verb(3, "Redirecting '%s' to fax extension\n", chan->name);
pbx_builtin_setvar_helper(chan, "FAXEXTEN", chan->exten);
if (ast_async_goto(chan, target_context, "fax", 1)) {
ast_log(LOG_WARNING, "Failed to async goto '%s' into fax of '%s'\n", chan->name, target_context);
}
} else {
ast_log(LOG_NOTICE, "Fax detected but no fax extension.\n");
}
} else {
ast_channel_unlock(chan);
ast_debug(1, "Already in a fax extension (or T38 was not enabled, state: '%d'), not redirecting.\n", state);
}
}
}
/*! \brief Set the global T38 capabilities on a SIP dialog structure */
static void set_t38_capabilities(struct sip_pvt *p)
{
p->t38.capability = global_t38_capability;
if (p->udptl) {
if (ast_udptl_get_error_correction_scheme(p->udptl) == UDPTL_ERROR_CORRECTION_FEC )
p->t38.capability |= T38FAX_UDP_EC_FEC;
else if (ast_udptl_get_error_correction_scheme(p->udptl) == UDPTL_ERROR_CORRECTION_REDUNDANCY )
p->t38.capability |= T38FAX_UDP_EC_REDUNDANCY;
else if (ast_udptl_get_error_correction_scheme(p->udptl) == UDPTL_ERROR_CORRECTION_NONE )
p->t38.capability |= T38FAX_UDP_EC_NONE;
p->t38.capability |= T38FAX_RATE_MANAGEMENT_TRANSFERED_TCF;
}
}
static void copy_socket_data(struct sip_socket *to_sock, const struct sip_socket *from_sock)
{
if (to_sock->tcptls_session) {
ao2_ref(to_sock->tcptls_session, -1);
to_sock->tcptls_session = NULL;
}
if (from_sock->tcptls_session) {
ao2_ref(from_sock->tcptls_session, +1);
}
*to_sock = *from_sock;
}
/*! \brief Initialize RTP portion of a dialog
* \returns -1 on failure, 0 on success
*/
static int dialog_initialize_rtp(struct sip_pvt *dialog)
{
if (!sip_methods[dialog->method].need_rtp) {
return 0;
}
if (!(dialog->rtp = ast_rtp_instance_new(dialog->engine, sched, &bindaddr, NULL))) {
return -1;
}
if (ast_test_flag(&dialog->flags[1], SIP_PAGE2_VIDEOSUPPORT) && (dialog->capability & AST_FORMAT_VIDEO_MASK)) {
if (!(dialog->vrtp = ast_rtp_instance_new(dialog->engine, sched, &bindaddr, NULL))) {
return -1;
}
ast_rtp_instance_set_timeout(dialog->vrtp, global_rtptimeout);
ast_rtp_instance_set_hold_timeout(dialog->vrtp, global_rtpholdtimeout);
ast_rtp_instance_set_prop(dialog->vrtp, AST_RTP_PROPERTY_RTCP, 1);
}
if (ast_test_flag(&dialog->flags[1], SIP_PAGE2_TEXTSUPPORT)) {
if (!(dialog->trtp = ast_rtp_instance_new(dialog->engine, sched, &bindaddr, NULL))) {
return -1;
}
ast_rtp_instance_set_timeout(dialog->trtp, global_rtptimeout);
ast_rtp_instance_set_hold_timeout(dialog->trtp, global_rtpholdtimeout);
ast_rtp_instance_set_prop(dialog->trtp, AST_RTP_PROPERTY_RTCP, 1);
}
ast_rtp_instance_set_timeout(dialog->rtp, global_rtptimeout);
ast_rtp_instance_set_hold_timeout(dialog->rtp, global_rtpholdtimeout);
ast_rtp_instance_set_prop(dialog->rtp, AST_RTP_PROPERTY_RTCP, 1);
ast_rtp_instance_set_prop(dialog->rtp, AST_RTP_PROPERTY_DTMF, ast_test_flag(&dialog->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833);
ast_rtp_instance_set_prop(dialog->rtp, AST_RTP_PROPERTY_DTMF_COMPENSATE, ast_test_flag(&dialog->flags[1], SIP_PAGE2_RFC2833_COMPENSATE));
ast_rtp_instance_set_qos(dialog->rtp, global_tos_audio, 0, "SIP RTP");
do_setnat(dialog, ast_test_flag(&dialog->flags[0], SIP_NAT) & SIP_NAT_ROUTE);
return 0;
}
/*! \brief Create address structure from peer reference.
* This function copies data from peer to the dialog, so we don't have to look up the peer
* again from memory or database during the life time of the dialog.
*
* \return -1 on error, 0 on success.
*
*/
static int create_addr_from_peer(struct sip_pvt *dialog, struct sip_peer *peer)
{
/* this checks that the dialog is contacting the peer on a valid
* transport type based on the peers transport configuration,
* otherwise, this function bails out */
if (dialog->socket.type && check_request_transport(peer, dialog))
return -1;
copy_socket_data(&dialog->socket, &peer->socket);
if ((peer->addr.sin_addr.s_addr || peer->defaddr.sin_addr.s_addr) &&
(!peer->maxms || ((peer->lastms >= 0) && (peer->lastms <= peer->maxms)))) {
dialog->sa = (peer->addr.sin_addr.s_addr) ? peer->addr : peer->defaddr;
dialog->recv = dialog->sa;
} else
return -1;
ast_copy_flags(&dialog->flags[0], &peer->flags[0], SIP_FLAGS_TO_COPY);
ast_copy_flags(&dialog->flags[1], &peer->flags[1], SIP_PAGE2_FLAGS_TO_COPY);
dialog->capability = peer->capability;
dialog->prefs = peer->prefs;
if (ast_test_flag(&dialog->flags[1], SIP_PAGE2_T38SUPPORT)) {
if (!dialog->udptl) {
/* t38pt_udptl was enabled in the peer and not in [general] */
dialog->udptl = ast_udptl_new_with_bindaddr(sched, io, 0, bindaddr.sin_addr);
}
ast_copy_flags(&dialog->t38.t38support, &peer->flags[1], SIP_PAGE2_T38SUPPORT);
set_t38_capabilities(dialog);
dialog->t38.jointcapability = dialog->t38.capability;
} else if (dialog->udptl) {
ast_udptl_destroy(dialog->udptl);
dialog->udptl = NULL;
}
ast_string_field_set(dialog, engine, peer->engine);
if (dialog_initialize_rtp(dialog)) {
return -1;
}
if (dialog->rtp) { /* Audio */
ast_rtp_instance_set_prop(dialog->rtp, AST_RTP_PROPERTY_DTMF, ast_test_flag(&dialog->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833);
ast_rtp_instance_set_prop(dialog->rtp, AST_RTP_PROPERTY_DTMF_COMPENSATE, ast_test_flag(&dialog->flags[1], SIP_PAGE2_RFC2833_COMPENSATE));
ast_rtp_instance_set_timeout(dialog->rtp, peer->rtptimeout);
ast_rtp_instance_set_hold_timeout(dialog->rtp, peer->rtpholdtimeout);
/* Set Frame packetization */
ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(dialog->rtp), dialog->rtp, &dialog->prefs);
dialog->autoframing = peer->autoframing;
}
if (dialog->vrtp) { /* Video */
ast_rtp_instance_set_timeout(dialog->vrtp, peer->rtptimeout);
ast_rtp_instance_set_hold_timeout(dialog->vrtp, peer->rtpholdtimeout);
}
if (dialog->trtp) { /* Realtime text */
ast_rtp_instance_set_timeout(dialog->trtp, peer->rtptimeout);
ast_rtp_instance_set_hold_timeout(dialog->trtp, peer->rtpholdtimeout);
}
ast_string_field_set(dialog, peername, peer->name);
ast_string_field_set(dialog, authname, peer->username);
ast_string_field_set(dialog, username, peer->username);
ast_string_field_set(dialog, peersecret, peer->secret);
ast_string_field_set(dialog, peermd5secret, peer->md5secret);
ast_string_field_set(dialog, mohsuggest, peer->mohsuggest);
ast_string_field_set(dialog, mohinterpret, peer->mohinterpret);
ast_string_field_set(dialog, tohost, peer->tohost);
ast_string_field_set(dialog, fullcontact, peer->fullcontact);
ast_string_field_set(dialog, context, peer->context);
ast_string_field_set(dialog, cid_num, peer->cid_num);
ast_string_field_set(dialog, cid_name, peer->cid_name);
ast_string_field_set(dialog, mwi_from, peer->mwi_from);
ast_string_field_set(dialog, parkinglot, peer->parkinglot);
ast_string_field_set(dialog, engine, peer->engine);
ref_proxy(dialog, obproxy_get(dialog, peer));
dialog->callgroup = peer->callgroup;
dialog->pickupgroup = peer->pickupgroup;
dialog->allowtransfer = peer->allowtransfer;
dialog->jointnoncodeccapability = dialog->noncodeccapability;
dialog->rtptimeout = peer->rtptimeout;
dialog->peerauth = peer->auth;
dialog->maxcallbitrate = peer->maxcallbitrate;
if (ast_strlen_zero(dialog->tohost))
ast_string_field_set(dialog, tohost, ast_inet_ntoa(dialog->sa.sin_addr));
if (!ast_strlen_zero(peer->fromdomain)) {
ast_string_field_set(dialog, fromdomain, peer->fromdomain);
if (!dialog->initreq.headers) {
char *c;
char *tmpcall = ast_strdupa(dialog->callid);
/* this sure looks to me like we are going to change the callid on this dialog!! */
c = strchr(tmpcall, '@');
if (c) {
*c = '\0';
ao2_t_unlink(dialogs, dialog, "About to change the callid -- remove the old name");
ast_string_field_build(dialog, callid, "%s@%s", tmpcall, peer->fromdomain);
ao2_t_link(dialogs, dialog, "New dialog callid -- inserted back into table");
}
}
}
if (!ast_strlen_zero(peer->fromuser))
ast_string_field_set(dialog, fromuser, peer->fromuser);
if (!ast_strlen_zero(peer->language))
ast_string_field_set(dialog, language, peer->language);
/* Set timer T1 to RTT for this peer (if known by qualify=) */
/* Minimum is settable or default to 100 ms */
/* If there is a maxms and lastms from a qualify use that over a manual T1
value. Otherwise, use the peer's T1 value. */
if (peer->maxms && peer->lastms)
dialog->timer_t1 = peer->lastms < global_t1min ? global_t1min : peer->lastms;
else
dialog->timer_t1 = peer->timer_t1;
/* Set timer B to control transaction timeouts, the peer setting is the default and overrides
the known timer */
if (peer->timer_b)
dialog->timer_b = peer->timer_b;
else
dialog->timer_b = 64 * dialog->timer_t1;
if ((ast_test_flag(&dialog->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833) ||
(ast_test_flag(&dialog->flags[0], SIP_DTMF) == SIP_DTMF_AUTO))
dialog->noncodeccapability |= AST_RTP_DTMF;
else
dialog->noncodeccapability &= ~AST_RTP_DTMF;
if (peer->call_limit)
ast_set_flag(&dialog->flags[0], SIP_CALL_LIMIT);
dialog->chanvars = copy_vars(peer->chanvars);
return 0;
}
/*! \brief create address structure from device name
* Or, if peer not found, find it in the global DNS
* returns TRUE (-1) on failure, FALSE on success */
static int create_addr(struct sip_pvt *dialog, const char *opeer, struct sockaddr_in *sin, int newdialog)
{
struct hostent *hp;
struct ast_hostent ahp;
struct sip_peer *peer;
char *port;
int portno = 0;
char host[MAXHOSTNAMELEN], *hostn;
char peername[256];
int srv_ret = 0;
ast_copy_string(peername, opeer, sizeof(peername));
port = strchr(peername, ':');
if (port)
*port++ = '\0';
dialog->sa.sin_family = AF_INET;
dialog->timer_t1 = global_t1; /* Default SIP retransmission timer T1 (RFC 3261) */
dialog->timer_b = global_timer_b; /* Default SIP transaction timer B (RFC 3261) */
peer = find_peer(peername, NULL, TRUE, FINDPEERS, FALSE);
if (peer) {
int res;
if (newdialog)
dialog->socket.type = 0;
res = create_addr_from_peer(dialog, peer);
if (!ast_strlen_zero(port)) {
if ((portno = atoi(port))) {
dialog->sa.sin_port = dialog->recv.sin_port = htons(portno);
}
}
unref_peer(peer, "create_addr: unref peer from find_peer hashtab lookup");
return res;
}
if (dialog_initialize_rtp(dialog)) {
return -1;
}
ast_string_field_set(dialog, tohost, peername);
/* Get the outbound proxy information */
ref_proxy(dialog, obproxy_get(dialog, NULL));
if (sin) {
/* This address should be updated using dnsmgr */
memcpy(&dialog->sa.sin_addr, &sin->sin_addr, sizeof(dialog->sa.sin_addr));
if (!sin->sin_port) {
if (ast_strlen_zero(port) || sscanf(port, "%u", &portno) != 1) {
portno = (dialog->socket.type & SIP_TRANSPORT_TLS) ?
STANDARD_TLS_PORT : STANDARD_SIP_PORT;
}
} else {
portno = ntohs(sin->sin_port);
}
} else {
/* Let's see if we can find the host in DNS. First try DNS SRV records,
then hostname lookup */
/*! \todo Fix this function. When we ask for SRV, we should check all transports
In the future, we should first check NAPTR to find out transport preference
*/
hostn = peername;
/* Section 4.2 of RFC 3263 specifies that if a port number is specified, then
* an A record lookup should be used instead of SRV.
*/
if (!port && sip_cfg.srvlookup) {
char service[MAXHOSTNAMELEN];
int tportno;
snprintf(service, sizeof(service), "_sip._%s.%s", get_transport(dialog->socket.type), peername);
srv_ret = ast_get_srv(NULL, host, sizeof(host), &tportno, service);
if (srv_ret > 0) {
hostn = host;
portno = tportno;
}
}
if (!portno)
portno = port ? atoi(port) : (dialog->socket.type & SIP_TRANSPORT_TLS) ? STANDARD_TLS_PORT : STANDARD_SIP_PORT;
hp = ast_gethostbyname(hostn, &ahp);
if (!hp) {
ast_log(LOG_WARNING, "No such host: %s\n", peername);
return -1;
}
memcpy(&dialog->sa.sin_addr, hp->h_addr, sizeof(dialog->sa.sin_addr));
}
if (!dialog->socket.type)
dialog->socket.type = SIP_TRANSPORT_UDP;
if (!dialog->socket.port)
dialog->socket.port = bindaddr.sin_port;
dialog->sa.sin_port = htons(portno);
dialog->recv = dialog->sa;
return 0;
}
/*! \brief Scheduled congestion on a call.
* Only called by the scheduler, must return the reference when done.
*/
static int auto_congest(const void *arg)
{
struct sip_pvt *p = (struct sip_pvt *)arg;
sip_pvt_lock(p);
p->initid = -1; /* event gone, will not be rescheduled */
if (p->owner) {
/* XXX fails on possible deadlock */
if (!ast_channel_trylock(p->owner)) {
append_history(p, "Cong", "Auto-congesting (timer)");
ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
ast_channel_unlock(p->owner);
}
}
sip_pvt_unlock(p);
dialog_unref(p, "unreffing arg passed into auto_congest callback (p->initid)");
return 0;
}
/*! \brief Initiate SIP call from PBX
* used from the dial() application */
static int sip_call(struct ast_channel *ast, char *dest, int timeout)
{
int res;
struct sip_pvt *p = ast->tech_pvt; /* chan is locked, so the reference cannot go away */
struct varshead *headp;
struct ast_var_t *current;
const char *referer = NULL; /* SIP referrer */
if ((ast->_state != AST_STATE_DOWN) && (ast->_state != AST_STATE_RESERVED)) {
ast_log(LOG_WARNING, "sip_call called on %s, neither down nor reserved\n", ast->name);
return -1;
}
/* Check whether there is vxml_url, distinctive ring variables */
headp=&ast->varshead;
AST_LIST_TRAVERSE(headp, current, entries) {
/* Check whether there is a VXML_URL variable */
if (!p->options->vxml_url && !strcasecmp(ast_var_name(current), "VXML_URL")) {
p->options->vxml_url = ast_var_value(current);
} else if (!p->options->uri_options && !strcasecmp(ast_var_name(current), "SIP_URI_OPTIONS")) {
p->options->uri_options = ast_var_value(current);
} else if (!p->options->addsipheaders && !strncasecmp(ast_var_name(current), "SIPADDHEADER", strlen("SIPADDHEADER"))) {
/* Check whether there is a variable with a name starting with SIPADDHEADER */
p->options->addsipheaders = 1;
} else if (!strcasecmp(ast_var_name(current), "SIPFROMDOMAIN")) {
ast_string_field_set(p, fromdomain, ast_var_value(current));
} else if (!strcasecmp(ast_var_name(current), "SIPTRANSFER")) {
/* This is a transfered call */
p->options->transfer = 1;
} else if (!strcasecmp(ast_var_name(current), "SIPTRANSFER_REFERER")) {
/* This is the referrer */
referer = ast_var_value(current);
} else if (!strcasecmp(ast_var_name(current), "SIPTRANSFER_REPLACES")) {
/* We're replacing a call. */
p->options->replaces = ast_var_value(current);
}
}
res = 0;
ast_set_flag(&p->flags[0], SIP_OUTGOING);
if (p->options->transfer) {
char buf[SIPBUFSIZE/2];
if (referer) {
if (sipdebug)
ast_debug(3, "Call for %s transfered by %s\n", p->username, referer);
snprintf(buf, sizeof(buf)-1, "-> %s (via %s)", p->cid_name, referer);
} else
snprintf(buf, sizeof(buf)-1, "-> %s", p->cid_name);
ast_string_field_set(p, cid_name, buf);
}
ast_debug(1, "Outgoing Call for %s\n", p->username);
res = update_call_counter(p, INC_CALL_RINGING);
if (res == -1) {
ast->hangupcause = AST_CAUSE_USER_BUSY;
return res;
}
p->callingpres = ast->cid.cid_pres;
p->jointcapability = ast_translate_available_formats(p->capability, p->prefcodec);
p->jointnoncodeccapability = p->noncodeccapability;
/* If there are no audio formats left to offer, punt */
if (!(p->jointcapability & AST_FORMAT_AUDIO_MASK)) {
ast_log(LOG_WARNING, "No audio format found to offer. Cancelling call to %s\n", p->username);
res = -1;
} else {
int xmitres;
p->t38.jointcapability = p->t38.capability;
ast_debug(2, "Our T38 capability (%d), joint T38 capability (%d)\n", p->t38.capability, p->t38.jointcapability);
sip_pvt_lock(p);
xmitres = transmit_invite(p, SIP_INVITE, 1, 2);
sip_pvt_unlock(p);
if (xmitres == XMIT_ERROR)
return -1;
p->invitestate = INV_CALLING;
/* Initialize auto-congest time */
AST_SCHED_REPLACE_UNREF(p->initid, sched, p->timer_b, auto_congest, p,
dialog_unref(_data, "dialog ptr dec when SCHED_REPLACE del op succeeded"),
dialog_unref(p, "dialog ptr dec when SCHED_REPLACE add failed"),
dialog_ref(p, "dialog ptr inc when SCHED_REPLACE add succeeded") );
}
return res;
}
/*! \brief Destroy registry object
Objects created with the register= statement in static configuration */
static void sip_registry_destroy(struct sip_registry *reg)
{
/* Really delete */
ast_debug(3, "Destroying registry entry for %s@%s\n", reg->username, reg->hostname);
if (reg->call) {
/* Clear registry before destroying to ensure
we don't get reentered trying to grab the registry lock */
reg->call->registry = registry_unref(reg->call->registry, "destroy reg->call->registry");
ast_debug(3, "Destroying active SIP dialog for registry %s@%s\n", reg->username, reg->hostname);
dialog_unlink_all(reg->call, TRUE, TRUE);
reg->call = dialog_unref(reg->call, "unref reg->call");
/* reg->call = sip_destroy(reg->call); */
}
AST_SCHED_DEL(sched, reg->expire);
AST_SCHED_DEL(sched, reg->timeout);
ast_string_field_free_memory(reg);
ast_atomic_fetchadd_int(&regobjs, -1);
ast_dnsmgr_release(reg->dnsmgr);
ast_free(reg);
}
/*! \brief Destroy MWI subscription object */
static void sip_subscribe_mwi_destroy(struct sip_subscription_mwi *mwi)
{
if (mwi->call) {
mwi->call->mwi = NULL;
sip_destroy(mwi->call);
}
AST_SCHED_DEL(sched, mwi->resub);
ast_string_field_free_memory(mwi);
ast_dnsmgr_release(mwi->dnsmgr);
ast_free(mwi);
}
/*! \brief Execute destruction of SIP dialog structure, release memory */
static void __sip_destroy(struct sip_pvt *p, int lockowner, int lockdialoglist)
{
struct sip_request *req;
if (sip_debug_test_pvt(p))
ast_verbose("Really destroying SIP dialog '%s' Method: %s\n", p->callid, sip_methods[p->method].text);
if (ast_test_flag(&p->flags[0], SIP_INC_COUNT) || ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD)) {
update_call_counter(p, DEC_CALL_LIMIT);
ast_debug(2, "This call did not properly clean up call limits. Call ID %s\n", p->callid);
}
/* Unlink us from the owner if we have one */
if (p->owner) {
if (lockowner)
ast_channel_lock(p->owner);
if (option_debug)
ast_log(LOG_DEBUG, "Detaching from %s\n", p->owner->name);
p->owner->tech_pvt = NULL;
/* Make sure that the channel knows its backend is going away */
p->owner->_softhangup |= AST_SOFTHANGUP_DEV;
if (lockowner)
ast_channel_unlock(p->owner);
/* Give the channel a chance to react before deallocation */
usleep(1);
}
/* Remove link from peer to subscription of MWI */
if (p->relatedpeer && p->relatedpeer->mwipvt)
p->relatedpeer->mwipvt = dialog_unref(p->relatedpeer->mwipvt, "delete ->relatedpeer->mwipvt");
if (p->relatedpeer && p->relatedpeer->call == p)
p->relatedpeer->call = dialog_unref(p->relatedpeer->call, "unset the relatedpeer->call field in tandem with relatedpeer field itself");
if (p->relatedpeer)
p->relatedpeer = unref_peer(p->relatedpeer,"unsetting a dialog relatedpeer field in sip_destroy");
if (p->registry) {
if (p->registry->call == p)
p->registry->call = dialog_unref(p->registry->call, "nulling out the registry's call dialog field in unlink_all");
p->registry = registry_unref(p->registry, "delete p->registry");
}
if (p->mwi) {
p->mwi->call = NULL;
}
if (dumphistory)
sip_dump_history(p);
if (p->options)
ast_free(p->options);
if (p->notify_headers) {
ast_variables_destroy(p->notify_headers);
p->notify_headers = NULL;
}
if (p->rtp) {
ast_rtp_instance_destroy(p->rtp);
}
if (p->vrtp) {
ast_rtp_instance_destroy(p->vrtp);
}
if (p->trtp) {
ast_rtp_instance_destroy(p->trtp);
}
if (p->udptl)
ast_udptl_destroy(p->udptl);
if (p->refer)
ast_free(p->refer);
if (p->route) {
free_old_route(p->route);
p->route = NULL;
}
if (p->initreq.data)
ast_free(p->initreq.data);
/* Destroy Session-Timers if allocated */
if (p->stimer) {
if (p->stimer->st_active == TRUE && p->stimer->st_schedid > -1) {
AST_SCHED_DEL_UNREF(sched, p->stimer->st_schedid,
dialog_unref(p, "removing session timer ref"));
}
ast_free(p->stimer);
p->stimer = NULL;
}
/* Clear history */
if (p->history) {
struct sip_history *hist;
while ( (hist = AST_LIST_REMOVE_HEAD(p->history, list)) ) {
ast_free(hist);
p->history_entries--;
}
ast_free(p->history);
p->history = NULL;
}
while ((req = AST_LIST_REMOVE_HEAD(&p->request_queue, next))) {
ast_free(req);
}
if (p->chanvars) {
ast_variables_destroy(p->chanvars);
p->chanvars = NULL;
}
ast_string_field_free_memory(p);
if (p->socket.tcptls_session) {
ao2_ref(p->socket.tcptls_session, -1);
p->socket.tcptls_session = NULL;
}
}
/*! \brief update_call_counter: Handle call_limit for SIP devices
* Setting a call-limit will cause calls above the limit not to be accepted.
*
* Remember that for a type=friend, there's one limit for the user and
* another for the peer, not a combined call limit.
* This will cause unexpected behaviour in subscriptions, since a "friend"
* is *two* devices in Asterisk, not one.
*
* Thought: For realtime, we should probably update storage with inuse counter...
*
* \return 0 if call is ok (no call limit, below threshold)
* -1 on rejection of call
*
*/
static int update_call_counter(struct sip_pvt *fup, int event)
{
char name[256];
int *inuse = NULL, *call_limit = NULL, *inringing = NULL;
int outgoing = fup->outgoing_call;
struct sip_peer *p = NULL;
ast_debug(3, "Updating call counter for %s call\n", outgoing ? "outgoing" : "incoming");
/* Test if we need to check call limits, in order to avoid
realtime lookups if we do not need it */
if (!ast_test_flag(&fup->flags[0], SIP_CALL_LIMIT) && !ast_test_flag(&fup->flags[1], SIP_PAGE2_CALL_ONHOLD))
return 0;
ast_copy_string(name, fup->username, sizeof(name));
/* Check the list of devices */
if ((p = find_peer(ast_strlen_zero(fup->peername) ? name : fup->peername, NULL, TRUE, FINDALLDEVICES, FALSE))) {
inuse = &p->inUse;
call_limit = &p->call_limit;
inringing = &p->inRinging;
ast_copy_string(name, fup->peername, sizeof(name));
}
if (!p) {
ast_debug(2, "%s is not a local device, no call limit\n", name);
return 0;
}
switch(event) {
/* incoming and outgoing affects the inUse counter */
case DEC_CALL_LIMIT:
/* Decrement inuse count if applicable */
if (inuse) {
sip_pvt_lock(fup);
ao2_lock(p);
if (*inuse > 0) {
if (ast_test_flag(&fup->flags[0], SIP_INC_COUNT)) {
(*inuse)--;
ast_clear_flag(&fup->flags[0], SIP_INC_COUNT);
}
} else {
*inuse = 0;
}
ao2_unlock(p);
sip_pvt_unlock(fup);
}
/* Decrement ringing count if applicable */
if (inringing) {
sip_pvt_lock(fup);
ao2_lock(p);
if (*inringing > 0) {
if (ast_test_flag(&fup->flags[0], SIP_INC_RINGING)) {
(*inringing)--;
ast_clear_flag(&fup->flags[0], SIP_INC_RINGING);
}
} else {
*inringing = 0;
}
ao2_unlock(p);
sip_pvt_unlock(fup);
}
/* Decrement onhold count if applicable */
sip_pvt_lock(fup);
ao2_lock(p);
if (ast_test_flag(&fup->flags[1], SIP_PAGE2_CALL_ONHOLD) && sip_cfg.notifyhold) {
ast_clear_flag(&fup->flags[1], SIP_PAGE2_CALL_ONHOLD);
ao2_unlock(p);
sip_pvt_unlock(fup);
sip_peer_hold(fup, FALSE);
} else {
ao2_unlock(p);
sip_pvt_unlock(fup);
}
if (sipdebug)
ast_debug(2, "Call %s %s '%s' removed from call limit %d\n", outgoing ? "to" : "from", "peer", name, *call_limit);
break;
case INC_CALL_RINGING:
case INC_CALL_LIMIT:
/* If call limit is active and we have reached the limit, reject the call */
if (*call_limit > 0 ) {
if (*inuse >= *call_limit) {
ast_log(LOG_NOTICE, "Call %s %s '%s' rejected due to usage limit of %d\n", outgoing ? "to" : "from", "peer", name, *call_limit);
unref_peer(p, "update_call_counter: unref peer p, call limit exceeded");
return -1;
}
}
if (inringing && (event == INC_CALL_RINGING)) {
sip_pvt_lock(fup);
ao2_lock(p);
if (!ast_test_flag(&fup->flags[0], SIP_INC_RINGING)) {
(*inringing)++;
ast_set_flag(&fup->flags[0], SIP_INC_RINGING);
}
ao2_unlock(p);
sip_pvt_unlock(fup);
}
if (inuse) {
sip_pvt_lock(fup);
ao2_lock(p);
if (!ast_test_flag(&fup->flags[0], SIP_INC_COUNT)) {
(*inuse)++;
ast_set_flag(&fup->flags[0], SIP_INC_COUNT);
}
ao2_unlock(p);
sip_pvt_unlock(fup);
}
if (sipdebug) {
ast_debug(2, "Call %s %s '%s' is %d out of %d\n", outgoing ? "to" : "from", "peer", name, *inuse, *call_limit);
}
break;
case DEC_CALL_RINGING:
if (inringing) {
sip_pvt_lock(fup);
ao2_lock(p);
if (ast_test_flag(&fup->flags[0], SIP_INC_RINGING)) {
if (*inringing > 0) {
(*inringing)--;
}
ast_clear_flag(&fup->flags[0], SIP_INC_RINGING);
}
ao2_unlock(p);
sip_pvt_unlock(fup);
}
break;
default:
ast_log(LOG_ERROR, "update_call_counter(%s, %d) called with no event!\n", name, event);
}
if (p) {
ast_devstate_changed(AST_DEVICE_UNKNOWN, "SIP/%s", p->name);
unref_peer(p, "update_call_counter: unref_peer from call counter");
}
return 0;
}
static void sip_destroy_fn(void *p)
{
sip_destroy(p);
}
/*! \brief Destroy SIP call structure.
* Make it return NULL so the caller can do things like
* foo = sip_destroy(foo);
* and reduce the chance of bugs due to dangling pointers.
*/
static struct sip_pvt * sip_destroy(struct sip_pvt *p)
{
ast_debug(3, "Destroying SIP dialog %s\n", p->callid);
__sip_destroy(p, TRUE, TRUE);
return NULL;
}
/*! \brief Convert SIP hangup causes to Asterisk hangup causes */
static int hangup_sip2cause(int cause)
{
/* Possible values taken from causes.h */
switch(cause) {
case 401: /* Unauthorized */
return AST_CAUSE_CALL_REJECTED;
case 403: /* Not found */
return AST_CAUSE_CALL_REJECTED;
case 404: /* Not found */
return AST_CAUSE_UNALLOCATED;
case 405: /* Method not allowed */
return AST_CAUSE_INTERWORKING;
case 407: /* Proxy authentication required */
return AST_CAUSE_CALL_REJECTED;
case 408: /* No reaction */
return AST_CAUSE_NO_USER_RESPONSE;
case 409: /* Conflict */
return AST_CAUSE_NORMAL_TEMPORARY_FAILURE;
case 410: /* Gone */
return AST_CAUSE_NUMBER_CHANGED;
case 411: /* Length required */
return AST_CAUSE_INTERWORKING;
case 413: /* Request entity too large */
return AST_CAUSE_INTERWORKING;
case 414: /* Request URI too large */
return AST_CAUSE_INTERWORKING;
case 415: /* Unsupported media type */
return AST_CAUSE_INTERWORKING;
case 420: /* Bad extension */
return AST_CAUSE_NO_ROUTE_DESTINATION;
case 480: /* No answer */
return AST_CAUSE_NO_ANSWER;
case 481: /* No answer */
return AST_CAUSE_INTERWORKING;
case 482: /* Loop detected */
return AST_CAUSE_INTERWORKING;
case 483: /* Too many hops */
return AST_CAUSE_NO_ANSWER;
case 484: /* Address incomplete */
return AST_CAUSE_INVALID_NUMBER_FORMAT;
case 485: /* Ambiguous */
return AST_CAUSE_UNALLOCATED;
case 486: /* Busy everywhere */
return AST_CAUSE_BUSY;
case 487: /* Request terminated */
return AST_CAUSE_INTERWORKING;
case 488: /* No codecs approved */
return AST_CAUSE_BEARERCAPABILITY_NOTAVAIL;
case 491: /* Request pending */
return AST_CAUSE_INTERWORKING;
case 493: /* Undecipherable */
return AST_CAUSE_INTERWORKING;
case 500: /* Server internal failure */
return AST_CAUSE_FAILURE;
case 501: /* Call rejected */
return AST_CAUSE_FACILITY_REJECTED;
case 502:
return AST_CAUSE_DESTINATION_OUT_OF_ORDER;
case 503: /* Service unavailable */
return AST_CAUSE_CONGESTION;
case 504: /* Gateway timeout */
return AST_CAUSE_RECOVERY_ON_TIMER_EXPIRE;
case 505: /* SIP version not supported */
return AST_CAUSE_INTERWORKING;
case 600: /* Busy everywhere */
return AST_CAUSE_USER_BUSY;
case 603: /* Decline */
return AST_CAUSE_CALL_REJECTED;
case 604: /* Does not exist anywhere */
return AST_CAUSE_UNALLOCATED;
case 606: /* Not acceptable */
return AST_CAUSE_BEARERCAPABILITY_NOTAVAIL;
default:
return AST_CAUSE_NORMAL;
}
/* Never reached */
return 0;
}
/*! \brief Convert Asterisk hangup causes to SIP codes
\verbatim
Possible values from causes.h
AST_CAUSE_NOTDEFINED AST_CAUSE_NORMAL AST_CAUSE_BUSY
AST_CAUSE_FAILURE AST_CAUSE_CONGESTION AST_CAUSE_UNALLOCATED
In addition to these, a lot of PRI codes is defined in causes.h
...should we take care of them too ?
Quote RFC 3398
ISUP Cause value SIP response
---------------- ------------
1 unallocated number 404 Not Found
2 no route to network 404 Not found
3 no route to destination 404 Not found
16 normal call clearing --- (*)
17 user busy 486 Busy here
18 no user responding 408 Request Timeout
19 no answer from the user 480 Temporarily unavailable
20 subscriber absent 480 Temporarily unavailable
21 call rejected 403 Forbidden (+)
22 number changed (w/o diagnostic) 410 Gone
22 number changed (w/ diagnostic) 301 Moved Permanently
23 redirection to new destination 410 Gone
26 non-selected user clearing 404 Not Found (=)
27 destination out of order 502 Bad Gateway
28 address incomplete 484 Address incomplete
29 facility rejected 501 Not implemented
31 normal unspecified 480 Temporarily unavailable
\endverbatim
*/
static const char *hangup_cause2sip(int cause)
{
switch (cause) {
case AST_CAUSE_UNALLOCATED: /* 1 */
case AST_CAUSE_NO_ROUTE_DESTINATION: /* 3 IAX2: Can't find extension in context */
case AST_CAUSE_NO_ROUTE_TRANSIT_NET: /* 2 */
return "404 Not Found";
case AST_CAUSE_CONGESTION: /* 34 */
case AST_CAUSE_SWITCH_CONGESTION: /* 42 */
return "503 Service Unavailable";
case AST_CAUSE_NO_USER_RESPONSE: /* 18 */
return "408 Request Timeout";
case AST_CAUSE_NO_ANSWER: /* 19 */
case AST_CAUSE_UNREGISTERED: /* 20 */
return "480 Temporarily unavailable";
case AST_CAUSE_CALL_REJECTED: /* 21 */
return "403 Forbidden";
case AST_CAUSE_NUMBER_CHANGED: /* 22 */
return "410 Gone";
case AST_CAUSE_NORMAL_UNSPECIFIED: /* 31 */
return "480 Temporarily unavailable";
case AST_CAUSE_INVALID_NUMBER_FORMAT:
return "484 Address incomplete";
case AST_CAUSE_USER_BUSY:
return "486 Busy here";
case AST_CAUSE_FAILURE:
return "500 Server internal failure";
case AST_CAUSE_FACILITY_REJECTED: /* 29 */
return "501 Not Implemented";
case AST_CAUSE_CHAN_NOT_IMPLEMENTED:
return "503 Service Unavailable";
/* Used in chan_iax2 */
case AST_CAUSE_DESTINATION_OUT_OF_ORDER:
return "502 Bad Gateway";
case AST_CAUSE_BEARERCAPABILITY_NOTAVAIL: /* Can't find codec to connect to host */
return "488 Not Acceptable Here";
case AST_CAUSE_NOTDEFINED:
default:
ast_debug(1, "AST hangup cause %d (no match found in SIP)\n", cause);
return NULL;
}
/* Never reached */
return 0;
}
/*! \brief sip_hangup: Hangup SIP call
* Part of PBX interface, called from ast_hangup */
static int sip_hangup(struct ast_channel *ast)
{
struct sip_pvt *p = ast->tech_pvt;
int needcancel = FALSE;
int needdestroy = 0;
struct ast_channel *oldowner = ast;
if (!p) {
ast_debug(1, "Asked to hangup channel that was not connected\n");
return 0;
}
if (ast_test_flag(ast, AST_FLAG_ANSWERED_ELSEWHERE) || ast->hangupcause == AST_CAUSE_ANSWERED_ELSEWHERE) {
ast_debug(1, "This call was answered elsewhere");
if (ast->hangupcause == AST_CAUSE_ANSWERED_ELSEWHERE) {
ast_debug(1, "####### It's the cause code, buddy. The cause code!!!\n");
}
append_history(p, "Cancel", "Call answered elsewhere");
p->answered_elsewhere = TRUE;
}
/* Store hangupcause locally in PVT so we still have it before disconnect */
if (p->owner)
p->hangupcause = p->owner->hangupcause;
if (ast_test_flag(&p->flags[0], SIP_DEFER_BYE_ON_TRANSFER)) {
if (ast_test_flag(&p->flags[0], SIP_INC_COUNT) || ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD)) {
if (sipdebug)
ast_debug(1, "update_call_counter(%s) - decrement call limit counter on hangup\n", p->username);
update_call_counter(p, DEC_CALL_LIMIT);
}
ast_debug(4, "SIP Transfer: Not hanging up right now... Rescheduling hangup for %s.\n", p->callid);
if (p->autokillid > -1 && sip_cancel_destroy(p))
ast_log(LOG_WARNING, "Unable to cancel SIP destruction. Expect bad things.\n");
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
ast_clear_flag(&p->flags[0], SIP_DEFER_BYE_ON_TRANSFER); /* Really hang up next time */
p->needdestroy = 0;
p->owner->tech_pvt = dialog_unref(p->owner->tech_pvt, "unref p->owner->tech_pvt");
sip_pvt_lock(p);
p->owner = NULL; /* Owner will be gone after we return, so take it away */
sip_pvt_unlock(p);
return 0;
}
if (ast_test_flag(ast, AST_FLAG_ZOMBIE)) {
if (p->refer)
ast_debug(1, "SIP Transfer: Hanging up Zombie channel %s after transfer ... Call-ID: %s\n", ast->name, p->callid);
else
ast_debug(1, "Hanging up zombie call. Be scared.\n");
} else
ast_debug(1, "Hangup call %s, SIP callid %s\n", ast->name, p->callid);
sip_pvt_lock(p);
if (ast_test_flag(&p->flags[0], SIP_INC_COUNT) || ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD)) {
if (sipdebug)
ast_debug(1, "update_call_counter(%s) - decrement call limit counter on hangup\n", p->username);
update_call_counter(p, DEC_CALL_LIMIT);
}
/* Determine how to disconnect */
if (p->owner != ast) {
ast_log(LOG_WARNING, "Huh? We aren't the owner? Can't hangup call.\n");
sip_pvt_unlock(p);
return 0;
}
/* If the call is not UP, we need to send CANCEL instead of BYE */
/* In case of re-invites, the call might be UP even though we have an incomplete invite transaction */
if (p->invitestate < INV_COMPLETED && p->owner->_state != AST_STATE_UP) {
needcancel = TRUE;
ast_debug(4, "Hanging up channel in state %s (not UP)\n", ast_state2str(ast->_state));
}
stop_media_flows(p); /* Immediately stop RTP, VRTP and UDPTL as applicable */
append_history(p, needcancel ? "Cancel" : "Hangup", "Cause %s", p->owner ? ast_cause2str(p->hangupcause) : "Unknown");
/* Disconnect */
if (p->vad)
ast_dsp_free(p->vad);
p->owner = NULL;
ast->tech_pvt = dialog_unref(ast->tech_pvt, "unref ast->tech_pvt");
ast_module_unref(ast_module_info->self);
/* Do not destroy this pvt until we have timeout or
get an answer to the BYE or INVITE/CANCEL
If we get no answer during retransmit period, drop the call anyway.
(Sorry, mother-in-law, you can't deny a hangup by sending
603 declined to BYE...)
*/
if (p->alreadygone)
needdestroy = 1; /* Set destroy flag at end of this function */
else if (p->invitestate != INV_CALLING)
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
/* Start the process if it's not already started */
if (!p->alreadygone && p->initreq.data && !ast_strlen_zero(p->initreq.data->str)) {
if (needcancel) { /* Outgoing call, not up */
if (ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
/* stop retransmitting an INVITE that has not received a response */
__sip_pretend_ack(p);
/* if we can't send right now, mark it pending */
if (p->invitestate == INV_CALLING) {
/* We can't send anything in CALLING state */
ast_set_flag(&p->flags[0], SIP_PENDINGBYE);
/* Do we need a timer here if we don't hear from them at all? */
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
append_history(p, "DELAY", "Not sending cancel, waiting for timeout");
} else {
p->invitestate = INV_CANCELLED;
/* Send a new request: CANCEL */
transmit_request(p, SIP_CANCEL, p->lastinvite, XMIT_RELIABLE, FALSE);
/* Actually don't destroy us yet, wait for the 487 on our original
INVITE, but do set an autodestruct just in case we never get it. */
needdestroy = 0;
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
}
} else { /* Incoming call, not up */
const char *res;
if (p->hangupcause && (res = hangup_cause2sip(p->hangupcause)))
transmit_response_reliable(p, res, &p->initreq);
else
transmit_response_reliable(p, "603 Declined", &p->initreq);
p->invitestate = INV_TERMINATED;
}
} else { /* Call is in UP state, send BYE */
if (p->stimer->st_active == TRUE) {
stop_session_timer(p);
}
if (!p->pendinginvite) {
struct ast_channel *bridge = ast_bridged_channel(oldowner);
char quality_buf[AST_MAX_USER_FIELD], *quality;
if (p->rtp) {
ast_rtp_instance_set_stats_vars(oldowner, p->rtp);
}
if (bridge) {
struct sip_pvt *q = bridge->tech_pvt;
if (IS_SIP_TECH(bridge->tech) && q && q->rtp) {
ast_rtp_instance_set_stats_vars(bridge, q->rtp);
}
}
if (p->do_history || oldowner) {
if (p->rtp && (quality = ast_rtp_instance_get_quality(p->rtp, AST_RTP_INSTANCE_STAT_FIELD_QUALITY, quality_buf, sizeof(quality_buf)))) {
if (p->do_history) {
append_history(p, "RTCPaudio", "Quality:%s", quality);
}
if (oldowner) {
pbx_builtin_setvar_helper(oldowner, "RTPAUDIOQOS", quality);
}
}
if (p->vrtp && (quality = ast_rtp_instance_get_quality(p->vrtp, AST_RTP_INSTANCE_STAT_FIELD_QUALITY, quality_buf, sizeof(quality_buf)))) {
if (p->do_history) {
append_history(p, "RTCPvideo", "Quality:%s", quality);
}
if (oldowner) {
pbx_builtin_setvar_helper(oldowner, "RTPVIDEOQOS", quality);
}
}
if (p->trtp && (quality = ast_rtp_instance_get_quality(p->trtp, AST_RTP_INSTANCE_STAT_FIELD_QUALITY, quality_buf, sizeof(quality_buf)))) {
if (p->do_history) {
append_history(p, "RTCPtext", "Quality:%s", quality);
}
if (oldowner) {
pbx_builtin_setvar_helper(oldowner, "RTPTEXTQOS", quality);
}
}
}
/* Send a hangup */
transmit_request_with_auth(p, SIP_BYE, 0, XMIT_RELIABLE, 1);
} else {
/* Note we will need a BYE when this all settles out
but we can't send one while we have "INVITE" outstanding. */
ast_set_flag(&p->flags[0], SIP_PENDINGBYE);
ast_clear_flag(&p->flags[0], SIP_NEEDREINVITE);
AST_SCHED_DEL_UNREF(sched, p->waitid, dialog_unref(p, "when you delete the waitid sched, you should dec the refcount for the stored dialog ptr"));
if (sip_cancel_destroy(p))
ast_log(LOG_WARNING, "Unable to cancel SIP destruction. Expect bad things.\n");
}
}
}
if (needdestroy) {
pvt_set_needdestroy(p, "hangup");
}
sip_pvt_unlock(p);
return 0;
}
/*! \brief Try setting codec suggested by the SIP_CODEC channel variable */
static void try_suggested_sip_codec(struct sip_pvt *p)
{
int fmt;
const char *codec;
if (p->outgoing_call) {
codec = pbx_builtin_getvar_helper(p->owner, "SIP_CODEC_OUTBOUND");
} else if (!(codec = pbx_builtin_getvar_helper(p->owner, "SIP_CODEC_INBOUND"))) {
codec = pbx_builtin_getvar_helper(p->owner, "SIP_CODEC");
}
if (!codec)
return;
fmt = ast_getformatbyname(codec);
if (fmt) {
ast_log(LOG_NOTICE, "Changing codec to '%s' for this call because of ${SIP_CODEC} variable\n", codec);
if (p->jointcapability & fmt) {
p->jointcapability &= fmt;
p->capability &= fmt;
} else
ast_log(LOG_NOTICE, "Ignoring ${SIP_CODEC} variable because it is not shared by both ends.\n");
} else
ast_log(LOG_NOTICE, "Ignoring ${SIP_CODEC} variable because of unrecognized/not configured codec (check allow/disallow in sip.conf): %s\n", codec);
return;
}
/*! \brief sip_answer: Answer SIP call , send 200 OK on Invite
* Part of PBX interface */
static int sip_answer(struct ast_channel *ast)
{
int res = 0;
struct sip_pvt *p = ast->tech_pvt;
sip_pvt_lock(p);
if (ast->_state != AST_STATE_UP) {
try_suggested_sip_codec(p);
ast_setstate(ast, AST_STATE_UP);
ast_debug(1, "SIP answering channel: %s\n", ast->name);
if (p->t38.state == T38_PEER_DIRECT) {
change_t38_state(p, T38_ENABLED);
}
ast_rtp_instance_new_source(p->rtp);
res = transmit_response_with_sdp(p, "200 OK", &p->initreq, XMIT_CRITICAL, FALSE, TRUE);
ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
}
sip_pvt_unlock(p);
return res;
}
/*! \brief Send frame to media channel (rtp) */
static int sip_write(struct ast_channel *ast, struct ast_frame *frame)
{
struct sip_pvt *p = ast->tech_pvt;
int res = 0;
switch (frame->frametype) {
case AST_FRAME_VOICE:
if (!(frame->subclass & ast->nativeformats)) {
char s1[512], s2[512], s3[512];
ast_log(LOG_WARNING, "Asked to transmit frame type %d, while native formats is %s(%d) read/write = %s(%d)/%s(%d)\n",
frame->subclass,
ast_getformatname_multiple(s1, sizeof(s1) - 1, ast->nativeformats & AST_FORMAT_AUDIO_MASK),
ast->nativeformats & AST_FORMAT_AUDIO_MASK,
ast_getformatname_multiple(s2, sizeof(s2) - 1, ast->readformat),
ast->readformat,
ast_getformatname_multiple(s3, sizeof(s3) - 1, ast->writeformat),
ast->writeformat);
return 0;
}
if (p) {
sip_pvt_lock(p);
if (p->rtp) {
/* If channel is not up, activate early media session */
if ((ast->_state != AST_STATE_UP) &&
!ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT) &&
!ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
ast_rtp_instance_new_source(p->rtp);
p->invitestate = INV_EARLY_MEDIA;
transmit_response_with_sdp(p, "183 Session Progress", &p->initreq, XMIT_UNRELIABLE, FALSE, FALSE);
ast_set_flag(&p->flags[0], SIP_PROGRESS_SENT);
} else if (p->t38.state == T38_ENABLED && !p->t38.direct) {
change_t38_state(p, T38_DISABLED);
transmit_reinvite_with_sdp(p, FALSE, FALSE);
} else {
p->lastrtptx = time(NULL);
res = ast_rtp_instance_write(p->rtp, frame);
}
}
sip_pvt_unlock(p);
}
break;
case AST_FRAME_VIDEO:
if (p) {
sip_pvt_lock(p);
if (p->vrtp) {
/* Activate video early media */
if ((ast->_state != AST_STATE_UP) &&
!ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT) &&
!ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
p->invitestate = INV_EARLY_MEDIA;
transmit_response_with_sdp(p, "183 Session Progress", &p->initreq, XMIT_UNRELIABLE, FALSE, FALSE);
ast_set_flag(&p->flags[0], SIP_PROGRESS_SENT);
}
p->lastrtptx = time(NULL);
res = ast_rtp_instance_write(p->vrtp, frame);
}
sip_pvt_unlock(p);
}
break;
case AST_FRAME_TEXT:
if (p) {
sip_pvt_lock(p);
if (p->red) {
ast_rtp_red_buffer(p->trtp, frame);
} else {
if (p->trtp) {
/* Activate text early media */
if ((ast->_state != AST_STATE_UP) &&
!ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT) &&
!ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
p->invitestate = INV_EARLY_MEDIA;
transmit_response_with_sdp(p, "183 Session Progress", &p->initreq, XMIT_UNRELIABLE, FALSE, FALSE);
ast_set_flag(&p->flags[0], SIP_PROGRESS_SENT);
}
p->lastrtptx = time(NULL);
res = ast_rtp_instance_write(p->trtp, frame);
}
}
sip_pvt_unlock(p);
}
break;
case AST_FRAME_IMAGE:
return 0;
break;
case AST_FRAME_MODEM:
if (p) {
sip_pvt_lock(p);
/* UDPTL requires two-way communication, so early media is not needed here.
we simply forget the frames if we get modem frames before the bridge is up.
Fax will re-transmit.
*/
if (ast->_state == AST_STATE_UP) {
if (ast_test_flag(&p->flags[1], SIP_PAGE2_T38SUPPORT) && p->t38.state == T38_DISABLED) {
if (!p->pendinginvite) {
change_t38_state(p, T38_LOCAL_REINVITE);
transmit_reinvite_with_sdp(p, TRUE, FALSE);
}
} else if (p->udptl && p->t38.state == T38_ENABLED) {
res = ast_udptl_write(p->udptl, frame);
}
}
sip_pvt_unlock(p);
}
break;
default:
ast_log(LOG_WARNING, "Can't send %d type frames with SIP write\n", frame->frametype);
return 0;
}
return res;
}
/*! \brief sip_fixup: Fix up a channel: If a channel is consumed, this is called.
Basically update any ->owner links */
static int sip_fixup(struct ast_channel *oldchan, struct ast_channel *newchan)
{
int ret = -1;
struct sip_pvt *p;
if (newchan && ast_test_flag(newchan, AST_FLAG_ZOMBIE))
ast_debug(1, "New channel is zombie\n");
if (oldchan && ast_test_flag(oldchan, AST_FLAG_ZOMBIE))
ast_debug(1, "Old channel is zombie\n");
if (!newchan || !newchan->tech_pvt) {
if (!newchan)
ast_log(LOG_WARNING, "No new channel! Fixup of %s failed.\n", oldchan->name);
else
ast_log(LOG_WARNING, "No SIP tech_pvt! Fixup of %s failed.\n", oldchan->name);
return -1;
}
p = newchan->tech_pvt;
sip_pvt_lock(p);
append_history(p, "Masq", "Old channel: %s\n", oldchan->name);
append_history(p, "Masq (cont)", "...new owner: %s\n", newchan->name);
if (p->owner != oldchan)
ast_log(LOG_WARNING, "old channel wasn't %p but was %p\n", oldchan, p->owner);
else {
p->owner = newchan;
/* Re-invite RTP back to Asterisk. Needed if channel is masqueraded out of a native
RTP bridge (i.e., RTP not going through Asterisk): RTP bridge code might not be
able to do this if the masquerade happens before the bridge breaks (e.g., AMI
redirect of both channels). Note that a channel can not be masqueraded *into*
a native bridge. So there is no danger that this breaks a native bridge that
should stay up. */
sip_set_rtp_peer(newchan, NULL, NULL, 0, 0, 0);
ret = 0;
}
ast_debug(3, "SIP Fixup: New owner for dialogue %s: %s (Old parent: %s)\n", p->callid, p->owner->name, oldchan->name);
sip_pvt_unlock(p);
return ret;
}
static int sip_senddigit_begin(struct ast_channel *ast, char digit)
{
struct sip_pvt *p = ast->tech_pvt;
int res = 0;
sip_pvt_lock(p);
switch (ast_test_flag(&p->flags[0], SIP_DTMF)) {
case SIP_DTMF_INBAND:
if (p->rtp && ast_rtp_instance_dtmf_mode_get(p->rtp) == AST_RTP_DTMF_MODE_INBAND) {
ast_rtp_instance_dtmf_begin(p->rtp, digit);
} else {
res = -1; /* Tell Asterisk to generate inband indications */
}
break;
case SIP_DTMF_RFC2833:
if (p->rtp)
ast_rtp_instance_dtmf_begin(p->rtp, digit);
break;
default:
break;
}
sip_pvt_unlock(p);
return res;
}
/*! \brief Send DTMF character on SIP channel
within one call, we're able to transmit in many methods simultaneously */
static int sip_senddigit_end(struct ast_channel *ast, char digit, unsigned int duration)
{
struct sip_pvt *p = ast->tech_pvt;
int res = 0;
sip_pvt_lock(p);
switch (ast_test_flag(&p->flags[0], SIP_DTMF)) {
case SIP_DTMF_INFO:
case SIP_DTMF_SHORTINFO:
transmit_info_with_digit(p, digit, duration);
break;
case SIP_DTMF_RFC2833:
if (p->rtp)
ast_rtp_instance_dtmf_end(p->rtp, digit);
break;
case SIP_DTMF_INBAND:
if (p->rtp && ast_rtp_instance_dtmf_mode_get(p->rtp) == AST_RTP_DTMF_MODE_INBAND) {
ast_rtp_instance_dtmf_end(p->rtp, digit);
} else {
res = -1; /* Tell Asterisk to stop inband indications */
}
break;
}
sip_pvt_unlock(p);
return res;
}
/*! \brief Transfer SIP call */
static int sip_transfer(struct ast_channel *ast, const char *dest)
{
struct sip_pvt *p = ast->tech_pvt;
int res;
if (dest == NULL) /* functions below do not take a NULL */
dest = "";
sip_pvt_lock(p);
if (ast->_state == AST_STATE_RING)
res = sip_sipredirect(p, dest);
else
res = transmit_refer(p, dest);
sip_pvt_unlock(p);
return res;
}
/*! \brief Play indication to user
* With SIP a lot of indications is sent as messages, letting the device play
the indication - busy signal, congestion etc
\return -1 to force ast_indicate to send indication in audio, 0 if SIP can handle the indication by sending a message
*/
static int sip_indicate(struct ast_channel *ast, int condition, const void *data, size_t datalen)
{
struct sip_pvt *p = ast->tech_pvt;
int res = 0;
sip_pvt_lock(p);
switch(condition) {
case AST_CONTROL_RINGING:
if (ast->_state == AST_STATE_RING) {
p->invitestate = INV_EARLY_MEDIA;
if (!ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT) ||
(ast_test_flag(&p->flags[0], SIP_PROG_INBAND) == SIP_PROG_INBAND_NEVER)) {
/* Send 180 ringing if out-of-band seems reasonable */
transmit_response(p, "180 Ringing", &p->initreq);
ast_set_flag(&p->flags[0], SIP_RINGING);
if (ast_test_flag(&p->flags[0], SIP_PROG_INBAND) != SIP_PROG_INBAND_YES)
break;
} else {
/* Well, if it's not reasonable, just send in-band */
}
}
res = -1;
break;
case AST_CONTROL_BUSY:
if (ast->_state != AST_STATE_UP) {
transmit_response_reliable(p, "486 Busy Here", &p->initreq);
p->invitestate = INV_COMPLETED;
sip_alreadygone(p);
ast_softhangup_nolock(ast, AST_SOFTHANGUP_DEV);
break;
}
res = -1;
break;
case AST_CONTROL_CONGESTION:
if (ast->_state != AST_STATE_UP) {
transmit_response_reliable(p, "503 Service Unavailable", &p->initreq);
p->invitestate = INV_COMPLETED;
sip_alreadygone(p);
ast_softhangup_nolock(ast, AST_SOFTHANGUP_DEV);
break;
}
res = -1;
break;
case AST_CONTROL_PROCEEDING:
if ((ast->_state != AST_STATE_UP) &&
!ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT) &&
!ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
transmit_response(p, "100 Trying", &p->initreq);
p->invitestate = INV_PROCEEDING;
break;
}
res = -1;
break;
case AST_CONTROL_PROGRESS:
if ((ast->_state != AST_STATE_UP) &&
!ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT) &&
!ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
p->invitestate = INV_EARLY_MEDIA;
transmit_response_with_sdp(p, "183 Session Progress", &p->initreq, XMIT_UNRELIABLE, FALSE, FALSE);
ast_set_flag(&p->flags[0], SIP_PROGRESS_SENT);
break;
}
res = -1;
break;
case AST_CONTROL_HOLD:
ast_rtp_instance_new_source(p->rtp);
ast_moh_start(ast, data, p->mohinterpret);
break;
case AST_CONTROL_UNHOLD:
ast_rtp_instance_new_source(p->rtp);
ast_moh_stop(ast);
break;
case AST_CONTROL_VIDUPDATE: /* Request a video frame update */
if (p->vrtp && !p->novideo) {
transmit_info_with_vidupdate(p);
/* ast_rtcp_send_h261fur(p->vrtp); */
} else
res = -1;
break;
case AST_CONTROL_T38: /* T38 control frame */
if (datalen != sizeof(enum ast_control_t38)) {
ast_log(LOG_ERROR, "Invalid datalen for AST_CONTROL_T38. Expected %d, got %d\n", (int)sizeof(enum ast_control_t38), (int)datalen);
} else {
switch (*((enum ast_control_t38 *) data)) {
case AST_T38_NEGOTIATED:
case AST_T38_REQUEST_NEGOTIATE: /* Request T38 */
if (p->t38.state == T38_PEER_REINVITE) {
AST_SCHED_DEL_UNREF(sched, p->t38id, dialog_unref(p, "when you delete the t38id sched, you should dec the refcount for the stored dialog ptr"));
change_t38_state(p, T38_ENABLED);
transmit_response_with_t38_sdp(p, "200 OK", &p->initreq, XMIT_CRITICAL);
} else if (ast_test_flag(&p->t38.t38support, SIP_PAGE2_T38SUPPORT) && p->t38.state != T38_ENABLED) {
change_t38_state(p, T38_LOCAL_REINVITE);
if (!p->pendinginvite) {
transmit_reinvite_with_sdp(p, TRUE, FALSE);
} else if (!ast_test_flag(&p->flags[0], SIP_PENDINGBYE)) {
ast_set_flag(&p->flags[0], SIP_NEEDREINVITE);
}
}
break;
case AST_T38_TERMINATED:
case AST_T38_REFUSED:
case AST_T38_REQUEST_TERMINATE: /* Shutdown T38 */
if (p->t38.state == T38_PEER_REINVITE) {
AST_SCHED_DEL_UNREF(sched, p->t38id, dialog_unref(p, "when you delete the t38id sched, you should dec the refcount for the stored dialog ptr"));
change_t38_state(p, T38_DISABLED);
transmit_response_reliable(p, "488 Not acceptable here", &p->initreq);
} else if (p->t38.state == T38_ENABLED)
transmit_reinvite_with_sdp(p, FALSE, FALSE);
break;
default:
break;
}
}
break;
case AST_CONTROL_SRCUPDATE:
ast_rtp_instance_new_source(p->rtp);
break;
case AST_CONTROL_CONNECTED_LINE:
update_connectedline(p, data, datalen);
break;
case AST_CONTROL_REDIRECTING:
update_redirecting(p, data, datalen);
break;
case -1:
res = -1;
break;
default:
ast_log(LOG_WARNING, "Don't know how to indicate condition %d\n", condition);
res = -1;
break;
}
sip_pvt_unlock(p);
return res;
}
/*! \brief Initiate a call in the SIP channel
called from sip_request_call (calls from the pbx ) for outbound channels
and from handle_request_invite for inbound channels
*/
static struct ast_channel *sip_new(struct sip_pvt *i, int state, const char *title)
{
struct ast_channel *tmp;
struct ast_variable *v = NULL;
int fmt;
int what;
int video;
int text;
int needvideo = 0;
int needtext = 0;
char buf[SIPBUFSIZE];
char *decoded_exten;
{
const char *my_name; /* pick a good name */
if (title) {
my_name = title;
} else {
char *port = NULL;
my_name = ast_strdupa(i->fromdomain);
if ((port = strchr(i->fromdomain, ':'))) {
*port = '\0';
}
}
sip_pvt_unlock(i);
/* Don't hold a sip pvt lock while we allocate a channel */
tmp = ast_channel_alloc(1, state, i->cid_num, i->cid_name, i->accountcode, i->exten, i->context, i->amaflags, "SIP/%s-%08x", my_name, (int)(long) i);
}
if (!tmp) {
ast_log(LOG_WARNING, "Unable to allocate AST channel structure for SIP channel\n");
sip_pvt_lock(i);
return NULL;
}
sip_pvt_lock(i);
tmp->tech = ( ast_test_flag(&i->flags[0], SIP_DTMF) == SIP_DTMF_INFO || ast_test_flag(&i->flags[0], SIP_DTMF) == SIP_DTMF_SHORTINFO) ? &sip_tech_info : &sip_tech;
/* Select our native format based on codec preference until we receive
something from another device to the contrary. */
if (i->jointcapability) { /* The joint capabilities of us and peer */
what = i->jointcapability;
video = i->jointcapability & AST_FORMAT_VIDEO_MASK;
text = i->jointcapability & AST_FORMAT_TEXT_MASK;
} else if (i->capability) { /* Our configured capability for this peer */
what = i->capability;
video = i->capability & AST_FORMAT_VIDEO_MASK;
text = i->capability & AST_FORMAT_TEXT_MASK;
} else {
what = global_capability; /* Global codec support */
video = global_capability & AST_FORMAT_VIDEO_MASK;
text = global_capability & AST_FORMAT_TEXT_MASK;
}
/* Set the native formats for audio and merge in video */
tmp->nativeformats = ast_codec_choose(&i->prefs, what, 1) | video | text;
ast_debug(3, "*** Our native formats are %s \n", ast_getformatname_multiple(buf, SIPBUFSIZE, tmp->nativeformats));
ast_debug(3, "*** Joint capabilities are %s \n", ast_getformatname_multiple(buf, SIPBUFSIZE, i->jointcapability));
ast_debug(3, "*** Our capabilities are %s \n", ast_getformatname_multiple(buf, SIPBUFSIZE, i->capability));
ast_debug(3, "*** AST_CODEC_CHOOSE formats are %s \n", ast_getformatname_multiple(buf, SIPBUFSIZE, ast_codec_choose(&i->prefs, what, 1)));
if (i->prefcodec)
ast_debug(3, "*** Our preferred formats from the incoming channel are %s \n", ast_getformatname_multiple(buf, SIPBUFSIZE, i->prefcodec));
/* XXX Why are we choosing a codec from the native formats?? */
fmt = ast_best_codec(tmp->nativeformats);
/* If we have a prefcodec setting, we have an inbound channel that set a
preferred format for this call. Otherwise, we check the jointcapability
We also check for vrtp. If it's not there, we are not allowed do any video anyway.
*/
if (i->vrtp) {
if (ast_test_flag(&i->flags[1], SIP_PAGE2_VIDEOSUPPORT))
needvideo = AST_FORMAT_VIDEO_MASK;
else if (i->prefcodec)
needvideo = i->prefcodec & AST_FORMAT_VIDEO_MASK; /* Outbound call */
else
needvideo = i->jointcapability & AST_FORMAT_VIDEO_MASK; /* Inbound call */
}
if (i->trtp) {
if (i->prefcodec)
needtext = i->prefcodec & AST_FORMAT_TEXT_MASK; /* Outbound call */
else
needtext = i->jointcapability & AST_FORMAT_TEXT_MASK; /* Inbound call */
}
if (needvideo)
ast_debug(3, "This channel can handle video! HOLLYWOOD next!\n");
else
ast_debug(3, "This channel will not be able to handle video.\n");
if ((ast_test_flag(&i->flags[0], SIP_DTMF) == SIP_DTMF_INBAND) || (ast_test_flag(&i->flags[0], SIP_DTMF) == SIP_DTMF_AUTO)) {
if (!i->rtp || ast_rtp_instance_dtmf_mode_set(i->rtp, AST_RTP_DTMF_MODE_INBAND)) {
i->vad = ast_dsp_new();
ast_dsp_set_features(i->vad, DSP_FEATURE_DIGIT_DETECT);
if (global_relaxdtmf)
ast_dsp_set_digitmode(i->vad, DSP_DIGITMODE_DTMF | DSP_DIGITMODE_RELAXDTMF);
}
} else if (ast_test_flag(&i->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833) {
if (i->rtp) {
ast_rtp_instance_dtmf_mode_set(i->rtp, AST_RTP_DTMF_MODE_RFC2833);
}
}
/* Set file descriptors for audio, video, realtime text and UDPTL as needed */
if (i->rtp) {
ast_channel_set_fd(tmp, 0, ast_rtp_instance_fd(i->rtp, 0));
ast_channel_set_fd(tmp, 1, ast_rtp_instance_fd(i->rtp, 1));
}
if (needvideo && i->vrtp) {
ast_channel_set_fd(tmp, 2, ast_rtp_instance_fd(i->vrtp, 0));
ast_channel_set_fd(tmp, 3, ast_rtp_instance_fd(i->vrtp, 1));
}
if (needtext && i->trtp)
ast_channel_set_fd(tmp, 4, ast_rtp_instance_fd(i->trtp, 0));
if (i->udptl)
ast_channel_set_fd(tmp, 5, ast_udptl_fd(i->udptl));
if (state == AST_STATE_RING)
tmp->rings = 1;
tmp->adsicpe = AST_ADSI_UNAVAILABLE;
tmp->writeformat = fmt;
tmp->rawwriteformat = fmt;
tmp->readformat = fmt;
tmp->rawreadformat = fmt;
tmp->tech_pvt = dialog_ref(i, "sip_new: set chan->tech_pvt to i");
tmp->callgroup = i->callgroup;
tmp->pickupgroup = i->pickupgroup;
tmp->cid.cid_pres = i->callingpres;
if (!ast_strlen_zero(i->accountcode))
ast_string_field_set(tmp, accountcode, i->accountcode);
if (i->amaflags)
tmp->amaflags = i->amaflags;
if (!ast_strlen_zero(i->language))
ast_string_field_set(tmp, language, i->language);
i->owner = tmp;
ast_module_ref(ast_module_info->self);
ast_copy_string(tmp->context, i->context, sizeof(tmp->context));
/*Since it is valid to have extensions in the dialplan that have unescaped characters in them
* we should decode the uri before storing it in the channel, but leave it encoded in the sip_pvt
* structure so that there aren't issues when forming URI's
*/
decoded_exten = ast_strdupa(i->exten);
ast_uri_decode(decoded_exten);
ast_copy_string(tmp->exten, decoded_exten, sizeof(tmp->exten));
/* Don't use ast_set_callerid() here because it will
* generate an unnecessary NewCallerID event */
tmp->cid.cid_ani = ast_strdup(i->cid_num);
if (!ast_strlen_zero(i->rdnis))
tmp->cid.cid_rdnis = ast_strdup(i->rdnis);
if (!ast_strlen_zero(i->exten) && strcmp(i->exten, "s"))
tmp->cid.cid_dnid = ast_strdup(i->exten);
tmp->priority = 1;
if (!ast_strlen_zero(i->uri))
pbx_builtin_setvar_helper(tmp, "SIPURI", i->uri);
if (!ast_strlen_zero(i->domain))
pbx_builtin_setvar_helper(tmp, "SIPDOMAIN", i->domain);
if (!ast_strlen_zero(i->callid))
pbx_builtin_setvar_helper(tmp, "SIPCALLID", i->callid);
if (i->rtp)
ast_jb_configure(tmp, &global_jbconf);
/* Set channel variables for this call from configuration */
for (v = i->chanvars ; v ; v = v->next) {
char valuebuf[1024];
pbx_builtin_setvar_helper(tmp, v->name, ast_get_encoded_str(v->value, valuebuf, sizeof(valuebuf)));
}
if (state != AST_STATE_DOWN && ast_pbx_start(tmp)) {
ast_log(LOG_WARNING, "Unable to start PBX on %s\n", tmp->name);
tmp->hangupcause = AST_CAUSE_SWITCH_CONGESTION;
ast_hangup(tmp);
tmp = NULL;
}
if (i->do_history)
append_history(i, "NewChan", "Channel %s - from %s", tmp->name, i->callid);
/* Inform manager user about new channel and their SIP call ID */
if (sip_cfg.callevents)
manager_event(EVENT_FLAG_SYSTEM, "ChannelUpdate",
"Channel: %s\r\nUniqueid: %s\r\nChanneltype: %s\r\nSIPcallid: %s\r\nSIPfullcontact: %s\r\n",
tmp->name, tmp->uniqueid, "SIP", i->callid, i->fullcontact);
return tmp;
}
/*! \brief Reads one line of SIP message body */
static char *get_body_by_line(const char *line, const char *name, int nameLen, char delimiter)
{
if (!strncasecmp(line, name, nameLen) && line[nameLen] == delimiter)
return ast_skip_blanks(line + nameLen + 1);
return "";
}
/*! \brief Lookup 'name' in the SDP starting
* at the 'start' line. Returns the matching line, and 'start'
* is updated with the next line number.
*/
static const char *get_sdp_iterate(int *start, struct sip_request *req, const char *name)
{
int len = strlen(name);
while (*start < req->sdp_end) {
const char *r = get_body_by_line(REQ_OFFSET_TO_STR(req, line[(*start)++]), name, len, '=');
if (r[0] != '\0')
return r;
}
return "";
}
/*! \brief Get a line from an SDP message body */
static const char *get_sdp(struct sip_request *req, const char *name)
{
int dummy = 0;
return get_sdp_iterate(&dummy, req, name);
}
/*! \brief Get a specific line from the message body */
static char *get_body(struct sip_request *req, char *name, char delimiter)
{
int x;
int len = strlen(name);
char *r;
for (x = 0; x < req->lines; x++) {
r = get_body_by_line(REQ_OFFSET_TO_STR(req, line[x]), name, len, delimiter);
if (r[0] != '\0')
return r;
}
return "";
}
/*! \brief Find compressed SIP alias */
static const char *find_alias(const char *name, const char *_default)
{
/*! \brief Structure for conversion between compressed SIP and "normal" SIP */
static const struct cfalias {
char * const fullname;
char * const shortname;
} aliases[] = {
{ "Content-Type", "c" },
{ "Content-Encoding", "e" },
{ "From", "f" },
{ "Call-ID", "i" },
{ "Contact", "m" },
{ "Content-Length", "l" },
{ "Subject", "s" },
{ "To", "t" },
{ "Supported", "k" },
{ "Refer-To", "r" },
{ "Referred-By", "b" },
{ "Allow-Events", "u" },
{ "Event", "o" },
{ "Via", "v" },
{ "Accept-Contact", "a" },
{ "Reject-Contact", "j" },
{ "Request-Disposition", "d" },
{ "Session-Expires", "x" },
{ "Identity", "y" },
{ "Identity-Info", "n" },
};
int x;
for (x = 0; x < ARRAY_LEN(aliases); x++) {
if (!strcasecmp(aliases[x].fullname, name))
return aliases[x].shortname;
}
return _default;
}
static const char *__get_header(const struct sip_request *req, const char *name, int *start)
{
int pass;
/*
* Technically you can place arbitrary whitespace both before and after the ':' in
* a header, although RFC3261 clearly says you shouldn't before, and place just
* one afterwards. If you shouldn't do it, what absolute idiot decided it was
* a good idea to say you can do it, and if you can do it, why in the hell would.
* you say you shouldn't.
* Anyways, pedanticsipchecking controls whether we allow spaces before ':',
* and we always allow spaces after that for compatibility.
*/
for (pass = 0; name && pass < 2;pass++) {
int x, len = strlen(name);
for (x = *start; x < req->headers; x++) {
char *header = REQ_OFFSET_TO_STR(req, header[x]);
if (!strncasecmp(header, name, len)) {
char *r = header + len; /* skip name */
if (sip_cfg.pedanticsipchecking)
r = ast_skip_blanks(r);
if (*r == ':') {
*start = x+1;
return ast_skip_blanks(r+1);
}
}
}
if (pass == 0) /* Try aliases */
name = find_alias(name, NULL);
}
/* Don't return NULL, so get_header is always a valid pointer */
return "";
}
/*! \brief Get header from SIP request
\return Always return something, so don't check for NULL because it won't happen :-)
*/
static const char *get_header(const struct sip_request *req, const char *name)
{
int start = 0;
return __get_header(req, name, &start);
}
/*! \brief Read RTP from network */
static struct ast_frame *sip_rtp_read(struct ast_channel *ast, struct sip_pvt *p, int *faxdetect)
{
/* Retrieve audio/etc from channel. Assumes p->lock is already held. */
struct ast_frame *f;
if (!p->rtp) {
/* We have no RTP allocated for this channel */
return &ast_null_frame;
}
switch(ast->fdno) {
case 0:
f = ast_rtp_instance_read(p->rtp, 0); /* RTP Audio */
break;
case 1:
f = ast_rtp_instance_read(p->rtp, 1); /* RTCP Control Channel */
break;
case 2:
f = ast_rtp_instance_read(p->vrtp, 0); /* RTP Video */
break;
case 3:
f = ast_rtp_instance_read(p->vrtp, 1); /* RTCP Control Channel for video */
break;
case 4:
f = ast_rtp_instance_read(p->trtp, 0); /* RTP Text */
if (sipdebug_text) {
int i;
unsigned char* arr = f->data.ptr;
for (i=0; i < f->datalen; i++)
ast_verbose("%c", (arr[i] > ' ' && arr[i] < '}') ? arr[i] : '.');
ast_verbose(" -> ");
for (i=0; i < f->datalen; i++)
ast_verbose("%02X ", arr[i]);
ast_verbose("\n");
}
break;
case 5:
f = ast_udptl_read(p->udptl); /* UDPTL for T.38 */
break;
default:
f = &ast_null_frame;
}
/* Don't forward RFC2833 if we're not supposed to */
if (f && (f->frametype == AST_FRAME_DTMF_BEGIN || f->frametype == AST_FRAME_DTMF_END) &&
(ast_test_flag(&p->flags[0], SIP_DTMF) != SIP_DTMF_RFC2833)) {
ast_debug(1, "Ignoring DTMF (%c) RTP frame because dtmfmode is not RFC2833\n", f->subclass);
return &ast_null_frame;
}
/* We already hold the channel lock */
if (!p->owner || (f && f->frametype != AST_FRAME_VOICE))
return f;
if (f && f->subclass != (p->owner->nativeformats & AST_FORMAT_AUDIO_MASK)) {
if (!(f->subclass & p->jointcapability)) {
ast_debug(1, "Bogus frame of format '%s' received from '%s'!\n",
ast_getformatname(f->subclass), p->owner->name);
return &ast_null_frame;
}
ast_debug(1, "Oooh, format changed to %d %s\n",
f->subclass, ast_getformatname(f->subclass));
p->owner->nativeformats = (p->owner->nativeformats & (AST_FORMAT_VIDEO_MASK | AST_FORMAT_TEXT_MASK)) | f->subclass;
ast_set_read_format(p->owner, p->owner->readformat);
ast_set_write_format(p->owner, p->owner->writeformat);
}
if (f && (ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_INBAND) && p->vad) {
f = ast_dsp_process(p->owner, p->vad, f);
if (f && f->frametype == AST_FRAME_DTMF) {
if (ast_test_flag(&p->t38.t38support, SIP_PAGE2_T38SUPPORT_UDPTL) && f->subclass == 'f') {
ast_debug(1, "Fax CNG detected on %s\n", ast->name);
*faxdetect = 1;
} else {
ast_debug(1, "* Detected inband DTMF '%c'\n", f->subclass);
}
}
}
return f;
}
/*! \brief Read SIP RTP from channel */
static struct ast_frame *sip_read(struct ast_channel *ast)
{
struct ast_frame *fr;
struct sip_pvt *p = ast->tech_pvt;
int faxdetected = FALSE;
sip_pvt_lock(p);
fr = sip_rtp_read(ast, p, &faxdetected);
p->lastrtprx = time(NULL);
/* If we are NOT bridged to another channel, and we have detected fax tone we issue T38 re-invite to a peer */
/* If we are bridged then it is the responsibility of the SIP device to issue T38 re-invite if it detects CNG or fax preamble */
if (faxdetected && ast_test_flag(&p->t38.t38support, SIP_PAGE2_T38SUPPORT_UDPTL) && (p->t38.state == T38_DISABLED) && !(ast_bridged_channel(ast))) {
if (!ast_test_flag(&p->flags[0], SIP_GOTREFER)) {
if (!p->pendinginvite) {
ast_debug(3, "Sending reinvite on SIP (%s) for T.38 negotiation.\n", ast->name);
change_t38_state(p, T38_LOCAL_REINVITE);
transmit_reinvite_with_sdp(p, TRUE, FALSE);
}
} else if (!ast_test_flag(&p->flags[0], SIP_PENDINGBYE)) {
ast_debug(3, "Deferring reinvite on SIP (%s) - it will be re-negotiated for T.38\n", ast->name);
ast_set_flag(&p->flags[0], SIP_NEEDREINVITE);
}
}
/* Only allow audio through if they sent progress with SDP, or if the channel is actually answered */
if (fr && fr->frametype == AST_FRAME_VOICE && p->invitestate != INV_EARLY_MEDIA && ast->_state != AST_STATE_UP) {
fr = &ast_null_frame;
}
sip_pvt_unlock(p);
return fr;
}
/*! \brief Generate 32 byte random string for callid's etc */
static char *generate_random_string(char *buf, size_t size)
{
long val[4];
int x;
for (x=0; x<4; x++)
val[x] = ast_random();
snprintf(buf, size, "%08lx%08lx%08lx%08lx", val[0], val[1], val[2], val[3]);
return buf;
}
/*! \brief Build SIP Call-ID value for a non-REGISTER transaction */
static void build_callid_pvt(struct sip_pvt *pvt)
{
char buf[33];
const char *host = S_OR(pvt->fromdomain, ast_inet_ntoa(pvt->ourip.sin_addr));
ast_string_field_build(pvt, callid, "%s@%s", generate_random_string(buf, sizeof(buf)), host);
}
/*! \brief Build SIP Call-ID value for a REGISTER transaction */
static void build_callid_registry(struct sip_registry *reg, struct in_addr ourip, const char *fromdomain)
{
char buf[33];
const char *host = S_OR(fromdomain, ast_inet_ntoa(ourip));
ast_string_field_build(reg, callid, "%s@%s", generate_random_string(buf, sizeof(buf)), host);
}
/*! \brief Make our SIP dialog tag */
static void make_our_tag(char *tagbuf, size_t len)
{
snprintf(tagbuf, len, "as%08lx", ast_random());
}
/*! \brief Allocate Session-Timers struct w/in dialog */
static struct sip_st_dlg* sip_st_alloc(struct sip_pvt *const p)
{
struct sip_st_dlg *stp;
if (p->stimer) {
ast_log(LOG_ERROR, "Session-Timer struct already allocated\n");
return p->stimer;
}
if (!(stp = ast_calloc(1, sizeof(struct sip_st_dlg))))
return NULL;
p->stimer = stp;
stp->st_schedid = -1; /* Session-Timers ast_sched scheduler id */
return p->stimer;
}
/*! \brief Allocate sip_pvt structure, set defaults and link in the container.
* Returns a reference to the object so whoever uses it later must
* remember to release the reference.
*/
static struct sip_pvt *sip_alloc(ast_string_field callid, struct sockaddr_in *sin,
int useglobal_nat, const int intended_method)
{
struct sip_pvt *p;
if (!(p = ao2_t_alloc(sizeof(*p), sip_destroy_fn, "allocate a dialog(pvt) struct")))
return NULL;
if (ast_string_field_init(p, 512)) {
ao2_t_ref(p, -1, "failed to string_field_init, drop p");
return NULL;
}
p->socket.fd = -1;
p->socket.type = SIP_TRANSPORT_UDP;
p->method = intended_method;
p->initid = -1;
p->waitid = -1;
p->autokillid = -1;
p->request_queue_sched_id = -1;
p->t38id = -1;
p->subscribed = NONE;
p->stateid = -1;
p->sessionversion_remote = -1;
p->session_modify = TRUE;
p->stimer = NULL;
p->prefs = default_prefs; /* Set default codecs for this call */
if (intended_method != SIP_OPTIONS) { /* Peerpoke has it's own system */
p->timer_t1 = global_t1; /* Default SIP retransmission timer T1 (RFC 3261) */
p->timer_b = global_timer_b; /* Default SIP transaction timer B (RFC 3261) */
}
if (!sin)
p->ourip = internip;
else {
p->sa = *sin;
ast_sip_ouraddrfor(&p->sa.sin_addr, &p->ourip);
}
/* Copy global flags to this PVT at setup. */
ast_copy_flags(&p->flags[0], &global_flags[0], SIP_FLAGS_TO_COPY);
ast_copy_flags(&p->flags[1], &global_flags[1], SIP_PAGE2_FLAGS_TO_COPY);
p->do_history = recordhistory;
p->branch = ast_random();
make_our_tag(p->tag, sizeof(p->tag));
p->ocseq = INITIAL_CSEQ;
if (sip_methods[intended_method].need_rtp) {
if (ast_test_flag(&p->flags[1], SIP_PAGE2_T38SUPPORT) && (p->udptl = ast_udptl_new_with_bindaddr(sched, io, 0, bindaddr.sin_addr))) {
ast_udptl_setqos(p->udptl, global_tos_audio, global_cos_audio);
}
p->maxcallbitrate = default_maxcallbitrate;
p->autoframing = global_autoframing;
}
if (useglobal_nat && sin) {
/* Setup NAT structure according to global settings if we have an address */
ast_copy_flags(&p->flags[0], &global_flags[0], SIP_NAT);
p->recv = *sin;
do_setnat(p, ast_test_flag(&p->flags[0], SIP_NAT) & SIP_NAT_ROUTE);
}
if (p->method != SIP_REGISTER)
ast_string_field_set(p, fromdomain, default_fromdomain);
build_via(p);
if (!callid)
build_callid_pvt(p);
else
ast_string_field_set(p, callid, callid);
/* Assign default music on hold class */
ast_string_field_set(p, mohinterpret, default_mohinterpret);
ast_string_field_set(p, mohsuggest, default_mohsuggest);
p->capability = global_capability;
p->allowtransfer = sip_cfg.allowtransfer;
if ((ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833) ||
(ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_AUTO))
p->noncodeccapability |= AST_RTP_DTMF;
if (p->udptl) {
ast_copy_flags(&p->t38.t38support, &p->flags[1], SIP_PAGE2_T38SUPPORT);
set_t38_capabilities(p);
p->t38.jointcapability = p->t38.capability;
}
ast_string_field_set(p, context, sip_cfg.default_context);
ast_string_field_set(p, parkinglot, default_parkinglot);
ast_string_field_set(p, engine, default_engine);
AST_LIST_HEAD_INIT_NOLOCK(&p->request_queue);
/* Add to active dialog list */
ao2_t_link(dialogs, p, "link pvt into dialogs table");
ast_debug(1, "Allocating new SIP dialog for %s - %s (%s)\n", callid ? callid : p->callid, sip_methods[intended_method].text, p->rtp ? "With RTP" : "No RTP");
return p;
}
/*! \brief argument to the helper function to identify a call */
struct find_call_cb_arg {
enum sipmethod method;
const char *callid;
const char *fromtag;
const char *totag;
const char *tag;
};
/*!
* code to determine whether this is the pvt that we are looking for.
* Return FALSE if not found, true otherwise. p is unlocked.
*/
static int find_call_cb(void *__pvt, void *__arg, int flags)
{
struct sip_pvt *p = __pvt;
struct find_call_cb_arg *arg = __arg;
/* In pedantic, we do not want packets with bad syntax to be connected to a PVT */
int found = FALSE;
if (!ast_strlen_zero(p->callid)) { /* XXX double check, do we allow match on empty p->callid ? */
if (arg->method == SIP_REGISTER)
found = (!strcmp(p->callid, arg->callid));
else {
found = !strcmp(p->callid, arg->callid);
if (sip_cfg.pedanticsipchecking && found) {
found = ast_strlen_zero(arg->tag) || ast_strlen_zero(p->theirtag) || !ast_test_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED) || !strcmp(p->theirtag, arg->tag);
}
}
ast_debug(5, "= %s Their Call ID: %s Their Tag %s Our tag: %s\n", found ? "Found" : "No match", p->callid, p->theirtag, p->tag);
/* If we get a new request within an existing to-tag - check the to tag as well */
if (sip_cfg.pedanticsipchecking && found && arg->method != SIP_RESPONSE) { /* SIP Request */
if (p->tag[0] == '\0' && arg->totag[0]) {
/* We have no to tag, but they have. Wrong dialog */
found = FALSE;
} else if (arg->totag[0]) { /* Both have tags, compare them */
if (strcmp(arg->totag, p->tag)) {
found = FALSE; /* This is not our packet */
}
}
if (!found)
ast_debug(5, "= Being pedantic: This is not our match on request: Call ID: %s Ourtag <null> Totag %s Method %s\n", p->callid, arg->totag, sip_methods[arg->method].text);
}
}
return found;
}
/*! \brief find or create a dialog structure for an incoming SIP message.
* Connect incoming SIP message to current dialog or create new dialog structure
* Returns a reference to the sip_pvt object, remember to give it back once done.
* Called by handle_incoming(), sipsock_read
*/
static struct sip_pvt *find_call(struct sip_request *req, struct sockaddr_in *sin, const int intended_method)
{
struct sip_pvt *p = NULL;
char *tag = ""; /* note, tag is never NULL */
char totag[128];
char fromtag[128];
struct find_call_cb_arg arg;
const char *callid = get_header(req, "Call-ID");
const char *from = get_header(req, "From");
const char *to = get_header(req, "To");
const char *cseq = get_header(req, "Cseq");
struct sip_pvt *sip_pvt_ptr;
/* Call-ID, to, from and Cseq are required by RFC 3261. (Max-forwards and via too - ignored now) */
/* get_header always returns non-NULL so we must use ast_strlen_zero() */
if (ast_strlen_zero(callid) || ast_strlen_zero(to) ||
ast_strlen_zero(from) || ast_strlen_zero(cseq))
return NULL; /* Invalid packet */
arg.method = req->method;
arg.callid = callid;
arg.fromtag = fromtag;
arg.totag = totag;
arg.tag = ""; /* make sure tag is never NULL */
if (sip_cfg.pedanticsipchecking) {
/* In principle Call-ID's uniquely identify a call, but with a forking SIP proxy
we need more to identify a branch - so we have to check branch, from
and to tags to identify a call leg.
For Asterisk to behave correctly, you need to turn on pedanticsipchecking
in sip.conf
*/
if (gettag(req, "To", totag, sizeof(totag)))
req->has_to_tag = 1; /* Used in handle_request/response */
gettag(req, "From", fromtag, sizeof(fromtag));
tag = (req->method == SIP_RESPONSE) ? totag : fromtag;
ast_debug(5, "= Looking for Call ID: %s (Checking %s) --From tag %s --To-tag %s \n", callid, req->method==SIP_RESPONSE ? "To" : "From", fromtag, totag);
/* All messages must always have From: tag */
if (ast_strlen_zero(fromtag)) {
ast_debug(5, "%s request has no from tag, dropping callid: %s from: %s\n", sip_methods[req->method].text , callid, from );
return NULL;
}
/* reject requests that must always have a To: tag */
if (ast_strlen_zero(totag) && (req->method == SIP_ACK || req->method == SIP_BYE || req->method == SIP_INFO )) {
ast_debug(5, "%s must have a to tag. dropping callid: %s from: %s\n", sip_methods[req->method].text , callid, from );
return NULL;
}
}
restartsearch:
if (!sip_cfg.pedanticsipchecking) {
struct sip_pvt tmp_dialog = {
.callid = callid,
};
sip_pvt_ptr = ao2_t_find(dialogs, &tmp_dialog, OBJ_POINTER, "ao2_find in dialogs");
if (sip_pvt_ptr) { /* well, if we don't find it-- what IS in there? */
/* Found the call */
sip_pvt_lock(sip_pvt_ptr);
return sip_pvt_ptr;
}
} else { /* in pedantic mode! -- do the fancy linear search */
ao2_lock(dialogs);
p = ao2_t_callback(dialogs, 0 /* single, data */, find_call_cb, &arg, "pedantic linear search for dialog");
if (p) {
if (sip_pvt_trylock(p)) {
ao2_unlock(dialogs);
usleep(1);
goto restartsearch;
}
ao2_unlock(dialogs);
return p;
}
ao2_unlock(dialogs);
}
/* See if the method is capable of creating a dialog */
if (sip_methods[intended_method].can_create == CAN_CREATE_DIALOG) {
if (intended_method == SIP_REFER) {
/* We do support REFER, but not outside of a dialog yet */
transmit_response_using_temp(callid, sin, 1, intended_method, req, "603 Declined (no dialog)");
} else if (intended_method == SIP_NOTIFY) {
/* We do not support out-of-dialog NOTIFY either,
like voicemail notification, so cancel that early */
transmit_response_using_temp(callid, sin, 1, intended_method, req, "489 Bad event");
} else {
/* Ok, time to create a new SIP dialog object, a pvt */
if ((p = sip_alloc(callid, sin, 1, intended_method))) {
/* Ok, we've created a dialog, let's go and process it */
sip_pvt_lock(p);
} else {
/* We have a memory or file/socket error (can't allocate RTP sockets or something) so we're not
getting a dialog from sip_alloc.
Without a dialog we can't retransmit and handle ACKs and all that, but at least
send an error message.
Sorry, we apologize for the inconvienience
*/
transmit_response_using_temp(callid, sin, 1, intended_method, req, "500 Server internal error");
ast_debug(4, "Failed allocating SIP dialog, sending 500 Server internal error and giving up\n");
}
}
return p; /* can be NULL */
} else if( sip_methods[intended_method].can_create == CAN_CREATE_DIALOG_UNSUPPORTED_METHOD) {
/* A method we do not support, let's take it on the volley */
transmit_response_using_temp(callid, sin, 1, intended_method, req, "501 Method Not Implemented");
ast_debug(2, "Got a request with unsupported SIP method.\n");
} else if (intended_method != SIP_RESPONSE && intended_method != SIP_ACK) {
/* This is a request outside of a dialog that we don't know about */
transmit_response_using_temp(callid, sin, 1, intended_method, req, "481 Call leg/transaction does not exist");
ast_debug(2, "That's odd... Got a request in unknown dialog. Callid %s\n", callid ? callid : "<unknown>");
}
/* We do not respond to responses for dialogs that we don't know about, we just drop
the session quickly */
if (intended_method == SIP_RESPONSE)
ast_debug(2, "That's odd... Got a response on a call we dont know about. Callid %s\n", callid ? callid : "<unknown>");
return NULL;
}
/*! \brief Parse register=> line in sip.conf and add to registry */
static int sip_register(const char *value, int lineno)
{
struct sip_registry *reg;
int portnum = 0;
enum sip_transport transport = SIP_TRANSPORT_UDP;
char buf[256] = "";
char *username = NULL;
char *hostname=NULL, *secret=NULL, *authuser=NULL, *expire=NULL;
char *callback=NULL;
if (!value)
return -1;
ast_copy_string(buf, value, sizeof(buf));
/* split [/contact][~expiry] */
expire = strchr(buf, '~');
if (expire)
*expire++ = '\0';
callback = strrchr(buf, '/');
if (callback)
*callback++ = '\0';
if (ast_strlen_zero(callback))
callback = "s";
sip_parse_host(buf, lineno, &username, &portnum, &transport);
/* First split around the last '@' then parse the two components. */
hostname = strrchr(username, '@'); /* allow @ in the first part */
if (hostname)
*hostname++ = '\0';
if (ast_strlen_zero(username) || ast_strlen_zero(hostname)) {
ast_log(LOG_WARNING, "Format for registration is [transport://]user[:secret[:authuser]]@domain[:port][/extension][~expiry] at line %d\n", lineno);
return -1;
}
/* split user[:secret[:authuser]] from the end to allow : character in user portion*/
authuser = strrchr(username, ':');
if (authuser) {
*authuser++ = '\0';
secret = strrchr(username, ':');
if (secret)
*secret++ = '\0';
else {
secret = authuser;
authuser = NULL;
}
}
if ((authuser) && (ast_strlen_zero(authuser)))
authuser = NULL;
if ((secret) && (ast_strlen_zero(secret)))
secret = NULL;
if (!(reg = ast_calloc(1, sizeof(*reg)))) {
ast_log(LOG_ERROR, "Out of memory. Can't allocate SIP registry entry\n");
return -1;
}
if (ast_string_field_init(reg, 256)) {
ast_log(LOG_ERROR, "Out of memory. Can't allocate SIP registry strings\n");
ast_free(reg);
return -1;
}
ast_atomic_fetchadd_int(&regobjs, 1);
ASTOBJ_INIT(reg);
ast_string_field_set(reg, callback, callback);
if (!ast_strlen_zero(username))
ast_string_field_set(reg, username, username);
if (hostname)
ast_string_field_set(reg, hostname, hostname);
if (authuser)
ast_string_field_set(reg, authuser, authuser);
if (secret)
ast_string_field_set(reg, secret, secret);
reg->transport = transport;
reg->expire = -1;
reg->expiry = (expire ? atoi(expire) : default_expiry);
reg->timeout = -1;
reg->refresh = reg->expiry;
reg->portno = portnum;
reg->callid_valid = FALSE;
reg->ocseq = INITIAL_CSEQ;
ASTOBJ_CONTAINER_LINK(&regl, reg); /* Add the new registry entry to the list */
registry_unref(reg, "unref the reg pointer"); /* release the reference given by ASTOBJ_INIT. The container has another reference */
return 0;
}
/*! \brief Parse mwi=> line in sip.conf and add to list */
static int sip_subscribe_mwi(const char *value, int lineno)
{
struct sip_subscription_mwi *mwi;
int portnum = 0;
enum sip_transport transport = SIP_TRANSPORT_UDP;
char buf[256] = "";
char *username = NULL, *hostname = NULL, *secret = NULL, *authuser = NULL, *porta = NULL, *mailbox = NULL;
if (!value) {
return -1;
}
ast_copy_string(buf, value, sizeof(buf));
sip_parse_host(buf, lineno, &username, &portnum, &transport);
if ((hostname = strrchr(username, '@'))) {
*hostname++ = '\0';
}
if ((secret = strchr(username, ':'))) {
*secret++ = '\0';
if ((authuser = strchr(secret, ':'))) {
*authuser++ = '\0';
}
}
if ((mailbox = strchr(hostname, '/'))) {
*mailbox++ = '\0';
}
if (ast_strlen_zero(username) || ast_strlen_zero(hostname) || ast_strlen_zero(mailbox)) {
ast_log(LOG_WARNING, "Format for MWI subscription is user[:secret[:authuser]]@host[:port][/mailbox] at line %d\n", lineno);
return -1;
}
if ((porta = strchr(hostname, ':'))) {
*porta++ = '\0';
if (!(portnum = atoi(porta))) {
ast_log(LOG_WARNING, "%s is not a valid port number at line %d\n", porta, lineno);
return -1;
}
}
if (!(mwi = ast_calloc(1, sizeof(*mwi)))) {
return -1;
}
if (ast_string_field_init(mwi, 256)) {
ast_free(mwi);
return -1;
}
ASTOBJ_INIT(mwi);
ast_string_field_set(mwi, username, username);
if (secret) {
ast_string_field_set(mwi, secret, secret);
}
if (authuser) {
ast_string_field_set(mwi, authuser, authuser);
}
ast_string_field_set(mwi, hostname, hostname);
ast_string_field_set(mwi, mailbox, mailbox);
mwi->resub = -1;
mwi->portno = portnum;
mwi->transport = transport;
ASTOBJ_CONTAINER_LINK(&submwil, mwi);
ASTOBJ_UNREF(mwi, sip_subscribe_mwi_destroy);
return 0;
}
/*! \brief Parse multiline SIP headers into one header
This is enabled if pedanticsipchecking is enabled */
static int lws2sws(char *msgbuf, int len)
{
int h = 0, t = 0;
int lws = 0;
for (; h < len;) {
/* Eliminate all CRs */
if (msgbuf[h] == '\r') {
h++;
continue;
}
/* Check for end-of-line */
if (msgbuf[h] == '\n') {
/* Check for end-of-message */
if (h + 1 == len)
break;
/* Check for a continuation line */
if (msgbuf[h + 1] == ' ' || msgbuf[h + 1] == '\t') {
/* Merge continuation line */
h++;
continue;
}
/* Propagate LF and start new line */
msgbuf[t++] = msgbuf[h++];
lws = 0;
continue;
}
if (msgbuf[h] == ' ' || msgbuf[h] == '\t') {
if (lws) {
h++;
continue;
}
msgbuf[t++] = msgbuf[h++];
lws = 1;
continue;
}
msgbuf[t++] = msgbuf[h++];
if (lws)
lws = 0;
}
msgbuf[t] = '\0';
return t;
}
/*! \brief Parse a SIP message
\note this function is used both on incoming and outgoing packets
*/
static int parse_request(struct sip_request *req)
{
char *c = req->data->str;
ptrdiff_t *dst = req->header;
int i = 0, lim = SIP_MAX_HEADERS - 1;
unsigned int skipping_headers = 0;
ptrdiff_t current_header_offset = 0;
char *previous_header = "";
req->header[0] = 0;
req->headers = -1; /* mark that we are working on the header */
for (; *c; c++) {
if (*c == '\r') { /* remove \r */
*c = '\0';
} else if (*c == '\n') { /* end of this line */
*c = '\0';
current_header_offset = (c + 1) - req->data->str;
previous_header = req->data->str + dst[i];
if (skipping_headers) {
/* check to see if this line is blank; if so, turn off
the skipping flag, so the next line will be processed
as a body line */
if (ast_strlen_zero(previous_header)) {
skipping_headers = 0;
}
dst[i] = current_header_offset; /* record start of next line */
continue;
}
if (sipdebug) {
ast_debug(4, "%7s %2d [%3d]: %s\n",
req->headers < 0 ? "Header" : "Body",
i, (int) strlen(previous_header), previous_header);
}
if (ast_strlen_zero(previous_header) && req->headers < 0) {
req->headers = i; /* record number of header lines */
dst = req->line; /* start working on the body */
i = 0;
lim = SIP_MAX_LINES - 1;
} else { /* move to next line, check for overflows */
if (i++ == lim) {
/* if we're processing headers, then skip any remaining
headers and move on to processing the body, otherwise
we're done */
if (req->headers != -1) {
break;
} else {
req->headers = i;
dst = req->line;
i = 0;
lim = SIP_MAX_LINES - 1;
skipping_headers = 1;
}
}
}
dst[i] = current_header_offset; /* record start of next line */
}
}
/* Check for last header or body line without CRLF. The RFC for SDP requires CRLF,
but since some devices send without, we'll be generous in what we accept. However,
if we've already reached the maximum number of lines for portion of the message
we were parsing, we can't accept any more, so just ignore it.
*/
previous_header = req->data->str + dst[i];
if ((i < lim) && !ast_strlen_zero(previous_header)) {
if (sipdebug) {
ast_debug(4, "%7s %2d [%3d]: %s\n",
req->headers < 0 ? "Header" : "Body",
i, (int) strlen(previous_header), previous_header );
}
i++;
}
/* update count of header or body lines */
if (req->headers >= 0) { /* we are in the body */
req->lines = i;
} else { /* no body */
req->headers = i;
req->lines = 0;
/* req->data->used will be a NULL byte */
req->line[0] = ast_str_strlen(req->data);
}
if (*c) {
ast_log(LOG_WARNING, "Too many lines, skipping <%s>\n", c);
}
/* Split up the first line parts */
return determine_firstline_parts(req);
}
/*!
\brief Determine whether a SIP message contains an SDP in its body
\param req the SIP request to process
\return 1 if SDP found, 0 if not found
Also updates req->sdp_start and req->sdp_end to indicate where the SDP
lives in the message body.
*/
static int find_sdp(struct sip_request *req)
{
const char *content_type;
const char *content_length;
const char *search;
char *boundary;
unsigned int x;
int boundaryisquoted = FALSE;
int found_application_sdp = FALSE;
int found_end_of_headers = FALSE;
content_length = get_header(req, "Content-Length");
if (!ast_strlen_zero(content_length)) {
if (sscanf(content_length, "%ud", &x) != 1) {
ast_log(LOG_WARNING, "Invalid Content-Length: %s\n", content_length);
return 0;
}
/* Content-Length of zero means there can't possibly be an
SDP here, even if the Content-Type says there is */
if (x == 0)
return 0;
}
content_type = get_header(req, "Content-Type");
/* if the body contains only SDP, this is easy */
if (!strncasecmp(content_type, "application/sdp", 15)) {
req->sdp_start = 0;
req->sdp_end = req->lines;
return req->lines ? 1 : 0;
}
/* if it's not multipart/mixed, there cannot be an SDP */
if (strncasecmp(content_type, "multipart/mixed", 15))
return 0;
/* if there is no boundary marker, it's invalid */
if ((search = strcasestr(content_type, ";boundary=")))
search += 10;
else if ((search = strcasestr(content_type, "; boundary=")))
search += 11;
else
return 0;
if (ast_strlen_zero(search))
return 0;
/* If the boundary is quoted with ", remove quote */
if (*search == '\"') {
search++;
boundaryisquoted = TRUE;
}
/* make a duplicate of the string, with two extra characters
at the beginning */
boundary = ast_strdupa(search - 2);
boundary[0] = boundary[1] = '-';
/* Remove final quote */
if (boundaryisquoted)
boundary[strlen(boundary) - 1] = '\0';
/* search for the boundary marker, the empty line delimiting headers from
sdp part and the end boundry if it exists */
for (x = 0; x < (req->lines); x++) {
char *line = REQ_OFFSET_TO_STR(req, line[x]);
if (!strncasecmp(line, boundary, strlen(boundary))){
if (found_application_sdp && found_end_of_headers) {
req->sdp_end = x-1;
return 1;
}
found_application_sdp = FALSE;
}
if (!strcasecmp(line, "Content-Type: application/sdp"))
found_application_sdp = TRUE;
if (ast_strlen_zero(line)) {
if (found_application_sdp && !found_end_of_headers){
req->sdp_start = x;
found_end_of_headers = TRUE;
}
}
}
if (found_application_sdp && found_end_of_headers) {
req->sdp_end = x;
return TRUE;
}
return FALSE;
}
/*! \brief Process SIP SDP offer, select formats and activate RTP channels
If offer is rejected, we will not change any properties of the call
Return 0 on success, a negative value on errors.
Must be called after find_sdp().
*/
static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action)
{
const char *m; /* SDP media offer */
const char *c;
const char *a;
const char *o; /* Pointer to o= line */
char *o_copy; /* Copy of o= line */
char *token;
char host[258];
int len = -1;
int portno = -1; /*!< RTP Audio port number */
int vportno = -1; /*!< RTP Video port number */
int tportno = -1; /*!< RTP Text port number */
int udptlportno = -1;
int peert38capability = 0;
char s[256];
int old = 0;
/* Peer capability is the capability in the SDP, non codec is RFC2833 DTMF (101) */
int peercapability = 0, peernoncodeccapability = 0;
int vpeercapability = 0, vpeernoncodeccapability = 0;
int tpeercapability = 0, tpeernoncodeccapability = 0;
struct sockaddr_in sin; /*!< media socket address */
struct sockaddr_in vsin; /*!< Video socket address */
struct sockaddr_in tsin; /*!< Text socket address */
const char *codecs;
struct hostent *hp; /*!< RTP Audio host IP */
struct hostent *vhp = NULL; /*!< RTP video host IP */
struct hostent *thp = NULL; /*!< RTP text host IP */
struct ast_hostent audiohp;
struct ast_hostent videohp;
struct ast_hostent texthp;
int codec;
int destiterator = 0;
int iterator;
int sendonly = -1;
int numberofports;
struct ast_rtp_codecs newaudiortp, newvideortp, newtextrtp;
int newjointcapability; /* Negotiated capability */
int newpeercapability;
int newnoncodeccapability;
int numberofmediastreams = 0;
int debug = sip_debug_test_pvt(p);
int found_rtpmap_codecs[SDP_MAX_RTPMAP_CODECS];
int last_rtpmap_codec=0;
char buf[SIPBUFSIZE];
uint64_t rua_version;
int red_data_pt[10];
int red_num_gen = 0;
int red_pt = 0;
char *red_cp; /* For T.140 red */
char red_fmtp[100] = "empty"; /* For T.140 red */
if (!p->rtp) {
ast_log(LOG_ERROR, "Got SDP but have no RTP session allocated.\n");
return -1;
}
/* Make sure that the codec structures are all cleared out */
ast_rtp_codecs_payloads_clear(&newaudiortp, NULL);
ast_rtp_codecs_payloads_clear(&newvideortp, NULL);
ast_rtp_codecs_payloads_clear(&newtextrtp, NULL);
/* Update our last rtprx when we receive an SDP, too */
p->lastrtprx = p->lastrtptx = time(NULL); /* XXX why both ? */
/* Store the SDP version number of remote UA. This will allow us to
distinguish between session modifications and session refreshes. If
the remote UA does not send an incremented SDP version number in a
subsequent RE-INVITE then that means its not changing media session.
The RE-INVITE may have been sent to update connected party, remote
target or to refresh the session (Session-Timers). Asterisk must not
change media session and increment its own version number in answer
SDP in this case. */
o = get_sdp(req, "o");
if (ast_strlen_zero(o)) {
ast_log(LOG_WARNING, "SDP syntax error. SDP without an o= line\n");
return -1;
}
o_copy = ast_strdupa(o);
token = strsep(&o_copy, " "); /* Skip username */
if (!o_copy) {
ast_log(LOG_WARNING, "SDP syntax error in o= line username\n");
return -1;
}
token = strsep(&o_copy, " "); /* Skip session-id */
if (!o_copy) {
ast_log(LOG_WARNING, "SDP syntax error in o= line session-id\n");
return -1;
}
token = strsep(&o_copy, " "); /* Version */
if (!o_copy) {
ast_log(LOG_WARNING, "SDP syntax error in o= line\n");
return -1;
}
if (!sscanf(token, "%" SCNu64, &rua_version)) {
ast_log(LOG_WARNING, "SDP syntax error in o= line version\n");
return -1;
}
if (ast_test_flag(&p->flags[1], SIP_PAGE2_IGNORESDPVERSION)
|| p->sessionversion_remote < 0
|| p->sessionversion_remote != rua_version) {
p->sessionversion_remote = rua_version;
p->session_modify = TRUE;
} else if (p->sessionversion_remote == rua_version) {
p->session_modify = FALSE;
ast_debug(2, "SDP version number same as previous SDP. Not parsing this SDP.\n");
return 0;
}
/* Try to find first media stream */
m = get_sdp(req, "m");
destiterator = req->sdp_start;
c = get_sdp_iterate(&destiterator, req, "c");
if (ast_strlen_zero(m) || ast_strlen_zero(c)) {
ast_log(LOG_WARNING, "Insufficient information for SDP (m = '%s', c = '%s')\n", m, c);
return -1;
}
/* Check for IPv4 address (not IPv6 yet) */
if (sscanf(c, "IN IP4 %256s", host) != 1) {
ast_log(LOG_WARNING, "Invalid host in c= line, '%s'\n", c);
return -1;
}
/* XXX This could block for a long time, and block the main thread! XXX */
hp = ast_gethostbyname(host, &audiohp);
if (!hp) {
ast_log(LOG_WARNING, "Unable to lookup host in c= line, '%s'\n", c);
return -1;
}
vhp = hp; /* Copy to video address as default too */
thp = hp; /* Copy to text address as default too */
iterator = req->sdp_start;
/* default: novideo and notext set */
p->novideo = TRUE;
p->notext = TRUE;
if (p->vrtp) {
ast_rtp_codecs_payloads_clear(&newvideortp, NULL);
}
if (p->trtp) {
ast_rtp_codecs_payloads_clear(&newtextrtp, NULL);
}
/* Find media streams in this SDP offer */
while ((m = get_sdp_iterate(&iterator, req, "m"))[0] != '\0') {
int x;
int audio = FALSE;
int video = FALSE;
int text = FALSE;
numberofports = 1;
len = -1;
if ((sscanf(m, "audio %d/%d RTP/AVP %n", &x, &numberofports, &len) == 2 && len > 0) ||
(sscanf(m, "audio %d RTP/AVP %n", &x, &len) == 1 && len > 0)) {
audio = TRUE;
numberofmediastreams++;
/* Found audio stream in this media definition */
portno = x;
/* Scan through the RTP payload types specified in a "m=" line: */
for (codecs = m + len; !ast_strlen_zero(codecs); codecs = ast_skip_blanks(codecs + len)) {
if (sscanf(codecs, "%d%n", &codec, &len) != 1) {
ast_log(LOG_WARNING, "Error in codec string '%s'\n", codecs);
return -1;
}
if (debug)
ast_verbose("Found RTP audio format %d\n", codec);
ast_rtp_codecs_payloads_set_m_type(&newaudiortp, NULL, codec);
}
} else if ((sscanf(m, "video %d/%d RTP/AVP %n", &x, &numberofports, &len) == 2 && len > 0) ||
(sscanf(m, "video %d RTP/AVP %n", &x, &len) == 1 && len >= 0)) {
video = TRUE;
p->novideo = FALSE;
numberofmediastreams++;
vportno = x;
/* Scan through the RTP payload types specified in a "m=" line: */
for (codecs = m + len; !ast_strlen_zero(codecs); codecs = ast_skip_blanks(codecs + len)) {
if (sscanf(codecs, "%d%n", &codec, &len) != 1) {
ast_log(LOG_WARNING, "Error in codec string '%s'\n", codecs);
return -1;
}
if (debug)
ast_verbose("Found RTP video format %d\n", codec);
ast_rtp_codecs_payloads_set_m_type(&newvideortp, NULL, codec);
}
} else if ((sscanf(m, "text %d/%d RTP/AVP %n", &x, &numberofports, &len) == 2 && len > 0) ||
(sscanf(m, "text %d RTP/AVP %n", &x, &len) == 1 && len > 0)) {
text = TRUE;
p->notext = FALSE;
numberofmediastreams++;
tportno = x;
/* Scan through the RTP payload types specified in a "m=" line: */
for (codecs = m + len; !ast_strlen_zero(codecs); codecs = ast_skip_blanks(codecs + len)) {
if (sscanf(codecs, "%d%n", &codec, &len) != 1) {
ast_log(LOG_WARNING, "Error in codec string '%s'\n", codecs);
return -1;
}
if (debug)
ast_verbose("Found RTP text format %d\n", codec);
ast_rtp_codecs_payloads_set_m_type(&newtextrtp, NULL, codec);
}
} else if (p->udptl && ( (sscanf(m, "image %d udptl t38%n", &x, &len) == 1 && len > 0) ||
(sscanf(m, "image %d UDPTL t38%n", &x, &len) == 1 && len > 0) )) {
if (debug)
ast_verbose("Got T.38 offer in SDP in dialog %s\n", p->callid);
udptlportno = x;
numberofmediastreams++;
} else
ast_log(LOG_WARNING, "Unsupported SDP media type in offer: %s\n", m);
if (numberofports > 1)
ast_log(LOG_WARNING, "SDP offered %d ports for media, not supported by Asterisk. Will try anyway...\n", numberofports);
/* Check for Media-description-level-address for audio */
c = get_sdp_iterate(&destiterator, req, "c");
if (!ast_strlen_zero(c)) {
if (sscanf(c, "IN IP4 %256s", host) != 1) {
ast_log(LOG_WARNING, "Invalid secondary host in c= line, '%s'\n", c);
} else {
/* XXX This could block for a long time, and block the main thread! XXX */
if (audio) {
if ( !(hp = ast_gethostbyname(host, &audiohp))) {
ast_log(LOG_WARNING, "Unable to lookup RTP Audio host in secondary c= line, '%s'\n", c);
return -2;
}
} else if (video) {
if (!(vhp = ast_gethostbyname(host, &videohp))) {
ast_log(LOG_WARNING, "Unable to lookup RTP video host in secondary c= line, '%s'\n", c);
return -2;
}
} else if (text) {
if (!(thp = ast_gethostbyname(host, &texthp))) {
ast_log(LOG_WARNING, "Unable to lookup RTP text host in secondary c= line, '%s'\n", c);
return -2;
}
}
}
}
}
if (portno == -1 && vportno == -1 && udptlportno == -1 && tportno == -1)
/* No acceptable offer found in SDP - we have no ports */
/* Do not change RTP or VRTP if this is a re-invite */
return -2;
if (numberofmediastreams > 3)
/* We have too many fax, audio and/or video and/or text media streams, fail this offer */
return -3;
/* RTP addresses and ports for audio and video */
sin.sin_family = AF_INET;
vsin.sin_family = AF_INET;
tsin.sin_family = AF_INET;
memcpy(&sin.sin_addr, hp->h_addr, sizeof(sin.sin_addr));
if (vhp)
memcpy(&vsin.sin_addr, vhp->h_addr, sizeof(vsin.sin_addr));
if (thp)
memcpy(&tsin.sin_addr, thp->h_addr, sizeof(tsin.sin_addr));
/* Setup UDPTL port number */
if (p->udptl) {
if (udptlportno > 0) {
sin.sin_port = htons(udptlportno);
if (ast_test_flag(&p->flags[0], SIP_NAT) && ast_test_flag(&p->flags[1], SIP_PAGE2_UDPTL_DESTINATION)) {
struct sockaddr_in remote_address = { 0, };
ast_rtp_instance_get_remote_address(p->rtp, &remote_address);
if (remote_address.sin_addr.s_addr) {
memcpy(&sin, &remote_address, sizeof(sin));
if (debug) {
ast_log(LOG_DEBUG, "Peer T.38 UDPTL is set behind NAT and with destination, destination address now %s\n", ast_inet_ntoa(sin.sin_addr));
}
}
}
ast_udptl_set_peer(p->udptl, &sin);
if (debug)
ast_debug(1, "Peer T.38 UDPTL is at port %s:%d\n", ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port));
} else {
ast_udptl_stop(p->udptl);
if (debug)
ast_debug(1, "Peer doesn't provide T.38 UDPTL\n");
}
}
if (p->rtp) {
if (portno > 0) {
sin.sin_port = htons(portno);
ast_rtp_instance_set_remote_address(p->rtp, &sin);
if (debug)
ast_verbose("Peer audio RTP is at port %s:%d\n", ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port));
} else {
if (udptlportno > 0) {
if (debug)
ast_verbose("Got T.38 Re-invite without audio. Keeping RTP active during T.38 session. Callid %s\n", p->callid);
} else {
ast_rtp_instance_stop(p->rtp);
if (debug)
ast_verbose("Peer doesn't provide audio. Callid %s\n", p->callid);
}
}
}
/* Setup video port number, assumes we have audio */
if (vportno != -1)
vsin.sin_port = htons(vportno);
/* Setup text port number, assumes we have audio */
if (tportno != -1)
tsin.sin_port = htons(tportno);
/* Next, scan through each "a=xxxx:" line, noting each
* specified RTP payload type (with corresponding MIME subtype):
*/
/* XXX This needs to be done per media stream, since it's media stream specific */
iterator = req->sdp_start;
while ((a = get_sdp_iterate(&iterator, req, "a"))[0] != '\0') {
char mimeSubtype[128];
char fmtp_string[64];
unsigned int sample_rate;
if (option_debug > 1) {
int breakout = FALSE;
/* If we're debugging, check for unsupported sdp options */
if (!strncasecmp(a, "rtcp:", (size_t) 5)) {
if (debug)
ast_verbose("Got unsupported a:rtcp in SDP offer \n");
breakout = TRUE;
} else if (!strncasecmp(a, "framerate:", (size_t) 10)) {
/* Video stuff: Not supported */
if (debug)
ast_verbose("Got unsupported a:framerate in SDP offer \n");
breakout = TRUE;
} else if (!strncasecmp(a, "maxprate:", (size_t) 9)) {
/* Video stuff: Not supported */
if (debug)
ast_verbose("Got unsupported a:maxprate in SDP offer \n");
breakout = TRUE;
} else if (!strncasecmp(a, "crypto:", (size_t) 7)) {
/* SRTP stuff, not yet supported */
if (debug)
ast_verbose("Got unsupported a:crypto in SDP offer \n");
breakout = TRUE;
}
if (breakout) /* We have a match, skip to next header */
continue;
}
if (!strcasecmp(a, "sendonly")) {
if (sendonly == -1)
sendonly = 1;
continue;
}
if (!strcasecmp(a, "inactive")) {
if (sendonly == -1)
sendonly = 2;
continue;
}
if (!strcasecmp(a, "sendrecv")) {
if (sendonly == -1)
sendonly = 0;
continue;
}
if (!strncasecmp(a, "ptime", 5)) {
char *tmp = strrchr(a, ':');
long int framing = 0;
if (tmp) {
tmp++;
framing = strtol(tmp, NULL, 10);
if (framing == LONG_MIN || framing == LONG_MAX) {
framing = 0;
ast_debug(1, "Can't read framing from SDP: %s\n", a);
}
}
if (framing && p->autoframing) {
struct ast_codec_pref *pref = &ast_rtp_instance_get_codecs(p->rtp)->pref;
int codec_n;
for (codec_n = 0; codec_n < AST_RTP_MAX_PT; codec_n++) {
struct ast_rtp_payload_type format = ast_rtp_codecs_payload_lookup(ast_rtp_instance_get_codecs(p->rtp), codec_n);
if (!format.asterisk_format || !format.code) /* non-codec or not found */
continue;
if (option_debug)
ast_log(LOG_DEBUG, "Setting framing for %d to %ld\n", format.code, framing);
ast_codec_pref_setsize(pref, format.code, framing);
}
ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(p->rtp), p->rtp, pref);
}
continue;
}
if (!strncmp(a, red_fmtp, strlen(red_fmtp))) {
/* count numbers of generations in fmtp */
red_cp = &red_fmtp[strlen(red_fmtp)];
strncpy(red_fmtp, a, 100);
sscanf(red_cp, "%u", &red_data_pt[red_num_gen]);
red_cp = strtok(red_cp, "/");
while (red_cp && red_num_gen++ < AST_RED_MAX_GENERATION) {
sscanf(red_cp, "%u", &red_data_pt[red_num_gen]);
red_cp = strtok(NULL, "/");
}
red_cp = red_fmtp;
continue;
}
if (sscanf(a, "fmtp: %u %63s", &codec, fmtp_string) == 2) {
struct ast_rtp_payload_type payload;
unsigned int handled = 0;
payload = ast_rtp_codecs_payload_lookup(&newaudiortp, codec);
if (!payload.code) {
/* it wasn't found, try the video rtp */
payload = ast_rtp_codecs_payload_lookup(&newvideortp, codec);
}
if (payload.code && payload.asterisk_format) {
unsigned int bit_rate;
switch (payload.code) {
case AST_FORMAT_SIREN7:
if (sscanf(fmtp_string, "bitrate=%u", &bit_rate) == 1) {
if (bit_rate != 32000) {
ast_log(LOG_WARNING, "Got Siren7 offer at %d bps, but only 32000 bps supported; ignoring.\n", bit_rate);
ast_rtp_codecs_payloads_unset(&newaudiortp, NULL, codec);
} else {
handled = 1;
}
}
break;
case AST_FORMAT_SIREN14:
if (sscanf(fmtp_string, "bitrate=%u", &bit_rate) == 1) {
if (bit_rate != 48000) {
ast_log(LOG_WARNING, "Got Siren14 offer at %d bps, but only 48000 bps supported; ignoring.\n", bit_rate);
ast_rtp_codecs_payloads_unset(&newaudiortp, NULL, codec);
} else {
handled = 1;
}
}
break;
}
}
if (!handled) {
ast_debug(1, "Got unsupported a:%s in SDP offer\n", a);
}
continue;
}
if (sscanf(a, "rtpmap: %u %127[^/]/%u", &codec, mimeSubtype, &sample_rate) == 3) {
/* We have a rtpmap to handle */
if (last_rtpmap_codec < SDP_MAX_RTPMAP_CODECS) {
/* Note: should really look at the '#chans' params too */
/* Note: This should all be done in the context of the m= above */
if (!strncasecmp(mimeSubtype, "H26", 3) || !strncasecmp(mimeSubtype, "MP4", 3)) { /* Video */
if (ast_rtp_codecs_payloads_set_rtpmap_type_rate(&newvideortp, NULL, codec, "video", mimeSubtype, 0, sample_rate) != -1) {
if (debug)
ast_verbose("Found video description format %s for ID %d\n", mimeSubtype, codec);
found_rtpmap_codecs[last_rtpmap_codec] = codec;
last_rtpmap_codec++;
} else {
ast_rtp_codecs_payloads_unset(&newvideortp, NULL, codec);
if (debug)
ast_verbose("Found unknown media description format %s for ID %d\n", mimeSubtype, codec);
}
} else if (!strncasecmp(mimeSubtype, "T140", 4)) { /* Text */
if (p->trtp) {
/* ast_verbose("Adding t140 mimeSubtype to textrtp struct\n"); */
ast_rtp_codecs_payloads_set_rtpmap_type_rate(&newtextrtp, NULL, codec, "text", mimeSubtype, 0, sample_rate);
}
} else if (!strncasecmp(mimeSubtype, "RED", 3)) { /* Text with Redudancy */
if (p->trtp) {
ast_rtp_codecs_payloads_set_rtpmap_type_rate(&newtextrtp, NULL, codec, "text", mimeSubtype, 0, sample_rate);
red_pt = codec;
sprintf(red_fmtp, "fmtp:%d ", red_pt);
if (debug)
ast_verbose("RED submimetype has payload type: %d\n", red_pt);
}
} else { /* Must be audio?? */
if (ast_rtp_codecs_payloads_set_rtpmap_type_rate(&newaudiortp, NULL, codec, "audio", mimeSubtype,
ast_test_flag(&p->flags[0], SIP_G726_NONSTANDARD) ? AST_RTP_OPT_G726_NONSTANDARD : 0, sample_rate) != -1) {
if (debug)
ast_verbose("Found audio description format %s for ID %d\n", mimeSubtype, codec);
found_rtpmap_codecs[last_rtpmap_codec] = codec;
last_rtpmap_codec++;
} else {
ast_rtp_codecs_payloads_unset(&newaudiortp, NULL, codec);
if (debug)
ast_verbose("Found unknown media description format %s for ID %d\n", mimeSubtype, codec);
}
}
} else {
if (debug)
ast_verbose("Discarded description format %s for ID %d\n", mimeSubtype, codec);
}
}
}
if (udptlportno != -1) {
int found = 0, x;
old = 0;
/* Scan trough the a= lines for T38 attributes and set apropriate fileds */
iterator = req->sdp_start;
while ((a = get_sdp_iterate(&iterator, req, "a"))[0] != '\0') {
if ((sscanf(a, "T38FaxMaxBuffer:%d", &x) == 1)) {
found = 1;
ast_debug(3, "MaxBufferSize:%d\n", x);
} else if ((sscanf(a, "T38MaxBitRate:%d", &x) == 1) || (sscanf(a, "T38FaxMaxRate:%d", &x) == 1)) {
found = 1;
ast_debug(3, "T38MaxBitRate: %d\n", x);
switch (x) {
case 14400:
peert38capability |= T38FAX_RATE_14400 | T38FAX_RATE_12000 | T38FAX_RATE_9600 | T38FAX_RATE_7200 | T38FAX_RATE_4800 | T38FAX_RATE_2400;
break;
case 12000:
peert38capability |= T38FAX_RATE_12000 | T38FAX_RATE_9600 | T38FAX_RATE_7200 | T38FAX_RATE_4800 | T38FAX_RATE_2400;
break;
case 9600:
peert38capability |= T38FAX_RATE_9600 | T38FAX_RATE_7200 | T38FAX_RATE_4800 | T38FAX_RATE_2400;
break;
case 7200:
peert38capability |= T38FAX_RATE_7200 | T38FAX_RATE_4800 | T38FAX_RATE_2400;
break;
case 4800:
peert38capability |= T38FAX_RATE_4800 | T38FAX_RATE_2400;
break;
case 2400:
peert38capability |= T38FAX_RATE_2400;
break;
}
} else if ((sscanf(a, "T38FaxVersion:%d", &x) == 1)) {
found = 1;
ast_debug(3, "FaxVersion: %d\n", x);
if (x == 0)
peert38capability |= T38FAX_VERSION_0;
else if (x == 1)
peert38capability |= T38FAX_VERSION_1;
} else if ((sscanf(a, "T38FaxMaxDatagram:%d", &x) == 1) || (sscanf(a, "T38MaxDatagram:%d", &x) == 1)) {
found = 1;
ast_debug(3, "FaxMaxDatagram: %d\n", x);
ast_udptl_set_far_max_datagram(p->udptl, x);
ast_udptl_set_local_max_datagram(p->udptl, x);
} else if ((strncmp(a, "T38FaxFillBitRemoval", 20) == 0)) {
found = 1;
if(sscanf(a, "T38FaxFillBitRemoval:%d", &x) == 1) {
ast_debug(3, "FillBitRemoval: %d\n", x);
if(x == 1)
peert38capability |= T38FAX_FILL_BIT_REMOVAL;
} else {
ast_debug(3, "FillBitRemoval\n");
peert38capability |= T38FAX_FILL_BIT_REMOVAL;
}
} else if ((strncmp(a, "T38FaxTranscodingMMR", 20) == 0)) {
found = 1;
if(sscanf(a, "T38FaxTranscodingMMR:%d", &x) == 1) {
ast_debug(3, "Transcoding MMR: %d\n", x);
if(x == 1)
peert38capability |= T38FAX_TRANSCODING_MMR;
} else {
ast_debug(3, "Transcoding MMR\n");
peert38capability |= T38FAX_TRANSCODING_MMR;
}
} else if ((strncmp(a, "T38FaxTranscodingJBIG", 21) == 0)) {
found = 1;
if(sscanf(a, "T38FaxTranscodingJBIG:%d", &x) == 1) {
ast_debug(3, "Transcoding JBIG: %d\n", x);
if(x == 1)
peert38capability |= T38FAX_TRANSCODING_JBIG;
} else {
ast_debug(3, "Transcoding JBIG\n");
peert38capability |= T38FAX_TRANSCODING_JBIG;
}
} else if ((sscanf(a, "T38FaxRateManagement:%255s", s) == 1)) {
found = 1;
ast_debug(3, "RateManagement: %s\n", s);
if (!strcasecmp(s, "localTCF"))
peert38capability |= T38FAX_RATE_MANAGEMENT_LOCAL_TCF;
else if (!strcasecmp(s, "transferredTCF"))
peert38capability |= T38FAX_RATE_MANAGEMENT_TRANSFERED_TCF;
} else if ((sscanf(a, "T38FaxUdpEC:%255s", s) == 1)) {
found = 1;
ast_debug(3, "UDP EC: %s\n", s);
if (!strcasecmp(s, "t38UDPRedundancy")) {
peert38capability |= T38FAX_UDP_EC_REDUNDANCY;
ast_udptl_set_error_correction_scheme(p->udptl, UDPTL_ERROR_CORRECTION_REDUNDANCY);
} else if (!strcasecmp(s, "t38UDPFEC")) {
peert38capability |= T38FAX_UDP_EC_FEC;
ast_udptl_set_error_correction_scheme(p->udptl, UDPTL_ERROR_CORRECTION_FEC);
} else {
peert38capability |= T38FAX_UDP_EC_NONE;
ast_udptl_set_error_correction_scheme(p->udptl, UDPTL_ERROR_CORRECTION_NONE);
}
}
}
if (found) { /* Some cisco equipment returns nothing beside c= and m= lines in 200 OK T38 SDP */
p->t38.peercapability = peert38capability;
p->t38.jointcapability = (peert38capability & 255); /* Put everything beside supported speeds settings */
peert38capability &= (T38FAX_RATE_14400 | T38FAX_RATE_12000 | T38FAX_RATE_9600 | T38FAX_RATE_7200 | T38FAX_RATE_4800 | T38FAX_RATE_2400);
p->t38.jointcapability |= (peert38capability & p->t38.capability); /* Put the lower of our's and peer's speed */
}
if (debug)
ast_debug(1, "Our T38 capability = (%d), peer T38 capability (%d), joint T38 capability (%d)\n",
p->t38.capability,
p->t38.peercapability,
p->t38.jointcapability);
/* Remote party offers T38, we need to update state */
if (t38action == SDP_T38_ACCEPT) {
if (p->t38.state == T38_LOCAL_REINVITE)
change_t38_state(p, T38_ENABLED);
} else if (t38action == SDP_T38_INITIATE) {
if (p->owner && p->lastinvite) {
change_t38_state(p, T38_PEER_REINVITE); /* T38 Offered in re-invite from remote party */
} else {
change_t38_state(p, T38_PEER_DIRECT); /* T38 Offered directly from peer in first invite */
}
}
} else {
change_t38_state(p, T38_DISABLED);
}
/* Now gather all of the codecs that we are asked for: */
ast_rtp_codecs_payload_formats(&newaudiortp, &peercapability, &peernoncodeccapability);
ast_rtp_codecs_payload_formats(&newvideortp, &vpeercapability, &vpeernoncodeccapability);
ast_rtp_codecs_payload_formats(&newtextrtp, &tpeercapability, &tpeernoncodeccapability);
newjointcapability = p->capability & (peercapability | vpeercapability | tpeercapability);
newpeercapability = (peercapability | vpeercapability | tpeercapability);
newnoncodeccapability = p->noncodeccapability & peernoncodeccapability;
if (debug) {
/* shame on whoever coded this.... */
char s1[SIPBUFSIZE], s2[SIPBUFSIZE], s3[SIPBUFSIZE], s4[SIPBUFSIZE], s5[SIPBUFSIZE];
ast_verbose("Capabilities: us - %s, peer - audio=%s/video=%s/text=%s, combined - %s\n",
ast_getformatname_multiple(s1, SIPBUFSIZE, p->capability),
ast_getformatname_multiple(s2, SIPBUFSIZE, peercapability),
ast_getformatname_multiple(s3, SIPBUFSIZE, vpeercapability),
ast_getformatname_multiple(s4, SIPBUFSIZE, tpeercapability),
ast_getformatname_multiple(s5, SIPBUFSIZE, newjointcapability));
}
if (debug) {
struct ast_str *s1 = ast_str_alloca(SIPBUFSIZE);
struct ast_str *s2 = ast_str_alloca(SIPBUFSIZE);
struct ast_str *s3 = ast_str_alloca(SIPBUFSIZE);
ast_verbose("Non-codec capabilities (dtmf): us - %s, peer - %s, combined - %s\n",
ast_rtp_lookup_mime_multiple2(s1, p->noncodeccapability, 0, 0),
ast_rtp_lookup_mime_multiple2(s2, peernoncodeccapability, 0, 0),
ast_rtp_lookup_mime_multiple2(s3, newnoncodeccapability, 0, 0));
}
if (!newjointcapability) {
/* If T.38 was not negotiated either, totally bail out... */
if (!p->t38.jointcapability || !udptlportno) {
ast_log(LOG_NOTICE, "No compatible codecs, not accepting this offer!\n");
/* Do NOT Change current setting */
return -1;
} else {
ast_debug(3, "Have T.38 but no audio codecs, accepting offer anyway\n");
return 0;
}
}
/* We are now ready to change the sip session and p->rtp and p->vrtp with the offered codecs, since
they are acceptable */
p->jointcapability = newjointcapability; /* Our joint codec profile for this call */
p->peercapability = newpeercapability; /* The other sides capability in latest offer */
p->jointnoncodeccapability = newnoncodeccapability; /* DTMF capabilities */
if (ast_test_flag(&p->flags[1], SIP_PAGE2_PREFERRED_CODEC)) { /* respond with single most preferred joint codec, limiting the other side's choice */
p->jointcapability = ast_codec_choose(&p->prefs, p->jointcapability, 1);
}
if (p->jointcapability & AST_FORMAT_T140RED) {
p->red = 1;
ast_rtp_red_init(p->trtp, 300, red_data_pt, 2);
} else {
p->red = 0;
}
ast_rtp_codecs_payloads_copy(&newaudiortp, ast_rtp_instance_get_codecs(p->rtp), p->rtp);
if (p->vrtp) {
ast_rtp_codecs_payloads_copy(&newvideortp, ast_rtp_instance_get_codecs(p->vrtp), p->vrtp);
}
if (p->trtp) {
ast_rtp_codecs_payloads_copy(&newtextrtp, ast_rtp_instance_get_codecs(p->trtp), p->trtp);
}
if (ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_AUTO) {
ast_clear_flag(&p->flags[0], SIP_DTMF);
if (newnoncodeccapability & AST_RTP_DTMF) {
/* XXX Would it be reasonable to drop the DSP at this point? XXX */
ast_set_flag(&p->flags[0], SIP_DTMF_RFC2833);
/* Since RFC2833 is now negotiated we need to change some properties of the RTP stream */
ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_DTMF, 1);
ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_DTMF_COMPENSATE, ast_test_flag(&p->flags[1], SIP_PAGE2_RFC2833_COMPENSATE));
} else {
ast_set_flag(&p->flags[0], SIP_DTMF_INBAND);
}
}
/* Setup audio port number */
if (p->rtp && sin.sin_port) {
ast_rtp_instance_set_remote_address(p->rtp, &sin);
if (debug)
ast_verbose("Peer audio RTP is at port %s:%d\n", ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port));
}
/* Setup video port number */
if (p->vrtp && vsin.sin_port) {
ast_rtp_instance_set_remote_address(p->vrtp, &vsin);
if (debug)
ast_verbose("Peer video RTP is at port %s:%d\n", ast_inet_ntoa(vsin.sin_addr), ntohs(vsin.sin_port));
}
/* Setup text port number */
if (p->trtp && tsin.sin_port) {
ast_rtp_instance_set_remote_address(p->trtp, &tsin);
if (debug)
ast_verbose("Peer text RTP is at port %s:%d\n", ast_inet_ntoa(tsin.sin_addr), ntohs(tsin.sin_port));
}
/* Ok, we're going with this offer */
ast_debug(2, "We're settling with these formats: %s\n", ast_getformatname_multiple(buf, SIPBUFSIZE, p->jointcapability));
if (!p->owner) /* There's no open channel owning us so we can return here. For a re-invite or so, we proceed */
return 0;
ast_debug(4, "We have an owner, now see if we need to change this call\n");
if (!(p->owner->nativeformats & p->jointcapability) && (p->jointcapability & AST_FORMAT_AUDIO_MASK)) {
if (debug) {
char s1[SIPBUFSIZE], s2[SIPBUFSIZE];
ast_debug(1, "Oooh, we need to change our audio formats since our peer supports only %s and not %s\n",
ast_getformatname_multiple(s1, SIPBUFSIZE, p->jointcapability),
ast_getformatname_multiple(s2, SIPBUFSIZE, p->owner->nativeformats));
}
p->owner->nativeformats = ast_codec_choose(&p->prefs, p->jointcapability, 1) | (p->capability & vpeercapability) | (p->capability & tpeercapability);
ast_set_read_format(p->owner, p->owner->readformat);
ast_set_write_format(p->owner, p->owner->writeformat);
}
if (ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD) && sin.sin_addr.s_addr && (!sendonly || sendonly == -1)) {
ast_queue_control(p->owner, AST_CONTROL_UNHOLD);
/* Activate a re-invite */
ast_queue_frame(p->owner, &ast_null_frame);
/* Queue Manager Unhold event */
append_history(p, "Unhold", "%s", req->data->str);
if (sip_cfg.callevents)
manager_event(EVENT_FLAG_CALL, "Hold",
"Status: Off\r\n"
"Channel: %s\r\n"
"Uniqueid: %s\r\n",
p->owner->name,
p->owner->uniqueid);
if (sip_cfg.notifyhold)
sip_peer_hold(p, FALSE);
ast_clear_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD); /* Clear both flags */
} else if (!sin.sin_addr.s_addr || (sendonly && sendonly != -1)) {
int already_on_hold = ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD);
ast_queue_control_data(p->owner, AST_CONTROL_HOLD,
S_OR(p->mohsuggest, NULL),
!ast_strlen_zero(p->mohsuggest) ? strlen(p->mohsuggest) + 1 : 0);
if (sendonly)
ast_rtp_instance_stop(p->rtp);
/* RTCP needs to go ahead, even if we're on hold!!! */
/* Activate a re-invite */
ast_queue_frame(p->owner, &ast_null_frame);
/* Queue Manager Hold event */
append_history(p, "Hold", "%s", req->data->str);
if (sip_cfg.callevents && !ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD)) {
manager_event(EVENT_FLAG_CALL, "Hold",
"Status: On\r\n"
"Channel: %s\r\n"
"Uniqueid: %s\r\n",
p->owner->name,
p->owner->uniqueid);
}
if (sendonly == 1) /* One directional hold (sendonly/recvonly) */
ast_set_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD_ONEDIR);
else if (sendonly == 2) /* Inactive stream */
ast_set_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD_INACTIVE);
else
ast_set_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD_ACTIVE);
if (sip_cfg.notifyhold && !already_on_hold)
sip_peer_hold(p, TRUE);
}
return 0;
}
#ifdef LOW_MEMORY
static void ts_ast_rtp_destroy(void *data)
{
struct ast_rtp *tmp = data;
ast_rtp_destroy(tmp);
}
#endif
/*! \brief Add header to SIP message */
static int add_header(struct sip_request *req, const char *var, const char *value)
{
if (req->headers == SIP_MAX_HEADERS) {
ast_log(LOG_WARNING, "Out of SIP header space\n");
return -1;
}
if (req->lines) {
ast_log(LOG_WARNING, "Can't add more headers when lines have been added\n");
return -1;
}
if (sip_cfg.compactheaders) {
var = find_alias(var, var);
}
ast_str_append(&req->data, 0, "%s: %s\r\n", var, value);
req->header[req->headers] = req->len;
req->len = ast_str_strlen(req->data);
req->headers++;
return 0;
}
/*! \brief Add 'Content-Length' header to SIP message */
static int add_header_contentLength(struct sip_request *req, int len)
{
char clen[10];
snprintf(clen, sizeof(clen), "%d", len);
return add_header(req, "Content-Length", clen);
}
/*! \brief Add content (not header) to SIP message */
static int add_line(struct sip_request *req, const char *line)
{
if (req->lines == SIP_MAX_LINES) {
ast_log(LOG_WARNING, "Out of SIP line space\n");
return -1;
}
if (!req->lines)
/* Add extra empty return */
req->len += ast_str_append(&req->data, 0, "\r\n");
req->line[req->lines] = req->len;
ast_str_append(&req->data, 0, "%s", line);
req->len = ast_str_strlen(req->data);
req->lines++;
return 0;
}
/*! \brief Copy one header field from one request to another */
static int copy_header(struct sip_request *req, const struct sip_request *orig, const char *field)
{
const char *tmp = get_header(orig, field);
if (!ast_strlen_zero(tmp)) /* Add what we're responding to */
return add_header(req, field, tmp);
ast_log(LOG_NOTICE, "No field '%s' present to copy\n", field);
return -1;
}
/*! \brief Copy all headers from one request to another */
static int copy_all_header(struct sip_request *req, const struct sip_request *orig, const char *field)
{
int start = 0;
int copied = 0;
for (;;) {
const char *tmp = __get_header(orig, field, &start);
if (ast_strlen_zero(tmp))
break;
/* Add what we're responding to */
add_header(req, field, tmp);
copied++;
}
return copied ? 0 : -1;
}
/*! \brief Copy SIP VIA Headers from the request to the response
\note If the client indicates that it wishes to know the port we received from,
it adds ;rport without an argument to the topmost via header. We need to
add the port number (from our point of view) to that parameter.
\verbatim
We always add ;received=<ip address> to the topmost via header.
\endverbatim
Received: RFC 3261, rport RFC 3581 */
static int copy_via_headers(struct sip_pvt *p, struct sip_request *req, const struct sip_request *orig, const char *field)
{
int copied = 0;
int start = 0;
for (;;) {
char new[512];
const char *oh = __get_header(orig, field, &start);
if (ast_strlen_zero(oh))
break;
if (!copied) { /* Only check for empty rport in topmost via header */
char leftmost[512], *others, *rport;
/* Only work on leftmost value */
ast_copy_string(leftmost, oh, sizeof(leftmost));
others = strchr(leftmost, ',');
if (others)
*others++ = '\0';
/* Find ;rport; (empty request) */
rport = strstr(leftmost, ";rport");
if (rport && *(rport+6) == '=')
rport = NULL; /* We already have a parameter to rport */
/* Check rport if NAT=yes or NAT=rfc3581 (which is the default setting) */
if (rport && ((ast_test_flag(&p->flags[0], SIP_NAT) == SIP_NAT_ALWAYS) || (ast_test_flag(&p->flags[0], SIP_NAT) == SIP_NAT_RFC3581))) {
/* We need to add received port - rport */
char *end;
rport = strstr(leftmost, ";rport");
if (rport) {
end = strchr(rport + 1, ';');
if (end)
memmove(rport, end, strlen(end) + 1);
else
*rport = '\0';
}
/* Add rport to first VIA header if requested */
snprintf(new, sizeof(new), "%s;received=%s;rport=%d%s%s",
leftmost, ast_inet_ntoa(p->recv.sin_addr),
ntohs(p->recv.sin_port),
others ? "," : "", others ? others : "");
} else {
/* We should *always* add a received to the topmost via */
snprintf(new, sizeof(new), "%s;received=%s%s%s",
leftmost, ast_inet_ntoa(p->recv.sin_addr),
others ? "," : "", others ? others : "");
}
oh = new; /* the header to copy */
} /* else add the following via headers untouched */
add_header(req, field, oh);
copied++;
}
if (!copied) {
ast_log(LOG_NOTICE, "No header field '%s' present to copy\n", field);
return -1;
}
return 0;
}
/*! \brief Add route header into request per learned route */
static void add_route(struct sip_request *req, struct sip_route *route)
{
char r[SIPBUFSIZE*2], *p;
int n, rem = sizeof(r);
if (!route)
return;
p = r;
for (;route ; route = route->next) {
n = strlen(route->hop);
if (rem < n+3) /* we need room for ",<route>" */
break;
if (p != r) { /* add a separator after fist route */
*p++ = ',';
--rem;
}
*p++ = '<';
ast_copy_string(p, route->hop, rem); /* cannot fail */
p += n;
*p++ = '>';
rem -= (n+2);
}
*p = '\0';
add_header(req, "Route", r);
}
/*! \brief Set destination from SIP URI
*
* Parse uri to h (host) and port - uri is already just the part inside the <>
* general form we are expecting is sip[s]:username[:password][;parameter]@host[:port][;...]
* If there's a port given, turn NAPTR/SRV off. NAPTR might indicate SIPS preference even
* for SIP: uri's
*
* If there's a sips: uri scheme, TLS will be required.
*/
static void set_destination(struct sip_pvt *p, char *uri)
{
char *h, *maddr, hostname[256];
int port, hn;
struct hostent *hp;
struct ast_hostent ahp;
int debug=sip_debug_test_pvt(p);
int tls_on = FALSE;
int use_dns = sip_cfg.srvlookup;
if (debug)
ast_verbose("set_destination: Parsing <%s> for address/port to send to\n", uri);
/* Find and parse hostname */
h = strchr(uri, '@');
if (h)
++h;
else {
h = uri;
if (!strncasecmp(h, "sip:", 4)) {
h += 4;
} else if (!strncasecmp(h, "sips:", 5)) {
h += 5;
tls_on = TRUE;
}
}
hn = strcspn(h, ":;>") + 1;
if (hn > sizeof(hostname))
hn = sizeof(hostname);
ast_copy_string(hostname, h, hn);
/* XXX bug here if string has been trimmed to sizeof(hostname) */
h += hn - 1;
/* Is "port" present? if not default to STANDARD_SIP_PORT */
if (*h == ':') {
/* Parse port */
++h;
port = strtol(h, &h, 10);
use_dns = FALSE;
} else
port = tls_on ? STANDARD_TLS_PORT : STANDARD_SIP_PORT;
/* Got the hostname:port - but maybe there's a "maddr=" to override address? */
maddr = strstr(h, "maddr=");
if (maddr) {
maddr += 6;
hn = strspn(maddr, "0123456789.") + 1;
if (hn > sizeof(hostname))
hn = sizeof(hostname);
ast_copy_string(hostname, maddr, hn);
}
/*! \todo XXX If we have use_dns on, then look for NAPTR/SRV, otherwise, just look for A records */
hp = ast_gethostbyname(hostname, &ahp);
if (hp == NULL) {
ast_log(LOG_WARNING, "Can't find address for host '%s'\n", hostname);
return;
}
p->sa.sin_family = AF_INET;
memcpy(&p->sa.sin_addr, hp->h_addr, sizeof(p->sa.sin_addr));
p->sa.sin_port = htons(port);
if (debug)
ast_verbose("set_destination: set destination to %s, port %d\n", ast_inet_ntoa(p->sa.sin_addr), port);
}
/*! \brief Initialize SIP response, based on SIP request */
static int init_resp(struct sip_request *resp, const char *msg)
{
/* Initialize a response */
memset(resp, 0, sizeof(*resp));
resp->method = SIP_RESPONSE;
if (!(resp->data = ast_str_create(SIP_MIN_PACKET)))
return -1;
resp->header[0] = 0;
ast_str_set(&resp->data, 0, "SIP/2.0 %s\r\n", msg);
resp->len = resp->data->used;
resp->headers++;
return 0;
}
/*! \brief Initialize SIP request */
static int init_req(struct sip_request *req, int sipmethod, const char *recip)
{
/* Initialize a request */
memset(req, 0, sizeof(*req));
if (!(req->data = ast_str_create(SIP_MIN_PACKET)))
return -1;
req->method = sipmethod;
req->header[0] = 0;
ast_str_set(&req->data, 0, "%s %s SIP/2.0\r\n", sip_methods[sipmethod].text, recip);
req->len = ast_str_strlen(req->data);
req->headers++;
return 0;
}
/*! \brief Test if this response needs a contact header */
static inline int resp_needs_contact(const char *msg, enum sipmethod method) {
/* Requirements for Contact header inclusion in responses generated
* from the header tables found in the following RFCs. Where the
* Contact header was marked mandatory (m) or optional (o) this
* function returns 1.
*
* - RFC 3261 (ACK, BYE, CANCEL, INVITE, OPTIONS, REGISTER)
* - RFC 2976 (INFO)
* - RFC 3262 (PRACK)
* - RFC 3265 (SUBSCRIBE, NOTIFY)
* - RFC 3311 (UPDATE)
* - RFC 3428 (MESSAGE)
* - RFC 3515 (REFER)
* - RFC 3903 (PUBLISH)
*/
switch (method) {
/* 1xx, 2xx, 3xx, 485 */
case SIP_INVITE:
case SIP_UPDATE:
case SIP_SUBSCRIBE:
case SIP_NOTIFY:
if ((msg[0] >= '1' && msg[0] <= '3') || !strncmp(msg, "485", 3))
return 1;
break;
/* 2xx, 3xx, 485 */
case SIP_REGISTER:
case SIP_OPTIONS:
if (msg[0] == '2' || msg[0] == '3' || !strncmp(msg, "485", 3))
return 1;
break;
/* 3xx, 485 */
case SIP_BYE:
case SIP_PRACK:
case SIP_MESSAGE:
case SIP_PUBLISH:
if (msg[0] == '3' || !strncmp(msg, "485", 3))
return 1;
break;
/* 2xx, 3xx, 4xx, 5xx, 6xx */
case SIP_REFER:
if (msg[0] >= '2' && msg[0] <= '6')
return 1;
break;
/* contact will not be included for everything else */
case SIP_ACK:
case SIP_CANCEL:
case SIP_INFO:
case SIP_PING:
default:
return 0;
}
return 0;
}
/*! \brief Prepare SIP response packet */
static int respprep(struct sip_request *resp, struct sip_pvt *p, const char *msg, const struct sip_request *req)
{
char newto[256];
const char *ot;
init_resp(resp, msg);
copy_via_headers(p, resp, req, "Via");
if (msg[0] == '1' || msg[0] == '2')
copy_all_header(resp, req, "Record-Route");
copy_header(resp, req, "From");
ot = get_header(req, "To");
if (!strcasestr(ot, "tag=") && strncmp(msg, "100", 3)) {
/* Add the proper tag if we don't have it already. If they have specified
their tag, use it. Otherwise, use our own tag */
if (!ast_strlen_zero(p->theirtag) && ast_test_flag(&p->flags[0], SIP_OUTGOING))
snprintf(newto, sizeof(newto), "%s;tag=%s", ot, p->theirtag);
else if (p->tag && !ast_test_flag(&p->flags[0], SIP_OUTGOING))
snprintf(newto, sizeof(newto), "%s;tag=%s", ot, p->tag);
else
ast_copy_string(newto, ot, sizeof(newto));
ot = newto;
}
add_header(resp, "To", ot);
copy_header(resp, req, "Call-ID");
copy_header(resp, req, "CSeq");
if (!ast_strlen_zero(global_useragent))
add_header(resp, "Server", global_useragent);
add_header(resp, "Allow", ALLOWED_METHODS);
add_header(resp, "Supported", SUPPORTED_EXTENSIONS);
/* If this is an invite, add Session-Timers related headers if the feature is active for this session */
if (p->method == SIP_INVITE && p->stimer && p->stimer->st_active == TRUE && p->stimer->st_active_peer_ua == TRUE) {
char se_hdr[256];
snprintf(se_hdr, sizeof(se_hdr), "%d;refresher=%s", p->stimer->st_interval,
strefresher2str(p->stimer->st_ref));
add_header(resp, "Require", "timer");
add_header(resp, "Session-Expires", se_hdr);
}
if (msg[0] == '2' && (p->method == SIP_SUBSCRIBE || p->method == SIP_REGISTER)) {
/* For registration responses, we also need expiry and
contact info */
char tmp[256];
snprintf(tmp, sizeof(tmp), "%d", p->expiry);
add_header(resp, "Expires", tmp);
if (p->expiry) { /* Only add contact if we have an expiry time */
char contact[SIPBUFSIZE];
snprintf(contact, sizeof(contact), "%s;expires=%d", p->our_contact, p->expiry);
add_header(resp, "Contact", contact); /* Not when we unregister */
}
} else if (!ast_strlen_zero(p->our_contact) && resp_needs_contact(msg, p->method)) {
add_header(resp, "Contact", p->our_contact);
}
if (!ast_strlen_zero(p->url)) {
add_header(resp, "Access-URL", p->url);
ast_string_field_set(p, url, NULL);
}
return 0;
}
/*! \brief Initialize a SIP request message (not the initial one in a dialog) */
static int reqprep(struct sip_request *req, struct sip_pvt *p, int sipmethod, int seqno, int newbranch)
{
struct sip_request *orig = &p->initreq;
char stripped[80];
char tmp[80];
char newto[256];
const char *c;
const char *ot, *of;
int is_strict = FALSE; /*!< Strict routing flag */
int is_outbound = ast_test_flag(&p->flags[0], SIP_OUTGOING); /* Session direction */
memset(req, 0, sizeof(struct sip_request));
snprintf(p->lastmsg, sizeof(p->lastmsg), "Tx: %s", sip_methods[sipmethod].text);
if (!seqno) {
p->ocseq++;
seqno = p->ocseq;
}
if (sipmethod == SIP_CANCEL || sipmethod == SIP_INVITE) {
p->branch = p->invite_branch;
build_via(p);
} else if (newbranch) {
p->branch ^= ast_random();
build_via(p);
}
/* Check for strict or loose router */
if (p->route && !ast_strlen_zero(p->route->hop) && strstr(p->route->hop, ";lr") == NULL) {
is_strict = TRUE;
if (sipdebug)
ast_debug(1, "Strict routing enforced for session %s\n", p->callid);
}
if (sipmethod == SIP_CANCEL)
c = REQ_OFFSET_TO_STR(&p->initreq, rlPart2); /* Use original URI */
else if (sipmethod == SIP_ACK) {
/* Use URI from Contact: in 200 OK (if INVITE)
(we only have the contacturi on INVITEs) */
if (!ast_strlen_zero(p->okcontacturi))
c = is_strict ? p->route->hop : p->okcontacturi;
else
c = REQ_OFFSET_TO_STR(&p->initreq, rlPart2);
} else if (!ast_strlen_zero(p->okcontacturi))
c = is_strict ? p->route->hop : p->okcontacturi; /* Use for BYE or REINVITE */
else if (!ast_strlen_zero(p->uri))
c = p->uri;
else {
char *n;
/* We have no URI, use To: or From: header as URI (depending on direction) */
ast_copy_string(stripped, get_header(orig, is_outbound ? "To" : "From"),
sizeof(stripped));
n = get_in_brackets(stripped);
c = remove_uri_parameters(n);
}
init_req(req, sipmethod, c);
snprintf(tmp, sizeof(tmp), "%d %s", seqno, sip_methods[sipmethod].text);
add_header(req, "Via", p->via);
if (p->route) {
set_destination(p, p->route->hop);
add_route(req, is_strict ? p->route->next : p->route);
}
add_header(req, "Max-Forwards", DEFAULT_MAX_FORWARDS);
ot = get_header(orig, "To");
of = get_header(orig, "From");
/* Add tag *unless* this is a CANCEL, in which case we need to send it exactly
as our original request, including tag (or presumably lack thereof) */
if (!strcasestr(ot, "tag=") && sipmethod != SIP_CANCEL) {
/* Add the proper tag if we don't have it already. If they have specified
their tag, use it. Otherwise, use our own tag */
if (is_outbound && !ast_strlen_zero(p->theirtag))
snprintf(newto, sizeof(newto), "%s;tag=%s", ot, p->theirtag);
else if (!is_outbound)
snprintf(newto, sizeof(newto), "%s;tag=%s", ot, p->tag);
else
snprintf(newto, sizeof(newto), "%s", ot);
ot = newto;
}
if (is_outbound) {
add_header(req, "From", of);
add_header(req, "To", ot);
} else {
add_header(req, "From", ot);
add_header(req, "To", of);
}
/* Do not add Contact for MESSAGE, BYE and Cancel requests */
if (sipmethod != SIP_BYE && sipmethod != SIP_CANCEL && sipmethod != SIP_MESSAGE)
add_header(req, "Contact", p->our_contact);
copy_header(req, orig, "Call-ID");
add_header(req, "CSeq", tmp);
if (!ast_strlen_zero(global_useragent))
add_header(req, "User-Agent", global_useragent);
if (!ast_strlen_zero(p->url)) {
add_header(req, "Access-URL", p->url);
ast_string_field_set(p, url, NULL);
}
/* Add Session-Timers related headers if the feature is active for this session.
An exception to this behavior is the ACK request. Since Asterisk never requires
session-timers support from a remote end-point (UAS) in an INVITE, it must
not send 'Require: timer' header in the ACK request.
This should only be added in the INVITE transactions, not MESSAGE or REFER or other
in-dialog messages.
*/
if (p->stimer && p->stimer->st_active == TRUE && p->stimer->st_active_peer_ua == TRUE
&& sipmethod == SIP_INVITE) {
char se_hdr[256];
snprintf(se_hdr, sizeof(se_hdr), "%d;refresher=%s", p->stimer->st_interval,
strefresher2str(p->stimer->st_ref));
add_header(req, "Require", "timer");
add_header(req, "Session-Expires", se_hdr);
snprintf(se_hdr, sizeof(se_hdr), "%d", st_get_se(p, FALSE));
add_header(req, "Min-SE", se_hdr);
}
return 0;
}
/*! \brief Base transmit response function */
static int __transmit_response(struct sip_pvt *p, const char *msg, const struct sip_request *req, enum xmittype reliable)
{
struct sip_request resp;
int seqno = 0;
if (reliable && (sscanf(get_header(req, "CSeq"), "%d ", &seqno) != 1)) {
ast_log(LOG_WARNING, "Unable to determine sequence number from '%s'\n", get_header(req, "CSeq"));
return -1;
}
respprep(&resp, p, msg, req);
if (ast_test_flag(&p->flags[0], SIP_SENDRPID)
&& ast_test_flag(&p->flags[1], SIP_PAGE2_CONNECTLINEUPDATE_PEND)
&& (!strncmp(msg, "180", 3) || !strncmp(msg, "183", 3))) {
ast_clear_flag(&p->flags[1], SIP_PAGE2_CONNECTLINEUPDATE_PEND);
add_rpid(&resp, p);
}
add_header_contentLength(&resp, 0);
/* If we are cancelling an incoming invite for some reason, add information
about the reason why we are doing this in clear text */
if (p->method == SIP_INVITE && msg[0] != '1' && p->owner && p->owner->hangupcause) {
char buf[10];
add_header(&resp, "X-Asterisk-HangupCause", ast_cause2str(p->owner->hangupcause));
snprintf(buf, sizeof(buf), "%d", p->owner->hangupcause);
add_header(&resp, "X-Asterisk-HangupCauseCode", buf);
}
return send_response(p, &resp, reliable, seqno);
}
static int temp_pvt_init(void *data)
{
struct sip_pvt *p = data;
p->do_history = 0; /* XXX do we need it ? isn't already all 0 ? */
return ast_string_field_init(p, 512);
}
static void temp_pvt_cleanup(void *data)
{
struct sip_pvt *p = data;
ast_string_field_free_memory(p);
ast_free(data);
}
/*! \brief Transmit response, no retransmits, using a temporary pvt structure */
static int transmit_response_using_temp(ast_string_field callid, struct sockaddr_in *sin, int useglobal_nat, const int intended_method, const struct sip_request *req, const char *msg)
{
struct sip_pvt *p = NULL;
if (!(p = ast_threadstorage_get(&ts_temp_pvt, sizeof(*p)))) {
ast_log(LOG_ERROR, "Failed to get temporary pvt\n");
return -1;
}
/* XXX the structure may be dirty from previous usage.
* Here we should state clearly how we should reinitialize it
* before using it.
* E.g. certainly the threadstorage should be left alone,
* but other thihngs such as flags etc. maybe need cleanup ?
*/
/* Initialize the bare minimum */
p->method = intended_method;
if (!sin)
p->ourip = internip;
else {
p->sa = *sin;
ast_sip_ouraddrfor(&p->sa.sin_addr, &p->ourip);
}
p->branch = ast_random();
make_our_tag(p->tag, sizeof(p->tag));
p->ocseq = INITIAL_CSEQ;
if (useglobal_nat && sin) {
ast_copy_flags(&p->flags[0], &global_flags[0], SIP_NAT);
p->recv = *sin;
do_setnat(p, ast_test_flag(&p->flags[0], SIP_NAT) & SIP_NAT_ROUTE);
}
ast_string_field_set(p, fromdomain, default_fromdomain);
build_via(p);
ast_string_field_set(p, callid, callid);
copy_socket_data(&p->socket, &req->socket);
/* Use this temporary pvt structure to send the message */
__transmit_response(p, msg, req, XMIT_UNRELIABLE);
/* Free the string fields, but not the pool space */
ast_string_field_init(p, 0);
return 0;
}
/*! \brief Transmit response, no retransmits */
static int transmit_response(struct sip_pvt *p, const char *msg, const struct sip_request *req)
{
return __transmit_response(p, msg, req, XMIT_UNRELIABLE);
}
/*! \brief Transmit response, no retransmits */
static int transmit_response_with_unsupported(struct sip_pvt *p, const char *msg, const struct sip_request *req, const char *unsupported)
{
struct sip_request resp;
respprep(&resp, p, msg, req);
append_date(&resp);
add_header(&resp, "Unsupported", unsupported);
add_header_contentLength(&resp, 0);
return send_response(p, &resp, XMIT_UNRELIABLE, 0);
}
/*! \brief Transmit 422 response with Min-SE header (Session-Timers) */
static int transmit_response_with_minse(struct sip_pvt *p, const char *msg, const struct sip_request *req, int minse_int)
{
struct sip_request resp;
char minse_str[20];
respprep(&resp, p, msg, req);
append_date(&resp);
snprintf(minse_str, sizeof(minse_str), "%d", minse_int);
add_header(&resp, "Min-SE", minse_str);
add_header_contentLength(&resp, 0);
return send_response(p, &resp, XMIT_UNRELIABLE, 0);
}
/*! \brief Transmit response, Make sure you get an ACK
This is only used for responses to INVITEs, where we need to make sure we get an ACK
*/
static int transmit_response_reliable(struct sip_pvt *p, const char *msg, const struct sip_request *req)
{
return __transmit_response(p, msg, req, req->ignore ? XMIT_UNRELIABLE : XMIT_CRITICAL);
}
/*! \brief Append date to SIP message */
static void append_date(struct sip_request *req)
{
char tmpdat[256];
struct tm tm;
time_t t = time(NULL);
gmtime_r(&t, &tm);
strftime(tmpdat, sizeof(tmpdat), "%a, %d %b %Y %T GMT", &tm);
add_header(req, "Date", tmpdat);
}
/*! \brief Append date and content length before transmitting response */
static int transmit_response_with_date(struct sip_pvt *p, const char *msg, const struct sip_request *req)
{
struct sip_request resp;
respprep(&resp, p, msg, req);
append_date(&resp);
add_header_contentLength(&resp, 0);
return send_response(p, &resp, XMIT_UNRELIABLE, 0);
}
/*! \brief Append Accept header, content length before transmitting response */
static int transmit_response_with_allow(struct sip_pvt *p, const char *msg, const struct sip_request *req, enum xmittype reliable)
{
struct sip_request resp;
respprep(&resp, p, msg, req);
add_header(&resp, "Accept", "application/sdp");
add_header_contentLength(&resp, 0);
return send_response(p, &resp, reliable, 0);
}
/*! \brief Respond with authorization request */
static int transmit_response_with_auth(struct sip_pvt *p, const char *msg, const struct sip_request *req, const char *randdata, enum xmittype reliable, const char *header, int stale)
{
struct sip_request resp;
char tmp[512];
int seqno = 0;
if (reliable && (sscanf(get_header(req, "CSeq"), "%d ", &seqno) != 1)) {
ast_log(LOG_WARNING, "Unable to determine sequence number from '%s'\n", get_header(req, "CSeq"));
return -1;
}
/* Stale means that they sent us correct authentication, but
based it on an old challenge (nonce) */
snprintf(tmp, sizeof(tmp), "Digest algorithm=MD5, realm=\"%s\", nonce=\"%s\"%s", sip_cfg.realm, randdata, stale ? ", stale=true" : "");
respprep(&resp, p, msg, req);
add_header(&resp, header, tmp);
add_header_contentLength(&resp, 0);
append_history(p, "AuthChal", "Auth challenge sent for %s - nc %d", p->username, p->noncecount);
return send_response(p, &resp, reliable, seqno);
}
/*! \brief Add text body to SIP message */
static int add_text(struct sip_request *req, const char *text)
{
/* XXX Convert \n's to \r\n's XXX */
add_header(req, "Content-Type", "text/plain;charset=UTF-8");
add_header_contentLength(req, strlen(text));
add_line(req, text);
return 0;
}
/*! \brief Add DTMF INFO tone to sip message
Mode = 0 for application/dtmf-relay (Cisco)
1 for application/dtmf
*/
static int add_digit(struct sip_request *req, char digit, unsigned int duration, int mode)
{
char tmp[256];
int event;
if (mode) {
/* Application/dtmf short version used by some implementations */
if (digit == '*')
event = 10;
else if (digit == '#')
event = 11;
else if ((digit >= 'A') && (digit <= 'D'))
event = 12 + digit - 'A';
else
event = atoi(&digit);
snprintf(tmp, sizeof(tmp), "%d\r\n", event);
add_header(req, "Content-Type", "application/dtmf");
add_header_contentLength(req, strlen(tmp));
add_line(req, tmp);
} else {
/* Application/dtmf-relay as documented by Cisco */
snprintf(tmp, sizeof(tmp), "Signal=%c\r\nDuration=%u\r\n", digit, duration);
add_header(req, "Content-Type", "application/dtmf-relay");
add_header_contentLength(req, strlen(tmp));
add_line(req, tmp);
}
return 0;
}
/*!
* \pre if p->owner exists, it must be locked
* \brief Add Remote-Party-ID header to SIP message
*/
static int add_rpid(struct sip_request *req, struct sip_pvt *p)
{
struct ast_str *tmp = ast_str_alloca(256);
char *lid_num = NULL;
char *lid_name = NULL;
int lid_pres;
const char *fromdomain;
const char *privacy = NULL;
const char *screen = NULL;
const char *anonymous_string = "\"Anonymous\" <anonymous@anonymous.invalid>";
if (!ast_test_flag(&p->flags[0], SIP_SENDRPID)) {
return 0;
}
if (p->owner && p->owner->connected.id.number)
lid_num = p->owner->connected.id.number;
if (p->owner && p->owner->connected.id.name)
lid_name = p->owner->connected.id.name;
lid_pres = (p->owner) ? p->owner->connected.id.number_presentation : AST_PRES_NUMBER_NOT_AVAILABLE;
if (ast_strlen_zero(lid_num))
return 0;
if (ast_strlen_zero(lid_name))
lid_name = lid_num;
fromdomain = S_OR(p->fromdomain, ast_inet_ntoa(p->ourip.sin_addr));
if (ast_test_flag(&p->flags[0], SIP_SENDRPID_PAI)) {
if ((lid_pres & AST_PRES_RESTRICTION) != AST_PRES_ALLOWED) {
ast_str_set(&tmp, -1, "%s", anonymous_string);
} else {
ast_str_set(&tmp, -1, "\"%s\" <sip:%s@%s>", lid_name, lid_num, fromdomain);
}
add_header(req, "P-Asserted-Identity", ast_str_buffer(tmp));
} else {
ast_str_set(&tmp, -1, "\"%s\" <sip:%s@%s>;party=%s", lid_name, lid_num, fromdomain, ast_test_flag(&p->flags[0], SIP_OUTGOING) ? "calling" : "called");
switch (lid_pres) {
case AST_PRES_ALLOWED_USER_NUMBER_NOT_SCREENED:
case AST_PRES_ALLOWED_USER_NUMBER_FAILED_SCREEN:
privacy = "off";
screen = "no";
break;
case AST_PRES_ALLOWED_USER_NUMBER_PASSED_SCREEN:
case AST_PRES_ALLOWED_NETWORK_NUMBER:
privacy = "off";
screen = "yes";
break;
case AST_PRES_PROHIB_USER_NUMBER_NOT_SCREENED:
case AST_PRES_PROHIB_USER_NUMBER_FAILED_SCREEN:
privacy = "full";
screen = "no";
break;
case AST_PRES_PROHIB_USER_NUMBER_PASSED_SCREEN:
case AST_PRES_PROHIB_NETWORK_NUMBER:
privacy = "full";
screen = "yes";
break;
case AST_PRES_NUMBER_NOT_AVAILABLE:
break;
default:
if ((lid_pres & AST_PRES_RESTRICTION) != AST_PRES_ALLOWED) {
privacy = "full";
}
else
privacy = "off";
screen = "no";
break;
}
if (!ast_strlen_zero(privacy) && !ast_strlen_zero(screen)) {
ast_str_append(&tmp, -1, ";privacy=%s;screen=%s", privacy, screen);
}
add_header(req, "Remote-Party-ID", ast_str_buffer(tmp));
}
return 0;
}
/*! \brief add XML encoded media control with update
\note XML: The only way to turn 0 bits of information into a few hundred. (markster) */
static int add_vidupdate(struct sip_request *req)
{
const char *xml_is_a_huge_waste_of_space =
"<?xml version=\"1.0\" encoding=\"utf-8\" ?>\r\n"
" <media_control>\r\n"
" <vc_primitive>\r\n"
" <to_encoder>\r\n"
" <picture_fast_update>\r\n"
" </picture_fast_update>\r\n"
" </to_encoder>\r\n"
" </vc_primitive>\r\n"
" </media_control>\r\n";
add_header(req, "Content-Type", "application/media_control+xml");
add_header_contentLength(req, strlen(xml_is_a_huge_waste_of_space));
add_line(req, xml_is_a_huge_waste_of_space);
return 0;
}
/*! \brief Add codec offer to SDP offer/answer body in INVITE or 200 OK */
static void add_codec_to_sdp(const struct sip_pvt *p, int codec,
struct ast_str **m_buf, struct ast_str **a_buf,
int debug, int *min_packet_size)
{
int rtp_code;
struct ast_format_list fmt;
if (debug)
ast_verbose("Adding codec 0x%x (%s) to SDP\n", codec, ast_getformatname(codec));
if ((rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(p->rtp), 1, codec)) == -1)
return;
if (p->rtp) {
struct ast_codec_pref *pref = &ast_rtp_instance_get_codecs(p->rtp)->pref;
fmt = ast_codec_pref_getsize(pref, codec);
} else /* I dont see how you couldn't have p->rtp, but good to check for and error out if not there like earlier code */
return;
ast_str_append(m_buf, 0, " %d", rtp_code);
ast_str_append(a_buf, 0, "a=rtpmap:%d %s/%d\r\n", rtp_code,
ast_rtp_lookup_mime_subtype2(1, codec,
ast_test_flag(&p->flags[0], SIP_G726_NONSTANDARD) ? AST_RTP_OPT_G726_NONSTANDARD : 0),
ast_rtp_lookup_sample_rate2(1, codec));
switch (codec) {
case AST_FORMAT_G729A:
/* Indicate that we don't support VAD (G.729 annex B) */
ast_str_append(a_buf, 0, "a=fmtp:%d annexb=no\r\n", rtp_code);
break;
case AST_FORMAT_G723_1:
/* Indicate that we don't support VAD (G.723.1 annex A) */
ast_str_append(a_buf, 0, "a=fmtp:%d annexa=no\r\n", rtp_code);
break;
case AST_FORMAT_ILBC:
/* Add information about us using only 20/30 ms packetization */
ast_str_append(a_buf, 0, "a=fmtp:%d mode=%d\r\n", rtp_code, fmt.cur_ms);
break;
case AST_FORMAT_SIREN7:
/* Indicate that we only expect 32Kbps */
ast_str_append(a_buf, 0, "a=fmtp:%d bitrate=32000\r\n", rtp_code);
break;
case AST_FORMAT_SIREN14:
/* Indicate that we only expect 48Kbps */
ast_str_append(a_buf, 0, "a=fmtp:%d bitrate=48000\r\n", rtp_code);
break;
}
if (fmt.cur_ms && (fmt.cur_ms < *min_packet_size))
*min_packet_size = fmt.cur_ms;
/* Our first codec packetization processed cannot be zero */
if ((*min_packet_size)==0 && fmt.cur_ms)
*min_packet_size = fmt.cur_ms;
}
/*! \brief Add video codec offer to SDP offer/answer body in INVITE or 200 OK */
/* This is different to the audio one now so we can add more caps later */
static void add_vcodec_to_sdp(const struct sip_pvt *p, int codec,
struct ast_str **m_buf, struct ast_str **a_buf,
int debug, int *min_packet_size)
{
int rtp_code;
if (!p->vrtp)
return;
if (debug)
ast_verbose("Adding video codec 0x%x (%s) to SDP\n", codec, ast_getformatname(codec));
if ((rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(p->vrtp), 1, codec)) == -1)
return;
ast_str_append(m_buf, 0, " %d", rtp_code);
ast_str_append(a_buf, 0, "a=rtpmap:%d %s/%d\r\n", rtp_code,
ast_rtp_lookup_mime_subtype2(1, codec, 0),
ast_rtp_lookup_sample_rate2(1, codec));
/* Add fmtp code here */
}
/*! \brief Add text codec offer to SDP offer/answer body in INVITE or 200 OK */
static void add_tcodec_to_sdp(const struct sip_pvt *p, int codec,
struct ast_str **m_buf, struct ast_str **a_buf,
int debug, int *min_packet_size)
{
int rtp_code;
if (!p->trtp)
return;
if (debug)
ast_verbose("Adding text codec 0x%x (%s) to SDP\n", codec, ast_getformatname(codec));
if ((rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(p->trtp), 1, codec)) == -1)
return;
ast_str_append(m_buf, 0, " %d", rtp_code);
ast_str_append(a_buf, 0, "a=rtpmap:%d %s/%d\r\n", rtp_code,
ast_rtp_lookup_mime_subtype2(1, codec, 0),
ast_rtp_lookup_sample_rate2(1, codec));
/* Add fmtp code here */
if (codec == AST_FORMAT_T140RED) {
int t140code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(p->trtp), 1, AST_FORMAT_T140);
ast_str_append(a_buf, 0, "a=fmtp:%d %d/%d/%d\r\n", rtp_code,
t140code,
t140code,
t140code);
}
}
/*! \brief Get Max T.38 Transmission rate from T38 capabilities */
static int t38_get_rate(int t38cap)
{
int maxrate = (t38cap & (T38FAX_RATE_14400 | T38FAX_RATE_12000 | T38FAX_RATE_9600 | T38FAX_RATE_7200 | T38FAX_RATE_4800 | T38FAX_RATE_2400));
if (maxrate & T38FAX_RATE_14400) {
ast_debug(2, "T38MaxBitRate 14400 found\n");
return 14400;
} else if (maxrate & T38FAX_RATE_12000) {
ast_debug(2, "T38MaxBitRate 12000 found\n");
return 12000;
} else if (maxrate & T38FAX_RATE_9600) {
ast_debug(2, "T38MaxBitRate 9600 found\n");
return 9600;
} else if (maxrate & T38FAX_RATE_7200) {
ast_debug(2, "T38MaxBitRate 7200 found\n");
return 7200;
} else if (maxrate & T38FAX_RATE_4800) {
ast_debug(2, "T38MaxBitRate 4800 found\n");
return 4800;
} else if (maxrate & T38FAX_RATE_2400) {
ast_debug(2, "T38MaxBitRate 2400 found\n");
return 2400;
} else {
ast_debug(2, "Strange, T38MaxBitRate NOT found in peers T38 SDP.\n");
return 0;
}
}
/*! \brief Add RFC 2833 DTMF offer to SDP */
static void add_noncodec_to_sdp(const struct sip_pvt *p, int format,
struct ast_str **m_buf, struct ast_str **a_buf,
int debug)
{
int rtp_code;
if (debug)
ast_verbose("Adding non-codec 0x%x (%s) to SDP\n", format, ast_rtp_lookup_mime_subtype2(0, format, 0));
if ((rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(p->rtp), 0, format)) == -1)
return;
ast_str_append(m_buf, 0, " %d", rtp_code);
ast_str_append(a_buf, 0, "a=rtpmap:%d %s/%d\r\n", rtp_code,
ast_rtp_lookup_mime_subtype2(0, format, 0),
ast_rtp_lookup_sample_rate2(0, format));
if (format == AST_RTP_DTMF) /* Indicate we support DTMF and FLASH... */
ast_str_append(a_buf, 0, "a=fmtp:%d 0-16\r\n", rtp_code);
}
/*! \brief Set all IP media addresses for this call
\note called from add_sdp()
*/
static void get_our_media_address(struct sip_pvt *p, int needvideo,
struct sockaddr_in *sin, struct sockaddr_in *vsin, struct sockaddr_in *tsin,
struct sockaddr_in *dest, struct sockaddr_in *vdest)
{
/* First, get our address */
ast_rtp_instance_get_local_address(p->rtp, sin);
if (p->vrtp)
ast_rtp_instance_get_local_address(p->vrtp, vsin);
if (p->trtp)
ast_rtp_instance_get_local_address(p->trtp, tsin);
/* Now, try to figure out where we want them to send data */
/* Is this a re-invite to move the media out, then use the original offer from caller */
if (p->redirip.sin_addr.s_addr) { /* If we have a redirection IP, use it */
dest->sin_port = p->redirip.sin_port;
dest->sin_addr = p->redirip.sin_addr;
} else {
dest->sin_addr = p->ourip.sin_addr;
dest->sin_port = sin->sin_port;
}
if (needvideo) {
/* Determine video destination */
if (p->vredirip.sin_addr.s_addr) {
vdest->sin_addr = p->vredirip.sin_addr;
vdest->sin_port = p->vredirip.sin_port;
} else {
vdest->sin_addr = p->ourip.sin_addr;
vdest->sin_port = vsin->sin_port;
}
}
}
/*! \brief Add Session Description Protocol message
If oldsdp is TRUE, then the SDP version number is not incremented. This mechanism
is used in Session-Timers where RE-INVITEs are used for refreshing SIP sessions
without modifying the media session in any way.
*/
static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int oldsdp, int add_audio, int add_t38)
{
int len = 0;
int alreadysent = 0;
struct sockaddr_in sin = { 0, };
struct sockaddr_in vsin = { 0, };
struct sockaddr_in tsin = { 0, };
struct sockaddr_in dest = { 0, };
struct sockaddr_in udptlsin = { 0, };
struct sockaddr_in vdest = { 0, };
struct sockaddr_in tdest = { 0, };
struct sockaddr_in udptldest = { 0, };
/* SDP fields */
char *version = "v=0\r\n"; /* Protocol version */
char subject[256]; /* Subject of the session */
char owner[256]; /* Session owner/creator */
char connection[256]; /* Connection data */
char *session_time = "t=0 0\r\n"; /* Time the session is active */
char bandwidth[256] = ""; /* Max bitrate */
char *hold = "";
struct ast_str *m_audio = ast_str_alloca(256); /* Media declaration line for audio */
struct ast_str *m_video = ast_str_alloca(256); /* Media declaration line for video */
struct ast_str *m_text = ast_str_alloca(256); /* Media declaration line for text */
struct ast_str *m_modem = ast_str_alloca(256); /* Media declaration line for modem */
struct ast_str *a_audio = ast_str_alloca(1024); /* Attributes for audio */
struct ast_str *a_video = ast_str_alloca(1024); /* Attributes for video */
struct ast_str *a_text = ast_str_alloca(1024); /* Attributes for text */
struct ast_str *a_modem = ast_str_alloca(1024); /* Attributes for modem */
int x;
int capability = 0;
int needaudio = FALSE;
int needvideo = FALSE;
int needtext = FALSE;
int debug = sip_debug_test_pvt(p);
int min_audio_packet_size = 0;
int min_video_packet_size = 0;
int min_text_packet_size = 0;
char codecbuf[SIPBUFSIZE];
char buf[SIPBUFSIZE];
/* Set the SDP session name */
snprintf(subject, sizeof(subject), "s=%s\r\n", ast_strlen_zero(global_sdpsession) ? "-" : global_sdpsession);
if (!p->rtp) {
ast_log(LOG_WARNING, "No way to add SDP without an RTP structure\n");
return AST_FAILURE;
}
/* XXX We should not change properties in the SIP dialog until
we have acceptance of the offer if this is a re-invite */
/* Set RTP Session ID and version */
if (!p->sessionid) {
p->sessionid = (int)ast_random();
p->sessionversion = p->sessionid;
} else {
if (oldsdp == FALSE)
p->sessionversion++;
}
get_our_media_address(p, needvideo, &sin, &vsin, &tsin, &dest, &vdest);
snprintf(owner, sizeof(owner), "o=%s %d %d IN IP4 %s\r\n", ast_strlen_zero(global_sdpowner) ? "-" : global_sdpowner, p->sessionid, p->sessionversion, ast_inet_ntoa(dest.sin_addr));
snprintf(connection, sizeof(connection), "c=IN IP4 %s\r\n", ast_inet_ntoa(dest.sin_addr));
if (add_audio) {
capability = p->jointcapability;
/* XXX note, Video and Text are negated - 'true' means 'no' */
ast_debug(1, "** Our capability: %s Video flag: %s Text flag: %s\n", ast_getformatname_multiple(codecbuf, sizeof(codecbuf), capability),
p->novideo ? "True" : "False", p->notext ? "True" : "False");
ast_debug(1, "** Our prefcodec: %s \n", ast_getformatname_multiple(codecbuf, sizeof(codecbuf), p->prefcodec));
#ifdef WHEN_WE_HAVE_T38_FOR_OTHER_TRANSPORTS
if (ast_test_flag(&p->t38.t38support, SIP_PAGE2_T38SUPPORT_RTP)) {
ast_str_append(&m_audio, 0, " %d", 191);
ast_str_append(&a_audio, 0, "a=rtpmap:%d %s/%d\r\n", 191, "t38", 8000);
}
#endif
/* Check if we need audio */
if (capability & AST_FORMAT_AUDIO_MASK)
needaudio = TRUE;
/* Check if we need video in this call */
if ((capability & AST_FORMAT_VIDEO_MASK) && !p->novideo) {
if (p->vrtp) {
needvideo = TRUE;
ast_debug(2, "This call needs video offers!\n");
} else
ast_debug(2, "This call needs video offers, but there's no video support enabled!\n");
}
if (debug)
ast_verbose("Audio is at %s port %d\n", ast_inet_ntoa(p->ourip.sin_addr), ntohs(sin.sin_port));
/* Ok, we need video. Let's add what we need for video and set codecs.
Video is handled differently than audio since we can not transcode. */
if (needvideo) {
ast_str_append(&m_video, 0, "m=video %d RTP/AVP", ntohs(vdest.sin_port));
/* Build max bitrate string */
if (p->maxcallbitrate)
snprintf(bandwidth, sizeof(bandwidth), "b=CT:%d\r\n", p->maxcallbitrate);
if (debug)
ast_verbose("Video is at %s port %d\n", ast_inet_ntoa(p->ourip.sin_addr), ntohs(vsin.sin_port));
}
/* Check if we need text in this call */
if((capability & AST_FORMAT_TEXT_MASK) && !p->notext) {
if (sipdebug_text)
ast_verbose("We think we can do text\n");
if (p->trtp) {
if (sipdebug_text)
ast_verbose("And we have a text rtp object\n");
needtext = TRUE;
ast_debug(2, "This call needs text offers! \n");
} else
ast_debug(2, "This call needs text offers, but there's no text support enabled ! \n");
}
/* Ok, we need text. Let's add what we need for text and set codecs.
Text is handled differently than audio since we can not transcode. */
if (needtext) {
if (sipdebug_text)
ast_verbose("Lets set up the text sdp\n");
/* Determine text destination */
if (p->tredirip.sin_addr.s_addr) {
tdest.sin_addr = p->tredirip.sin_addr;
tdest.sin_port = p->tredirip.sin_port;
} else {
tdest.sin_addr = p->ourip.sin_addr;
tdest.sin_port = tsin.sin_port;
}
ast_str_append(&m_text, 0, "m=text %d RTP/AVP", ntohs(tdest.sin_port));
if (debug) /* XXX should I use tdest below ? */
ast_verbose("Text is at %s port %d\n", ast_inet_ntoa(p->ourip.sin_addr), ntohs(tsin.sin_port));
}
/* Start building generic SDP headers */
/* We break with the "recommendation" and send our IP, in order that our
peer doesn't have to ast_gethostbyname() us */
ast_str_append(&m_audio, 0, "m=audio %d RTP/AVP", ntohs(dest.sin_port));
if (ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD) == SIP_PAGE2_CALL_ONHOLD_ONEDIR)
hold = "a=recvonly\r\n";
else if (ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD) == SIP_PAGE2_CALL_ONHOLD_INACTIVE)
hold = "a=inactive\r\n";
else
hold = "a=sendrecv\r\n";
/* Now, start adding audio codecs. These are added in this order:
- First what was requested by the calling channel
- Then preferences in order from sip.conf device config for this peer/user
- Then other codecs in capabilities, including video
*/
/* Prefer the audio codec we were requested to use, first, no matter what
Note that p->prefcodec can include video codecs, so mask them out
*/
if (capability & p->prefcodec) {
int codec = p->prefcodec & AST_FORMAT_AUDIO_MASK;
add_codec_to_sdp(p, codec, &m_audio, &a_audio, debug, &min_audio_packet_size);
alreadysent |= codec;
}
/* Start by sending our preferred audio/video codecs */
for (x = 0; x < 32; x++) {
int codec;
if (!(codec = ast_codec_pref_index(&p->prefs, x)))
break;
if (!(capability & codec))
continue;
if (alreadysent & codec)
continue;
add_codec_to_sdp(p, codec, &m_audio, &a_audio, debug, &min_audio_packet_size);
alreadysent |= codec;
}
/* Now send any other common audio and video codecs, and non-codec formats: */
for (x = 1; x <= (needtext ? AST_FORMAT_TEXT_MASK : (needvideo ? AST_FORMAT_VIDEO_MASK : AST_FORMAT_AUDIO_MASK)); x <<= 1) {
if (!(capability & x)) /* Codec not requested */
continue;
if (alreadysent & x) /* Already added to SDP */
continue;
if (x & AST_FORMAT_AUDIO_MASK)
add_codec_to_sdp(p, x, &m_audio, &a_audio, debug, &min_audio_packet_size);
else if (x & AST_FORMAT_VIDEO_MASK)
add_vcodec_to_sdp(p, x, &m_video, &a_video, debug, &min_video_packet_size);
else if (x & AST_FORMAT_TEXT_MASK)
add_tcodec_to_sdp(p, x, &m_text, &a_text, debug, &min_text_packet_size);
}
/* Now add DTMF RFC2833 telephony-event as a codec */
for (x = 1; x <= AST_RTP_MAX; x <<= 1) {
if (!(p->jointnoncodeccapability & x))
continue;
add_noncodec_to_sdp(p, x, &m_audio, &a_audio, debug);
}
ast_debug(3, "-- Done with adding codecs to SDP\n");
if (!p->owner || !ast_internal_timing_enabled(p->owner))
ast_str_append(&a_audio, 0, "a=silenceSupp:off - - - -\r\n");
if (min_audio_packet_size)
ast_str_append(&a_audio, 0, "a=ptime:%d\r\n", min_audio_packet_size);
/* XXX don't think you can have ptime for video */
if (min_video_packet_size)
ast_str_append(&a_video, 0, "a=ptime:%d\r\n", min_video_packet_size);
/* XXX don't think you can have ptime for text */
if (min_text_packet_size)
ast_str_append(&a_text, 0, "a=ptime:%d\r\n", min_text_packet_size);
if (m_audio->len - m_audio->used < 2 || m_video->len - m_video->used < 2 ||
m_text->len - m_text->used < 2 || a_text->len - a_text->used < 2 ||
a_audio->len - a_audio->used < 2 || a_video->len - a_video->used < 2)
ast_log(LOG_WARNING, "SIP SDP may be truncated due to undersized buffer!!\n");
}
if (add_t38) {
/* Our T.38 end is */
ast_udptl_get_us(p->udptl, &udptlsin);
/* Determine T.38 UDPTL destination */
if (p->udptlredirip.sin_addr.s_addr) {
udptldest.sin_port = p->udptlredirip.sin_port;
udptldest.sin_addr = p->udptlredirip.sin_addr;
} else {
udptldest.sin_addr = p->ourip.sin_addr;
udptldest.sin_port = udptlsin.sin_port;
}
if (debug)
ast_debug(1, "T.38 UDPTL is at %s port %d\n", ast_inet_ntoa(p->ourip.sin_addr), ntohs(udptlsin.sin_port));
/* We break with the "recommendation" and send our IP, in order that our
peer doesn't have to ast_gethostbyname() us */
if (debug) {
ast_debug(1, "Our T38 capability (%d), peer T38 capability (%d), joint capability (%d)\n",
p->t38.capability,
p->t38.peercapability,
p->t38.jointcapability);
}
ast_str_append(&m_modem, 0, "m=image %d udptl t38", ntohs(udptldest.sin_port));
if ((p->t38.jointcapability & T38FAX_VERSION) == T38FAX_VERSION_0)
ast_str_append(&a_modem, 0, "a=T38FaxVersion:0\r\n");
if ((p->t38.jointcapability & T38FAX_VERSION) == T38FAX_VERSION_1)
ast_str_append(&a_modem, 0, "a=T38FaxVersion:1\r\n");
if ((x = t38_get_rate(p->t38.jointcapability)))
ast_str_append(&a_modem, 0, "a=T38MaxBitRate:%d\r\n", x);
if ((p->t38.jointcapability & T38FAX_FILL_BIT_REMOVAL) == T38FAX_FILL_BIT_REMOVAL)
ast_str_append(&a_modem, 0, "a=T38FaxFillBitRemoval\r\n");
if ((p->t38.jointcapability & T38FAX_TRANSCODING_MMR) == T38FAX_TRANSCODING_MMR)
ast_str_append(&a_modem, 0, "a=T38FaxTranscodingMMR\r\n");
if ((p->t38.jointcapability & T38FAX_TRANSCODING_JBIG) == T38FAX_TRANSCODING_JBIG)
ast_str_append(&a_modem, 0, "a=T38FaxTranscodingJBIG\r\n");
ast_str_append(&a_modem, 0, "a=T38FaxRateManagement:%s\r\n", (p->t38.jointcapability & T38FAX_RATE_MANAGEMENT_LOCAL_TCF) ? "localTCF" : "transferredTCF");
x = ast_udptl_get_local_max_datagram(p->udptl);
ast_str_append(&a_modem, 0, "a=T38FaxMaxBuffer:%d\r\n", x);
ast_str_append(&a_modem, 0, "a=T38FaxMaxDatagram:%d\r\n", x);
if (p->t38.jointcapability != T38FAX_UDP_EC_NONE)
ast_str_append(&a_modem, 0, "a=T38FaxUdpEC:%s\r\n", (p->t38.jointcapability & T38FAX_UDP_EC_REDUNDANCY) ? "t38UDPRedundancy" : "t38UDPFEC");
}
if (needaudio)
ast_str_append(&m_audio, 0, "\r\n");
if (needvideo)
ast_str_append(&m_video, 0, "\r\n");
if (needtext)
ast_str_append(&m_text, 0, "\r\n");
if (add_t38)
ast_str_append(&m_modem, 0, "\r\n");
len = strlen(version) + strlen(subject) + strlen(owner) +
strlen(connection) + strlen(session_time);
if (needaudio)
len += m_audio->used + a_audio->used + strlen(hold);
if (needvideo) /* only if video response is appropriate */
len += m_video->used + a_video->used + strlen(bandwidth) + strlen(hold);
if (needtext) /* only if text response is appropriate */
len += m_text->used + a_text->used + strlen(hold);
if (add_t38)
len += m_modem->used + a_modem->used;
add_header(resp, "Content-Type", "application/sdp");
add_header_contentLength(resp, len);
add_line(resp, version);
add_line(resp, owner);
add_line(resp, subject);
add_line(resp, connection);
if (needvideo) /* only if video response is appropriate */
add_line(resp, bandwidth);
add_line(resp, session_time);
if (needaudio) {
add_line(resp, m_audio->str);
add_line(resp, a_audio->str);
add_line(resp, hold);
}
if (needvideo) { /* only if video response is appropriate */
add_line(resp, m_video->str);
add_line(resp, a_video->str);
add_line(resp, hold); /* Repeat hold for the video stream */
}
if (needtext) { /* only if text response is appropriate */
add_line(resp, m_text->str);
add_line(resp, a_text->str);
add_line(resp, hold); /* Repeat hold for the text stream */
}
if (add_t38) {
add_line(resp, m_modem->str);
add_line(resp, a_modem->str);
}
/* Update lastrtprx when we send our SDP */
p->lastrtprx = p->lastrtptx = time(NULL); /* XXX why both ? */
ast_debug(3, "Done building SDP. Settling with this capability: %s\n", ast_getformatname_multiple(buf, SIPBUFSIZE, capability));
return AST_SUCCESS;
}
/*! \brief Used for 200 OK and 183 early media */
static int transmit_response_with_t38_sdp(struct sip_pvt *p, char *msg, struct sip_request *req, int retrans)
{
struct sip_request resp;
int seqno;
if (sscanf(get_header(req, "CSeq"), "%d ", &seqno) != 1) {
ast_log(LOG_WARNING, "Unable to get seqno from '%s'\n", get_header(req, "CSeq"));
return -1;
}
respprep(&resp, p, msg, req);
if (p->udptl) {
ast_udptl_offered_from_local(p->udptl, 0);
add_sdp(&resp, p, 0, 0, 1);
} else
ast_log(LOG_ERROR, "Can't add SDP to response, since we have no UDPTL session allocated. Call-ID %s\n", p->callid);
if (retrans && !p->pendinginvite)
p->pendinginvite = seqno; /* Buggy clients sends ACK on RINGING too */
return send_response(p, &resp, retrans, seqno);
}
/*! \brief copy SIP request (mostly used to save request for responses) */
static void copy_request(struct sip_request *dst, const struct sip_request *src)
{
struct ast_str *duplicate = dst->data;
/* First copy stuff */
memcpy(dst, src, sizeof(*dst));
dst->data = duplicate;
/* All these + 1's are to account for the need to include the NULL terminator
* Using typical string functions like ast_copy_string or ast_str_set will not
* work in this case because the src's data string is riddled with \0's all over
* the place and so a memcpy is the only way to accurately copy the string
*/
if (!dst->data && !(dst->data = ast_str_create(src->data->used + 1)))
return;
else if (dst->data->len < src->data->used + 1)
ast_str_make_space(&dst->data, src->data->used + 1);
memcpy(dst->data->str, src->data->str, src->data->used + 1);
dst->data->used = src->data->used;
}
/*! \brief Used for 200 OK and 183 early media
\return Will return XMIT_ERROR for network errors.
*/
static int transmit_response_with_sdp(struct sip_pvt *p, const char *msg, const struct sip_request *req, enum xmittype reliable, int oldsdp, int rpid)
{
struct sip_request resp;
int seqno;
if (sscanf(get_header(req, "CSeq"), "%d ", &seqno) != 1) {
ast_log(LOG_WARNING, "Unable to get seqno from '%s'\n", get_header(req, "CSeq"));
return -1;
}
respprep(&resp, p, msg, req);
if (rpid == TRUE) {
add_rpid(&resp, p);
}
if (p->rtp) {
if (!p->autoframing && !ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
ast_debug(1, "Setting framing from config on incoming call\n");
ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(p->rtp), p->rtp, &p->prefs);
}
try_suggested_sip_codec(p);
if (p->t38.state == T38_PEER_DIRECT || p->t38.state == T38_ENABLED) {
add_sdp(&resp, p, oldsdp, TRUE, TRUE);
} else {
add_sdp(&resp, p, oldsdp, TRUE, FALSE);
}
} else
ast_log(LOG_ERROR, "Can't add SDP to response, since we have no RTP session allocated. Call-ID %s\n", p->callid);
if (reliable && !p->pendinginvite)
p->pendinginvite = seqno; /* Buggy clients sends ACK on RINGING too */
return send_response(p, &resp, reliable, seqno);
}
/*! \brief Parse first line of incoming SIP request */
static int determine_firstline_parts(struct sip_request *req)
{
char *e = ast_skip_blanks(req->data->str); /* there shouldn't be any */
char *local_rlPart1;
if (!*e)
return -1;
req->rlPart1 = e - req->data->str; /* method or protocol */
local_rlPart1 = e;
e = ast_skip_nonblanks(e);
if (*e)
*e++ = '\0';
/* Get URI or status code */
e = ast_skip_blanks(e);
if ( !*e )
return -1;
ast_trim_blanks(e);
if (!strcasecmp(local_rlPart1, "SIP/2.0") ) { /* We have a response */
if (strlen(e) < 3) /* status code is 3 digits */
return -1;
req->rlPart2 = e - req->data->str;
} else { /* We have a request */
if ( *e == '<' ) { /* XXX the spec says it must not be in <> ! */
ast_debug(3, "Oops. Bogus uri in <> %s\n", e);
e++;
if (!*e)
return -1;
}
req->rlPart2 = e - req->data->str; /* URI */
e = ast_skip_nonblanks(e);
if (*e)
*e++ = '\0';
e = ast_skip_blanks(e);
if (strcasecmp(e, "SIP/2.0") ) {
ast_debug(3, "Skipping packet - Bad request protocol %s\n", e);
return -1;
}
}
return 1;
}
/*! \brief Transmit reinvite with SDP
\note A re-invite is basically a new INVITE with the same CALL-ID and TAG as the
INVITE that opened the SIP dialogue
We reinvite so that the audio stream (RTP) go directly between
the SIP UAs. SIP Signalling stays with * in the path.
If t38version is TRUE, we send T38 SDP for re-invite from audio/video to
T38 UDPTL transmission on the channel
If oldsdp is TRUE then the SDP version number is not incremented. This
is needed for Session-Timers so we can send a re-invite to refresh the
SIP session without modifying the media session.
*/
static int transmit_reinvite_with_sdp(struct sip_pvt *p, int t38version, int oldsdp)
{
struct sip_request req;
reqprep(&req, p, ast_test_flag(&p->flags[0], SIP_REINVITE_UPDATE) ? SIP_UPDATE : SIP_INVITE, 0, 1);
add_header(&req, "Allow", ALLOWED_METHODS);
add_header(&req, "Supported", SUPPORTED_EXTENSIONS);
if (sipdebug) {
if (oldsdp == TRUE)
add_header(&req, "X-asterisk-Info", "SIP re-invite (Session-Timers)");
else
add_header(&req, "X-asterisk-Info", "SIP re-invite (External RTP bridge)");
}
if (p->do_history)
append_history(p, "ReInv", "Re-invite sent");
try_suggested_sip_codec(p);
if (t38version)
add_sdp(&req, p, oldsdp, FALSE, TRUE);
else
add_sdp(&req, p, oldsdp, TRUE, FALSE);
/* Use this as the basis */
initialize_initreq(p, &req);
p->lastinvite = p->ocseq;
ast_set_flag(&p->flags[0], SIP_OUTGOING); /* Change direction of this dialog */
return send_request(p, &req, XMIT_CRITICAL, p->ocseq);
}
/* \brief Remove URI parameters at end of URI, not in username part though */
static char *remove_uri_parameters(char *uri)
{
char *atsign;
atsign = strchr(uri, '@'); /* First, locate the at sign */
if (!atsign)
atsign = uri; /* Ok hostname only, let's stick with the rest */
atsign = strchr(atsign, ';'); /* Locate semi colon */
if (atsign)
*atsign = '\0'; /* Kill at the semi colon */
return uri;
}
/*! \brief Check Contact: URI of SIP message */
static void extract_uri(struct sip_pvt *p, struct sip_request *req)
{
char stripped[SIPBUFSIZE];
char *c;
ast_copy_string(stripped, get_header(req, "Contact"), sizeof(stripped));
c = get_in_brackets(stripped);
/* Cut the URI at the at sign after the @, not in the username part */
c = remove_uri_parameters(c);
if (!ast_strlen_zero(c))
ast_string_field_set(p, uri, c);
}
/*! \brief Build contact header - the contact header we send out */
static void build_contact(struct sip_pvt *p)
{
int ourport = ntohs(p->ourip.sin_port);
if (p->socket.type & SIP_TRANSPORT_UDP) {
if (!sip_standard_port(p->socket.type, ourport))
ast_string_field_build(p, our_contact, "<sip:%s%s%s:%d>", p->exten, ast_strlen_zero(p->exten) ? "" : "@", ast_inet_ntoa(p->ourip.sin_addr), ourport);
else
ast_string_field_build(p, our_contact, "<sip:%s%s%s>", p->exten, ast_strlen_zero(p->exten) ? "" : "@", ast_inet_ntoa(p->ourip.sin_addr));
} else {
/*! \todo We should not always add port here. Port is only added if it's non-standard (see code above) */
ast_string_field_build(p, our_contact, "<sip:%s%s%s:%d;transport=%s>", p->exten, ast_strlen_zero(p->exten) ? "" : "@", ast_inet_ntoa(p->ourip.sin_addr), ourport, get_transport(p->socket.type));
}
}
/*! \brief Initiate new SIP request to peer/user */
static void initreqprep(struct sip_request *req, struct sip_pvt *p, int sipmethod)
{
struct ast_str *invite = ast_str_alloca(256);
char from[256];
char to[256];
char tmp_n[SIPBUFSIZE/2]; /* build a local copy of 'n' if needed */
char tmp_l[SIPBUFSIZE/2]; /* build a local copy of 'l' if needed */
const char *l = NULL; /* XXX what is this, exactly ? */
const char *n = NULL; /* XXX what is this, exactly ? */
const char *urioptions = "";
int ourport;
if (ast_test_flag(&p->flags[0], SIP_USEREQPHONE)) {
const char *s = p->username; /* being a string field, cannot be NULL */
/* Test p->username against allowed characters in AST_DIGIT_ANY
If it matches the allowed characters list, then sipuser = ";user=phone"
If not, then sipuser = ""
*/
/* + is allowed in first position in a tel: uri */
if (*s == '+')
s++;
for (; *s; s++) {
if (!strchr(AST_DIGIT_ANYNUM, *s) )
break;
}
/* If we have only digits, add ;user=phone to the uri */
if (!*s)
urioptions = ";user=phone";
}
snprintf(p->lastmsg, sizeof(p->lastmsg), "Init: %s", sip_methods[sipmethod].text);
if (p->owner && (p->owner->connected.id.number_presentation & AST_PRES_RESTRICTION) == AST_PRES_ALLOWED) {
l = p->owner->connected.id.number;
n = p->owner->connected.id.name;
}
/* Hey, it's a NOTIFY! See if they've configured a mwi_from.
* XXX Right now, this logic works because the only place that mwi_from
* is set on the sip_pvt is in sip_send_mwi_to_peer. If things changed, then
* we might end up putting the mwi_from setting into other types of NOTIFY
* messages as well.
*/
if (sipmethod == SIP_NOTIFY && !ast_strlen_zero(p->mwi_from)) {
l = p->mwi_from;
}
if (ast_strlen_zero(l))
l = default_callerid;
if (ast_strlen_zero(n))
n = l;
/* Allow user to be overridden */
if (!ast_strlen_zero(p->fromuser))
l = p->fromuser;
else /* Save for any further attempts */
ast_string_field_set(p, fromuser, l);
/* Allow user to be overridden */
if (!ast_strlen_zero(p->fromname))
n = p->fromname;
else /* Save for any further attempts */
ast_string_field_set(p, fromname, n);
if (sip_cfg.pedanticsipchecking) {
ast_uri_encode(n, tmp_n, sizeof(tmp_n), 0);
n = tmp_n;
ast_uri_encode(l, tmp_l, sizeof(tmp_l), 0);
l = tmp_l;
}
ourport = ntohs(p->ourip.sin_port);
if (!sip_standard_port(p->socket.type, ourport) && ast_strlen_zero(p->fromdomain))
snprintf(from, sizeof(from), "\"%s\" <sip:%s@%s:%d>;tag=%s", n, l, ast_inet_ntoa(p->ourip.sin_addr), ourport, p->tag);
else
snprintf(from, sizeof(from), "\"%s\" <sip:%s@%s>;tag=%s", n, l, S_OR(p->fromdomain, ast_inet_ntoa(p->ourip.sin_addr)), p->tag);
/* If we're calling a registered SIP peer, use the fullcontact to dial to the peer */
if (!ast_strlen_zero(p->fullcontact)) {
/* If we have full contact, trust it */
ast_str_append(&invite, 0, "%s", p->fullcontact);
} else {
/* Otherwise, use the username while waiting for registration */
ast_str_append(&invite, 0, "sip:");
if (!ast_strlen_zero(p->username)) {
n = p->username;
if (sip_cfg.pedanticsipchecking) {
ast_uri_encode(n, tmp_n, sizeof(tmp_n), 0);
n = tmp_n;
}
ast_str_append(&invite, 0, "%s@", n);
}
ast_str_append(&invite, 0, "%s", p->tohost);
if (ntohs(p->sa.sin_port) != STANDARD_SIP_PORT)
ast_str_append(&invite, 0, ":%d", ntohs(p->sa.sin_port));
ast_str_append(&invite, 0, "%s", urioptions);
}
/* If custom URI options have been provided, append them */
if (p->options && !ast_strlen_zero(p->options->uri_options))
ast_str_append(&invite, 0, ";%s", p->options->uri_options);
/* This is the request URI, which is the next hop of the call
which may or may not be the destination of the call
*/
ast_string_field_set(p, uri, invite->str);
if (!ast_strlen_zero(p->todnid)) {
/*! \todo Need to add back the VXML URL here at some point, possibly use build_string for all this junk */
if (!strchr(p->todnid, '@')) {
/* We have no domain in the dnid */
snprintf(to, sizeof(to), "<sip:%s@%s>%s%s", p->todnid, p->tohost, ast_strlen_zero(p->theirtag) ? "" : ";tag=", p->theirtag);
} else {
snprintf(to, sizeof(to), "<sip:%s>%s%s", p->todnid, ast_strlen_zero(p->theirtag) ? "" : ";tag=", p->theirtag);
}
} else {
if (sipmethod == SIP_NOTIFY && !ast_strlen_zero(p->theirtag)) {
/* If this is a NOTIFY, use the From: tag in the subscribe (RFC 3265) */
snprintf(to, sizeof(to), "<%s%s>;tag=%s", (!strncasecmp(p->uri, "sip:", 4) ? "sip:" : ""), p->uri, p->theirtag);
} else if (p->options && p->options->vxml_url) {
/* If there is a VXML URL append it to the SIP URL */
snprintf(to, sizeof(to), "<%s>;%s", p->uri, p->options->vxml_url);
} else
snprintf(to, sizeof(to), "<%s>", p->uri);
}
init_req(req, sipmethod, p->uri);
/* now tmp_n is available so reuse it to build the CSeq */
snprintf(tmp_n, sizeof(tmp_n), "%d %s", ++p->ocseq, sip_methods[sipmethod].text);
add_header(req, "Via", p->via);
add_header(req, "Max-Forwards", DEFAULT_MAX_FORWARDS);
/* SLD: FIXME?: do Route: here too? I think not cos this is the first request.
* OTOH, then we won't have anything in p->route anyway */
add_header(req, "From", from);
add_header(req, "To", to);
ast_string_field_set(p, exten, l);
build_contact(p);
add_header(req, "Contact", p->our_contact);
add_header(req, "Call-ID", p->callid);
add_header(req, "CSeq", tmp_n);
if (!ast_strlen_zero(global_useragent))
add_header(req, "User-Agent", global_useragent);
}
/*! \brief Add "Diversion" header to outgoing message
*
* We need to add a Diversion header if the owner channel of
* this dialog has redirecting information associated with it.
*
* \param req The request/response to which we will add the header
* \param pvt The sip_pvt which represents the call-leg
* \param apr Redirecting data used to make the diversion header
*/
static void add_diversion_header(struct sip_request *req, struct sip_pvt *pvt)
{
const char *diverting_number;
const char *diverting_name;
const char *reason;
char header_text[256];
if (!pvt->owner) {
return;
}
diverting_number = pvt->owner->cid.cid_rdnis;
diverting_name = pvt->owner->redirecting.from.name;
reason = sip_reason_code_to_str(pvt->owner->redirecting.reason);
if (ast_strlen_zero(diverting_number)) {
return;
}
/* We at least have a number to place in the Diversion header, which is enough */
if (ast_strlen_zero(diverting_name)) {
snprintf(header_text, sizeof(header_text), "<sip:%s@%s>;reason=%s", diverting_number, ast_inet_ntoa(pvt->ourip.sin_addr), reason);
} else {
snprintf(header_text, sizeof(header_text), "\"%s\" <sip:%s@%s>;reason=%s", diverting_name, diverting_number, ast_inet_ntoa(pvt->ourip.sin_addr), reason);
}
add_header(req, "Diversion", header_text);
}
/*! \brief Build REFER/INVITE/OPTIONS/SUBSCRIBE message and transmit it
\param init 0 = Prepare request within dialog, 1= prepare request, new branch, 2= prepare new request and new dialog. do_proxy_auth calls this with init!=2
\param p sip_pvt structure
\param sdp unknown
\param sipmethod unknown
*/
static int transmit_invite(struct sip_pvt *p, int sipmethod, int sdp, int init)
{
struct sip_request req;
struct ast_variable *var;
req.method = sipmethod;
if (init) {/* Bump branch even on initial requests */
p->branch ^= ast_random();
p->invite_branch = p->branch;
build_via(p);
}
if (init > 1)
initreqprep(&req, p, sipmethod);
else
/* If init=1, we should not generate a new branch. If it's 0, we need a new branch. */
reqprep(&req, p, sipmethod, 0, init ? 0 : 1);
if (p->options && p->options->auth)
add_header(&req, p->options->authheader, p->options->auth);
append_date(&req);
if (sipmethod == SIP_REFER) { /* Call transfer */
if (p->refer) {
char buf[SIPBUFSIZE];
if (!ast_strlen_zero(p->refer->refer_to))
add_header(&req, "Refer-To", p->refer->refer_to);
if (!ast_strlen_zero(p->refer->referred_by)) {
snprintf(buf, sizeof(buf), "%s <%s>", p->refer->referred_by_name, p->refer->referred_by);
add_header(&req, "Referred-By", buf);
}
}
} else if (sipmethod == SIP_SUBSCRIBE) { /* We only support sending MWI subscriptions right now */
char buf[SIPBUFSIZE];
add_header(&req, "Event", "message-summary");
add_header(&req, "Accept", "application/simple-message-summary");
snprintf(buf, sizeof(buf), "%d", mwi_expiry);
add_header(&req, "Expires", buf);
}
/* This new INVITE is part of an attended transfer. Make sure that the
other end knows and replace the current call with this new call */
if (p->options && !ast_strlen_zero(p->options->replaces)) {
add_header(&req, "Replaces", p->options->replaces);
add_header(&req, "Require", "replaces");
}
/* Add Session-Timers related headers */
if (st_get_mode(p) == SESSION_TIMER_MODE_ORIGINATE) {
char i2astr[10];
if (!p->stimer->st_interval)
p->stimer->st_interval = st_get_se(p, TRUE);
p->stimer->st_active = TRUE;
snprintf(i2astr, sizeof(i2astr), "%d", p->stimer->st_interval);
add_header(&req, "Session-Expires", i2astr);
snprintf(i2astr, sizeof(i2astr), "%d", st_get_se(p, FALSE));
add_header(&req, "Min-SE", i2astr);
}
add_header(&req, "Allow", ALLOWED_METHODS);
add_header(&req, "Supported", SUPPORTED_EXTENSIONS);
if(p->notify_headers) {
char buf[512];
for (var = p->notify_headers; var; var = var->next) {
ast_copy_string(buf, var->value, sizeof(buf));
add_header(&req, var->name, ast_unescape_semicolon(buf));
}
}
if (p->options && p->options->addsipheaders && p->owner) {
struct ast_channel *chan = p->owner; /* The owner channel */
struct varshead *headp;
ast_channel_lock(chan);
headp = &chan->varshead;
if (!headp)
ast_log(LOG_WARNING, "No Headp for the channel...ooops!\n");
else {
const struct ast_var_t *current;
AST_LIST_TRAVERSE(headp, current, entries) {
/* SIPADDHEADER: Add SIP header to outgoing call */
if (!strncasecmp(ast_var_name(current), "SIPADDHEADER", strlen("SIPADDHEADER"))) {
char *content, *end;
const char *header = ast_var_value(current);
char *headdup = ast_strdupa(header);
/* Strip of the starting " (if it's there) */
if (*headdup == '"')
headdup++;
if ((content = strchr(headdup, ':'))) {
*content++ = '\0';
content = ast_skip_blanks(content); /* Skip white space */
/* Strip the ending " (if it's there) */
end = content + strlen(content) -1;
if (*end == '"')
*end = '\0';
add_header(&req, headdup, content);
if (sipdebug)
ast_debug(1, "Adding SIP Header \"%s\" with content :%s: \n", headdup, content);
}
}
}
}
ast_channel_unlock(chan);
}
if ((sipmethod == SIP_INVITE || sipmethod == SIP_UPDATE) && ast_test_flag(&p->flags[0], SIP_SENDRPID))
add_rpid(&req, p);
if (sipmethod == SIP_INVITE) {
add_diversion_header(&req, p);
}
if (sdp) {
if (p->udptl && p->t38.state == T38_LOCAL_REINVITE) {
ast_udptl_offered_from_local(p->udptl, 1);
ast_debug(1, "T38 is in state %d on channel %s\n", p->t38.state, p->owner ? p->owner->name : "<none>");
add_sdp(&req, p, FALSE, FALSE, TRUE);
} else if (p->rtp) {
try_suggested_sip_codec(p);
add_sdp(&req, p, FALSE, TRUE, FALSE);
}
} else {
if (!p->notify_headers) {
add_header_contentLength(&req, 0);
}
}
if (!p->initreq.headers || init > 2)
initialize_initreq(p, &req);
p->lastinvite = p->ocseq;
return send_request(p, &req, init ? XMIT_CRITICAL : XMIT_RELIABLE, p->ocseq);
}
/*! \brief Send a subscription or resubscription for MWI */
static int sip_subscribe_mwi_do(const void *data)
{
struct sip_subscription_mwi *mwi = (struct sip_subscription_mwi*)data;
if (!mwi) {
return -1;
}
mwi->resub = -1;
__sip_subscribe_mwi_do(mwi);
ASTOBJ_UNREF(mwi, sip_subscribe_mwi_destroy);
return 0;
}
/*! \brief Actually setup an MWI subscription or resubscribe */
static int __sip_subscribe_mwi_do(struct sip_subscription_mwi *mwi)
{
/* If we have no DNS manager let's do a lookup */
if (!mwi->dnsmgr) {
char transport[MAXHOSTNAMELEN];
snprintf(transport, sizeof(transport), "_sip._%s", get_transport(mwi->transport));
ast_dnsmgr_lookup(mwi->hostname, &mwi->us, &mwi->dnsmgr, sip_cfg.srvlookup ? transport : NULL);
}
/* If we already have a subscription up simply send a resubscription */
if (mwi->call) {
transmit_invite(mwi->call, SIP_SUBSCRIBE, 0, 0);
return 0;
}
/* Create a dialog that we will use for the subscription */
if (!(mwi->call = sip_alloc(NULL, NULL, 0, SIP_SUBSCRIBE))) {
return -1;
}
ref_proxy(mwi->call, obproxy_get(mwi->call, NULL));
if (!mwi->us.sin_port && mwi->portno) {
mwi->us.sin_port = htons(mwi->portno);
}
/* Setup the destination of our subscription */
if (create_addr(mwi->call, mwi->hostname, &mwi->us, 0)) {
dialog_unlink_all(mwi->call, TRUE, TRUE);
mwi->call = dialog_unref(mwi->call, "unref dialog after unlink_all");
return 0;
}
if (!mwi->dnsmgr && mwi->portno) {
mwi->call->sa.sin_port = htons(mwi->portno);
mwi->call->recv.sin_port = htons(mwi->portno);
} else {
mwi->portno = ntohs(mwi->call->sa.sin_port);
}
/* Set various other information */
if (!ast_strlen_zero(mwi->authuser)) {
ast_string_field_set(mwi->call, peername, mwi->authuser);
ast_string_field_set(mwi->call, authname, mwi->authuser);
ast_string_field_set(mwi->call, fromuser, mwi->authuser);
} else {
ast_string_field_set(mwi->call, peername, mwi->username);
ast_string_field_set(mwi->call, authname, mwi->username);
ast_string_field_set(mwi->call, fromuser, mwi->username);
}
ast_string_field_set(mwi->call, username, mwi->username);
if (!ast_strlen_zero(mwi->secret)) {
ast_string_field_set(mwi->call, peersecret, mwi->secret);
}
mwi->call->socket.type = mwi->transport;
mwi->call->socket.port = htons(mwi->portno);
ast_sip_ouraddrfor(&mwi->call->sa.sin_addr, &mwi->call->ourip);
build_contact(mwi->call);
build_via(mwi->call);
build_callid_pvt(mwi->call);
ast_set_flag(&mwi->call->flags[0], SIP_OUTGOING);
/* Associate the call with us */
mwi->call->mwi = ASTOBJ_REF(mwi);
/* Actually send the packet */
transmit_invite(mwi->call, SIP_SUBSCRIBE, 0, 2);
return 0;
}
static int find_calling_channel(void *obj, void *arg, void *data, int flags)
{
struct ast_channel *c = obj;
struct sip_pvt *p = data;
int res;
ast_channel_lock(c);
res = (c->pbx &&
(!strcasecmp(c->macroexten, p->exten) || !strcasecmp(c->exten, p->exten)) &&
(sip_cfg.notifycid == IGNORE_CONTEXT || !strcasecmp(c->context, p->context)));
ast_channel_unlock(c);
return res ? CMP_MATCH | CMP_STOP : 0;
}
/*! \brief Builds XML portion of state NOTIFY messages */
static void state_notify_build_xml(int state, int full, const char *exten, const char *context, struct ast_str **tmp, struct sip_pvt *p, int subscribed, const char *mfrom, const char *mto)
{
enum state { NOTIFY_OPEN, NOTIFY_INUSE, NOTIFY_CLOSED } local_state = NOTIFY_OPEN;
const char *statestring = "terminated";
const char *pidfstate = "--";
const char *pidfnote= "Ready";
char hint[AST_MAX_EXTENSION];
switch (state) {
case (AST_EXTENSION_RINGING | AST_EXTENSION_INUSE):
statestring = (sip_cfg.notifyringing) ? "early" : "confirmed";
local_state = NOTIFY_INUSE;
pidfstate = "busy";
pidfnote = "Ringing";
break;
case AST_EXTENSION_RINGING:
statestring = "early";
local_state = NOTIFY_INUSE;
pidfstate = "busy";
pidfnote = "Ringing";
break;
case AST_EXTENSION_INUSE:
statestring = "confirmed";
local_state = NOTIFY_INUSE;
pidfstate = "busy";
pidfnote = "On the phone";
break;
case AST_EXTENSION_BUSY:
statestring = "confirmed";
local_state = NOTIFY_CLOSED;
pidfstate = "busy";
pidfnote = "On the phone";
break;
case AST_EXTENSION_UNAVAILABLE:
statestring = "terminated";
local_state = NOTIFY_CLOSED;
pidfstate = "away";
pidfnote = "Unavailable";
break;
case AST_EXTENSION_ONHOLD:
statestring = "confirmed";
local_state = NOTIFY_CLOSED;
pidfstate = "busy";
pidfnote = "On hold";
break;
case AST_EXTENSION_NOT_INUSE:
default:
/* Default setting */
break;
}
/* Check which device/devices we are watching and if they are registered */
if (ast_get_hint(hint, sizeof(hint), NULL, 0, NULL, context, exten)) {
char *hint2 = hint, *individual_hint = NULL;
int hint_count = 0, unavailable_count = 0;
while ((individual_hint = strsep(&hint2, "&"))) {
hint_count++;
if (ast_device_state(individual_hint) == AST_DEVICE_UNAVAILABLE)
unavailable_count++;
}
/* If none of the hinted devices are registered, we will
* override notification and show no availability.
*/
if (hint_count > 0 && hint_count == unavailable_count) {
local_state = NOTIFY_CLOSED;
pidfstate = "away";
pidfnote = "Not online";
}
}
switch (subscribed) {
case XPIDF_XML:
case CPIM_PIDF_XML:
ast_str_append(tmp, 0,
"<?xml version=\"1.0\"?>\n"
"<!DOCTYPE presence PUBLIC \"-//IETF//DTD RFCxxxx XPIDF 1.0//EN\" \"xpidf.dtd\">\n"
"<presence>\n");
ast_str_append(tmp, 0, "<presentity uri=\"%s;method=SUBSCRIBE\" />\n", mfrom);
ast_str_append(tmp, 0, "<atom id=\"%s\">\n", exten);
ast_str_append(tmp, 0, "<address uri=\"%s;user=ip\" priority=\"0.800000\">\n", mto);
ast_str_append(tmp, 0, "<status status=\"%s\" />\n", (local_state == NOTIFY_OPEN) ? "open" : (local_state == NOTIFY_INUSE) ? "inuse" : "closed");
ast_str_append(tmp, 0, "<msnsubstatus substatus=\"%s\" />\n", (local_state == NOTIFY_OPEN) ? "online" : (local_state == NOTIFY_INUSE) ? "onthephone" : "offline");
ast_str_append(tmp, 0, "</address>\n</atom>\n</presence>\n");
break;
case PIDF_XML: /* Eyebeam supports this format */
ast_str_append(tmp, 0,
"<?xml version=\"1.0\" encoding=\"ISO-8859-1\"?>\n"
"<presence xmlns=\"urn:ietf:params:xml:ns:pidf\" \nxmlns:pp=\"urn:ietf:params:xml:ns:pidf:person\"\nxmlns:es=\"urn:ietf:params:xml:ns:pidf:rpid:status:rpid-status\"\nxmlns:ep=\"urn:ietf:params:xml:ns:pidf:rpid:rpid-person\"\nentity=\"%s\">\n", mfrom);
ast_str_append(tmp, 0, "<pp:person><status>\n");
if (pidfstate[0] != '-')
ast_str_append(tmp, 0, "<ep:activities><ep:%s/></ep:activities>\n", pidfstate);
ast_str_append(tmp, 0, "</status></pp:person>\n");
ast_str_append(tmp, 0, "<note>%s</note>\n", pidfnote); /* Note */
ast_str_append(tmp, 0, "<tuple id=\"%s\">\n", exten); /* Tuple start */
ast_str_append(tmp, 0, "<contact priority=\"1\">%s</contact>\n", mto);
if (pidfstate[0] == 'b') /* Busy? Still open ... */
ast_str_append(tmp, 0, "<status><basic>open</basic></status>\n");
else
ast_str_append(tmp, 0, "<status><basic>%s</basic></status>\n", (local_state != NOTIFY_CLOSED) ? "open" : "closed");
ast_str_append(tmp, 0, "</tuple>\n</presence>\n");
break;
case DIALOG_INFO_XML: /* SNOM subscribes in this format */
ast_str_append(tmp, 0, "<?xml version=\"1.0\"?>");
ast_str_append(tmp, 0, "<dialog-info xmlns=\"urn:ietf:params:xml:ns:dialog-info\" version=\"%d\" state=\"%s\" entity=\"%s\">", p->dialogver, full ? "full" : "partial", mto);
if ((state & AST_EXTENSION_RINGING) && sip_cfg.notifyringing) {
const char *local_display = exten;
char *local_target = ast_strdupa(mto);
/* There are some limitations to how this works. The primary one is that the
callee must be dialing the same extension that is being monitored. Simply dialing
the hint'd device is not sufficient. */
if (sip_cfg.notifycid) {
struct ast_channel *caller;
if ((caller = ast_channel_callback(find_calling_channel, NULL, p, 0))) {
int need = strlen(caller->cid.cid_num) + strlen(p->fromdomain) + sizeof("sip:@");
local_target = alloca(need);
ast_channel_lock(caller);
snprintf(local_target, need, "sip:%s@%s", caller->cid.cid_num, p->fromdomain);
local_display = ast_strdupa(caller->cid.cid_name);
ast_channel_unlock(caller);
caller = ast_channel_unref(caller);
}
}
/* We create a fake call-id which the phone will send back in an INVITE
Replaces header which we can grab and do some magic with. */
ast_str_append(tmp, 0,
"<dialog id=\"%s\" call-id=\"pickup-%s\" direction=\"recipient\">\n"
"<remote>\n"
/* See the limitations of this above. Luckily the phone seems to still be
happy when these values are not correct. */
"<identity display=\"%s\">%s</identity>\n"
"<target uri=\"%s\"/>\n"
"</remote>\n"
"<local>\n"
"<identity>%s</identity>\n"
"<target uri=\"%s\"/>\n"
"</local>\n",
exten, p->callid, local_display, local_target, local_target, mto, mto);
} else {
ast_str_append(tmp, 0, "<dialog id=\"%s\">", exten);
}
ast_str_append(tmp, 0, "<state>%s</state>\n", statestring);
if (state == AST_EXTENSION_ONHOLD) {
ast_str_append(tmp, 0, "<local>\n<target uri=\"%s\">\n"
"<param pname=\"+sip.rendering\" pvalue=\"no\"/>\n"
"</target>\n</local>\n", mto);
}
ast_str_append(tmp, 0, "</dialog>\n</dialog-info>\n");
break;
case NONE:
default:
break;
}
}
/*! \brief Used in the SUBSCRIBE notification subsystem (RFC3265) */
static int transmit_state_notify(struct sip_pvt *p, int state, int full, int timeout)
{
struct ast_str *tmp = ast_str_alloca(4000);
char from[256], to[256];
char *c, *mfrom, *mto;
struct sip_request req;
const struct cfsubscription_types *subscriptiontype;
memset(from, 0, sizeof(from));
memset(to, 0, sizeof(to));
subscriptiontype = find_subscription_type(p->subscribed);
ast_copy_string(from, get_header(&p->initreq, "From"), sizeof(from));
c = get_in_brackets(from);
if (strncasecmp(c, "sip:", 4) && strncasecmp(c, "sips:", 5)) {
ast_log(LOG_WARNING, "Huh? Not a SIP header (%s)?\n", c);
return -1;
}
mfrom = remove_uri_parameters(c);
ast_copy_string(to, get_header(&p->initreq, "To"), sizeof(to));
c = get_in_brackets(to);
if (strncasecmp(c, "sip:", 4) && strncasecmp(c, "sips:", 5)) {
ast_log(LOG_WARNING, "Huh? Not a SIP header (%s)?\n", c);
return -1;
}
mto = remove_uri_parameters(c);
reqprep(&req, p, SIP_NOTIFY, 0, 1);
add_header(&req, "Event", subscriptiontype->event);
add_header(&req, "Content-Type", subscriptiontype->mediatype);
switch(state) {
case AST_EXTENSION_DEACTIVATED:
if (timeout)
add_header(&req, "Subscription-State", "terminated;reason=timeout");
else {
add_header(&req, "Subscription-State", "terminated;reason=probation");
add_header(&req, "Retry-After", "60");
}
break;
case AST_EXTENSION_REMOVED:
add_header(&req, "Subscription-State", "terminated;reason=noresource");
break;
default:
if (p->expiry)
add_header(&req, "Subscription-State", "active");
else /* Expired */
add_header(&req, "Subscription-State", "terminated;reason=timeout");
}
switch (p->subscribed) {
case XPIDF_XML:
case CPIM_PIDF_XML:
add_header(&req, "Event", subscriptiontype->event);
state_notify_build_xml(state, full, p->exten, p->context, &tmp, p, p->subscribed, mfrom, mto);
add_header(&req, "Content-Type", subscriptiontype->mediatype);
p->dialogver++;
break;
case PIDF_XML: /* Eyebeam supports this format */
add_header(&req, "Event", subscriptiontype->event);
state_notify_build_xml(state, full, p->exten, p->context, &tmp, p, p->subscribed, mfrom, mto);
add_header(&req, "Content-Type", subscriptiontype->mediatype);
p->dialogver++;
break;
case DIALOG_INFO_XML: /* SNOM subscribes in this format */
add_header(&req, "Event", subscriptiontype->event);
state_notify_build_xml(state, full, p->exten, p->context, &tmp, p, p->subscribed, mfrom, mto);
add_header(&req, "Content-Type", subscriptiontype->mediatype);
p->dialogver++;
break;
case NONE:
default:
break;
}
add_header_contentLength(&req, tmp->used);
add_line(&req, tmp->str);
p->pendinginvite = p->ocseq; /* Remember that we have a pending NOTIFY in order not to confuse the NOTIFY subsystem */
return send_request(p, &req, XMIT_RELIABLE, p->ocseq);
}
/*! \brief Notify user of messages waiting in voicemail (RFC3842)
\note - Notification only works for registered peers with mailbox= definitions
in sip.conf
- We use the SIP Event package message-summary
MIME type defaults to "application/simple-message-summary";
*/
static int transmit_notify_with_mwi(struct sip_pvt *p, int newmsgs, int oldmsgs, const char *vmexten)
{
struct sip_request req;
struct ast_str *out = ast_str_alloca(500);
initreqprep(&req, p, SIP_NOTIFY);
add_header(&req, "Event", "message-summary");
add_header(&req, "Content-Type", default_notifymime);
ast_str_append(&out, 0, "Messages-Waiting: %s\r\n", newmsgs ? "yes" : "no");
ast_str_append(&out, 0, "Message-Account: sip:%s@%s\r\n",
S_OR(vmexten, default_vmexten), S_OR(p->fromdomain, ast_inet_ntoa(p->ourip.sin_addr)));
/* Cisco has a bug in the SIP stack where it can't accept the
(0/0) notification. This can temporarily be disabled in
sip.conf with the "buggymwi" option */
ast_str_append(&out, 0, "Voice-Message: %d/%d%s\r\n",
newmsgs, oldmsgs, (ast_test_flag(&p->flags[1], SIP_PAGE2_BUGGY_MWI) ? "" : " (0/0)"));
if (p->subscribed) {
if (p->expiry)
add_header(&req, "Subscription-State", "active");
else /* Expired */
add_header(&req, "Subscription-State", "terminated;reason=timeout");
}
add_header_contentLength(&req, out->used);
add_line(&req, out->str);
if (!p->initreq.headers)
initialize_initreq(p, &req);
return send_request(p, &req, XMIT_RELIABLE, p->ocseq);
}
/*! \brief Notify a transferring party of the status of transfer (RFC3515) */
static int transmit_notify_with_sipfrag(struct sip_pvt *p, int cseq, char *message, int terminate)
{
struct sip_request req;
char tmp[SIPBUFSIZE/2];
reqprep(&req, p, SIP_NOTIFY, 0, 1);
snprintf(tmp, sizeof(tmp), "refer;id=%d", cseq);
add_header(&req, "Event", tmp);
add_header(&req, "Subscription-state", terminate ? "terminated;reason=noresource" : "active");
add_header(&req, "Content-Type", "message/sipfrag;version=2.0");
add_header(&req, "Allow", ALLOWED_METHODS);
add_header(&req, "Supported", SUPPORTED_EXTENSIONS);
snprintf(tmp, sizeof(tmp), "SIP/2.0 %s\r\n", message);
add_header_contentLength(&req, strlen(tmp));
add_line(&req, tmp);
if (!p->initreq.headers)
initialize_initreq(p, &req);
p->lastnoninvite = p->ocseq;
return send_request(p, &req, XMIT_RELIABLE, p->ocseq);
}
/*! \brief Notify device with custom headers from sip_notify.conf */
static int transmit_notify_custom(struct sip_pvt *p, struct ast_variable *vars) {
struct sip_request req;
struct ast_variable *var, *newvar;
initreqprep(&req, p, SIP_NOTIFY);
/* Copy notify vars and add headers */
p->notify_headers = newvar = ast_variable_new("Subscription-State", "terminated", "");
add_header(&req, newvar->name, newvar->value);
for (var = vars; var; var = var->next) {
char buf[512];
ast_debug(2, " Adding pair %s=%s\n", var->name, var->value);
ast_copy_string(buf, var->value, sizeof(buf));
add_header(&req, var->name, ast_unescape_semicolon(buf));
newvar->next = ast_variable_new(var->name, var->value, "");
newvar = newvar->next;
}
if (!p->initreq.headers) { /* Initialize first request before sending */
initialize_initreq(p, &req);
}
return send_request(p, &req, XMIT_UNRELIABLE, p->ocseq);
}
static int manager_sipnotify(struct mansession *s, const struct message *m)
{
const char *channame = astman_get_header(m, "Channel");
struct ast_variable *vars = astman_get_variables(m);
struct sip_pvt *p;
if (ast_strlen_zero(channame)) {
astman_send_error(s, m, "SIPNotify requires a channel name");
return 0;
}
if (!strncasecmp(channame, "sip/", 4)) {
channame += 4;
}
if (!(p = sip_alloc(NULL, NULL, 0, SIP_NOTIFY))) {
astman_send_error(s, m, "Unable to build sip pvt data for notify (memory/socket error)");
return 0;
}
if (create_addr(p, channame, NULL, 0)) {
/* Maybe they're not registered, etc. */
dialog_unlink_all(p, TRUE, TRUE);
dialog_unref(p, "unref dialog inside for loop" );
/* sip_destroy(p); */
astman_send_error(s, m, "Could not create address");
return 0;
}
/* Notify is outgoing call */
ast_set_flag(&p->flags[0], SIP_OUTGOING);
/* Recalculate our side, and recalculate Call ID */
ast_sip_ouraddrfor(&p->sa.sin_addr, &p->ourip);
build_via(p);
ao2_t_unlink(dialogs, p, "About to change the callid -- remove the old name");
build_callid_pvt(p);
ao2_t_link(dialogs, p, "Linking in new name");
dialog_ref(p, "bump the count of p, which transmit_sip_request will decrement.");
sip_scheddestroy(p, SIP_TRANS_TIMEOUT);
if (!transmit_notify_custom(p, vars)) {
astman_send_ack(s, m, "Notify Sent");
} else {
astman_send_error(s, m, "Unable to send notify");
}
ast_variables_destroy(vars);
return 0;
}
static char mandescr_sipnotify[] =
"Description: Sends a SIP Notify event\n"
"All parameters for this event must be specified in the body of this request\n"
"via multiple Variable: name=value sequences.\n"
"Variables: \n"
" *Channel: <peername> Peer to receive the notify. Required.\n"
" *Variable: <name>=<value> At least one variable pair must be specified.\n"
" ActionID: <id> Action ID for this transaction. Will be returned.\n";
/*! \brief Send a provisional response indicating that a call was redirected
*/
static void update_redirecting(struct sip_pvt *p, const void *data, size_t datalen)
{
struct sip_request resp;
if (p->owner->_state == AST_STATE_UP || ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
return;
}
if (!ast_strlen_zero(p->owner->redirecting.to.number)) {
ast_string_field_set(p, exten, p->owner->redirecting.to.number);
build_contact(p);
}
respprep(&resp, p, "181 Call is being forwarded", &p->initreq);
add_diversion_header(&resp, p);
send_response(p, &resp, XMIT_UNRELIABLE, 0);
}
/*! \brief Notify peer that the connected line has changed */
static void update_connectedline(struct sip_pvt *p, const void *data, size_t datalen)
{
if (!ast_test_flag(&p->flags[0], SIP_SENDRPID))
return;
if (ast_strlen_zero(p->owner->connected.id.number))
return;
append_history(p, "ConnectedLine", "%s party is now %s <%s>", ast_test_flag(&p->flags[0], SIP_OUTGOING) ? "Calling" : "Called", p->owner->connected.id.name, p->owner->connected.id.number);
if (p->owner->_state == AST_STATE_UP || ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
struct sip_request req;
if (p->invitestate == INV_CONFIRMED || p->invitestate == INV_TERMINATED) {
reqprep(&req, p, ast_test_flag(&p->flags[0], SIP_REINVITE_UPDATE) ? SIP_UPDATE : SIP_INVITE, 0, 1);
add_header(&req, "Allow", ALLOWED_METHODS);
add_header(&req, "Supported", SUPPORTED_EXTENSIONS);
add_rpid(&req, p);
add_sdp(&req, p, FALSE, TRUE, FALSE);
initialize_initreq(p, &req);
p->lastinvite = p->ocseq;
ast_set_flag(&p->flags[0], SIP_OUTGOING);
send_request(p, &req, XMIT_CRITICAL, p->ocseq);
} else {
reqprep(&req, p, SIP_UPDATE, 0, 1);
add_rpid(&req, p);
add_header_contentLength(&req, 0);
send_request(p, &req, XMIT_CRITICAL, p->ocseq);
}
} else {
if (ast_test_flag(&p->flags[1], SIP_PAGE2_RPID_IMMEDIATE)) {
struct sip_request resp;
if ((p->owner->_state == AST_STATE_RING) && !ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT)) {
respprep(&resp, p, "180 Ringing", &p->initreq);
add_rpid(&resp, p);
send_response(p, &resp, XMIT_UNRELIABLE, 0);
ast_set_flag(&p->flags[0], SIP_RINGING);
} else if (p->owner->_state == AST_STATE_RINGING) {
respprep(&resp, p, "183 Session Progress", &p->initreq);
add_rpid(&resp, p);
send_response(p, &resp, XMIT_UNRELIABLE, 0);
ast_set_flag(&p->flags[0], SIP_PROGRESS_SENT);
} else {
ast_log(LOG_DEBUG, "Unable able to send update to '%s' in state '%s'\n", p->owner->name, ast_state2str(p->owner->_state));
}
} else {
ast_set_flag(&p->flags[1], SIP_PAGE2_CONNECTLINEUPDATE_PEND);
}
}
}
static const struct _map_x_s regstatestrings[] = {
{ REG_STATE_FAILED, "Failed" },
{ REG_STATE_UNREGISTERED, "Unregistered"},
{ REG_STATE_REGSENT, "Request Sent"},
{ REG_STATE_AUTHSENT, "Auth. Sent"},
{ REG_STATE_REGISTERED, "Registered"},
{ REG_STATE_REJECTED, "Rejected"},
{ REG_STATE_TIMEOUT, "Timeout"},
{ REG_STATE_NOAUTH, "No Authentication"},
{ -1, NULL } /* terminator */
};
/*! \brief Convert registration state status to string */
static const char *regstate2str(enum sipregistrystate regstate)
{
return map_x_s(regstatestrings, regstate, "Unknown");
}
/*! \brief Update registration with SIP Proxy.
* Called from the scheduler when the previous registration expires,
* so we don't have to cancel the pending event.
* We assume the reference so the sip_registry is valid, since it
* is stored in the scheduled event anyways.
*/
static int sip_reregister(const void *data)
{
/* if we are here, we know that we need to reregister. */
struct sip_registry *r = (struct sip_registry *) data;
/* if we couldn't get a reference to the registry object, punt */
if (!r)
return 0;
if (r->call && r->call->do_history)
append_history(r->call, "RegistryRenew", "Account: %s@%s", r->username, r->hostname);
/* Since registry's are only added/removed by the the monitor thread, this
may be overkill to reference/dereference at all here */
if (sipdebug)
ast_log(LOG_NOTICE, " -- Re-registration for %s@%s\n", r->username, r->hostname);
r->expire = -1;
__sip_do_register(r);
registry_unref(r, "unreg the re-registered");
return 0;
}
/*! \brief Register with SIP proxy */
static int __sip_do_register(struct sip_registry *r)
{
int res;
res = transmit_register(r, SIP_REGISTER, NULL, NULL);
return res;
}
/*! \brief Registration timeout, register again
* Registered as a timeout handler during transmit_register(),
* to retransmit the packet if a reply does not come back.
* This is called by the scheduler so the event is not pending anymore when
* we are called.
*/
static int sip_reg_timeout(const void *data)
{
/* if we are here, our registration timed out, so we'll just do it over */
struct sip_registry *r = (struct sip_registry *)data; /* the ref count should have been bumped when the sched item was added */
struct sip_pvt *p;
int res;
/* if we couldn't get a reference to the registry object, punt */
if (!r)
return 0;
if (r->dnsmgr) {
/* If the registration has timed out, maybe the IP changed. Force a refresh. */
ast_dnsmgr_refresh(r->dnsmgr);
}
ast_log(LOG_NOTICE, " -- Registration for '%s@%s' timed out, trying again (Attempt #%d)\n", r->username, r->hostname, r->regattempts);
/* If the initial tranmission failed, we may not have an existing dialog,
* so it is possible that r->call == NULL.
* Otherwise destroy it, as we have a timeout so we don't want it.
*/
if (r->call) {
/* Unlink us, destroy old call. Locking is not relevant here because all this happens
in the single SIP manager thread. */
p = r->call;
sip_pvt_lock(p);
pvt_set_needdestroy(p, "registration timeout");
/* Pretend to ACK anything just in case */
__sip_pretend_ack(p);
sip_pvt_unlock(p);
/* decouple the two objects */
/* p->registry == r, so r has 2 refs, and the unref won't take the object away */
if (p->registry)
p->registry = registry_unref(p->registry, "p->registry unreffed");
r->call = dialog_unref(r->call, "unrefing r->call");
}
/* If we have a limit, stop registration and give up */
if (global_regattempts_max && r->regattempts > global_regattempts_max) {
/* Ok, enough is enough. Don't try any more */
/* We could add an external notification here...
steal it from app_voicemail :-) */
ast_log(LOG_NOTICE, " -- Giving up forever trying to register '%s@%s'\n", r->username, r->hostname);
r->regstate = REG_STATE_FAILED;
} else {
r->regstate = REG_STATE_UNREGISTERED;
r->timeout = -1;
res=transmit_register(r, SIP_REGISTER, NULL, NULL);
}
manager_event(EVENT_FLAG_SYSTEM, "Registry", "ChannelType: SIP\r\nUsername: %s\r\nDomain: %s\r\nStatus: %s\r\n", r->username, r->hostname, regstate2str(r->regstate));
registry_unref(r, "unreffing registry_unref r");
return 0;
}
/*! \brief Transmit register to SIP proxy or UA
* auth = NULL on the initial registration (from sip_reregister())
*/
static int transmit_register(struct sip_registry *r, int sipmethod, const char *auth, const char *authheader)
{
struct sip_request req;
char from[256];
char to[256];
char tmp[80];
char addr[80];
struct sip_pvt *p;
int res;
char *fromdomain;
char *domainport = NULL;
/* exit if we are already in process with this registrar ?*/
if (r == NULL || ((auth == NULL) && (r->regstate == REG_STATE_REGSENT || r->regstate == REG_STATE_AUTHSENT))) {
if (r) {
ast_log(LOG_NOTICE, "Strange, trying to register %s@%s when registration already pending\n", r->username, r->hostname);
}
return 0;
}
if (r->dnsmgr == NULL) {
char transport[MAXHOSTNAMELEN];
snprintf(transport, sizeof(transport), "_sip._%s", get_transport(r->transport)); /* have to use static get_transport function */
ast_dnsmgr_lookup(r->hostname, &r->us, &r->dnsmgr, sip_cfg.srvlookup ? transport : NULL);
}
if (r->call) { /* We have a registration */
if (!auth) {
ast_log(LOG_WARNING, "Already have a REGISTER going on to %s@%s?? \n", r->username, r->hostname);
return 0;
} else {
p = dialog_ref(r->call, "getting a copy of the r->call dialog in transmit_register");
make_our_tag(p->tag, sizeof(p->tag)); /* create a new local tag for every register attempt */
ast_string_field_set(p, theirtag, NULL); /* forget their old tag, so we don't match tags when getting response */
}
} else {
/* Build callid for registration if we haven't registered before */
if (!r->callid_valid) {
build_callid_registry(r, internip.sin_addr, default_fromdomain);
r->callid_valid = TRUE;
}
/* Allocate SIP dialog for registration */
if (!(p = sip_alloc( r->callid, NULL, 0, SIP_REGISTER))) {
ast_log(LOG_WARNING, "Unable to allocate registration transaction (memory or socket error)\n");
return 0;
}
if (p->do_history)
append_history(p, "RegistryInit", "Account: %s@%s", r->username, r->hostname);
ref_proxy(p, obproxy_get(p, NULL));
/* Use port number specified if no SRV record was found */
if (!r->us.sin_port && r->portno)
r->us.sin_port = htons(r->portno);
/* Find address to hostname */
if (create_addr(p, r->hostname, &r->us, 0)) {
/* we have what we hope is a temporary network error,
* probably DNS. We need to reschedule a registration try */
dialog_unlink_all(p, TRUE, TRUE);
p = dialog_unref(p, "unref dialog after unlink_all");
if (r->timeout > -1) {
AST_SCHED_REPLACE_UNREF(r->timeout, sched, global_reg_timeout * 1000, sip_reg_timeout, r,
registry_unref(_data, "del for REPLACE of registry ptr"),
registry_unref(r, "object ptr dec when SCHED_REPLACE add failed"),
registry_addref(r,"add for REPLACE registry ptr"));
ast_log(LOG_WARNING, "Still have a registration timeout for %s@%s (create_addr() error), %d\n", r->username, r->hostname, r->timeout);
} else {
r->timeout = ast_sched_add(sched, global_reg_timeout * 1000, sip_reg_timeout, registry_addref(r, "add for REPLACE registry ptr"));
ast_log(LOG_WARNING, "Probably a DNS error for registration to %s@%s, trying REGISTER again (after %d seconds)\n", r->username, r->hostname, global_reg_timeout);
}
r->regattempts++;
return 0;
}
/* Copy back Call-ID in case create_addr changed it */
ast_string_field_set(r, callid, p->callid);
if (!r->dnsmgr && r->portno) {
p->sa.sin_port = htons(r->portno);
p->recv.sin_port = htons(r->portno);
} else { /* Set registry port to the port set from the peer definition/srv or default */
r->portno = ntohs(p->sa.sin_port);
}
ast_set_flag(&p->flags[0], SIP_OUTGOING); /* Registration is outgoing call */
r->call = dialog_ref(p, "copying dialog into registry r->call"); /* Save pointer to SIP dialog */
p->registry = registry_addref(r, "transmit_register: addref to p->registry in transmit_register"); /* Add pointer to registry in packet */
if (!ast_strlen_zero(r->secret)) { /* Secret (password) */
ast_string_field_set(p, peersecret, r->secret);
}
if (!ast_strlen_zero(r->md5secret))
ast_string_field_set(p, peermd5secret, r->md5secret);
/* User name in this realm
- if authuser is set, use that, otherwise use username */
if (!ast_strlen_zero(r->authuser)) {
ast_string_field_set(p, peername, r->authuser);
ast_string_field_set(p, authname, r->authuser);
} else if (!ast_strlen_zero(r->username)) {
ast_string_field_set(p, peername, r->username);
ast_string_field_set(p, authname, r->username);
ast_string_field_set(p, fromuser, r->username);
}
if (!ast_strlen_zero(r->username))
ast_string_field_set(p, username, r->username);
/* Save extension in packet */
if (!ast_strlen_zero(r->callback))
ast_string_field_set(p, exten, r->callback);
/* Set transport and port so the correct contact is built */
p->socket.type = r->transport;
if (r->transport == SIP_TRANSPORT_TLS || r->transport == SIP_TRANSPORT_TCP) {
p->socket.port = sip_tcp_desc.local_address.sin_port;
}
/*
check which address we should use in our contact header
based on whether the remote host is on the external or
internal network so we can register through nat
*/
ast_sip_ouraddrfor(&p->sa.sin_addr, &p->ourip);
build_contact(p);
}
/* set up a timeout */
if (auth == NULL) {
if (r->timeout > -1)
ast_log(LOG_WARNING, "Still have a registration timeout, #%d - deleting it\n", r->timeout);
AST_SCHED_REPLACE_UNREF(r->timeout, sched, global_reg_timeout * 1000, sip_reg_timeout, r,
registry_unref(_data,"reg ptr unrefed from del in SCHED_REPLACE"),
registry_unref(r,"reg ptr unrefed from add failure in SCHED_REPLACE"),
registry_addref(r,"reg ptr reffed from add in SCHED_REPLACE"));
ast_debug(1, "Scheduled a registration timeout for %s id #%d \n", r->hostname, r->timeout);
}
if ((fromdomain = strchr(r->username, '@'))) {
/* the domain name is just behind '@' */
fromdomain++ ;
/* We have a domain in the username for registration */
snprintf(from, sizeof(from), "<sip:%s>;tag=%s", r->username, p->tag);
if (!ast_strlen_zero(p->theirtag))
snprintf(to, sizeof(to), "<sip:%s>;tag=%s", r->username, p->theirtag);
else
snprintf(to, sizeof(to), "<sip:%s>", r->username);
/* If the registration username contains '@', then the domain should be used as
the equivalent of "fromdomain" for the registration */
if (ast_strlen_zero(p->fromdomain)) {
ast_string_field_set(p, fromdomain, fromdomain);
}
} else {
snprintf(from, sizeof(from), "<sip:%s@%s>;tag=%s", r->username, p->tohost, p->tag);
if (!ast_strlen_zero(p->theirtag))
snprintf(to, sizeof(to), "<sip:%s@%s>;tag=%s", r->username, p->tohost, p->theirtag);
else
snprintf(to, sizeof(to), "<sip:%s@%s>", r->username, p->tohost);
}
/* Fromdomain is what we are registering to, regardless of actual
host name from SRV */
if (!ast_strlen_zero(p->fromdomain)) {
domainport = strrchr(p->fromdomain, ':');
if (domainport) {
*domainport++ = '\0'; /* trim off domainport from p->fromdomain */
if (ast_strlen_zero(domainport))
domainport = NULL;
}
if (domainport) {
if (atoi(domainport) != STANDARD_SIP_PORT)
snprintf(addr, sizeof(addr), "sip:%s:%s", p->fromdomain, domainport);
else
snprintf(addr, sizeof(addr), "sip:%s", p->fromdomain);
} else {
if (r->portno && r->portno != STANDARD_SIP_PORT)
snprintf(addr, sizeof(addr), "sip:%s:%d", p->fromdomain, r->portno);
else
snprintf(addr, sizeof(addr), "sip:%s", p->fromdomain);
}
} else {
if (r->portno && r->portno != STANDARD_SIP_PORT)
snprintf(addr, sizeof(addr), "sip:%s:%d", r->hostname, r->portno);
else
snprintf(addr, sizeof(addr), "sip:%s", r->hostname);
}
ast_string_field_set(p, uri, addr);
p->branch ^= ast_random();
init_req(&req, sipmethod, addr);
/* Add to CSEQ */
snprintf(tmp, sizeof(tmp), "%u %s", ++r->ocseq, sip_methods[sipmethod].text);
p->ocseq = r->ocseq;
build_via(p);
add_header(&req, "Via", p->via);
add_header(&req, "Max-Forwards", DEFAULT_MAX_FORWARDS);
add_header(&req, "From", from);
add_header(&req, "To", to);
add_header(&req, "Call-ID", p->callid);
add_header(&req, "CSeq", tmp);
if (!ast_strlen_zero(global_useragent))
add_header(&req, "User-Agent", global_useragent);
if (auth) /* Add auth header */
add_header(&req, authheader, auth);
else if (!ast_strlen_zero(r->nonce)) {
char digest[1024];
/* We have auth data to reuse, build a digest header.
* Note, this is not always useful because some parties do not
* like nonces to be reused (for good reasons!) so they will
* challenge us anyways.
*/
if (sipdebug)
ast_debug(1, " >>> Re-using Auth data for %s@%s\n", r->username, r->hostname);
ast_string_field_set(p, realm, r->realm);
ast_string_field_set(p, nonce, r->nonce);
ast_string_field_set(p, domain, r->domain);
ast_string_field_set(p, opaque, r->opaque);
ast_string_field_set(p, qop, r->qop);
p->noncecount = ++r->noncecount;
memset(digest, 0, sizeof(digest));
if(!build_reply_digest(p, sipmethod, digest, sizeof(digest)))
add_header(&req, "Authorization", digest);
else
ast_log(LOG_NOTICE, "No authorization available for authentication of registration to %s@%s\n", r->username, r->hostname);
}
snprintf(tmp, sizeof(tmp), "%d", r->expiry);
add_header(&req, "Expires", tmp);
add_header(&req, "Contact", p->our_contact);
add_header_contentLength(&req, 0);
initialize_initreq(p, &req);
if (sip_debug_test_pvt(p)) {
ast_verbose("REGISTER %d headers, %d lines\n", p->initreq.headers, p->initreq.lines);
}
r->regstate = auth ? REG_STATE_AUTHSENT : REG_STATE_REGSENT;
r->regattempts++; /* Another attempt */
ast_debug(4, "REGISTER attempt %d to %s@%s\n", r->regattempts, r->username, r->hostname);
res = send_request(p, &req, XMIT_CRITICAL, p->ocseq);
dialog_unref(p, "p is finished here at the end of transmit_register");
return res;
}
/*! \brief Transmit text with SIP MESSAGE method */
static int transmit_message_with_text(struct sip_pvt *p, const char *text)
{
struct sip_request req;
reqprep(&req, p, SIP_MESSAGE, 0, 1);
add_text(&req, text);
return send_request(p, &req, XMIT_RELIABLE, p->ocseq);
}
/*! \brief Allocate SIP refer structure */
static int sip_refer_allocate(struct sip_pvt *p)
{
p->refer = ast_calloc(1, sizeof(struct sip_refer));
return p->refer ? 1 : 0;
}
/*! \brief Transmit SIP REFER message (initiated by the transfer() dialplan application
\note this is currently broken as we have no way of telling the dialplan
engine whether a transfer succeeds or fails.
\todo Fix the transfer() dialplan function so that a transfer may fail
*/
static int transmit_refer(struct sip_pvt *p, const char *dest)
{
struct sip_request req = {
.headers = 0,
};
char from[256];
const char *of;
char *c;
char referto[256];
char *ttag, *ftag;
char *theirtag = ast_strdupa(p->theirtag);
int use_tls=FALSE;
if (sipdebug)
ast_debug(1, "SIP transfer of %s to %s\n", p->callid, dest);
/* Are we transfering an inbound or outbound call ? */
if (ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
of = get_header(&p->initreq, "To");
ttag = theirtag;
ftag = p->tag;
} else {
of = get_header(&p->initreq, "From");
ftag = theirtag;
ttag = p->tag;
}
ast_copy_string(from, of, sizeof(from));
of = get_in_brackets(from);
ast_string_field_set(p, from, of);
if (!strncasecmp(of, "sip:", 4)) {
of += 4;
}else if (!strncasecmp(of, "sips:", 5)) {
of += 5;
use_tls = TRUE;
} else {
ast_log(LOG_NOTICE, "From address missing 'sip(s):', assuming sip:\n");
}
/* Get just the username part */
if ((c = strchr(dest, '@')))
c = NULL;
else if ((c = strchr(of, '@')))
*c++ = '\0';
if (c)
snprintf(referto, sizeof(referto), "<sip%s:%s@%s>", use_tls ? "s" : "", dest, c);
else
snprintf(referto, sizeof(referto), "<sip%s:%s>", use_tls ? "s" : "", dest);
/* save in case we get 407 challenge */
sip_refer_allocate(p);
ast_copy_string(p->refer->refer_to, referto, sizeof(p->refer->refer_to));
ast_copy_string(p->refer->referred_by, p->our_contact, sizeof(p->refer->referred_by));
p->refer->status = REFER_SENT; /* Set refer status */
reqprep(&req, p, SIP_REFER, 0, 1);
add_header(&req, "Refer-To", referto);
add_header(&req, "Allow", ALLOWED_METHODS);
add_header(&req, "Supported", SUPPORTED_EXTENSIONS);
if (!ast_strlen_zero(p->our_contact))
add_header(&req, "Referred-By", p->our_contact);
return send_request(p, &req, XMIT_RELIABLE, p->ocseq);
/* We should propably wait for a NOTIFY here until we ack the transfer */
/* Maybe fork a new thread and wait for a STATUS of REFER_200OK on the refer status before returning to app_transfer */
/*! \todo In theory, we should hang around and wait for a reply, before
returning to the dial plan here. Don't know really how that would
affect the transfer() app or the pbx, but, well, to make this
useful we should have a STATUS code on transfer().
*/
}
/*! \brief Send SIP INFO dtmf message, see Cisco documentation on cisco.com */
static int transmit_info_with_digit(struct sip_pvt *p, const char digit, unsigned int duration)
{
struct sip_request req;
reqprep(&req, p, SIP_INFO, 0, 1);
add_digit(&req, digit, duration, (ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_SHORTINFO));
return send_request(p, &req, XMIT_RELIABLE, p->ocseq);
}
/*! \brief Send SIP INFO with video update request */
static int transmit_info_with_vidupdate(struct sip_pvt *p)
{
struct sip_request req;
reqprep(&req, p, SIP_INFO, 0, 1);
add_vidupdate(&req);
return send_request(p, &req, XMIT_RELIABLE, p->ocseq);
}
/*! \brief Transmit generic SIP request
returns XMIT_ERROR if transmit failed with a critical error (don't retry)
*/
static int transmit_request(struct sip_pvt *p, int sipmethod, int seqno, enum xmittype reliable, int newbranch)
{
struct sip_request resp;
if (sipmethod == SIP_ACK)
p->invitestate = INV_CONFIRMED;
reqprep(&resp, p, sipmethod, seqno, newbranch);
if (sipmethod == SIP_CANCEL && p->answered_elsewhere)
add_header(&resp, "Reason", "SIP;cause=200;text=\"Call completed elsewhere\"");
add_header_contentLength(&resp, 0);
return send_request(p, &resp, reliable, seqno ? seqno : p->ocseq);
}
/*! \brief return the request and response heade for a 401 or 407 code */
static void auth_headers(enum sip_auth_type code, char **header, char **respheader)
{
if (code == WWW_AUTH) { /* 401 */
*header = "WWW-Authenticate";
*respheader = "Authorization";
} else if (code == PROXY_AUTH) { /* 407 */
*header = "Proxy-Authenticate";
*respheader = "Proxy-Authorization";
} else {
ast_verbose("-- wrong response code %d\n", code);
*header = *respheader = "Invalid";
}
}
/*! \brief Transmit SIP request, auth added */
static int transmit_request_with_auth(struct sip_pvt *p, int sipmethod, int seqno, enum xmittype reliable, int newbranch)
{
struct sip_request resp;
reqprep(&resp, p, sipmethod, seqno, newbranch);
if (!ast_strlen_zero(p->realm)) {
char digest[1024];
memset(digest, 0, sizeof(digest));
if(!build_reply_digest(p, sipmethod, digest, sizeof(digest))) {
char *dummy, *response;
enum sip_auth_type code = p->options ? p->options->auth_type : PROXY_AUTH; /* XXX force 407 if unknown */
auth_headers(code, &dummy, &response);
add_header(&resp, response, digest);
} else
ast_log(LOG_WARNING, "No authentication available for call %s\n", p->callid);
}
/* If we are hanging up and know a cause for that, send it in clear text to make
debugging easier. */
if (sipmethod == SIP_BYE) {
char buf[10];
add_header(&resp, "X-Asterisk-HangupCause", ast_cause2str(p->hangupcause));
snprintf(buf, sizeof(buf), "%d", p->hangupcause);
add_header(&resp, "X-Asterisk-HangupCauseCode", buf);
}
add_header_contentLength(&resp, 0);
return send_request(p, &resp, reliable, seqno ? seqno : p->ocseq);
}
/*! \brief Remove registration data from realtime database or AST/DB when registration expires */
static void destroy_association(struct sip_peer *peer)
{
int realtimeregs = ast_check_realtime("sipregs");
char *tablename = (realtimeregs) ? "sipregs" : "sippeers";
if (!sip_cfg.ignore_regexpire) {
if (peer->rt_fromcontact && sip_cfg.peer_rtupdate) {
ast_update_realtime(tablename, "name", peer->name, "fullcontact", "", "ipaddr", "", "port", "", "regseconds", "0", peer->deprecated_username ? "username" : "defaultuser", "", "regserver", "", "useragent", "", "lastms", "", SENTINEL);
} else {
ast_db_del("SIP/Registry", peer->name);
}
}
}
/*! \brief Expire registration of SIP peer */
static int expire_register(const void *data)
{
struct sip_peer *peer = (struct sip_peer *)data;
if (!peer) /* Hmmm. We have no peer. Weird. */
return 0;
peer->expire = -1;
memset(&peer->addr, 0, sizeof(peer->addr));
destroy_association(peer); /* remove registration data from storage */
manager_event(EVENT_FLAG_SYSTEM, "PeerStatus", "ChannelType: SIP\r\nPeer: SIP/%s\r\nPeerStatus: Unregistered\r\nCause: Expired\r\n", peer->name);
register_peer_exten(peer, FALSE); /* Remove regexten */
ast_devstate_changed(AST_DEVICE_UNKNOWN, "SIP/%s", peer->name);
/* Do we need to release this peer from memory?
Only for realtime peers and autocreated peers
*/
if (peer->is_realtime)
ast_debug(3, "-REALTIME- peer expired registration. Name: %s. Realtime peer objects now %d\n", peer->name, rpeerobjs);
if (peer->selfdestruct ||
ast_test_flag(&peer->flags[1], SIP_PAGE2_RTAUTOCLEAR)) {
ao2_t_unlink(peers, peer, "ao2_unlink of peer from peers table");
if (peer->addr.sin_addr.s_addr) {
ao2_t_unlink(peers_by_ip, peer, "ao2_unlink of peer from peers_by_ip table");
}
}
unref_peer(peer, "removing peer ref for expire_register");
return 0;
}
/*! \brief Poke peer (send qualify to check if peer is alive and well) */
static int sip_poke_peer_s(const void *data)
{
struct sip_peer *peer = (struct sip_peer *)data;
peer->pokeexpire = -1;
sip_poke_peer(peer, 0);
unref_peer(peer, "removing poke peer ref");
return 0;
}
/*! \brief Get registration details from Asterisk DB */
static void reg_source_db(struct sip_peer *peer)
{
char data[256];
struct in_addr in;
int expire;
int port;
char *scan, *addr, *port_str, *expiry_str, *username, *contact;
if (peer->rt_fromcontact)
return;
if (ast_db_get("SIP/Registry", peer->name, data, sizeof(data)))
return;
scan = data;
addr = strsep(&scan, ":");
port_str = strsep(&scan, ":");
expiry_str = strsep(&scan, ":");
username = strsep(&scan, ":");
contact = scan; /* Contact include sip: and has to be the last part of the database entry as long as we use : as a separator */
if (!inet_aton(addr, &in))
return;
if (port_str)
port = atoi(port_str);
else
return;
if (expiry_str)
expire = atoi(expiry_str);
else
return;
if (username)
ast_string_field_set(peer, username, username);
if (contact)
ast_string_field_set(peer, fullcontact, contact);
ast_debug(2, "SIP Seeding peer from astdb: '%s' at %s@%s:%d for %d\n",
peer->name, peer->username, ast_inet_ntoa(in), port, expire);
memset(&peer->addr, 0, sizeof(peer->addr));
peer->addr.sin_family = AF_INET;
peer->addr.sin_addr = in;
peer->addr.sin_port = htons(port);
if (sipsock < 0) {
/* SIP isn't up yet, so schedule a poke only, pretty soon */
AST_SCHED_REPLACE_UNREF(peer->pokeexpire, sched, ast_random() % 5000 + 1, sip_poke_peer_s, peer,
unref_peer(_data, "removing poke peer ref"),
unref_peer(peer, "removing poke peer ref"),
ref_peer(peer, "adding poke peer ref"));
} else {
sip_poke_peer(peer, 0);
}
AST_SCHED_REPLACE_UNREF(peer->expire, sched, (expire + 10) * 1000, expire_register, peer,
unref_peer(_data, "remove registration ref"),
unref_peer(peer, "remove registration ref"),
ref_peer(peer, "add registration ref"));
register_peer_exten(peer, TRUE);
}
/*! \brief Save contact header for 200 OK on INVITE */
static int parse_ok_contact(struct sip_pvt *pvt, struct sip_request *req)
{
char contact[SIPBUFSIZE];
char *c;
/* Look for brackets */
ast_copy_string(contact, get_header(req, "Contact"), sizeof(contact));
c = get_in_brackets(contact);
/* Save full contact to call pvt for later bye or re-invite */
ast_string_field_set(pvt, fullcontact, c);
/* Save URI for later ACKs, BYE or RE-invites */
ast_string_field_set(pvt, okcontacturi, c);
/* We should return false for URI:s we can't handle,
like tel:, mailto:,ldap: etc */
return TRUE;
}
static int __set_address_from_contact(const char *fullcontact, struct sockaddr_in *sin, int tcp)
{
struct hostent *hp;
struct ast_hostent ahp;
int port;
char *host, *pt;
char contact_buf[256];
char contact2_buf[256];
char *contact, *contact2;
int use_tls = FALSE;
/* Work on a copy */
ast_copy_string(contact_buf, fullcontact, sizeof(contact_buf));
ast_copy_string(contact2_buf, fullcontact, sizeof(contact2_buf));
contact = contact_buf;
contact2 = contact2_buf;
/* We have only the part in <brackets> here so we just need to parse a SIP URI.*/
/*! \brief This code is wrong, it assumes that the contact we receive will use the
same transport as the request. It's not a valid assumption. The contact for
a udp connection can be a SIPS uri, or request ;transport=tcp
\todo Fix this buggy code. It doesn't even parse transport!!!!
Note: The outbound proxy could be using UDP between the proxy and Asterisk.
We still need to be able to send to the remote agent through the proxy.
*/
if (tcp) {
if (!parse_uri(contact, "sips:", &contact, NULL, &host, &pt, NULL)) {
use_tls = TRUE;
} else {
if (parse_uri(contact2, "sip:", &contact, NULL, &host, &pt, NULL))
ast_log(LOG_NOTICE, "'%s' is not a valid SIP contact (missing sip:) trying to use anyway\n", contact);
}
port = !ast_strlen_zero(pt) ? atoi(pt) : STANDARD_TLS_PORT;
/*! \todo XXX why are we setting TLS port if there's no port given? parse_uri needs to return the transport. */
} else {
if (parse_uri(contact, "sip:", &contact, NULL, &host, &pt, NULL))
ast_log(LOG_NOTICE, "'%s' is not a valid SIP contact (missing sip:) trying to use anyway\n", contact);
port = !ast_strlen_zero(pt) ? atoi(pt) : STANDARD_SIP_PORT;
}
/* XXX This could block for a long time XXX */
/* We should only do this if it's a name, not an IP */
/* \todo - if there's no PORT number in contact - we are required to check NAPTR/SRV records
to find transport, port address and hostname. If there's a port number, we have to
assume that the domain part is a host name and only look for an A/AAAA record in DNS.
*/
hp = ast_gethostbyname(host, &ahp);
if (!hp) {
ast_log(LOG_WARNING, "Invalid host name in Contact: (can't resolve in DNS) : '%s'\n", host);
return -1;
}
sin->sin_family = AF_INET;
memcpy(&sin->sin_addr, hp->h_addr, sizeof(sin->sin_addr));
sin->sin_port = htons(port);
return 0;
}
/*! \brief Change the other partys IP address based on given contact */
static int set_address_from_contact(struct sip_pvt *pvt)
{
if (ast_test_flag(&pvt->flags[0], SIP_NAT_ROUTE)) {
/* NAT: Don't trust the contact field. Just use what they came to us
with. */
/*! \todo We need to save the TRANSPORT here too */
pvt->sa = pvt->recv;
return 0;
}
return __set_address_from_contact(pvt->fullcontact, &pvt->sa, pvt->socket.type == SIP_TRANSPORT_TLS ? 1 : 0);
}
/*! \brief Parse contact header and save registration (peer registration) */
static enum parse_register_result parse_register_contact(struct sip_pvt *pvt, struct sip_peer *peer, struct sip_request *req)
{
char contact[SIPBUFSIZE];
char data[SIPBUFSIZE];
const char *expires = get_header(req, "Expires");
int expire = atoi(expires);
char *curi, *host, *pt, *curi2;
int port;
const char *useragent;
struct hostent *hp;
struct ast_hostent ahp;
struct sockaddr_in oldsin, testsin;
ast_copy_string(contact, get_header(req, "Contact"), sizeof(contact));
if (ast_strlen_zero(expires)) { /* No expires header, try look in Contact: */
char *s = strcasestr(contact, ";expires=");
if (s) {
expires = strsep(&s, ";"); /* trim ; and beyond */
if (sscanf(expires + 9, "%d", &expire) != 1)
expire = default_expiry;
} else {
/* Nothing has been specified */
expire = default_expiry;
}
}
if (peer->socket.type == req->socket.type)
copy_socket_data(&peer->socket, &req->socket);
copy_socket_data(&pvt->socket, &req->socket);
/* Look for brackets */
curi = contact;
if (strchr(contact, '<') == NULL) /* No <, check for ; and strip it */
strsep(&curi, ";"); /* This is Header options, not URI options */
curi = get_in_brackets(contact);
curi2 = ast_strdupa(curi);
/* if they did not specify Contact: or Expires:, they are querying
what we currently have stored as their contact address, so return
it
*/
if (ast_strlen_zero(curi) && ast_strlen_zero(expires)) {
/* If we have an active registration, tell them when the registration is going to expire */
if (peer->expire > -1 && !ast_strlen_zero(peer->fullcontact))
pvt->expiry = ast_sched_when(sched, peer->expire);
return PARSE_REGISTER_QUERY;
} else if (!strcasecmp(curi, "*") || !expire) { /* Unregister this peer */
/* This means remove all registrations and return OK */
memset(&peer->addr, 0, sizeof(peer->addr));
AST_SCHED_DEL_UNREF(sched, peer->expire,
unref_peer(peer, "remove register expire ref"));
destroy_association(peer);
register_peer_exten(peer, FALSE); /* Remove extension from regexten= setting in sip.conf */
ast_string_field_set(peer, fullcontact, "");
ast_string_field_set(peer, useragent, "");
peer->sipoptions = 0;
peer->lastms = 0;
pvt->expiry = 0;
ast_verb(3, "Unregistered SIP '%s'\n", peer->name);
manager_event(EVENT_FLAG_SYSTEM, "PeerStatus", "ChannelType: SIP\r\nPeer: SIP/%s\r\nPeerStatus: Unregistered\r\n", peer->name);
return PARSE_REGISTER_UPDATE;
}
/* Store whatever we got as a contact from the client */
ast_string_field_set(peer, fullcontact, curi);
/* For the 200 OK, we should use the received contact */
ast_string_field_build(pvt, our_contact, "<%s>", curi);
/* Make sure it's a SIP URL */
/*! \todo This code assumes that the Contact is using the same transport as the
REGISTER request. That might not be true at all. You can receive
sips: requests over any transport. Needs to be fixed.
Does not parse the ;transport uri parameter at this point, which might be handy
in some situations.
*/
if (pvt->socket.type == SIP_TRANSPORT_TLS) {
if (parse_uri(curi, "sips:", &curi, NULL, &host, &pt, NULL)) {
if (parse_uri(curi2, "sip:", &curi, NULL, &host, &pt, NULL))
ast_log(LOG_NOTICE, "Not a valid SIP contact (missing sip:) trying to use anyway\n");
}
port = !ast_strlen_zero(pt) ? atoi(pt) : STANDARD_TLS_PORT;
} else {
if (parse_uri(curi, "sip:", &curi, NULL, &host, &pt, NULL))
ast_log(LOG_NOTICE, "Not a valid SIP contact (missing sip:) trying to use anyway\n");
port = !ast_strlen_zero(pt) ? atoi(pt) : STANDARD_SIP_PORT;
}
oldsin = peer->addr;
/* If we were already linked into the peers_by_ip container unlink ourselves so nobody can find us */
if (peer->addr.sin_addr.s_addr) {
ao2_t_unlink(peers_by_ip, peer, "ao2_unlink of peer from peers_by_ip table");
}
/* Check that they're allowed to register at this IP */
/* XXX This could block for a long time XXX */
/*! \todo Check NAPTR/SRV if we have not got a port in the URI */
hp = ast_gethostbyname(host, &ahp);
if (!hp) {
ast_log(LOG_WARNING, "Invalid host '%s'\n", host);
ast_string_field_set(peer, fullcontact, "");
ast_string_field_set(pvt, our_contact, "");
return PARSE_REGISTER_FAILED;
}
memcpy(&testsin.sin_addr, hp->h_addr, sizeof(testsin.sin_addr));
if (ast_apply_ha(global_contact_ha, &testsin) != AST_SENSE_ALLOW ||
ast_apply_ha(peer->contactha, &testsin) != AST_SENSE_ALLOW) {
ast_log(LOG_WARNING, "Host '%s' disallowed by rule\n", host);
ast_string_field_set(peer, fullcontact, "");
ast_string_field_set(pvt, our_contact, "");
return PARSE_REGISTER_FAILED;
}
/*! \todo This could come before the checking of DNS earlier on, to avoid
DNS lookups where we don't need it... */
if (!ast_test_flag(&peer->flags[0], SIP_NAT_ROUTE)) {
peer->addr.sin_family = AF_INET;
memcpy(&peer->addr.sin_addr, hp->h_addr, sizeof(peer->addr.sin_addr));
peer->addr.sin_port = htons(port);
} else {
/* Don't trust the contact field. Just use what they came to us
with */
peer->addr = pvt->recv;
}
/* Now that our address has been updated put ourselves back into the container for lookups */
ao2_t_link(peers_by_ip, peer, "ao2_link into peers_by_ip table");
/* Save SIP options profile */
peer->sipoptions = pvt->sipoptions;
if (!ast_strlen_zero(curi) && ast_strlen_zero(peer->username))
ast_string_field_set(peer, username, curi);
AST_SCHED_DEL_UNREF(sched, peer->expire,
unref_peer(peer, "remove register expire ref"));
if (expire > max_expiry)
expire = max_expiry;
if (expire < min_expiry)
expire = min_expiry;
if (peer->is_realtime && !ast_test_flag(&peer->flags[1], SIP_PAGE2_RTCACHEFRIENDS)) {
peer->expire = -1;
} else {
peer->expire = ast_sched_add(sched, (expire + 10) * 1000, expire_register,
ref_peer(peer, "add registration ref"));
if (peer->expire == -1) {
unref_peer(peer, "remote registration ref");
}
}
pvt->expiry = expire;
snprintf(data, sizeof(data), "%s:%d:%d:%s:%s", ast_inet_ntoa(peer->addr.sin_addr), ntohs(peer->addr.sin_port), expire, peer->username, peer->fullcontact);
/* Saving TCP connections is useless, we won't be able to reconnect
XXX WHY???? XXX
\todo Fix this immediately.
*/
if (!peer->rt_fromcontact && (peer->socket.type & SIP_TRANSPORT_UDP))
ast_db_put("SIP/Registry", peer->name, data);
manager_event(EVENT_FLAG_SYSTEM, "PeerStatus", "ChannelType: SIP\r\nPeer: SIP/%s\r\nPeerStatus: Registered\r\nAddress: %s\r\nPort: %d\r\n", peer->name, ast_inet_ntoa(peer->addr.sin_addr), ntohs(peer->addr.sin_port));
/* Is this a new IP address for us? */
if (VERBOSITY_ATLEAST(2) && inaddrcmp(&peer->addr, &oldsin)) {
ast_verbose(VERBOSE_PREFIX_3 "Registered SIP '%s' at %s port %d\n", peer->name, ast_inet_ntoa(peer->addr.sin_addr), ntohs(peer->addr.sin_port));
}
sip_poke_peer(peer, 0);
register_peer_exten(peer, 1);
/* Save User agent */
useragent = get_header(req, "User-Agent");
if (strcasecmp(useragent, peer->useragent)) {
ast_string_field_set(peer, useragent, useragent);
ast_verb(4, "Saved useragent \"%s\" for peer %s\n", peer->useragent, peer->name);
}
return PARSE_REGISTER_UPDATE;
}
/*! \brief Remove route from route list */
static void free_old_route(struct sip_route *route)
{
struct sip_route *next;
while (route) {
next = route->next;
ast_free(route);
route = next;
}
}
/*! \brief List all routes - mostly for debugging */
static void list_route(struct sip_route *route)
{
if (!route)
ast_verbose("list_route: no route\n");
else {
for (;route; route = route->next)
ast_verbose("list_route: hop: <%s>\n", route->hop);
}
}
/*! \brief Build route list from Record-Route header */
static void build_route(struct sip_pvt *p, struct sip_request *req, int backwards)
{
struct sip_route *thishop, *head, *tail;
int start = 0;
int len;
const char *rr, *contact, *c;
/* Once a persistant route is set, don't fool with it */
if (p->route && p->route_persistant) {
ast_debug(1, "build_route: Retaining previous route: <%s>\n", p->route->hop);
return;
}
if (p->route) {
free_old_route(p->route);
p->route = NULL;
}
/* We only want to create the route set the first time this is called */
p->route_persistant = 1;
/* Build a tailq, then assign it to p->route when done.
* If backwards, we add entries from the head so they end up
* in reverse order. However, we do need to maintain a correct
* tail pointer because the contact is always at the end.
*/
head = NULL;
tail = head;
/* 1st we pass through all the hops in any Record-Route headers */
for (;;) {
/* Each Record-Route header */
rr = __get_header(req, "Record-Route", &start);
if (*rr == '\0')
break;
for (; (rr = strchr(rr, '<')) ; rr += len) { /* Each route entry */
++rr;
len = strcspn(rr, ">") + 1;
/* Make a struct route */
if ((thishop = ast_malloc(sizeof(*thishop) + len))) {
/* ast_calloc is not needed because all fields are initialized in this block */
ast_copy_string(thishop->hop, rr, len);
ast_debug(2, "build_route: Record-Route hop: <%s>\n", thishop->hop);
/* Link in */
if (backwards) {
/* Link in at head so they end up in reverse order */
thishop->next = head;
head = thishop;
/* If this was the first then it'll be the tail */
if (!tail)
tail = thishop;
} else {
thishop->next = NULL;
/* Link in at the end */
if (tail)
tail->next = thishop;
else
head = thishop;
tail = thishop;
}
}
}
}
/* Only append the contact if we are dealing with a strict router */
if (!head || (!ast_strlen_zero(head->hop) && strstr(head->hop, ";lr") == NULL) ) {
/* 2nd append the Contact: if there is one */
/* Can be multiple Contact headers, comma separated values - we just take the first */
contact = get_header(req, "Contact");
if (!ast_strlen_zero(contact)) {
ast_debug(2, "build_route: Contact hop: %s\n", contact);
/* Look for <: delimited address */
c = strchr(contact, '<');
if (c) {
/* Take to > */
++c;
len = strcspn(c, ">") + 1;
} else {
/* No <> - just take the lot */
c = contact;
len = strlen(contact) + 1;
}
if ((thishop = ast_malloc(sizeof(*thishop) + len))) {
/* ast_calloc is not needed because all fields are initialized in this block */
ast_copy_string(thishop->hop, c, len);
thishop->next = NULL;
/* Goes at the end */
if (tail)
tail->next = thishop;
else
head = thishop;
}
}
}
/* Store as new route */
p->route = head;
/* For debugging dump what we ended up with */
if (sip_debug_test_pvt(p))
list_route(p->route);
}
AST_THREADSTORAGE(check_auth_buf);
#define CHECK_AUTH_BUF_INITLEN 256
/*! \brief Check user authorization from peer definition
Some actions, like REGISTER and INVITEs from peers require
authentication (if peer have secret set)
\return 0 on success, non-zero on error
*/
static enum check_auth_result check_auth(struct sip_pvt *p, struct sip_request *req, const char *username,
const char *secret, const char *md5secret, int sipmethod,
char *uri, enum xmittype reliable, int ignore)
{
const char *response;
char *reqheader, *respheader;
const char *authtoken;
char a1_hash[256];
char resp_hash[256]="";
char *c;
int wrongnonce = FALSE;
int good_response;
const char *usednonce = p->randdata;
struct ast_str *buf;
int res;
/* table of recognised keywords, and their value in the digest */
enum keys { K_RESP, K_URI, K_USER, K_NONCE, K_LAST };
struct x {
const char *key;
const char *s;
} *i, keys[] = {
[K_RESP] = { "response=", "" },
[K_URI] = { "uri=", "" },
[K_USER] = { "username=", "" },
[K_NONCE] = { "nonce=", "" },
[K_LAST] = { NULL, NULL}
};
/* Always OK if no secret */
if (ast_strlen_zero(secret) && ast_strlen_zero(md5secret))
return AUTH_SUCCESSFUL;
/* Always auth with WWW-auth since we're NOT a proxy */
/* Using proxy-auth in a B2BUA may block proxy authorization in the same transaction */
response = "401 Unauthorized";
/*
* Note the apparent swap of arguments below, compared to other
* usages of auth_headers().
*/
auth_headers(WWW_AUTH, &respheader, &reqheader);
authtoken = get_header(req, reqheader);
if (ignore && !ast_strlen_zero(p->randdata) && ast_strlen_zero(authtoken)) {
/* This is a retransmitted invite/register/etc, don't reconstruct authentication
information */
if (!reliable) {
/* Resend message if this was NOT a reliable delivery. Otherwise the
retransmission should get it */
transmit_response_with_auth(p, response, req, p->randdata, reliable, respheader, 0);
/* Schedule auto destroy in 32 seconds (according to RFC 3261) */
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
}
return AUTH_CHALLENGE_SENT;
} else if (ast_strlen_zero(p->randdata) || ast_strlen_zero(authtoken)) {
/* We have no auth, so issue challenge and request authentication */
ast_string_field_build(p, randdata, "%08lx", ast_random()); /* Create nonce for challenge */
transmit_response_with_auth(p, response, req, p->randdata, reliable, respheader, 0);
/* Schedule auto destroy in 32 seconds */
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
return AUTH_CHALLENGE_SENT;
}
/* --- We have auth, so check it */
/* Whoever came up with the authentication section of SIP can suck my %&#$&* for not putting
an example in the spec of just what it is you're doing a hash on. */
if (!(buf = ast_str_thread_get(&check_auth_buf, CHECK_AUTH_BUF_INITLEN)))
return AUTH_SECRET_FAILED; /*! XXX \todo need a better return code here */
/* Make a copy of the response and parse it */
res = ast_str_set(&buf, 0, "%s", authtoken);
if (res == AST_DYNSTR_BUILD_FAILED)
return AUTH_SECRET_FAILED; /*! XXX \todo need a better return code here */
c = buf->str;
while(c && *(c = ast_skip_blanks(c)) ) { /* lookup for keys */
for (i = keys; i->key != NULL; i++) {
const char *separator = ","; /* default */
if (strncasecmp(c, i->key, strlen(i->key)) != 0)
continue;
/* Found. Skip keyword, take text in quotes or up to the separator. */
c += strlen(i->key);
if (*c == '"') { /* in quotes. Skip first and look for last */
c++;
separator = "\"";
}
i->s = c;
strsep(&c, separator);
break;
}
if (i->key == NULL) /* not found, jump after space or comma */
strsep(&c, " ,");
}
/* Verify that digest username matches the username we auth as */
if (strcmp(username, keys[K_USER].s)) {
ast_log(LOG_WARNING, "username mismatch, have <%s>, digest has <%s>\n",
username, keys[K_USER].s);
/* Oops, we're trying something here */
return AUTH_USERNAME_MISMATCH;
}
/* Verify nonce from request matches our nonce. If not, send 401 with new nonce */
if (strcasecmp(p->randdata, keys[K_NONCE].s)) { /* XXX it was 'n'casecmp ? */
wrongnonce = TRUE;
usednonce = keys[K_NONCE].s;
}
if (!ast_strlen_zero(md5secret))
ast_copy_string(a1_hash, md5secret, sizeof(a1_hash));
else {
char a1[256];
snprintf(a1, sizeof(a1), "%s:%s:%s", username, sip_cfg.realm, secret);
ast_md5_hash(a1_hash, a1);
}
/* compute the expected response to compare with what we received */
{
char a2[256];
char a2_hash[256];
char resp[256];
snprintf(a2, sizeof(a2), "%s:%s", sip_methods[sipmethod].text,
S_OR(keys[K_URI].s, uri));
ast_md5_hash(a2_hash, a2);
snprintf(resp, sizeof(resp), "%s:%s:%s", a1_hash, usednonce, a2_hash);
ast_md5_hash(resp_hash, resp);
}
good_response = keys[K_RESP].s &&
!strncasecmp(keys[K_RESP].s, resp_hash, strlen(resp_hash));
if (wrongnonce) {
if (good_response) {
if (sipdebug)
ast_log(LOG_NOTICE, "Correct auth, but based on stale nonce received from '%s'\n", get_header(req, "To"));
/* We got working auth token, based on stale nonce . */
ast_string_field_build(p, randdata, "%08lx", ast_random());
transmit_response_with_auth(p, response, req, p->randdata, reliable, respheader, TRUE);
} else {
/* Everything was wrong, so give the device one more try with a new challenge */
if (!req->ignore) {
if (sipdebug)
ast_log(LOG_NOTICE, "Bad authentication received from '%s'\n", get_header(req, "To"));
ast_string_field_build(p, randdata, "%08lx", ast_random());
} else {
if (sipdebug)
ast_log(LOG_NOTICE, "Duplicate authentication received from '%s'\n", get_header(req, "To"));
}
transmit_response_with_auth(p, response, req, p->randdata, reliable, respheader, FALSE);
}
/* Schedule auto destroy in 32 seconds */
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
return AUTH_CHALLENGE_SENT;
}
if (good_response) {
append_history(p, "AuthOK", "Auth challenge succesful for %s", username);
return AUTH_SUCCESSFUL;
}
/* Ok, we have a bad username/secret pair */
/* Tell the UAS not to re-send this authentication data, because
it will continue to fail
*/
return AUTH_SECRET_FAILED;
}
/*! \brief Change onhold state of a peer using a pvt structure */
static void sip_peer_hold(struct sip_pvt *p, int hold)
{
struct sip_peer *peer = find_peer(p->peername, NULL, 1, FINDALLDEVICES, FALSE);
if (!peer)
return;
/* If they put someone on hold, increment the value... otherwise decrement it */
ast_atomic_fetchadd_int(&peer->onHold, (hold ? +1 : -1));
/* Request device state update */
ast_devstate_changed(AST_DEVICE_UNKNOWN, "SIP/%s", peer->name);
unref_peer(peer, "sip_peer_hold: from find_peer operation");
return;
}
/*! \brief Receive MWI events that we have subscribed to */
static void mwi_event_cb(const struct ast_event *event, void *userdata)
{
struct sip_peer *peer = userdata;
ao2_lock(peer);
sip_send_mwi_to_peer(peer, event, 0);
ao2_unlock(peer);
}
/*! \brief Callback for the devicestate notification (SUBSCRIBE) support subsystem
\note If you add an "hint" priority to the extension in the dial plan,
you will get notifications on device state changes */
static int cb_extensionstate(char *context, char* exten, int state, void *data)
{
struct sip_pvt *p = data;
sip_pvt_lock(p);
switch(state) {
case AST_EXTENSION_DEACTIVATED: /* Retry after a while */
case AST_EXTENSION_REMOVED: /* Extension is gone */
if (p->autokillid > -1 && sip_cancel_destroy(p)) /* Remove subscription expiry for renewals */
ast_log(LOG_WARNING, "Unable to cancel SIP destruction. Expect bad things.\n");
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT); /* Delete subscription in 32 secs */
ast_verb(2, "Extension state: Watcher for hint %s %s. Notify User %s\n", exten, state == AST_EXTENSION_DEACTIVATED ? "deactivated" : "removed", p->username);
p->stateid = -1;
p->subscribed = NONE;
append_history(p, "Subscribestatus", "%s", state == AST_EXTENSION_REMOVED ? "HintRemoved" : "Deactivated");
break;
default: /* Tell user */
p->laststate = state;
break;
}
if (p->subscribed != NONE) { /* Only send state NOTIFY if we know the format */
if (!p->pendinginvite) {
transmit_state_notify(p, state, 1, FALSE);
} else {
/* We already have a NOTIFY sent that is not answered. Queue the state up.
if many state changes happen meanwhile, we will only send a notification of the last one */
ast_set_flag(&p->flags[1], SIP_PAGE2_STATECHANGEQUEUE);
}
}
ast_verb(2, "Extension Changed %s[%s] new state %s for Notify User %s %s\n", exten, context, ast_extension_state2str(state), p->username,
ast_test_flag(&p->flags[1], SIP_PAGE2_STATECHANGEQUEUE) ? "(queued)" : "");
sip_pvt_unlock(p);
return 0;
}
/*! \brief Send a fake 401 Unauthorized response when the administrator
wants to hide the names of local devices from fishers
*/
static void transmit_fake_auth_response(struct sip_pvt *p, int sipmethod, struct sip_request *req, enum xmittype reliable)
{
/* We have to emulate EXACTLY what we'd get with a good peer
* and a bad password, or else we leak information. */
const char *response = "407 Proxy Authentication Required";
const char *reqheader = "Proxy-Authorization";
const char *respheader = "Proxy-Authenticate";
const char *authtoken;
struct ast_str *buf;
char *c;
/* table of recognised keywords, and their value in the digest */
enum keys { K_NONCE, K_LAST };
struct x {
const char *key;
const char *s;
} *i, keys[] = {
[K_NONCE] = { "nonce=", "" },
[K_LAST] = { NULL, NULL}
};
if (sipmethod == SIP_REGISTER || sipmethod == SIP_SUBSCRIBE) {
response = "401 Unauthorized";
reqheader = "Authorization";
respheader = "WWW-Authenticate";
}
authtoken = get_header(req, reqheader);
if (req->ignore && !ast_strlen_zero(p->randdata) && ast_strlen_zero(authtoken)) {
/* This is a retransmitted invite/register/etc, don't reconstruct authentication
* information */
transmit_response_with_auth(p, response, req, p->randdata, 0, respheader, 0);
/* Schedule auto destroy in 32 seconds (according to RFC 3261) */
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
return;
} else if (ast_strlen_zero(p->randdata) || ast_strlen_zero(authtoken)) {
/* We have no auth, so issue challenge and request authentication */
ast_string_field_build(p, randdata, "%08lx", ast_random()); /* Create nonce for challenge */
transmit_response_with_auth(p, response, req, p->randdata, 0, respheader, 0);
/* Schedule auto destroy in 32 seconds */
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
return;
}
if (!(buf = ast_str_thread_get(&check_auth_buf, CHECK_AUTH_BUF_INITLEN))) {
transmit_response(p, "403 Forbidden (Bad auth)", &p->initreq);
return;
}
/* Make a copy of the response and parse it */
if (ast_str_set(&buf, 0, "%s", authtoken) == AST_DYNSTR_BUILD_FAILED) {
transmit_response(p, "403 Forbidden (Bad auth)", &p->initreq);
return;
}
c = buf->str;
while (c && *(c = ast_skip_blanks(c))) { /* lookup for keys */
for (i = keys; i->key != NULL; i++) {
const char *separator = ","; /* default */
if (strncasecmp(c, i->key, strlen(i->key)) != 0) {
continue;
}
/* Found. Skip keyword, take text in quotes or up to the separator. */
c += strlen(i->key);
if (*c == '"') { /* in quotes. Skip first and look for last */
c++;
separator = "\"";
}
i->s = c;
strsep(&c, separator);
break;
}
if (i->key == NULL) { /* not found, jump after space or comma */
strsep(&c, " ,");
}
}
/* Verify nonce from request matches our nonce. If not, send 401 with new nonce */
if (strcasecmp(p->randdata, keys[K_NONCE].s)) {
if (!req->ignore) {
ast_string_field_build(p, randdata, "%08lx", ast_random());
}
transmit_response_with_auth(p, response, req, p->randdata, reliable, respheader, FALSE);
/* Schedule auto destroy in 32 seconds */
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
} else {
transmit_response(p, "403 Forbidden (Bad auth)", &p->initreq);
}
}
/*!
* Terminate the uri at the first ';' or space.
* Technically we should ignore escaped space per RFC3261 (19.1.1 etc)
* but don't do it for the time being. Remember the uri format is:
* (User-parameters was added after RFC 3261)
*\verbatim
*
* sip:user:password;user-parameters@host:port;uri-parameters?headers
* sips:user:password;user-parameters@host:port;uri-parameters?headers
*
*\endverbatim
* \todo As this function does not support user-parameters, it's considered broken
* and needs fixing.
*/
static char *terminate_uri(char *uri)
{
char *t = uri;
while (*t && *t > ' ' && *t != ';')
t++;
*t = '\0';
return uri;
}
/*! \brief Verify registration of user
- Registration is done in several steps, first a REGISTER without auth
to get a challenge (nonce) then a second one with auth
- Registration requests are only matched with peers that are marked as "dynamic"
*/
static enum check_auth_result register_verify(struct sip_pvt *p, struct sockaddr_in *sin,
struct sip_request *req, char *uri)
{
enum check_auth_result res = AUTH_NOT_FOUND;
struct sip_peer *peer;
char tmp[256];
char *name, *c;
char *domain;
terminate_uri(uri); /* warning, overwrite the string */
ast_copy_string(tmp, get_header(req, "To"), sizeof(tmp));
if (sip_cfg.pedanticsipchecking)
ast_uri_decode(tmp);
c = get_in_brackets(tmp);
c = remove_uri_parameters(c);
if (!strncasecmp(c, "sip:", 4)) {
name = c + 4;
} else if (!strncasecmp(c, "sips:", 5)) {
name = c + 5;
} else {
name = c;
ast_log(LOG_NOTICE, "Invalid to address: '%s' from %s (missing sip:) trying to use anyway...\n", c, ast_inet_ntoa(sin->sin_addr));
}
/*! \todo XXX here too we interpret a missing @domain as a name-only
* URI, whereas the RFC says this is a domain-only uri.
*/
/* Strip off the domain name */
if ((c = strchr(name, '@'))) {
*c++ = '\0';
domain = c;
if ((c = strchr(domain, ':'))) /* Remove :port */
*c = '\0';
if (!AST_LIST_EMPTY(&domain_list)) {
if (!check_sip_domain(domain, NULL, 0)) {
transmit_response(p, "404 Not found (unknown domain)", &p->initreq);
return AUTH_UNKNOWN_DOMAIN;
}
}
}
c = strchr(name, ';'); /* Remove any Username parameters */
if (c)
*c = '\0';
ast_string_field_set(p, exten, name);
build_contact(p);
peer = find_peer(name, NULL, TRUE, FINDPEERS, FALSE);
if (!(peer && ast_apply_ha(peer->ha, sin))) {
/* Peer fails ACL check */
if (peer) {
unref_peer(peer, "register_verify: unref_peer: from find_peer operation");
peer = NULL;
res = AUTH_ACL_FAILED;
} else
res = AUTH_NOT_FOUND;
}
if (peer) {
if (!peer->host_dynamic) {
ast_log(LOG_ERROR, "Peer '%s' is trying to register, but not configured as host=dynamic\n", peer->name);
res = AUTH_PEER_NOT_DYNAMIC;
} else {
ast_copy_flags(&p->flags[0], &peer->flags[0], SIP_NAT);
if (ast_test_flag(&p->flags[1], SIP_PAGE2_REGISTERTRYING))
transmit_response(p, "100 Trying", req);
if (!(res = check_auth(p, req, peer->name, peer->secret, peer->md5secret, SIP_REGISTER, uri, XMIT_UNRELIABLE, req->ignore))) {
if (sip_cancel_destroy(p))
ast_log(LOG_WARNING, "Unable to cancel SIP destruction. Expect bad things.\n");
if (check_request_transport(peer, req)) {
ast_set_flag(&p->flags[0], SIP_PENDINGBYE);
transmit_response_with_date(p, "403 Forbidden", req);
res = AUTH_BAD_TRANSPORT;
} else {
/* We have a successful registration attempt with proper authentication,
now, update the peer */
switch (parse_register_contact(p, peer, req)) {
case PARSE_REGISTER_FAILED:
ast_log(LOG_WARNING, "Failed to parse contact info\n");
transmit_response_with_date(p, "400 Bad Request", req);
peer->lastmsgssent = -1;
res = 0;
break;
case PARSE_REGISTER_QUERY:
transmit_response_with_date(p, "200 OK", req);
peer->lastmsgssent = -1;
res = 0;
break;
case PARSE_REGISTER_UPDATE:
update_peer(peer, p->expiry);
/* Say OK and ask subsystem to retransmit msg counter */
transmit_response_with_date(p, "200 OK", req);
if (!ast_test_flag((&peer->flags[1]), SIP_PAGE2_SUBSCRIBEMWIONLY))
peer->lastmsgssent = -1;
res = 0;
break;
}
}
}
}
}
if (!peer && sip_cfg.autocreatepeer) {
/* Create peer if we have autocreate mode enabled */
peer = temp_peer(name);
if (peer) {
ao2_t_link(peers, peer, "link peer into peer table");
if (peer->addr.sin_addr.s_addr) {
ao2_t_link(peers_by_ip, peer, "link peer into peers-by-ip table");
}
if (sip_cancel_destroy(p))
ast_log(LOG_WARNING, "Unable to cancel SIP destruction. Expect bad things.\n");
switch (parse_register_contact(p, peer, req)) {
case PARSE_REGISTER_FAILED:
ast_log(LOG_WARNING, "Failed to parse contact info\n");
transmit_response_with_date(p, "400 Bad Request", req);
peer->lastmsgssent = -1;
res = 0;
break;
case PARSE_REGISTER_QUERY:
transmit_response_with_date(p, "200 OK", req);
peer->lastmsgssent = -1;
res = 0;
break;
case PARSE_REGISTER_UPDATE:
/* Say OK and ask subsystem to retransmit msg counter */
transmit_response_with_date(p, "200 OK", req);
manager_event(EVENT_FLAG_SYSTEM, "PeerStatus", "ChannelType: SIP\r\nPeer: SIP/%s\r\nPeerStatus: Registered\r\nAddress: %s\r\nPort: %d\r\n", peer->name, ast_inet_ntoa(sin->sin_addr), ntohs(sin->sin_port));
peer->lastmsgssent = -1;
res = 0;
break;
}
}
}
if (!peer && sip_cfg.alwaysauthreject) {
/* If we found a peer, we transmit a 100 Trying. Therefore, if we're
* trying to avoid leaking information, we MUST also transmit the same
* response when we DON'T find a peer. */
transmit_response(p, "100 Trying", req);
/* Insert a fake delay between the 100 and the subsequent failure. */
sched_yield();
}
if (!res) {
ast_devstate_changed(AST_DEVICE_UNKNOWN, "SIP/%s", peer->name);
}
if (res < 0) {
switch (res) {
case AUTH_SECRET_FAILED:
/* Wrong password in authentication. Go away, don't try again until you fixed it */
transmit_response(p, "403 Forbidden (Bad auth)", &p->initreq);
if (global_authfailureevents)
manager_event(EVENT_FLAG_SYSTEM, "PeerStatus", "ChannelType: SIP\r\nPeer: SIP/%s\r\nPeerStatus: Rejected\r\nCause: AUTH_SECRET_FAILED\r\nAddress: %s\r\nPort: %d\r\n",
name, ast_inet_ntoa(sin->sin_addr), ntohs(sin->sin_port));
break;
case AUTH_USERNAME_MISMATCH:
/* Username and digest username does not match.
Asterisk uses the From: username for authentication. We need the
devices to use the same authentication user name until we support
proper authentication by digest auth name */
transmit_response(p, "403 Authentication user name does not match account name", &p->initreq);
if (global_authfailureevents)
manager_event(EVENT_FLAG_SYSTEM, "PeerStatus", "ChannelType: SIP\r\nPeer: SIP/%s\r\nPeerStatus: Rejected\r\nCause: AUTH_USERNAME_MISMATCH\r\nAddress: %s\r\nPort: %d\r\n",
name, ast_inet_ntoa(sin->sin_addr), ntohs(sin->sin_port));
break;
case AUTH_NOT_FOUND:
case AUTH_PEER_NOT_DYNAMIC:
case AUTH_ACL_FAILED:
if (sip_cfg.alwaysauthreject) {
transmit_fake_auth_response(p, SIP_REGISTER, &p->initreq, XMIT_UNRELIABLE);
if (global_authfailureevents) {
manager_event(EVENT_FLAG_SYSTEM, "PeerStatus", "ChannelType: SIP\r\nPeer: SIP/%s\r\nPeerStatus: Rejected\r\nCause: %s\r\nAddress: %s\r\nPort: %d\r\n",
name, res == AUTH_PEER_NOT_DYNAMIC ? "AUTH_PEER_NOT_DYNAMIC" : "URI_NOT_FOUND",
ast_inet_ntoa(sin->sin_addr), ntohs(sin->sin_port));
}
} else {
/* URI not found */
if (res == AUTH_PEER_NOT_DYNAMIC) {
transmit_response(p, "403 Forbidden", &p->initreq);
if (global_authfailureevents)
manager_event(EVENT_FLAG_SYSTEM, "PeerStatus", "ChannelType: SIP\r\nPeer: SIP/%s\r\nPeerStatus: Rejected\r\nCause: AUTH_PEER_NOT_DYNAMIC\r\nAddress: %s\r\nPort: %d\r\n",
name, ast_inet_ntoa(sin->sin_addr), ntohs(sin->sin_port));
}
else
transmit_response(p, "404 Not found", &p->initreq);
if (global_authfailureevents)
manager_event(EVENT_FLAG_SYSTEM, "PeerStatus", "ChannelType: SIP\r\nPeer: SIP/%s\r\nPeerStatus: Rejected\r\nCause: URI_NOT_FOUND\r\nAddress: %s\r\nPort: %d\r\n",
name, ast_inet_ntoa(sin->sin_addr), ntohs(sin->sin_port));
}
break;
case AUTH_BAD_TRANSPORT:
default:
break;
}
}
if (peer)
unref_peer(peer, "register_verify: unref_peer: tossing stack peer pointer at end of func");
return res;
}
/*! \brief Translate referring cause */
static void sip_set_redirstr(struct sip_pvt *p, char *reason) {
if (!strcmp(reason, "unknown")) {
ast_string_field_set(p, redircause, "UNKNOWN");
} else if (!strcmp(reason, "user-busy")) {
ast_string_field_set(p, redircause, "BUSY");
} else if (!strcmp(reason, "no-answer")) {
ast_string_field_set(p, redircause, "NOANSWER");
} else if (!strcmp(reason, "unavailable")) {
ast_string_field_set(p, redircause, "UNREACHABLE");
} else if (!strcmp(reason, "unconditional")) {
ast_string_field_set(p, redircause, "UNCONDITIONAL");
} else if (!strcmp(reason, "time-of-day")) {
ast_string_field_set(p, redircause, "UNKNOWN");
} else if (!strcmp(reason, "do-not-disturb")) {
ast_string_field_set(p, redircause, "UNKNOWN");
} else if (!strcmp(reason, "deflection")) {
ast_string_field_set(p, redircause, "UNKNOWN");
} else if (!strcmp(reason, "follow-me")) {
ast_string_field_set(p, redircause, "UNKNOWN");
} else if (!strcmp(reason, "out-of-service")) {
ast_string_field_set(p, redircause, "UNREACHABLE");
} else if (!strcmp(reason, "away")) {
ast_string_field_set(p, redircause, "UNREACHABLE");
} else {
ast_string_field_set(p, redircause, "UNKNOWN");
}
}
/*! \brief Parse the parts of the P-Asserted-Identity header
* on an incoming packet. Returns 1 if a valid header is found
* and it is different from the current caller id.
*/
static int get_pai(struct sip_pvt *p, struct sip_request *req)
{
char pai[256];
char privacy[64];
char *cid_num = "";
char *cid_name = "";
int callingpres = AST_PRES_ALLOWED_USER_NUMBER_NOT_SCREENED;
char *start = NULL, *end = NULL;
ast_copy_string(pai, get_header(req, "P-Asserted-Identity"), sizeof(pai));
if (ast_strlen_zero(pai)) {
return 0;
}
start = pai;
if (*start == '"') {
*start++ = '\0';
end = strchr(start, '"');
if (!end)
return 0;
*end++ = '\0';
cid_name = start;
start = ast_skip_blanks(end);
}
if (*start != '<')
return 0;
*start++ = '\0';
end = strchr(start, '@');
if (!end)
return 0;
*end++ = '\0';
if (!strncasecmp(start, "anonymous@anonymous.invalid", 27)) {
callingpres = AST_PRES_PROHIB_USER_NUMBER_NOT_SCREENED;
/*XXX Assume no change in cid_num. Perhaps it should be
* blanked?
*/
cid_num = (char *)p->cid_num;
} else if (!strncasecmp(start, "sip:", 4)) {
cid_num = start + 4;
if (ast_is_shrinkable_phonenumber(cid_num))
ast_shrink_phone_number(cid_num);
start = end;
end = strchr(start, '>');
if (!end)
return 0;
*end = '\0';
} else {
return 0;
}
ast_copy_string(privacy, get_header(req, "Privacy"), sizeof(privacy));
if (!ast_strlen_zero(privacy) && strncmp(privacy, "id", 2)) {
callingpres = AST_PRES_PROHIB_USER_NUMBER_NOT_SCREENED;
}
/* Only return true if the supplied caller id is different */
if (!strcasecmp(p->cid_num, cid_num) && !strcasecmp(p->cid_name, cid_name) && p->callingpres == callingpres)
return 0;
ast_string_field_set(p, cid_num, cid_num);
ast_string_field_set(p, cid_name, cid_name);
p->callingpres = callingpres;
if (p->owner) {
ast_set_callerid(p->owner, cid_num, cid_name, NULL);
p->owner->cid.cid_pres = callingpres;
}
return 1;
}
/*! \brief Get name, number and presentation from remote party id header,
* returns true if a valid header was found and it was different from the
* current caller id.
*/
static int get_rpid(struct sip_pvt *p, struct sip_request *oreq)
{
char tmp[256];
struct sip_request *req;
char *cid_num = "";
char *cid_name = "";
int callingpres = AST_PRES_ALLOWED_USER_NUMBER_NOT_SCREENED;
char *privacy = "";
char *screen = "";
char *start, *end;
if (!ast_test_flag(&p->flags[0], SIP_TRUSTRPID))
return 0;
req = oreq;
if (!req)
req = &p->initreq;
ast_copy_string(tmp, get_header(req, "Remote-Party-ID"), sizeof(tmp));
if (ast_strlen_zero(tmp)) {
return get_pai(p, req);
}
start = tmp;
if (*start == '"') {
*start++ = '\0';
end = strchr(start, '"');
if (!end)
return 0;
*end++ = '\0';
cid_name = start;
start = ast_skip_blanks(end);
}
if (*start != '<')
return 0;
*start++ = '\0';
end = strchr(start, '@');
if (!end)
return 0;
*end++ = '\0';
if (strncasecmp(start, "sip:", 4))
return 0;
cid_num = start + 4;
if (ast_is_shrinkable_phonenumber(cid_num))
ast_shrink_phone_number(cid_num);
start = end;
end = strchr(start, '>');
if (!end)
return 0;
*end++ = '\0';
if (*end) {
start = end;
if (*start != ';')
return 0;
*start++ = '\0';
while (!ast_strlen_zero(start)) {
end = strchr(start, ';');
if (end)
*end++ = '\0';
if (!strncasecmp(start, "privacy=", 8))
privacy = start + 8;
else if (!strncasecmp(start, "screen=", 7))
screen = start + 7;
start = end;
}
if (!strcasecmp(privacy, "full")) {
if (!strcasecmp(screen, "yes"))
callingpres = AST_PRES_PROHIB_USER_NUMBER_PASSED_SCREEN;
else if (!strcasecmp(screen, "no"))
callingpres = AST_PRES_PROHIB_USER_NUMBER_NOT_SCREENED;
} else {
if (!strcasecmp(screen, "yes"))
callingpres = AST_PRES_ALLOWED_USER_NUMBER_PASSED_SCREEN;
else if (!strcasecmp(screen, "no"))
callingpres = AST_PRES_ALLOWED_USER_NUMBER_NOT_SCREENED;
}
}
/* Only return true if the supplied caller id is different */
if (!strcasecmp(p->cid_num, cid_num) && !strcasecmp(p->cid_name, cid_name) && p->callingpres == callingpres)
return 0;
ast_string_field_set(p, cid_num, cid_num);
ast_string_field_set(p, cid_name, cid_name);
p->callingpres = callingpres;
if (p->owner) {
ast_set_callerid(p->owner, cid_num, cid_name, NULL);
p->owner->cid.cid_pres = callingpres;
}
return 1;
}
/*! \brief Get referring dnis */
static int get_rdnis(struct sip_pvt *p, struct sip_request *oreq, char **name, char **number, int *reason)
{
char tmp[256], *exten, *rexten, *rdomain, *rname = NULL;
char *params, *reason_param = NULL;
struct sip_request *req;
req = oreq ? oreq : &p->initreq;
ast_copy_string(tmp, get_header(req, "Diversion"), sizeof(tmp));
if (ast_strlen_zero(tmp))
return -1;
if ((params = strchr(tmp, '>'))) {
params = strchr(params, ';');
}
exten = get_in_brackets(tmp);
if (!strncasecmp(exten, "sip:", 4)) {
exten += 4;
} else if (!strncasecmp(exten, "sips:", 5)) {
exten += 5;
} else {
ast_log(LOG_WARNING, "Huh? Not an RDNIS SIP header (%s)?\n", exten);
return -1;
}
/* Get diversion-reason param if present */
if (params) {
*params = '\0'; /* Cut off parameters */
params++;
while (*params == ';' || *params == ' ')
params++;
/* Check if we have a reason parameter */
if ((reason_param = strcasestr(params, "reason="))) {
reason_param+=7;
/* Remove enclosing double-quotes */
if (*reason_param == '"')
ast_strip_quoted(reason_param, "\"", "\"");
if (!ast_strlen_zero(reason_param)) {
sip_set_redirstr(p, reason_param);
if (p->owner) {
pbx_builtin_setvar_helper(p->owner, "__PRIREDIRECTREASON", p->redircause);
pbx_builtin_setvar_helper(p->owner, "__SIPREDIRECTREASON", reason_param);
}
}
}
}
rdomain = exten;
rexten = strsep(&rdomain, "@"); /* trim anything after @ */
if (p->owner)
pbx_builtin_setvar_helper(p->owner, "__SIPRDNISDOMAIN", rdomain);
if (sip_debug_test_pvt(p))
ast_verbose("RDNIS for this call is %s (reason %s)\n", exten, reason ? reason_param : "");
/*ast_string_field_set(p, rdnis, rexten);*/
if (*tmp == '\"') {
char *end_quote;
rname = tmp + 1;
end_quote = strchr(rname, '\"');
*end_quote = '\0';
}
if (number) {
*number = ast_strdup(rexten);
}
if (name && rname) {
*name = ast_strdup(rname);
}
if (reason && !ast_strlen_zero(reason_param)) {
*reason = sip_reason_str_to_code(reason_param);
}
return 0;
}
/*! \brief Find out who the call is for.
We use the request uri as a destination.
This code assumes authentication has been done, so that the
device (peer/user) context is already set.
\return 0 on success (found a matching extension),
1 for pickup extension or overlap dialling support (if we support it),
-1 on error.
\note If the incoming uri is a SIPS: uri, we are required to carry this across
the dialplan, so that the outbound call also is a sips: call or encrypted
IAX2 call. If that's not available, the call should FAIL.
*/
static int get_destination(struct sip_pvt *p, struct sip_request *oreq)
{
char tmp[256] = "", *uri, *a;
char tmpf[256] = "", *from = NULL;
struct sip_request *req;
char *colon;
char *decoded_uri;
req = oreq;
if (!req)
req = &p->initreq;
/* Find the request URI */
if (req->rlPart2)
ast_copy_string(tmp, REQ_OFFSET_TO_STR(req, rlPart2), sizeof(tmp));
if (sip_cfg.pedanticsipchecking)
ast_uri_decode(tmp);
uri = get_in_brackets(tmp);
if (!strncasecmp(uri, "sip:", 4)) {
uri += 4;
} else if (!strncasecmp(uri, "sips:", 5)) {
uri += 5;
} else {
ast_log(LOG_WARNING, "Huh? Not a SIP header (%s)?\n", uri);
return -1;
}
/* Now find the From: caller ID and name */
/* XXX Why is this done in get_destination? Isn't it already done?
Needs to be checked
*/
ast_copy_string(tmpf, get_header(req, "From"), sizeof(tmpf));
if (!ast_strlen_zero(tmpf)) {
if (sip_cfg.pedanticsipchecking)
ast_uri_decode(tmpf);
from = get_in_brackets(tmpf);
}
if (!ast_strlen_zero(from)) {
if (!strncasecmp(from, "sip:", 4)) {
from += 4;
} else if (!strncasecmp(from, "sips:", 5)) {
from += 5;
} else {
ast_log(LOG_WARNING, "Huh? Not a SIP header (%s)?\n", from);
return -1;
}
if ((a = strchr(from, '@')))
*a++ = '\0';
else
a = from; /* just a domain */
from = strsep(&from, ";"); /* Remove userinfo options */
a = strsep(&a, ";"); /* Remove URI options */
ast_string_field_set(p, fromdomain, a);
}
/* Skip any options and find the domain */
/* Get the target domain */
if ((a = strchr(uri, '@'))) {
*a++ = '\0';
} else { /* No username part */
a = uri;
uri = "s"; /* Set extension to "s" */
}
colon = strchr(a, ':'); /* Remove :port */
if (colon)
*colon = '\0';
uri = strsep(&uri, ";"); /* Remove userinfo options */
a = strsep(&a, ";"); /* Remove URI options */
ast_string_field_set(p, domain, a);
if (!AST_LIST_EMPTY(&domain_list)) {
char domain_context[AST_MAX_EXTENSION];
domain_context[0] = '\0';
if (!check_sip_domain(p->domain, domain_context, sizeof(domain_context))) {
if (!sip_cfg.allow_external_domains && (req->method == SIP_INVITE || req->method == SIP_REFER)) {
ast_debug(1, "Got SIP %s to non-local domain '%s'; refusing request.\n", sip_methods[req->method].text, p->domain);
return -2;
}
}
/* If we have a context defined, overwrite the original context */
if (!ast_strlen_zero(domain_context))
ast_string_field_set(p, context, domain_context);
}
/* If the request coming in is a subscription and subscribecontext has been specified use it */
if (req->method == SIP_SUBSCRIBE && !ast_strlen_zero(p->subscribecontext))
ast_string_field_set(p, context, p->subscribecontext);
if (sip_debug_test_pvt(p))
ast_verbose("Looking for %s in %s (domain %s)\n", uri, p->context, p->domain);
/* If this is a subscription we actually just need to see if a hint exists for the extension */
if (req->method == SIP_SUBSCRIBE) {
char hint[AST_MAX_EXTENSION];
return (ast_get_hint(hint, sizeof(hint), NULL, 0, NULL, p->context, p->exten) ? 0 : -1);
} else {
decoded_uri = ast_strdupa(uri);
ast_uri_decode(decoded_uri);
/* Check the dialplan for the username part of the request URI,
the domain will be stored in the SIPDOMAIN variable
Since extensions.conf can have unescaped characters, try matching a decoded
uri in addition to the non-decoded uri
Return 0 if we have a matching extension */
if (ast_exists_extension(NULL, p->context, uri, 1, S_OR(p->cid_num, from)) || ast_exists_extension(NULL, p->context, decoded_uri, 1, S_OR(p->cid_num, from)) ||
!strcmp(decoded_uri, ast_pickup_ext())) {
if (!oreq)
ast_string_field_set(p, exten, decoded_uri);
return 0;
}
}
/* Return 1 for pickup extension or overlap dialling support (if we support it) */
if((ast_test_flag(&global_flags[1], SIP_PAGE2_ALLOWOVERLAP) &&
ast_canmatch_extension(NULL, p->context, decoded_uri, 1, S_OR(p->cid_num, from))) ||
!strncmp(decoded_uri, ast_pickup_ext(), strlen(decoded_uri))) {
return 1;
}
return -1;
}
/*! \brief Lock dialog lock and find matching pvt lock
\return a reference, remember to release it when done
*/
static struct sip_pvt *get_sip_pvt_byid_locked(const char *callid, const char *totag, const char *fromtag)
{
struct sip_pvt *sip_pvt_ptr;
struct sip_pvt tmp_dialog = {
.callid = callid,
};
if (totag)
ast_debug(4, "Looking for callid %s (fromtag %s totag %s)\n", callid, fromtag ? fromtag : "<no fromtag>", totag ? totag : "<no totag>");
/* Search dialogs and find the match */
sip_pvt_ptr = ao2_t_find(dialogs, &tmp_dialog, OBJ_POINTER, "ao2_find of dialog in dialogs table");
if (sip_pvt_ptr) {
/* Go ahead and lock it (and its owner) before returning */
sip_pvt_lock(sip_pvt_ptr);
if (sip_cfg.pedanticsipchecking) {
unsigned char frommismatch = 0, tomismatch = 0;
if (ast_strlen_zero(fromtag)) {
sip_pvt_unlock(sip_pvt_ptr);
ast_debug(4, "Matched %s call for callid=%s - no from tag specified, pedantic check fails\n",
sip_pvt_ptr->outgoing_call == TRUE ? "OUTGOING": "INCOMING", sip_pvt_ptr->callid);
return NULL;
}
if (ast_strlen_zero(totag)) {
sip_pvt_unlock(sip_pvt_ptr);
ast_debug(4, "Matched %s call for callid=%s - no to tag specified, pedantic check fails\n",
sip_pvt_ptr->outgoing_call == TRUE ? "OUTGOING": "INCOMING", sip_pvt_ptr->callid);
return NULL;
}
/* RFC 3891
* > 3. User Agent Server Behavior: Receiving a Replaces Header
* > The Replaces header contains information used to match an existing
* > SIP dialog (call-id, to-tag, and from-tag). Upon receiving an INVITE
* > with a Replaces header, the User Agent (UA) attempts to match this
* > information with a confirmed or early dialog. The User Agent Server
* > (UAS) matches the to-tag and from-tag parameters as if they were tags
* > present in an incoming request. In other words, the to-tag parameter
* > is compared to the local tag, and the from-tag parameter is compared
* > to the remote tag.
*
* Thus, the totag is always compared to the local tag, regardless if
* this our call is an incoming or outgoing call.
*/
frommismatch = !!strcmp(fromtag, sip_pvt_ptr->theirtag);
tomismatch = !!strcmp(totag, sip_pvt_ptr->tag);
if (frommismatch || tomismatch) {
sip_pvt_unlock(sip_pvt_ptr);
if (frommismatch) {
ast_debug(4, "Matched %s call for callid=%s - pedantic from tag check fails; their tag is %s our tag is %s\n",
sip_pvt_ptr->outgoing_call == TRUE ? "OUTGOING": "INCOMING", sip_pvt_ptr->callid,
fromtag, sip_pvt_ptr->theirtag);
}
if (tomismatch) {
ast_debug(4, "Matched %s call for callid=%s - pedantic to tag check fails; their tag is %s our tag is %s\n",
sip_pvt_ptr->outgoing_call == TRUE ? "OUTGOING": "INCOMING", sip_pvt_ptr->callid,
totag, sip_pvt_ptr->tag);
}
return NULL;
}
}
if (totag)
ast_debug(4, "Matched %s call - their tag is %s Our tag is %s\n",
sip_pvt_ptr->outgoing_call == TRUE ? "OUTGOING": "INCOMING",
sip_pvt_ptr->theirtag, sip_pvt_ptr->tag);
/* deadlock avoidance... */
while (sip_pvt_ptr->owner && ast_channel_trylock(sip_pvt_ptr->owner)) {
sip_pvt_unlock(sip_pvt_ptr);
usleep(1);
sip_pvt_lock(sip_pvt_ptr);
}
}
return sip_pvt_ptr;
}
/*! \brief Call transfer support (the REFER method)
* Extracts Refer headers into pvt dialog structure
*
* \note If we get a SIPS uri in the refer-to header, we're required to set up a secure signalling path
* to that extension. As a minimum, this needs to be added to a channel variable, if not a channel
* flag.
*/
static int get_refer_info(struct sip_pvt *transferer, struct sip_request *outgoing_req)
{
const char *p_referred_by = NULL;
char *h_refer_to = NULL;
char *h_referred_by = NULL;
char *refer_to;
const char *p_refer_to;
char *referred_by_uri = NULL;
char *ptr;
struct sip_request *req = NULL;
const char *transfer_context = NULL;
struct sip_refer *referdata;
req = outgoing_req;
referdata = transferer->refer;
if (!req)
req = &transferer->initreq;
p_refer_to = get_header(req, "Refer-To");
if (ast_strlen_zero(p_refer_to)) {
ast_log(LOG_WARNING, "Refer-To Header missing. Skipping transfer.\n");
return -2; /* Syntax error */
}
h_refer_to = ast_strdupa(p_refer_to);
refer_to = get_in_brackets(h_refer_to);
if (sip_cfg.pedanticsipchecking)
ast_uri_decode(refer_to);
if (!strncasecmp(refer_to, "sip:", 4)) {
refer_to += 4; /* Skip sip: */
} else if (!strncasecmp(refer_to, "sips:", 5)) {
refer_to += 5;
} else {
ast_log(LOG_WARNING, "Can't transfer to non-sip: URI. (Refer-to: %s)?\n", refer_to);
return -3;
}
/* Get referred by header if it exists */
p_referred_by = get_header(req, "Referred-By");
/* Give useful transfer information to the dialplan */
if (transferer->owner) {
struct ast_channel *peer = ast_bridged_channel(transferer->owner);
if (peer) {
pbx_builtin_setvar_helper(peer, "SIPREFERRINGCONTEXT", transferer->context);
pbx_builtin_setvar_helper(peer, "SIPREFERREDBYHDR", p_referred_by);
}
}
if (!ast_strlen_zero(p_referred_by)) {
char *lessthan;
h_referred_by = ast_strdupa(p_referred_by);
if (sip_cfg.pedanticsipchecking)
ast_uri_decode(h_referred_by);
/* Store referrer's caller ID name */
ast_copy_string(referdata->referred_by_name, h_referred_by, sizeof(referdata->referred_by_name));
if ((lessthan = strchr(referdata->referred_by_name, '<'))) {
*(lessthan - 1) = '\0'; /* Space */
}
referred_by_uri = get_in_brackets(h_referred_by);
if (!strncasecmp(referred_by_uri, "sip:", 4)) {
referred_by_uri += 4; /* Skip sip: */
} else if (!strncasecmp(referred_by_uri, "sips:", 5)) {
referred_by_uri += 5; /* Skip sips: */
} else {
ast_log(LOG_WARNING, "Huh? Not a sip: header (Referred-by: %s). Skipping.\n", referred_by_uri);
referred_by_uri = NULL;
}
}
/* Check for arguments in the refer_to header */
if ((ptr = strcasestr(refer_to, "replaces="))) {
char *to = NULL, *from = NULL;
/* This is an attended transfer */
referdata->attendedtransfer = 1;
ast_copy_string(referdata->replaces_callid, ptr+9, sizeof(referdata->replaces_callid));
ast_uri_decode(referdata->replaces_callid);
if ((ptr = strchr(referdata->replaces_callid, ';'))) /* Find options */ {
*ptr++ = '\0';
}
if (ptr) {
/* Find the different tags before we destroy the string */
to = strcasestr(ptr, "to-tag=");
from = strcasestr(ptr, "from-tag=");
}
/* Grab the to header */
if (to) {
ptr = to + 7;
if ((to = strchr(ptr, '&')))
*to = '\0';
if ((to = strchr(ptr, ';')))
*to = '\0';
ast_copy_string(referdata->replaces_callid_totag, ptr, sizeof(referdata->replaces_callid_totag));
}
if (from) {
ptr = from + 9;
if ((to = strchr(ptr, '&')))
*to = '\0';
if ((to = strchr(ptr, ';')))
*to = '\0';
ast_copy_string(referdata->replaces_callid_fromtag, ptr, sizeof(referdata->replaces_callid_fromtag));
}
if (!sip_cfg.pedanticsipchecking)
ast_debug(2, "Attended transfer: Will use Replace-Call-ID : %s (No check of from/to tags)\n", referdata->replaces_callid );
else
ast_debug(2, "Attended transfer: Will use Replace-Call-ID : %s F-tag: %s T-tag: %s\n", referdata->replaces_callid, referdata->replaces_callid_fromtag ? referdata->replaces_callid_fromtag : "<none>", referdata->replaces_callid_totag ? referdata->replaces_callid_totag : "<none>" );
}
if ((ptr = strchr(refer_to, '@'))) { /* Separate domain */
char *urioption = NULL, *domain;
*ptr++ = '\0';
if ((urioption = strchr(ptr, ';'))) /* Separate urioptions */
*urioption++ = '\0';
domain = ptr;
if ((ptr = strchr(domain, ':'))) /* Remove :port */
*ptr = '\0';
/* Save the domain for the dial plan */
ast_copy_string(referdata->refer_to_domain, domain, sizeof(referdata->refer_to_domain));
if (urioption)
ast_copy_string(referdata->refer_to_urioption, urioption, sizeof(referdata->refer_to_urioption));
}
if ((ptr = strchr(refer_to, ';'))) /* Remove options */
*ptr = '\0';
ast_copy_string(referdata->refer_to, refer_to, sizeof(referdata->refer_to));
if (referred_by_uri) {
if ((ptr = strchr(referred_by_uri, ';'))) /* Remove options */
*ptr = '\0';
ast_copy_string(referdata->referred_by, referred_by_uri, sizeof(referdata->referred_by));
} else {
referdata->referred_by[0] = '\0';
}
/* Determine transfer context */
if (transferer->owner) /* Mimic behaviour in res_features.c */
transfer_context = pbx_builtin_getvar_helper(transferer->owner, "TRANSFER_CONTEXT");
/* By default, use the context in the channel sending the REFER */
if (ast_strlen_zero(transfer_context)) {
transfer_context = S_OR(transferer->owner->macrocontext,
S_OR(transferer->context, sip_cfg.default_context));
}
ast_copy_string(referdata->refer_to_context, transfer_context, sizeof(referdata->refer_to_context));
/* Either an existing extension or the parking extension */
if (referdata->attendedtransfer || ast_exists_extension(NULL, transfer_context, refer_to, 1, NULL) ) {
if (sip_debug_test_pvt(transferer)) {
ast_verbose("SIP transfer to extension %s@%s by %s\n", refer_to, transfer_context, referred_by_uri);
}
/* We are ready to transfer to the extension */
return 0;
}
if (sip_debug_test_pvt(transferer))
ast_verbose("Failed SIP Transfer to non-existing extension %s in context %s\n n", refer_to, transfer_context);
/* Failure, we can't find this extension */
return -1;
}
/*! \brief Call transfer support (old way, deprecated by the IETF)
* \note does not account for SIPS: uri requirements, nor check transport
*/
static int get_also_info(struct sip_pvt *p, struct sip_request *oreq)
{
char tmp[256] = "", *c, *a;
struct sip_request *req = oreq ? oreq : &p->initreq;
struct sip_refer *referdata = NULL;
const char *transfer_context = NULL;
if (!p->refer && !sip_refer_allocate(p))
return -1;
referdata = p->refer;
ast_copy_string(tmp, get_header(req, "Also"), sizeof(tmp));
c = get_in_brackets(tmp);
if (sip_cfg.pedanticsipchecking)
ast_uri_decode(c);
if (!strncasecmp(c, "sip:", 4)) {
c += 4;
} else if (!strncasecmp(c, "sips:", 5)) {
c += 5;
} else {
ast_log(LOG_WARNING, "Huh? Not a SIP header in Also: transfer (%s)?\n", c);
return -1;
}
if ((a = strchr(c, ';'))) /* Remove arguments */
*a = '\0';
if ((a = strchr(c, '@'))) { /* Separate Domain */
*a++ = '\0';
ast_copy_string(referdata->refer_to_domain, a, sizeof(referdata->refer_to_domain));
}
if (sip_debug_test_pvt(p))
ast_verbose("Looking for %s in %s\n", c, p->context);
if (p->owner) /* Mimic behaviour in res_features.c */
transfer_context = pbx_builtin_getvar_helper(p->owner, "TRANSFER_CONTEXT");
/* By default, use the context in the channel sending the REFER */
if (ast_strlen_zero(transfer_context)) {
transfer_context = S_OR(p->owner->macrocontext,
S_OR(p->context, sip_cfg.default_context));
}
if (ast_exists_extension(NULL, transfer_context, c, 1, NULL)) {
/* This is a blind transfer */
ast_debug(1, "SIP Bye-also transfer to Extension %s@%s \n", c, transfer_context);
ast_copy_string(referdata->refer_to, c, sizeof(referdata->refer_to));
ast_copy_string(referdata->referred_by, "", sizeof(referdata->referred_by));
ast_copy_string(referdata->refer_contact, "", sizeof(referdata->refer_contact));
referdata->refer_call = dialog_unref(referdata->refer_call, "unreffing referdata->refer_call");
/* Set new context */
ast_string_field_set(p, context, transfer_context);
return 0;
} else if (ast_canmatch_extension(NULL, p->context, c, 1, NULL)) {
return 1;
}
return -1;
}
/*! \brief check received= and rport= in a SIP response.
* If we get a response with received= and/or rport= in the Via:
* line, use them as 'p->ourip' (see RFC 3581 for rport,
* and RFC 3261 for received).
* Using these two fields SIP can produce the correct
* address and port in the SIP headers without the need for STUN.
* The address part is also reused for the media sessions.
* Note that ast_sip_ouraddrfor() still rewrites p->ourip
* if you specify externip/seternaddr/stunaddr.
*/
static attribute_unused void check_via_response(struct sip_pvt *p, struct sip_request *req)
{
char via[256];
char *cur, *opts;
ast_copy_string(via, get_header(req, "Via"), sizeof(via));
/* Work on the leftmost value of the topmost Via header */
opts = strchr(via, ',');
if (opts)
*opts = '\0';
/* parse all relevant options */
opts = strchr(via, ';');
if (!opts)
return; /* no options to parse */
*opts++ = '\0';
while ( (cur = strsep(&opts, ";")) ) {
if (!strncmp(cur, "rport=", 6)) {
int port = strtol(cur+6, NULL, 10);
/* XXX add error checking */
p->ourip.sin_port = ntohs(port);
} else if (!strncmp(cur, "received=", 9)) {
if (ast_parse_arg(cur+9, PARSE_INADDR, &p->ourip))
; /* XXX add error checking */
}
}
}
/*! \brief check Via: header for hostname, port and rport request/answer */
static void check_via(struct sip_pvt *p, struct sip_request *req)
{
char via[512];
char *c, *pt;
struct hostent *hp;
struct ast_hostent ahp;
ast_copy_string(via, get_header(req, "Via"), sizeof(via));
/* Work on the leftmost value of the topmost Via header */
c = strchr(via, ',');
if (c)
*c = '\0';
/* Check for rport */
c = strstr(via, ";rport");
if (c && (c[6] != '=')) /* rport query, not answer */
ast_set_flag(&p->flags[0], SIP_NAT_ROUTE);
c = strchr(via, ';');
if (c)
*c = '\0';
c = strchr(via, ' ');
if (c) {
*c = '\0';
c = ast_skip_blanks(c+1);
if (strcasecmp(via, "SIP/2.0/UDP") && strcasecmp(via, "SIP/2.0/TCP") && strcasecmp(via, "SIP/2.0/TLS")) {
ast_log(LOG_WARNING, "Don't know how to respond via '%s'\n", via);
return;
}
pt = strchr(c, ':');
if (pt)
*pt++ = '\0'; /* remember port pointer */
hp = ast_gethostbyname(c, &ahp);
if (!hp) {
ast_log(LOG_WARNING, "'%s' is not a valid host\n", c);
return;
}
memset(&p->sa, 0, sizeof(p->sa));
p->sa.sin_family = AF_INET;
memcpy(&p->sa.sin_addr, hp->h_addr, sizeof(p->sa.sin_addr));
p->sa.sin_port = htons(pt ? atoi(pt) : STANDARD_SIP_PORT);
if (sip_debug_test_pvt(p)) {
const struct sockaddr_in *dst = sip_real_dst(p);
ast_verbose("Sending to %s : %d (%s)\n", ast_inet_ntoa(dst->sin_addr), ntohs(dst->sin_port), sip_nat_mode(p));
}
}
}
/*! \brief Get caller id name from SIP headers */
static char *get_calleridname(const char *input, char *output, size_t outputsize)
{
const char *end = strchr(input, '<'); /* first_bracket */
const char *tmp = strchr(input, '"'); /* first quote */
int bytes = 0;
int maxbytes = outputsize - 1;
if (!end || end == input) /* we require a part in brackets */
return NULL;
end--; /* move just before "<" */
if (tmp && tmp <= end) {
/* The quote (tmp) precedes the bracket (end+1).
* Find the matching quote and return the content.
*/
end = strchr(tmp+1, '"');
if (!end)
return NULL;
bytes = (int) (end - tmp);
/* protect the output buffer */
if (bytes > maxbytes)
bytes = maxbytes;
ast_copy_string(output, tmp + 1, bytes);
} else {
/* No quoted string, or it is inside brackets. */
/* clear the empty characters in the begining*/
input = ast_skip_blanks(input);
/* clear the empty characters in the end */
while(*end && *end < 33 && end > input)
end--;
if (end >= input) {
bytes = (int) (end - input) + 2;
/* protect the output buffer */
if (bytes > maxbytes)
bytes = maxbytes;
ast_copy_string(output, input, bytes);
} else
return NULL;
}
return output;
}
/*! \brief Validate device authentication */
static enum check_auth_result check_peer_ok(struct sip_pvt *p, char *of,
struct sip_request *req, int sipmethod, struct sockaddr_in *sin,
struct sip_peer **authpeer,
enum xmittype reliable, char *calleridname, char *uri2)
{
enum check_auth_result res;
int debug=sip_debug_test_addr(sin);
struct sip_peer *peer;
if (sipmethod == SIP_SUBSCRIBE) {
/* For subscribes, match on device name only; for other methods,
* match on IP address-port of the incoming request.
*/
peer = find_peer(of, NULL, TRUE, FINDALLDEVICES, FALSE);
} else {
/* First find devices based on username (avoid all type=peer's) */
peer = find_peer(of, NULL, TRUE, FINDUSERS, FALSE);
/* Then find devices based on IP */
if (!peer) {
peer = find_peer(NULL, &p->recv, TRUE, FINDPEERS, FALSE);
}
}
if (!peer) {
if (debug)
ast_verbose("No matching peer for '%s' from '%s:%d'\n",
of, ast_inet_ntoa(p->recv.sin_addr), ntohs(p->recv.sin_port));
return AUTH_DONT_KNOW;
}
if (debug)
ast_verbose("Found peer '%s' for '%s' from %s:%d\n",
peer->name, of, ast_inet_ntoa(p->recv.sin_addr), ntohs(p->recv.sin_port));
/* XXX what about p->prefs = peer->prefs; ? */
/* Set Frame packetization */
if (p->rtp) {
ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(p->rtp), p->rtp, &peer->prefs);
p->autoframing = peer->autoframing;
}
/* Take the peer */
ast_copy_flags(&p->flags[0], &peer->flags[0], SIP_FLAGS_TO_COPY);
ast_copy_flags(&p->flags[1], &peer->flags[1], SIP_PAGE2_FLAGS_TO_COPY);
/* Copy SIP extensions profile to peer */
/* XXX is this correct before a successful auth ? */
if (p->sipoptions)
peer->sipoptions = p->sipoptions;
do_setnat(p, ast_test_flag(&p->flags[0], SIP_NAT_ROUTE));
ast_string_field_set(p, peersecret, peer->secret);
ast_string_field_set(p, peermd5secret, peer->md5secret);
ast_string_field_set(p, subscribecontext, peer->subscribecontext);
ast_string_field_set(p, mohinterpret, peer->mohinterpret);
ast_string_field_set(p, mohsuggest, peer->mohsuggest);
ast_string_field_set(p, parkinglot, peer->parkinglot);
ast_string_field_set(p, engine, peer->engine);
if (peer->callingpres) /* Peer calling pres setting will override RPID */
p->callingpres = peer->callingpres;
if (peer->maxms && peer->lastms)
p->timer_t1 = peer->lastms < global_t1min ? global_t1min : peer->lastms;
else
p->timer_t1 = peer->timer_t1;
/* Set timer B to control transaction timeouts */
if (peer->timer_b)
p->timer_b = peer->timer_b;
else
p->timer_b = 64 * p->timer_t1;
if (ast_test_flag(&peer->flags[0], SIP_INSECURE_INVITE)) {
/* Pretend there is no required authentication */
ast_string_field_set(p, peersecret, NULL);
ast_string_field_set(p, peermd5secret, NULL);
}
if (!(res = check_auth(p, req, peer->name, p->peersecret, p->peermd5secret, sipmethod, uri2, reliable, req->ignore))) {
ast_copy_flags(&p->flags[0], &peer->flags[0], SIP_FLAGS_TO_COPY);
ast_copy_flags(&p->flags[1], &peer->flags[1], SIP_PAGE2_FLAGS_TO_COPY);
/* If we have a call limit, set flag */
if (peer->call_limit)
ast_set_flag(&p->flags[0], SIP_CALL_LIMIT);
ast_string_field_set(p, peername, peer->name);
ast_string_field_set(p, authname, peer->name);
if (sipmethod == SIP_INVITE) {
/* copy channel vars */
p->chanvars = copy_vars(peer->chanvars);
}
if (authpeer) {
ao2_t_ref(peer, 1, "copy pointer into (*authpeer)");
(*authpeer) = peer; /* Add a ref to the object here, to keep it in memory a bit longer if it is realtime */
}
if (!ast_strlen_zero(peer->username)) {
ast_string_field_set(p, username, peer->username);
/* Use the default username for authentication on outbound calls */
/* XXX this takes the name from the caller... can we override ? */
ast_string_field_set(p, authname, peer->username);
}
if (!get_rpid(p, req)) {
if (!ast_strlen_zero(peer->cid_num)) {
char *tmp = ast_strdupa(peer->cid_num);
if (ast_is_shrinkable_phonenumber(tmp))
ast_shrink_phone_number(tmp);
ast_string_field_set(p, cid_num, tmp);
}
if (!ast_strlen_zero(peer->cid_name))
ast_string_field_set(p, cid_name, peer->cid_name);
if (peer->callingpres)
p->callingpres = peer->callingpres;
}
ast_string_field_set(p, fullcontact, peer->fullcontact);
if (!ast_strlen_zero(peer->context))
ast_string_field_set(p, context, peer->context);
ast_string_field_set(p, peersecret, peer->secret);
ast_string_field_set(p, peermd5secret, peer->md5secret);
ast_string_field_set(p, language, peer->language);
ast_string_field_set(p, accountcode, peer->accountcode);
p->amaflags = peer->amaflags;
p->callgroup = peer->callgroup;
p->pickupgroup = peer->pickupgroup;
p->capability = peer->capability;
p->prefs = peer->prefs;
p->jointcapability = peer->capability;
if (p->peercapability)
p->jointcapability &= p->peercapability;
p->maxcallbitrate = peer->maxcallbitrate;
if ((ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833) ||
(ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_AUTO))
p->noncodeccapability |= AST_RTP_DTMF;
else
p->noncodeccapability &= ~AST_RTP_DTMF;
p->jointnoncodeccapability = p->noncodeccapability;
if (p->t38.peercapability)
p->t38.jointcapability &= p->t38.peercapability;
if (!dialog_initialize_rtp(p)) {
if (p->rtp) {
ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(p->rtp), p->rtp, &peer->prefs);
p->autoframing = peer->autoframing;
}
} else {
res = AUTH_RTP_FAILED;
}
}
unref_peer(peer, "check_peer_ok: unref_peer: tossing temp ptr to peer from find_peer");
return res;
}
/*! \brief Check if matching user or peer is defined
Match user on From: user name and peer on IP/port
This is used on first invite (not re-invites) and subscribe requests
\return 0 on success, non-zero on failure
*/
static enum check_auth_result check_user_full(struct sip_pvt *p, struct sip_request *req,
int sipmethod, char *uri, enum xmittype reliable,
struct sockaddr_in *sin, struct sip_peer **authpeer)
{
char from[256];
char *dummy; /* dummy return value for parse_uri */
char *domain; /* dummy return value for parse_uri */
char *of, *of2;
enum check_auth_result res;
char calleridname[50];
char *uri2 = ast_strdupa(uri);
terminate_uri(uri2); /* trim extra stuff */
ast_copy_string(from, get_header(req, "From"), sizeof(from));
if (sip_cfg.pedanticsipchecking)
ast_uri_decode(from);
/* XXX here tries to map the username for invite things */
memset(calleridname, 0, sizeof(calleridname));
get_calleridname(from, calleridname, sizeof(calleridname));
if (calleridname[0])
ast_string_field_set(p, cid_name, calleridname);
of = get_in_brackets(from);
if (ast_strlen_zero(p->exten)) {
char *t = uri2;
if (!strncasecmp(t, "sip:", 4))
t+= 4;
else if (!strncasecmp(t, "sips:", 5))
t += 5;
ast_string_field_set(p, exten, t);
t = strchr(p->exten, '@');
if (t)
*t = '\0';
if (ast_strlen_zero(p->our_contact))
build_contact(p);
}
/* save the URI part of the From header */
ast_string_field_set(p, from, of);
of2 = ast_strdupa(of);
/* ignore all fields but name */
/*! \todo Samme logical error as in many places above. Need a generic function for this.
*/
if (p->socket.type == SIP_TRANSPORT_TLS) {
if (parse_uri(of, "sips:", &of, &dummy, &domain, &dummy, &dummy)) {
if (parse_uri(of2, "sip:", &of, &dummy, &domain, &dummy, &dummy))
ast_log(LOG_NOTICE, "From address missing 'sip:', using it anyway\n");
}
} else {
if (parse_uri(of, "sip:", &of, &dummy, &domain, &dummy, &dummy))
ast_log(LOG_NOTICE, "From address missing 'sip:', using it anyway\n");
}
if (ast_strlen_zero(of)) {
/* XXX note: the original code considered a missing @host
* as a username-only URI. The SIP RFC (19.1.1) says that
* this is wrong, and it should be considered as a domain-only URI.
* For backward compatibility, we keep this block, but it is
* really a mistake and should go away.
*/
of = domain;
}
{
char *tmp = ast_strdupa(of);
/* We need to be able to handle auth-headers looking like
<sip:8164444422;phone-context=+1@1.2.3.4:5060;user=phone;tag=SDadkoa01-gK0c3bdb43>
*/
tmp = strsep(&tmp, ";");
if (ast_is_shrinkable_phonenumber(tmp))
ast_shrink_phone_number(tmp);
ast_string_field_set(p, cid_num, tmp);
}
if (global_match_auth_username) {
/*
* XXX This is experimental code to grab the search key from the
* Auth header's username instead of the 'From' name, if available.
* Do not enable this block unless you understand the side effects (if any!)
* Note, the search for "username" should be done in a more robust way.
* Note2, at the moment we check both fields, though maybe we should
* pick one or another depending on the request ? XXX
*/
const char *hdr = get_header(req, "Authorization");
if (ast_strlen_zero(hdr))
hdr = get_header(req, "Proxy-Authorization");
if ( !ast_strlen_zero(hdr) && (hdr = strstr(hdr, "username=\"")) ) {
ast_copy_string(from, hdr + strlen("username=\""), sizeof(from));
of = from;
of = strsep(&of, "\"");
}
}
res = check_peer_ok(p, of, req, sipmethod, sin,
authpeer, reliable, calleridname, uri2);
if (res != AUTH_DONT_KNOW)
return res;
/* Finally, apply the guest policy */
if (sip_cfg.allowguest) {
get_rpid(p, req);
if (!dialog_initialize_rtp(p)) {
res = AUTH_SUCCESSFUL;
} else {
res = AUTH_RTP_FAILED;
}
} else if (sip_cfg.alwaysauthreject)
res = AUTH_FAKE_AUTH; /* reject with fake authorization request */
else
res = AUTH_SECRET_FAILED; /* we don't want any guests, authentication will fail */
return res;
}
/*! \brief Find user
If we get a match, this will add a reference pointer to the user object in ASTOBJ, that needs to be unreferenced
*/
static int check_user(struct sip_pvt *p, struct sip_request *req, int sipmethod, char *uri, enum xmittype reliable, struct sockaddr_in *sin)
{
return check_user_full(p, req, sipmethod, uri, reliable, sin, NULL);
}
/*! \brief Get text out of a SIP MESSAGE packet */
static int get_msg_text(char *buf, int len, struct sip_request *req, int addnewline)
{
int x;
int y;
buf[0] = '\0';
/*XXX isn't strlen(buf) going to always be 0? */
y = len - strlen(buf) - 5;
if (y < 0)
y = 0;
for (x = 0; x < req->lines; x++) {
char *line = REQ_OFFSET_TO_STR(req, line[x]);
strncat(buf, line, y); /* safe */
y -= strlen(line) + 1;
if (y < 0)
y = 0;
if (y != 0 && addnewline)
strcat(buf, "\n"); /* safe */
}
return 0;
}
/*! \brief Receive SIP MESSAGE method messages
\note We only handle messages within current calls currently
Reference: RFC 3428 */
static void receive_message(struct sip_pvt *p, struct sip_request *req)
{
char buf[1400];
struct ast_frame f;
const char *content_type = get_header(req, "Content-Type");
if (strncmp(content_type, "text/plain", strlen("text/plain"))) { /* No text/plain attachment */
transmit_response(p, "415 Unsupported Media Type", req); /* Good enough, or? */
if (!p->owner)
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
return;
}
if (get_msg_text(buf, sizeof(buf), req, FALSE)) {
ast_log(LOG_WARNING, "Unable to retrieve text from %s\n", p->callid);
transmit_response(p, "202 Accepted", req);
if (!p->owner)
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
return;
}
if (p->owner) {
if (sip_debug_test_pvt(p))
ast_verbose("SIP Text message received: '%s'\n", buf);
memset(&f, 0, sizeof(f));
f.frametype = AST_FRAME_TEXT;
f.subclass = 0;
f.offset = 0;
f.data.ptr = buf;
f.datalen = strlen(buf);
ast_queue_frame(p->owner, &f);
transmit_response(p, "202 Accepted", req); /* We respond 202 accepted, since we relay the message */
return;
}
/* Message outside of a call, we do not support that */
ast_log(LOG_WARNING, "Received message to %s from %s, dropped it...\n Content-Type:%s\n Message: %s\n", get_header(req, "To"), get_header(req, "From"), content_type, buf);
transmit_response(p, "405 Method Not Allowed", req);
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
return;
}
/*! \brief CLI Command to show calls within limits set by call_limit */
static char *sip_show_inuse(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
#define FORMAT "%-25.25s %-15.15s %-15.15s \n"
#define FORMAT2 "%-25.25s %-15.15s %-15.15s \n"
char ilimits[40];
char iused[40];
int showall = FALSE;
struct ao2_iterator i;
struct sip_peer *peer;
switch (cmd) {
case CLI_INIT:
e->command = "sip show inuse";
e->usage =
"Usage: sip show inuse [all]\n"
" List all SIP devices usage counters and limits.\n"
" Add option \"all\" to show all devices, not only those with a limit.\n";
return NULL;
case CLI_GENERATE:
return NULL;
}
if (a->argc < 3)
return CLI_SHOWUSAGE;
if (a->argc == 4 && !strcmp(a->argv[3], "all"))
showall = TRUE;
ast_cli(a->fd, FORMAT, "* Peer name", "In use", "Limit");
i = ao2_iterator_init(peers, 0);
while ((peer = ao2_t_iterator_next(&i, "iterate thru peer table"))) {
ao2_lock(peer);
if (peer->call_limit)
snprintf(ilimits, sizeof(ilimits), "%d", peer->call_limit);
else
ast_copy_string(ilimits, "N/A", sizeof(ilimits));
snprintf(iused, sizeof(iused), "%d/%d/%d", peer->inUse, peer->inRinging, peer->onHold);
if (showall || peer->call_limit)
ast_cli(a->fd, FORMAT2, peer->name, iused, ilimits);
ao2_unlock(peer);
unref_peer(peer, "toss iterator pointer");
}
return CLI_SUCCESS;
#undef FORMAT
#undef FORMAT2
}
/*! \brief Convert transfer mode to text string */
static char *transfermode2str(enum transfermodes mode)
{
if (mode == TRANSFER_OPENFORALL)
return "open";
else if (mode == TRANSFER_CLOSED)
return "closed";
return "strict";
}
static struct _map_x_s natmodes[] = {
{ SIP_NAT_NEVER, "No"},
{ SIP_NAT_ROUTE, "Route"},
{ SIP_NAT_ALWAYS, "Always"},
{ SIP_NAT_RFC3581, "RFC3581"},
{ -1, NULL}, /* terminator */
};
/*! \brief Convert NAT setting to text string */
static const char *nat2str(int nat)
{
return map_x_s(natmodes, nat, "Unknown");
}
#ifdef NOTUSED
/* OEJ: This is not used, but may be useful in the future, so I don't want to
delete it. Keeping it enabled generates compiler warnings.
*/
static struct _map_x_s natcfgmodes[] = {
{ SIP_NAT_NEVER, "never"},
{ SIP_NAT_ROUTE, "route"},
{ SIP_NAT_ALWAYS, "yes"},
{ SIP_NAT_RFC3581, "no"},
{ -1, NULL}, /* terminator */
};
/*! \brief Convert NAT setting to text string appropriate for config files */
static const char *nat2strconfig(int nat)
{
return map_x_s(natcfgmodes, nat, "Unknown");
}
#endif
/*! \brief Report Peer status in character string
* \return 0 if peer is unreachable, 1 if peer is online, -1 if unmonitored
*/
/* Session-Timer Modes */
static struct _map_x_s stmodes[] = {
{ SESSION_TIMER_MODE_ACCEPT, "Accept"},
{ SESSION_TIMER_MODE_ORIGINATE, "Originate"},
{ SESSION_TIMER_MODE_REFUSE, "Refuse"},
{ -1, NULL},
};
static const char *stmode2str(enum st_mode m)
{
return map_x_s(stmodes, m, "Unknown");
}
static enum st_mode str2stmode(const char *s)
{
return map_s_x(stmodes, s, -1);
}
/* Session-Timer Refreshers */
static struct _map_x_s strefreshers[] = {
{ SESSION_TIMER_REFRESHER_AUTO, "auto"},
{ SESSION_TIMER_REFRESHER_UAC, "uac"},
{ SESSION_TIMER_REFRESHER_UAS, "uas"},
{ -1, NULL},
};
static const char *strefresher2str(enum st_refresher r)
{
return map_x_s(strefreshers, r, "Unknown");
}
static enum st_refresher str2strefresher(const char *s)
{
return map_s_x(strefreshers, s, -1);
}
static int peer_status(struct sip_peer *peer, char *status, int statuslen)
{
int res = 0;
if (peer->maxms) {
if (peer->lastms < 0) {
ast_copy_string(status, "UNREACHABLE", statuslen);
} else if (peer->lastms > peer->maxms) {
snprintf(status, statuslen, "LAGGED (%d ms)", peer->lastms);
res = 1;
} else if (peer->lastms) {
snprintf(status, statuslen, "OK (%d ms)", peer->lastms);
res = 1;
} else {
ast_copy_string(status, "UNKNOWN", statuslen);
}
} else {
ast_copy_string(status, "Unmonitored", statuslen);
/* Checking if port is 0 */
res = -1;
}
return res;
}
/*! \brief return Yes or No depending on the argument.
* This is used in many places in CLI command, having a function to generate
* this helps maintaining a consistent output (and possibly emitting the
* output in other languages, at some point).
*/
static const char *cli_yesno(int x)
{
return x ? "Yes" : "No";
}
/*! \brief Show active TCP connections */
static char *sip_show_tcp(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
struct sip_threadinfo *th;
#define FORMAT2 "%-30.30s %3.6s %9.9s %6.6s\n"
#define FORMAT "%-30.30s %-6d %-9.9s %-6.6s\n"
switch (cmd) {
case CLI_INIT:
e->command = "sip show tcp";
e->usage =
"Usage: sip show tcp\n"
" Lists all active TCP/TLS sessions.\n";
return NULL;
case CLI_GENERATE:
return NULL;
}
if (a->argc != 3)
return CLI_SHOWUSAGE;
ast_cli(a->fd, FORMAT2, "Host", "Port", "Transport", "Type");
AST_LIST_LOCK(&threadl);
AST_LIST_TRAVERSE(&threadl, th, list) {
ast_cli(a->fd, FORMAT, ast_inet_ntoa(th->tcptls_session->remote_address.sin_addr),
ntohs(th->tcptls_session->remote_address.sin_port),
get_transport(th->type),
(th->tcptls_session->client ? "Client" : "Server"));
}
AST_LIST_UNLOCK(&threadl);
return CLI_SUCCESS;
#undef FORMAT
#undef FORMAT2
}
/*! \brief CLI Command 'SIP Show Users' */
static char *sip_show_users(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
regex_t regexbuf;
int havepattern = FALSE;
struct ao2_iterator user_iter;
struct sip_peer *user;
#define FORMAT "%-25.25s %-15.15s %-15.15s %-15.15s %-5.5s%-10.10s\n"
switch (cmd) {
case CLI_INIT:
e->command = "sip show users";
e->usage =
"Usage: sip show users [like <pattern>]\n"
" Lists all known SIP users.\n"
" Optional regular expression pattern is used to filter the user list.\n";
return NULL;
case CLI_GENERATE:
return NULL;
}
switch (a->argc) {
case 5:
if (!strcasecmp(a->argv[3], "like")) {
if (regcomp(&regexbuf, a->argv[4], REG_EXTENDED | REG_NOSUB))
return CLI_SHOWUSAGE;
havepattern = TRUE;
} else
return CLI_SHOWUSAGE;
case 3:
break;
default:
return CLI_SHOWUSAGE;
}
ast_cli(a->fd, FORMAT, "Username", "Secret", "Accountcode", "Def.Context", "ACL", "NAT");
user_iter = ao2_iterator_init(peers, 0);
while ((user = ao2_iterator_next(&user_iter))) {
ao2_lock(user);
if (!(user->type & SIP_TYPE_USER)) {
ao2_unlock(user);
unref_peer(user, "sip show users");
continue;
}
if (havepattern && regexec(&regexbuf, user->name, 0, NULL, 0)) {
ao2_unlock(user);
unref_peer(user, "sip show users");
continue;
}
ast_cli(a->fd, FORMAT, user->name,
user->secret,
user->accountcode,
user->context,
cli_yesno(user->ha != NULL),
nat2str(ast_test_flag(&user->flags[0], SIP_NAT)));
ao2_unlock(user);
unref_peer(user, "sip show users");
}
if (havepattern)
regfree(&regexbuf);
return CLI_SUCCESS;
#undef FORMAT
}
/*! \brief Manager Action SIPShowRegistry description */
static char mandescr_show_registry[] =
"Description: Lists all registration requests and status\n"
"Registrations will follow as separate events. followed by a final event called\n"
"RegistrationsComplete.\n"
"Variables: \n"
" ActionID: <id> Action ID for this transaction. Will be returned.\n";
/*! \brief Show SIP registrations in the manager API */
static int manager_show_registry(struct mansession *s, const struct message *m)
{
const char *id = astman_get_header(m, "ActionID");
char idtext[256] = "";
int total = 0;
if (!ast_strlen_zero(id))
snprintf(idtext, sizeof(idtext), "ActionID: %s\r\n", id);
astman_send_listack(s, m, "Registrations will follow", "start");
ASTOBJ_CONTAINER_TRAVERSE(&regl, 1, do {
ASTOBJ_RDLOCK(iterator);
astman_append(s,
"Event: RegistryEntry\r\n"
"Host: %s\r\n"
"Port: %d\r\n"
"Username: %s\r\n"
"Refresh: %d\r\n"
"State: %s\r\n"
"RegistrationTime: %ld\r\n"
"\r\n", iterator->hostname, iterator->portno ? iterator->portno : STANDARD_SIP_PORT,
iterator->username, iterator->refresh, regstate2str(iterator->regstate), (long) iterator->regtime.tv_sec);
ASTOBJ_UNLOCK(iterator);
total++;
} while(0));
astman_append(s,
"Event: RegistrationsComplete\r\n"
"EventList: Complete\r\n"
"ListItems: %d\r\n"
"%s"
"\r\n", total, idtext);
return 0;
}
static char mandescr_show_peers[] =
"Description: Lists SIP peers in text format with details on current status.\n"
"Peerlist will follow as separate events, followed by a final event called\n"
"PeerlistComplete.\n"
"Variables: \n"
" ActionID: <id> Action ID for this transaction. Will be returned.\n";
/*! \brief Show SIP peers in the manager API */
/* Inspired from chan_iax2 */
static int manager_sip_show_peers(struct mansession *s, const struct message *m)
{
const char *id = astman_get_header(m, "ActionID");
const char *a[] = {"sip", "show", "peers"};
char idtext[256] = "";
int total = 0;
if (!ast_strlen_zero(id))
snprintf(idtext, sizeof(idtext), "ActionID: %s\r\n", id);
astman_send_listack(s, m, "Peer status list will follow", "start");
/* List the peers in separate manager events */
_sip_show_peers(-1, &total, s, m, 3, a);
/* Send final confirmation */
astman_append(s,
"Event: PeerlistComplete\r\n"
"EventList: Complete\r\n"
"ListItems: %d\r\n"
"%s"
"\r\n", total, idtext);
return 0;
}
/*! \brief CLI Show Peers command */
static char *sip_show_peers(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
switch (cmd) {
case CLI_INIT:
e->command = "sip show peers";
e->usage =
"Usage: sip show peers [like <pattern>]\n"
" Lists all known SIP peers.\n"
" Optional regular expression pattern is used to filter the peer list.\n";
return NULL;
case CLI_GENERATE:
return NULL;
}
return _sip_show_peers(a->fd, NULL, NULL, NULL, a->argc, (const char **) a->argv);
}
int peercomparefunc(const void *a, const void *b);
int peercomparefunc(const void *a, const void *b)
{
struct sip_peer **ap = (struct sip_peer **)a;
struct sip_peer **bp = (struct sip_peer **)b;
return strcmp((*ap)->name, (*bp)->name);
}
/*! \brief Execute sip show peers command */
static char *_sip_show_peers(int fd, int *total, struct mansession *s, const struct message *m, int argc, const char *argv[])
{
regex_t regexbuf;
int havepattern = FALSE;
struct sip_peer *peer;
struct ao2_iterator i;
/* the last argument is left-aligned, so we don't need a size anyways */
#define FORMAT2 "%-25.25s %-15.15s %-3.3s %-3.3s %-3.3s %-8s %-10s %s\n"
#define FORMAT "%-25.25s %-15.15s %-3.3s %-3.3s %-3.3s %-8d %-10s %s\n"
char name[256];
int total_peers = 0;
int peers_mon_online = 0;
int peers_mon_offline = 0;
int peers_unmon_offline = 0;
int peers_unmon_online = 0;
const char *id;
char idtext[256] = "";
int realtimepeers;
int objcount = ao2_container_count(peers);
struct sip_peer **peerarray;
int k;
realtimepeers = ast_check_realtime("sippeers");
peerarray = ast_calloc(sizeof(struct sip_peer *), objcount);
if (s) { /* Manager - get ActionID */
id = astman_get_header(m, "ActionID");
if (!ast_strlen_zero(id))
snprintf(idtext, sizeof(idtext), "ActionID: %s\r\n", id);
}
switch (argc) {
case 5:
if (!strcasecmp(argv[3], "like")) {
if (regcomp(&regexbuf, argv[4], REG_EXTENDED | REG_NOSUB))
return CLI_SHOWUSAGE;
havepattern = TRUE;
} else
return CLI_SHOWUSAGE;
case 3:
break;
default:
return CLI_SHOWUSAGE;
}
if (!s) /* Normal list */
ast_cli(fd, FORMAT2, "Name/username", "Host", "Dyn", "Nat", "ACL", "Port", "Status", (realtimepeers ? "Realtime" : ""));
i = ao2_iterator_init(peers, 0);
while ((peer = ao2_t_iterator_next(&i, "iterate thru peers table"))) {
ao2_lock(peer);
if (!(peer->type & SIP_TYPE_PEER)) {
ao2_unlock(peer);
unref_peer(peer, "unref peer because it's actually a user");
continue;
}
if (havepattern && regexec(&regexbuf, peer->name, 0, NULL, 0)) {
objcount--;
ao2_unlock(peer);
unref_peer(peer, "toss iterator peer ptr before continue");
continue;
}
peerarray[total_peers++] = peer;
ao2_unlock(peer);
}
qsort(peerarray, total_peers, sizeof(struct sip_peer *), peercomparefunc);
for(k=0; k < total_peers; k++) {
char status[20] = "";
char srch[2000];
char pstatus;
peer = peerarray[k];
ao2_lock(peer);
if (havepattern && regexec(&regexbuf, peer->name, 0, NULL, 0)) {
ao2_unlock(peer);
unref_peer(peer, "toss iterator peer ptr before continue");
continue;
}
if (!ast_strlen_zero(peer->username) && !s)
snprintf(name, sizeof(name), "%s/%s", peer->name, peer->username);
else
ast_copy_string(name, peer->name, sizeof(name));
pstatus = peer_status(peer, status, sizeof(status));
if (pstatus == 1)
peers_mon_online++;
else if (pstatus == 0)
peers_mon_offline++;
else {
if (peer->addr.sin_port == 0)
peers_unmon_offline++;
else
peers_unmon_online++;
}
snprintf(srch, sizeof(srch), FORMAT, name,
peer->addr.sin_addr.s_addr ? ast_inet_ntoa(peer->addr.sin_addr) : "(Unspecified)",
peer->host_dynamic ? " D " : " ", /* Dynamic or not? */
ast_test_flag(&peer->flags[0], SIP_NAT_ROUTE) ? " N " : " ", /* NAT=yes? */
peer->ha ? " A " : " ", /* permit/deny */
ntohs(peer->addr.sin_port), status,
realtimepeers ? (peer->is_realtime ? "Cached RT":"") : "");
if (!s) {/* Normal CLI list */
ast_cli(fd, FORMAT, name,
peer->addr.sin_addr.s_addr ? ast_inet_ntoa(peer->addr.sin_addr) : "(Unspecified)",
peer->host_dynamic ? " D " : " ", /* Dynamic or not? */
ast_test_flag(&peer->flags[0], SIP_NAT_ROUTE) ? " N " : " ", /* NAT=yes? */
peer->ha ? " A " : " ", /* permit/deny */
ntohs(peer->addr.sin_port), status,
realtimepeers ? (peer->is_realtime ? "Cached RT":"") : "");
} else { /* Manager format */
/* The names here need to be the same as other channels */
astman_append(s,
"Event: PeerEntry\r\n%s"
"Channeltype: SIP\r\n"
"ObjectName: %s\r\n"
"ChanObjectType: peer\r\n" /* "peer" or "user" */
"IPaddress: %s\r\n"
"IPport: %d\r\n"
"Dynamic: %s\r\n"
"Natsupport: %s\r\n"
"VideoSupport: %s\r\n"
"TextSupport: %s\r\n"
"ACL: %s\r\n"
"Status: %s\r\n"
"RealtimeDevice: %s\r\n\r\n",
idtext,
peer->name,
peer->addr.sin_addr.s_addr ? ast_inet_ntoa(peer->addr.sin_addr) : "-none-",
ntohs(peer->addr.sin_port),
peer->host_dynamic ? "yes" : "no", /* Dynamic or not? */
ast_test_flag(&peer->flags[0], SIP_NAT_ROUTE) ? "yes" : "no", /* NAT=yes? */
ast_test_flag(&peer->flags[1], SIP_PAGE2_VIDEOSUPPORT) ? "yes" : "no", /* VIDEOSUPPORT=yes? */
ast_test_flag(&peer->flags[1], SIP_PAGE2_TEXTSUPPORT) ? "yes" : "no", /* TEXTSUPPORT=yes? */
peer->ha ? "yes" : "no", /* permit/deny */
status,
realtimepeers ? (peer->is_realtime ? "yes":"no") : "no");
}
ao2_unlock(peer);
unref_peer(peer, "toss iterator peer ptr");
}
if (!s)
ast_cli(fd, "%d sip peers [Monitored: %d online, %d offline Unmonitored: %d online, %d offline]\n",
total_peers, peers_mon_online, peers_mon_offline, peers_unmon_online, peers_unmon_offline);
if (havepattern)
regfree(&regexbuf);
if (total)
*total = total_peers;
ast_free(peerarray);
return CLI_SUCCESS;
#undef FORMAT
#undef FORMAT2
}
static int peer_dump_func(void *userobj, void *arg, int flags)
{
struct sip_peer *peer = userobj;
int refc = ao2_t_ref(userobj, 0, "");
int *fd = arg;
ast_cli(*fd, "name: %s\ntype: peer\nobjflags: %d\nrefcount: %d\n\n",
peer->name, 0, refc);
return 0;
}
static int dialog_dump_func(void *userobj, void *arg, int flags)
{
struct sip_pvt *pvt = userobj;
int refc = ao2_t_ref(userobj, 0, "");
int *fd = arg;
ast_cli(*fd, "name: %s\ntype: dialog\nobjflags: %d\nrefcount: %d\n\n",
pvt->callid, 0, refc);
return 0;
}
/*! \brief List all allocated SIP Objects (realtime or static) */
static char *sip_show_objects(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
char tmp[256];
switch (cmd) {
case CLI_INIT:
e->command = "sip show objects";
e->usage =
"Usage: sip show objects\n"
" Lists status of known SIP objects\n";
return NULL;
case CLI_GENERATE:
return NULL;
}
if (a->argc != 3)
return CLI_SHOWUSAGE;
ast_cli(a->fd, "-= Peer objects: %d static, %d realtime, %d autocreate =-\n\n", speerobjs, rpeerobjs, apeerobjs);
ao2_t_callback(peers, OBJ_NODATA, peer_dump_func, &a->fd, "initiate ao2_callback to dump peers");
ast_cli(a->fd, "-= Registry objects: %d =-\n\n", regobjs);
ASTOBJ_CONTAINER_DUMP(a->fd, tmp, sizeof(tmp), &regl);
ast_cli(a->fd, "-= Dialog objects:\n\n");
ao2_t_callback(dialogs, OBJ_NODATA, dialog_dump_func, &a->fd, "initiate ao2_callback to dump dialogs");
return CLI_SUCCESS;
}
/*! \brief Print call group and pickup group */
static void print_group(int fd, ast_group_t group, int crlf)
{
char buf[256];
ast_cli(fd, crlf ? "%s\r\n" : "%s\n", ast_print_group(buf, sizeof(buf), group) );
}
/*! \brief mapping between dtmf flags and strings */
static struct _map_x_s dtmfstr[] = {
{ SIP_DTMF_RFC2833, "rfc2833" },
{ SIP_DTMF_INFO, "info" },
{ SIP_DTMF_SHORTINFO, "shortinfo" },
{ SIP_DTMF_INBAND, "inband" },
{ SIP_DTMF_AUTO, "auto" },
{ -1, NULL }, /* terminator */
};
/*! \brief Convert DTMF mode to printable string */
static const char *dtmfmode2str(int mode)
{
return map_x_s(dtmfstr, mode, "<error>");
}
/*! \brief maps a string to dtmfmode, returns -1 on error */
static int str2dtmfmode(const char *str)
{
return map_s_x(dtmfstr, str, -1);
}
static struct _map_x_s insecurestr[] = {
{ SIP_INSECURE_PORT, "port" },
{ SIP_INSECURE_INVITE, "invite" },
{ SIP_INSECURE_PORT | SIP_INSECURE_INVITE, "port,invite" },
{ 0, "no" },
{ -1, NULL }, /* terminator */
};
/*! \brief Convert Insecure setting to printable string */
static const char *insecure2str(int mode)
{
return map_x_s(insecurestr, mode, "<error>");
}
/*! \brief Destroy disused contexts between reloads
Only used in reload_config so the code for regcontext doesn't get ugly
*/
static void cleanup_stale_contexts(char *new, char *old)
{
char *oldcontext, *newcontext, *stalecontext, *stringp, newlist[AST_MAX_CONTEXT];
while ((oldcontext = strsep(&old, "&"))) {
stalecontext = '\0';
ast_copy_string(newlist, new, sizeof(newlist));
stringp = newlist;
while ((newcontext = strsep(&stringp, "&"))) {
if (!strcmp(newcontext, oldcontext)) {
/* This is not the context you're looking for */
stalecontext = '\0';
break;
} else if (strcmp(newcontext, oldcontext)) {
stalecontext = oldcontext;
}
}
if (stalecontext)
ast_context_destroy(ast_context_find(stalecontext), "SIP");
}
}
/*!
* \brief Match dialogs that need to be destroyed
*
* \details This is used with ao2_callback to unlink/delete all dialogs that
* are marked needdestroy. It will return CMP_MATCH for candidates, and they
* will be unlinked.
*
* \todo Re-work this to improve efficiency. Currently, this function is called
* on _every_ dialog after processing _every_ incoming SIP/UDP packet, or
* potentially even more often when the scheduler has entries to run.
*/
static int dialog_needdestroy(void *dialogobj, void *arg, int flags)
{
struct sip_pvt *dialog = dialogobj;
time_t *t = arg;
if (sip_pvt_trylock(dialog)) {
/* Don't block the monitor thread. This function is called often enough
* that we can wait for the next time around. */
return 0;
}
/* We absolutely cannot destroy the rtp struct while a bridge is active or we WILL crash */
if (dialog->rtp && ast_rtp_instance_get_bridged(dialog->rtp)) {
ast_debug(2, "Bridge still active. Delaying destroy of SIP dialog '%s' Method: %s\n", dialog->callid, sip_methods[dialog->method].text);
sip_pvt_unlock(dialog);
return 0;
}
if (dialog->vrtp && ast_rtp_instance_get_bridged(dialog->vrtp)) {
ast_debug(2, "Bridge still active. Delaying destroy of SIP dialog '%s' Method: %s\n", dialog->callid, sip_methods[dialog->method].text);
sip_pvt_unlock(dialog);
return 0;
}
/* Check RTP timeouts and kill calls if we have a timeout set and do not get RTP */
check_rtp_timeout(dialog, *t);
/* If we have sessions that needs to be destroyed, do it now */
/* Check if we have outstanding requests not responsed to or an active call
- if that's the case, wait with destruction */
if (dialog->needdestroy && !dialog->packets && !dialog->owner) {
/* We absolutely cannot destroy the rtp struct while a bridge is active or we WILL crash */
if (dialog->rtp && ast_rtp_instance_get_bridged(dialog->rtp)) {
ast_debug(2, "Bridge still active. Delaying destruction of SIP dialog '%s' Method: %s\n", dialog->callid, sip_methods[dialog->method].text);
sip_pvt_unlock(dialog);
return 0;
}
if (dialog->vrtp && ast_rtp_instance_get_bridged(dialog->vrtp)) {
ast_debug(2, "Bridge still active. Delaying destroy of SIP dialog '%s' Method: %s\n", dialog->callid, sip_methods[dialog->method].text);
sip_pvt_unlock(dialog);
return 0;
}
sip_pvt_unlock(dialog);
/* no, the unlink should handle this: dialog_unref(dialog, "needdestroy: one more refcount decrement to allow dialog to be destroyed"); */
/* the CMP_MATCH will unlink this dialog from the dialog hash table */
dialog_unlink_all(dialog, TRUE, FALSE);
return 0; /* the unlink_all should unlink this from the table, so.... no need to return a match */
}
sip_pvt_unlock(dialog);
return 0;
}
/* this func is used with ao2_callback to unlink/delete all marked
peers */
static int peer_is_marked(void *peerobj, void *arg, int flags)
{
struct sip_peer *peer = peerobj;
return peer->the_mark ? CMP_MATCH : 0;
}
/*! \brief Remove temporary realtime objects from memory (CLI) */
/*! \todo XXXX Propably needs an overhaul after removal of the devices */
static char *sip_prune_realtime(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
struct sip_peer *peer, *pi;
int prunepeer = FALSE;
int multi = FALSE;
char *name = NULL;
regex_t regexbuf;
struct ao2_iterator i;
static char *choices[] = { "all", "like", NULL };
char *cmplt;
if (cmd == CLI_INIT) {
e->command = "sip prune realtime [peer|all]";
e->usage =
"Usage: sip prune realtime [peer [<name>|all|like <pattern>]|all]\n"
" Prunes object(s) from the cache.\n"
" Optional regular expression pattern is used to filter the objects.\n";
return NULL;
} else if (cmd == CLI_GENERATE) {
if (a->pos == 4 && !strcasecmp(a->argv[3], "peer")) {
cmplt = ast_cli_complete(a->word, choices, a->n);
if (!cmplt)
cmplt = complete_sip_peer(a->word, a->n - sizeof(choices), SIP_PAGE2_RTCACHEFRIENDS);
return cmplt;
}
if (a->pos == 5 && !strcasecmp(a->argv[4], "like"))
return complete_sip_peer(a->word, a->n, SIP_PAGE2_RTCACHEFRIENDS);
return NULL;
}
switch (a->argc) {
case 4:
name = a->argv[3];
/* we accept a name in position 3, but keywords are not good. */
if (!strcasecmp(name, "peer") || !strcasecmp(name, "like"))
return CLI_SHOWUSAGE;
prunepeer = TRUE;
if (!strcasecmp(name, "all")) {
multi = TRUE;
name = NULL;
}
/* else a single name, already set */
break;
case 5:
/* sip prune realtime {peer|like} name */
name = a->argv[4];
if (!strcasecmp(a->argv[3], "peer"))
prunepeer = TRUE;
else if (!strcasecmp(a->argv[3], "like")) {
prunepeer = TRUE;
multi = TRUE;
} else
return CLI_SHOWUSAGE;
if (!strcasecmp(name, "like"))
return CLI_SHOWUSAGE;
if (!multi && !strcasecmp(name, "all")) {
multi = TRUE;
name = NULL;
}
break;
case 6:
name = a->argv[5];
multi = TRUE;
/* sip prune realtime {peer} like name */
if (strcasecmp(a->argv[4], "like"))
return CLI_SHOWUSAGE;
if (!strcasecmp(a->argv[3], "peer")) {
prunepeer = TRUE;
} else
return CLI_SHOWUSAGE;
break;
default:
return CLI_SHOWUSAGE;
}
if (multi && name) {
if (regcomp(&regexbuf, name, REG_EXTENDED | REG_NOSUB))
return CLI_SHOWUSAGE;
}
if (multi) {
if (prunepeer) {
int pruned = 0;
i = ao2_iterator_init(peers, 0);
while ((pi = ao2_t_iterator_next(&i, "iterate thru peers table"))) {
ao2_lock(pi);
if (name && regexec(&regexbuf, pi->name, 0, NULL, 0)) {
unref_peer(pi, "toss iterator peer ptr before continue");
ao2_unlock(pi);
continue;
};
if (ast_test_flag(&pi->flags[1], SIP_PAGE2_RTCACHEFRIENDS)) {
pi->the_mark = 1;
pruned++;
}
ao2_unlock(pi);
unref_peer(pi, "toss iterator peer ptr");
}
if (pruned) {
ao2_t_callback(peers, OBJ_NODATA | OBJ_UNLINK | OBJ_MULTIPLE, peer_is_marked, NULL,
"initiating callback to remove marked peers");
ast_cli(a->fd, "%d peers pruned.\n", pruned);
} else
ast_cli(a->fd, "No peers found to prune.\n");
}
} else {
if (prunepeer) {
struct sip_peer tmp;
ast_copy_string(tmp.name, name, sizeof(tmp.name));
if ((peer = ao2_t_find(peers, &tmp, OBJ_POINTER | OBJ_UNLINK, "finding to unlink from peers"))) {
if (peer->addr.sin_addr.s_addr) {
ao2_t_unlink(peers_by_ip, peer, "unlinking peer from peers_by_ip also");
}
if (!ast_test_flag(&peer->flags[1], SIP_PAGE2_RTCACHEFRIENDS)) {
ast_cli(a->fd, "Peer '%s' is not a Realtime peer, cannot be pruned.\n", name);
/* put it back! */
ao2_t_link(peers, peer, "link peer into peer table");
if (peer->addr.sin_addr.s_addr) {
ao2_t_link(peers_by_ip, peer, "link peer into peers_by_ip table");
}
} else
ast_cli(a->fd, "Peer '%s' pruned.\n", name);
unref_peer(peer, "sip_prune_realtime: unref_peer: tossing temp peer ptr");
} else
ast_cli(a->fd, "Peer '%s' not found.\n", name);
}
}
return CLI_SUCCESS;
}
/*! \brief Print codec list from preference to CLI/manager */
static void print_codec_to_cli(int fd, struct ast_codec_pref *pref)
{
int x, codec;
for(x = 0; x < 32 ; x++) {
codec = ast_codec_pref_index(pref, x);
if (!codec)
break;
ast_cli(fd, "%s", ast_getformatname(codec));
ast_cli(fd, ":%d", pref->framing[x]);
if (x < 31 && ast_codec_pref_index(pref, x + 1))
ast_cli(fd, ",");
}
if (!x)
ast_cli(fd, "none");
}
/*! \brief Print domain mode to cli */
static const char *domain_mode_to_text(const enum domain_mode mode)
{
switch (mode) {
case SIP_DOMAIN_AUTO:
return "[Automatic]";
case SIP_DOMAIN_CONFIG:
return "[Configured]";
}
return "";
}
/*! \brief CLI command to list local domains */
static char *sip_show_domains(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
struct domain *d;
#define FORMAT "%-40.40s %-20.20s %-16.16s\n"
switch (cmd) {
case CLI_INIT:
e->command = "sip show domains";
e->usage =
"Usage: sip show domains\n"
" Lists all configured SIP local domains.\n"
" Asterisk only responds to SIP messages to local domains.\n";
return NULL;
case CLI_GENERATE:
return NULL;
}
if (AST_LIST_EMPTY(&domain_list)) {
ast_cli(a->fd, "SIP Domain support not enabled.\n\n");
return CLI_SUCCESS;
} else {
ast_cli(a->fd, FORMAT, "Our local SIP domains:", "Context", "Set by");
AST_LIST_LOCK(&domain_list);
AST_LIST_TRAVERSE(&domain_list, d, list)
ast_cli(a->fd, FORMAT, d->domain, S_OR(d->context, "(default)"),
domain_mode_to_text(d->mode));
AST_LIST_UNLOCK(&domain_list);
ast_cli(a->fd, "\n");
return CLI_SUCCESS;
}
}
#undef FORMAT
static char mandescr_show_peer[] =
"Description: Show one SIP peer with details on current status.\n"
"Variables: \n"
" Peer: <name> The peer name you want to check.\n"
" ActionID: <id> Optional action ID for this AMI transaction.\n";
/*! \brief Show SIP peers in the manager API */
static int manager_sip_show_peer(struct mansession *s, const struct message *m)
{
const char *a[4];
const char *peer;
peer = astman_get_header(m, "Peer");
if (ast_strlen_zero(peer)) {
astman_send_error(s, m, "Peer: <name> missing.");
return 0;
}
a[0] = "sip";
a[1] = "show";
a[2] = "peer";
a[3] = peer;
_sip_show_peer(1, -1, s, m, 4, a);
astman_append(s, "\r\n\r\n" );
return 0;
}
/*! \brief Show one peer in detail */
static char *sip_show_peer(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
switch (cmd) {
case CLI_INIT:
e->command = "sip show peer";
e->usage =
"Usage: sip show peer <name> [load]\n"
" Shows all details on one SIP peer and the current status.\n"
" Option \"load\" forces lookup of peer in realtime storage.\n";
return NULL;
case CLI_GENERATE:
return complete_sip_show_peer(a->line, a->word, a->pos, a->n);
}
return _sip_show_peer(0, a->fd, NULL, NULL, a->argc, (const char **) a->argv);
}
/*! \brief Send qualify message to peer from cli or manager. Mostly for debugging. */
static char *_sip_qualify_peer(int type, int fd, struct mansession *s, const struct message *m, int argc, const char *argv[])
{
struct sip_peer *peer;
int load_realtime;
if (argc < 4)
return CLI_SHOWUSAGE;
load_realtime = (argc == 5 && !strcmp(argv[4], "load")) ? TRUE : FALSE;
if ((peer = find_peer(argv[3], NULL, load_realtime, FINDPEERS, FALSE))) {
sip_poke_peer(peer, 1);
unref_peer(peer, "qualify: done with peer");
} else if (type == 0) {
ast_cli(fd, "Peer '%s' not found\n", argv[3]);
} else {
astman_send_error(s, m, "Peer not found");
}
return CLI_SUCCESS;
}
/*! \brief Qualify SIP peers in the manager API */
static int manager_sip_qualify_peer(struct mansession *s, const struct message *m)
{
const char *a[4];
const char *peer;
peer = astman_get_header(m, "Peer");
if (ast_strlen_zero(peer)) {
astman_send_error(s, m, "Peer: <name> missing.");
return 0;
}
a[0] = "sip";
a[1] = "qualify";
a[2] = "peer";
a[3] = peer;
_sip_qualify_peer(1, -1, s, m, 4, a);
astman_append(s, "\r\n\r\n" );
return 0;
}
/*! \brief Send an OPTIONS packet to a SIP peer */
static char *sip_qualify_peer(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
switch (cmd) {
case CLI_INIT:
e->command = "sip qualify peer";
e->usage =
"Usage: sip qualify peer <name> [load]\n"
" Requests a response from one SIP peer and the current status.\n"
" Option \"load\" forces lookup of peer in realtime storage.\n";
return NULL;
case CLI_GENERATE:
return complete_sip_show_peer(a->line, a->word, a->pos, a->n);
}
return _sip_qualify_peer(0, a->fd, NULL, NULL, a->argc, (const char **) a->argv);
}
/*! \brief list peer mailboxes to CLI */
static void peer_mailboxes_to_str(struct ast_str **mailbox_str, struct sip_peer *peer)
{
struct sip_mailbox *mailbox;
AST_LIST_TRAVERSE(&peer->mailboxes, mailbox, entry) {
ast_str_append(mailbox_str, 0, "%s%s%s%s",
mailbox->mailbox,
ast_strlen_zero(mailbox->context) ? "" : "@",
S_OR(mailbox->context, ""),
AST_LIST_NEXT(mailbox, entry) ? "," : "");
}
}
/*! \brief Show one peer in detail (main function) */
static char *_sip_show_peer(int type, int fd, struct mansession *s, const struct message *m, int argc, const char *argv[])
{
char status[30] = "";
char cbuf[256];
struct sip_peer *peer;
char codec_buf[512];
struct ast_codec_pref *pref;
struct ast_variable *v;
struct sip_auth *auth;
int x = 0, codec = 0, load_realtime;
int realtimepeers;
realtimepeers = ast_check_realtime("sippeers");
if (argc < 4)
return CLI_SHOWUSAGE;
load_realtime = (argc == 5 && !strcmp(argv[4], "load")) ? TRUE : FALSE;
peer = find_peer(argv[3], NULL, load_realtime, FINDPEERS, FALSE);
if (s) { /* Manager */
if (peer) {
const char *id = astman_get_header(m, "ActionID");
astman_append(s, "Response: Success\r\n");
if (!ast_strlen_zero(id))
astman_append(s, "ActionID: %s\r\n", id);
} else {
snprintf (cbuf, sizeof(cbuf), "Peer %s not found.", argv[3]);
astman_send_error(s, m, cbuf);
return CLI_SUCCESS;
}
}
if (peer && type==0 ) { /* Normal listing */
struct ast_str *mailbox_str = ast_str_alloca(512);
ast_cli(fd, "\n\n");
ast_cli(fd, " * Name : %s\n", peer->name);
if (realtimepeers) { /* Realtime is enabled */
ast_cli(fd, " Realtime peer: %s\n", peer->is_realtime ? "Yes, cached" : "No");
}
ast_cli(fd, " Secret : %s\n", ast_strlen_zero(peer->secret)?"<Not set>":"<Set>");
ast_cli(fd, " MD5Secret : %s\n", ast_strlen_zero(peer->md5secret)?"<Not set>":"<Set>");
ast_cli(fd, " Remote Secret: %s\n", ast_strlen_zero(peer->remotesecret)?"<Not set>":"<Set>");
for (auth = peer->auth; auth; auth = auth->next) {
ast_cli(fd, " Realm-auth : Realm %-15.15s User %-10.20s ", auth->realm, auth->username);
ast_cli(fd, "%s\n", !ast_strlen_zero(auth->secret)?"<Secret set>":(!ast_strlen_zero(auth->md5secret)?"<MD5secret set>" : "<Not set>"));
}
ast_cli(fd, " Context : %s\n", peer->context);
ast_cli(fd, " Subscr.Cont. : %s\n", S_OR(peer->subscribecontext, "<Not set>") );
ast_cli(fd, " Language : %s\n", peer->language);
if (!ast_strlen_zero(peer->accountcode))
ast_cli(fd, " Accountcode : %s\n", peer->accountcode);
ast_cli(fd, " AMA flags : %s\n", ast_cdr_flags2str(peer->amaflags));
ast_cli(fd, " Transfer mode: %s\n", transfermode2str(peer->allowtransfer));
ast_cli(fd, " CallingPres : %s\n", ast_describe_caller_presentation(peer->callingpres));
if (!ast_strlen_zero(peer->fromuser))
ast_cli(fd, " FromUser : %s\n", peer->fromuser);
if (!ast_strlen_zero(peer->fromdomain))
ast_cli(fd, " FromDomain : %s\n", peer->fromdomain);
ast_cli(fd, " Callgroup : ");
print_group(fd, peer->callgroup, 0);
ast_cli(fd, " Pickupgroup : ");
print_group(fd, peer->pickupgroup, 0);
peer_mailboxes_to_str(&mailbox_str, peer);
ast_cli(fd, " Mailbox : %s\n", mailbox_str->str);
ast_cli(fd, " VM Extension : %s\n", peer->vmexten);
ast_cli(fd, " LastMsgsSent : %d/%d\n", (peer->lastmsgssent & 0x7fff0000) >> 16, peer->lastmsgssent & 0xffff);
ast_cli(fd, " Call limit : %d\n", peer->call_limit);
if (peer->busy_level)
ast_cli(fd, " Busy level : %d\n", peer->busy_level);
ast_cli(fd, " Dynamic : %s\n", cli_yesno(peer->host_dynamic));
ast_cli(fd, " Callerid : %s\n", ast_callerid_merge(cbuf, sizeof(cbuf), peer->cid_name, peer->cid_num, "<unspecified>"));
ast_cli(fd, " MaxCallBR : %d kbps\n", peer->maxcallbitrate);
ast_cli(fd, " Expire : %ld\n", ast_sched_when(sched, peer->expire));
ast_cli(fd, " Insecure : %s\n", insecure2str(ast_test_flag(&peer->flags[0], SIP_INSECURE)));
ast_cli(fd, " Nat : %s\n", nat2str(ast_test_flag(&peer->flags[0], SIP_NAT)));
ast_cli(fd, " ACL : %s\n", cli_yesno(peer->ha != NULL));
ast_cli(fd, " T38 pt UDPTL : %s\n", cli_yesno(ast_test_flag(&peer->flags[1], SIP_PAGE2_T38SUPPORT_UDPTL)));
#ifdef WHEN_WE_HAVE_T38_FOR_OTHER_TRANSPORTS
ast_cli(fd, " T38 pt RTP : %s\n", cli_yesno(ast_test_flag(&peer->flags[1], SIP_PAGE2_T38SUPPORT_RTP)));
ast_cli(fd, " T38 pt TCP : %s\n", cli_yesno(ast_test_flag(&peer->flags[1], SIP_PAGE2_T38SUPPORT_TCP)));
#endif
ast_cli(fd, " CanReinvite : %s\n", cli_yesno(ast_test_flag(&peer->flags[0], SIP_CAN_REINVITE)));
ast_cli(fd, " PromiscRedir : %s\n", cli_yesno(ast_test_flag(&peer->flags[0], SIP_PROMISCREDIR)));
ast_cli(fd, " User=Phone : %s\n", cli_yesno(ast_test_flag(&peer->flags[0], SIP_USEREQPHONE)));
ast_cli(fd, " Video Support: %s\n", cli_yesno(ast_test_flag(&peer->flags[1], SIP_PAGE2_VIDEOSUPPORT)));
ast_cli(fd, " Text Support : %s\n", cli_yesno(ast_test_flag(&peer->flags[1], SIP_PAGE2_TEXTSUPPORT)));
ast_cli(fd, " Ign SDP ver : %s\n", cli_yesno(ast_test_flag(&peer->flags[1], SIP_PAGE2_IGNORESDPVERSION)));
ast_cli(fd, " Trust RPID : %s\n", cli_yesno(ast_test_flag(&peer->flags[0], SIP_TRUSTRPID)));
ast_cli(fd, " Send RPID : %s\n", cli_yesno(ast_test_flag(&peer->flags[0], SIP_SENDRPID)));
ast_cli(fd, " Subscriptions: %s\n", cli_yesno(ast_test_flag(&peer->flags[1], SIP_PAGE2_ALLOWSUBSCRIBE)));
ast_cli(fd, " Overlap dial : %s\n", cli_yesno(ast_test_flag(&peer->flags[1], SIP_PAGE2_ALLOWOVERLAP)));
if (peer->outboundproxy)
ast_cli(fd, " Outb. proxy : %s %s\n", ast_strlen_zero(peer->outboundproxy->name) ? "<not set>" : peer->outboundproxy->name,
peer->outboundproxy->force ? "(forced)" : "");
/* - is enumerated */
ast_cli(fd, " DTMFmode : %s\n", dtmfmode2str(ast_test_flag(&peer->flags[0], SIP_DTMF)));
ast_cli(fd, " Timer T1 : %d\n", peer->timer_t1);
ast_cli(fd, " Timer B : %d\n", peer->timer_b);
ast_cli(fd, " ToHost : %s\n", peer->tohost);
ast_cli(fd, " Addr->IP : %s Port %d\n", peer->addr.sin_addr.s_addr ? ast_inet_ntoa(peer->addr.sin_addr) : "(Unspecified)", ntohs(peer->addr.sin_port));
ast_cli(fd, " Defaddr->IP : %s Port %d\n", ast_inet_ntoa(peer->defaddr.sin_addr), ntohs(peer->defaddr.sin_port));
ast_cli(fd, " Prim.Transp. : %s\n", get_transport(peer->socket.type));
ast_cli(fd, " Allowed.Trsp : %s\n", get_transport_list(peer->transports));
if (!ast_strlen_zero(global_regcontext))
ast_cli(fd, " Reg. exten : %s\n", peer->regexten);
ast_cli(fd, " Def. Username: %s\n", peer->username);
ast_cli(fd, " SIP Options : ");
if (peer->sipoptions) {
int lastoption = -1;
for (x = 0 ; x < ARRAY_LEN(sip_options); x++) {
if (sip_options[x].id != lastoption) {
if (peer->sipoptions & sip_options[x].id)
ast_cli(fd, "%s ", sip_options[x].text);
lastoption = x;
}
}
} else
ast_cli(fd, "(none)");
ast_cli(fd, "\n");
ast_cli(fd, " Codecs : ");
ast_getformatname_multiple(codec_buf, sizeof(codec_buf) -1, peer->capability);
ast_cli(fd, "%s\n", codec_buf);
ast_cli(fd, " Codec Order : (");
print_codec_to_cli(fd, &peer->prefs);
ast_cli(fd, ")\n");
ast_cli(fd, " Auto-Framing : %s \n", cli_yesno(peer->autoframing));
ast_cli(fd, " 100 on REG : %s\n", ast_test_flag(&peer->flags[1], SIP_PAGE2_REGISTERTRYING) ? "Yes" : "No");
ast_cli(fd, " Status : ");
peer_status(peer, status, sizeof(status));
ast_cli(fd, "%s\n", status);
ast_cli(fd, " Useragent : %s\n", peer->useragent);
ast_cli(fd, " Reg. Contact : %s\n", peer->fullcontact);
ast_cli(fd, " Qualify Freq : %d ms\n", peer->qualifyfreq);
if (peer->chanvars) {
ast_cli(fd, " Variables :\n");
for (v = peer->chanvars ; v ; v = v->next)
ast_cli(fd, " %s = %s\n", v->name, v->value);
}
ast_cli(fd, " Sess-Timers : %s\n", stmode2str(peer->stimer.st_mode_oper));
ast_cli(fd, " Sess-Refresh : %s\n", strefresher2str(peer->stimer.st_ref));
ast_cli(fd, " Sess-Expires : %d secs\n", peer->stimer.st_max_se);
ast_cli(fd, " Min-Sess : %d secs\n", peer->stimer.st_min_se);
ast_cli(fd, " RTP Engine : %s\n", peer->engine);
ast_cli(fd, "\n");
peer = unref_peer(peer, "sip_show_peer: unref_peer: done with peer ptr");
} else if (peer && type == 1) { /* manager listing */
char buffer[256];
struct ast_str *mailbox_str = ast_str_alloca(512);
astman_append(s, "Channeltype: SIP\r\n");
astman_append(s, "ObjectName: %s\r\n", peer->name);
astman_append(s, "ChanObjectType: peer\r\n");
astman_append(s, "SecretExist: %s\r\n", ast_strlen_zero(peer->secret)?"N":"Y");
astman_append(s, "RemoteSecretExist: %s\r\n", ast_strlen_zero(peer->remotesecret)?"N":"Y");
astman_append(s, "MD5SecretExist: %s\r\n", ast_strlen_zero(peer->md5secret)?"N":"Y");
astman_append(s, "Context: %s\r\n", peer->context);
astman_append(s, "Language: %s\r\n", peer->language);
if (!ast_strlen_zero(peer->accountcode))
astman_append(s, "Accountcode: %s\r\n", peer->accountcode);
astman_append(s, "AMAflags: %s\r\n", ast_cdr_flags2str(peer->amaflags));
astman_append(s, "CID-CallingPres: %s\r\n", ast_describe_caller_presentation(peer->callingpres));
if (!ast_strlen_zero(peer->fromuser))
astman_append(s, "SIP-FromUser: %s\r\n", peer->fromuser);
if (!ast_strlen_zero(peer->fromdomain))
astman_append(s, "SIP-FromDomain: %s\r\n", peer->fromdomain);
astman_append(s, "Callgroup: ");
astman_append(s, "%s\r\n", ast_print_group(buffer, sizeof(buffer), peer->callgroup));
astman_append(s, "Pickupgroup: ");
astman_append(s, "%s\r\n", ast_print_group(buffer, sizeof(buffer), peer->pickupgroup));
peer_mailboxes_to_str(&mailbox_str, peer);
astman_append(s, "VoiceMailbox: %s\r\n", mailbox_str->str);
astman_append(s, "TransferMode: %s\r\n", transfermode2str(peer->allowtransfer));
astman_append(s, "LastMsgsSent: %d\r\n", peer->lastmsgssent);
astman_append(s, "Call-limit: %d\r\n", peer->call_limit);
astman_append(s, "Busy-level: %d\r\n", peer->busy_level);
astman_append(s, "MaxCallBR: %d kbps\r\n", peer->maxcallbitrate);
astman_append(s, "Dynamic: %s\r\n", peer->host_dynamic?"Y":"N");
astman_append(s, "Callerid: %s\r\n", ast_callerid_merge(cbuf, sizeof(cbuf), peer->cid_name, peer->cid_num, ""));
astman_append(s, "RegExpire: %ld seconds\r\n", ast_sched_when(sched, peer->expire));
astman_append(s, "SIP-AuthInsecure: %s\r\n", insecure2str(ast_test_flag(&peer->flags[0], SIP_INSECURE)));
astman_append(s, "SIP-NatSupport: %s\r\n", nat2str(ast_test_flag(&peer->flags[0], SIP_NAT)));
astman_append(s, "ACL: %s\r\n", (peer->ha?"Y":"N"));
astman_append(s, "SIP-CanReinvite: %s\r\n", (ast_test_flag(&peer->flags[0], SIP_CAN_REINVITE)?"Y":"N"));
astman_append(s, "SIP-PromiscRedir: %s\r\n", (ast_test_flag(&peer->flags[0], SIP_PROMISCREDIR)?"Y":"N"));
astman_append(s, "SIP-UserPhone: %s\r\n", (ast_test_flag(&peer->flags[0], SIP_USEREQPHONE)?"Y":"N"));
astman_append(s, "SIP-VideoSupport: %s\r\n", (ast_test_flag(&peer->flags[1], SIP_PAGE2_VIDEOSUPPORT)?"Y":"N"));
astman_append(s, "SIP-TextSupport: %s\r\n", (ast_test_flag(&peer->flags[1], SIP_PAGE2_TEXTSUPPORT)?"Y":"N"));
astman_append(s, "SIP-Sess-Timers: %s\r\n", stmode2str(peer->stimer.st_mode_oper));
astman_append(s, "SIP-Sess-Refresh: %s\r\n", strefresher2str(peer->stimer.st_ref));
astman_append(s, "SIP-Sess-Expires: %d\r\n", peer->stimer.st_max_se);
astman_append(s, "SIP-Sess-Min: %d\r\n", peer->stimer.st_min_se);
astman_append(s, "SIP-RTP-Engine: %s\r\n", peer->engine);
/* - is enumerated */
astman_append(s, "SIP-DTMFmode: %s\r\n", dtmfmode2str(ast_test_flag(&peer->flags[0], SIP_DTMF)));
astman_append(s, "ToHost: %s\r\n", peer->tohost);
astman_append(s, "Address-IP: %s\r\nAddress-Port: %d\r\n", peer->addr.sin_addr.s_addr ? ast_inet_ntoa(peer->addr.sin_addr) : "", ntohs(peer->addr.sin_port));
astman_append(s, "Default-addr-IP: %s\r\nDefault-addr-port: %d\r\n", ast_inet_ntoa(peer->defaddr.sin_addr), ntohs(peer->defaddr.sin_port));
astman_append(s, "Default-Username: %s\r\n", peer->username);
if (!ast_strlen_zero(global_regcontext))
astman_append(s, "RegExtension: %s\r\n", peer->regexten);
astman_append(s, "Codecs: ");
ast_getformatname_multiple(codec_buf, sizeof(codec_buf) -1, peer->capability);
astman_append(s, "%s\r\n", codec_buf);
astman_append(s, "CodecOrder: ");
pref = &peer->prefs;
for(x = 0; x < 32 ; x++) {
codec = ast_codec_pref_index(pref, x);
if (!codec)
break;
astman_append(s, "%s", ast_getformatname(codec));
if (x < 31 && ast_codec_pref_index(pref, x+1))
astman_append(s, ",");
}
astman_append(s, "\r\n");
astman_append(s, "Status: ");
peer_status(peer, status, sizeof(status));
astman_append(s, "%s\r\n", status);
astman_append(s, "SIP-Useragent: %s\r\n", peer->useragent);
astman_append(s, "Reg-Contact: %s\r\n", peer->fullcontact);
astman_append(s, "QualifyFreq: %d ms\r\n", peer->qualifyfreq);
if (peer->chanvars) {
for (v = peer->chanvars ; v ; v = v->next) {
astman_append(s, "ChanVariable: %s=%s\r\n", v->name, v->value);
}
}
peer = unref_peer(peer, "sip_show_peer: unref_peer: done with peer");
} else {
ast_cli(fd, "Peer %s not found.\n", argv[3]);
ast_cli(fd, "\n");
}
return CLI_SUCCESS;
}
/*! \brief Do completion on user name */
static char *complete_sip_user(const char *word, int state)
{
char *result = NULL;
int wordlen = strlen(word);
int which = 0;
struct ao2_iterator user_iter;
struct sip_peer *user;
user_iter = ao2_iterator_init(peers, 0);
while ((user = ao2_iterator_next(&user_iter))) {
ao2_lock(user);
if (!(user->type & SIP_TYPE_USER)) {
ao2_unlock(user);
unref_peer(user, "complete sip user");
continue;
}
/* locking of the object is not required because only the name and flags are being compared */
if (!strncasecmp(word, user->name, wordlen) && ++which > state) {
result = ast_strdup(user->name);
}
ao2_unlock(user);
unref_peer(user, "complete sip user");
}
return result;
}
/*! \brief Support routine for 'sip show user' CLI */
static char *complete_sip_show_user(const char *line, const char *word, int pos, int state)
{
if (pos == 3)
return complete_sip_user(word, state);
return NULL;
}
/*! \brief Show one user in detail */
static char *sip_show_user(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
char cbuf[256];
struct sip_peer *user;
struct ast_variable *v;
int load_realtime;
switch (cmd) {
case CLI_INIT:
e->command = "sip show user";
e->usage =
"Usage: sip show user <name> [load]\n"
" Shows all details on one SIP user and the current status.\n"
" Option \"load\" forces lookup of peer in realtime storage.\n";
return NULL;
case CLI_GENERATE:
return complete_sip_show_user(a->line, a->word, a->pos, a->n);
}
if (a->argc < 4)
return CLI_SHOWUSAGE;
/* Load from realtime storage? */
load_realtime = (a->argc == 5 && !strcmp(a->argv[4], "load")) ? TRUE : FALSE;
if ((user = find_peer(a->argv[3], NULL, load_realtime, FINDUSERS, FALSE))) {
ao2_lock(user);
ast_cli(a->fd, "\n\n");
ast_cli(a->fd, " * Name : %s\n", user->name);
ast_cli(a->fd, " Secret : %s\n", ast_strlen_zero(user->secret)?"<Not set>":"<Set>");
ast_cli(a->fd, " MD5Secret : %s\n", ast_strlen_zero(user->md5secret)?"<Not set>":"<Set>");
ast_cli(a->fd, " Context : %s\n", user->context);
ast_cli(a->fd, " Language : %s\n", user->language);
if (!ast_strlen_zero(user->accountcode))
ast_cli(a->fd, " Accountcode : %s\n", user->accountcode);
ast_cli(a->fd, " AMA flags : %s\n", ast_cdr_flags2str(user->amaflags));
ast_cli(a->fd, " Transfer mode: %s\n", transfermode2str(user->allowtransfer));
ast_cli(a->fd, " MaxCallBR : %d kbps\n", user->maxcallbitrate);
ast_cli(a->fd, " CallingPres : %s\n", ast_describe_caller_presentation(user->callingpres));
ast_cli(a->fd, " Call limit : %d\n", user->call_limit);
ast_cli(a->fd, " Callgroup : ");
print_group(a->fd, user->callgroup, 0);
ast_cli(a->fd, " Pickupgroup : ");
print_group(a->fd, user->pickupgroup, 0);
ast_cli(a->fd, " Callerid : %s\n", ast_callerid_merge(cbuf, sizeof(cbuf), user->cid_name, user->cid_num, "<unspecified>"));
ast_cli(a->fd, " ACL : %s\n", cli_yesno(user->ha != NULL));
ast_cli(a->fd, " Sess-Timers : %s\n", stmode2str(user->stimer.st_mode_oper));
ast_cli(a->fd, " Sess-Refresh : %s\n", strefresher2str(user->stimer.st_ref));
ast_cli(a->fd, " Sess-Expires : %d secs\n", user->stimer.st_max_se);
ast_cli(a->fd, " Sess-Min-SE : %d secs\n", user->stimer.st_min_se);
ast_cli(a->fd, " RTP Engine : %s\n", user->engine);
ast_cli(a->fd, " Codec Order : (");
print_codec_to_cli(a->fd, &user->prefs);
ast_cli(a->fd, ")\n");
ast_cli(a->fd, " Auto-Framing: %s \n", cli_yesno(user->autoframing));
if (user->chanvars) {
ast_cli(a->fd, " Variables :\n");
for (v = user->chanvars ; v ; v = v->next)
ast_cli(a->fd, " %s = %s\n", v->name, v->value);
}
ast_cli(a->fd, "\n");
ao2_unlock(user);
unref_peer(user, "sip show user");
} else {
ast_cli(a->fd, "User %s not found.\n", a->argv[3]);
ast_cli(a->fd, "\n");
}
return CLI_SUCCESS;
}
static char *sip_show_sched(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
struct ast_str *cbuf;
struct ast_cb_names cbnames = {9, { "retrans_pkt",
"__sip_autodestruct",
"expire_register",
"auto_congest",
"sip_reg_timeout",
"sip_poke_peer_s",
"sip_poke_noanswer",
"sip_reregister",
"sip_reinvite_retry"},
{ retrans_pkt,
__sip_autodestruct,
expire_register,
auto_congest,
sip_reg_timeout,
sip_poke_peer_s,
sip_poke_noanswer,
sip_reregister,
sip_reinvite_retry}};
switch (cmd) {
case CLI_INIT:
e->command = "sip show sched";
e->usage =
"Usage: sip show sched\n"
" Shows stats on what's in the sched queue at the moment\n";
return NULL;
case CLI_GENERATE:
return NULL;
}
cbuf = ast_str_alloca(2048);
ast_cli(a->fd, "\n");
ast_sched_report(sched, &cbuf, &cbnames);
ast_cli(a->fd, "%s", cbuf->str);
return CLI_SUCCESS;
}
/*! \brief Show SIP Registry (registrations with other SIP proxies */
static char *sip_show_registry(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
#define FORMAT2 "%-30.30s %-6.6s %-12.12s %8.8s %-20.20s %-25.25s\n"
#define FORMAT "%-30.30s %-6.6s %-12.12s %8d %-20.20s %-25.25s\n"
char host[80];
char tmpdat[256];
struct ast_tm tm;
int counter = 0;
switch (cmd) {
case CLI_INIT:
e->command = "sip show registry";
e->usage =
"Usage: sip show registry\n"
" Lists all registration requests and status.\n";
return NULL;
case CLI_GENERATE:
return NULL;
}
if (a->argc != 3)
return CLI_SHOWUSAGE;
ast_cli(a->fd, FORMAT2, "Host", "dnsmgr", "Username", "Refresh", "State", "Reg.Time");
ASTOBJ_CONTAINER_TRAVERSE(&regl, 1, do {
ASTOBJ_RDLOCK(iterator);
snprintf(host, sizeof(host), "%s:%d", iterator->hostname, iterator->portno ? iterator->portno : STANDARD_SIP_PORT);
if (iterator->regtime.tv_sec) {
ast_localtime(&iterator->regtime, &tm, NULL);
ast_strftime(tmpdat, sizeof(tmpdat), "%a, %d %b %Y %T", &tm);
} else
tmpdat[0] = '\0';
ast_cli(a->fd, FORMAT, host, (iterator->dnsmgr) ? "Y" : "N", iterator->username, iterator->refresh, regstate2str(iterator->regstate), tmpdat);
ASTOBJ_UNLOCK(iterator);
counter++;
} while(0));
ast_cli(a->fd, "%d SIP registrations.\n", counter);
return CLI_SUCCESS;
#undef FORMAT
#undef FORMAT2
}
/*! \brief Unregister (force expiration) a SIP peer in the registry via CLI
\note This function does not tell the SIP device what's going on,
so use it with great care.
*/
static char *sip_unregister(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
struct sip_peer *peer;
int load_realtime = 0;
switch (cmd) {
case CLI_INIT:
e->command = "sip unregister";
e->usage =
"Usage: sip unregister <peer>\n"
" Unregister (force expiration) a SIP peer from the registry\n";
return NULL;
case CLI_GENERATE:
return complete_sip_unregister(a->line, a->word, a->pos, a->n);
}
if (a->argc != 3)
return CLI_SHOWUSAGE;
if ((peer = find_peer(a->argv[2], NULL, load_realtime, FINDPEERS, TRUE))) {
if (peer->expire > 0) {
expire_register(ref_peer(peer, "ref for expire_register"));
ast_cli(a->fd, "Unregistered peer \'%s\'\n\n", a->argv[2]);
} else {
ast_cli(a->fd, "Peer %s not registered\n", a->argv[2]);
}
unref_peer(peer, "sip_unregister: unref_peer via sip_unregister: done with peer from find_peer call");
} else {
ast_cli(a->fd, "Peer unknown: \'%s\'. Not unregistered.\n", a->argv[2]);
}
return CLI_SUCCESS;
}
/*! \brief Callback for show_chanstats */
static int show_chanstats_cb(void *__cur, void *__arg, int flags)
{
#define FORMAT2 "%-15.15s %-11.11s %-8.8s %-10.10s %-10.10s (%-2.2s) %-6.6s %-10.10s %-10.10s ( %%) %-6.6s\n"
#define FORMAT "%-15.15s %-11.11s %-8.8s %-10.10u%-1.1s %-10.10u (%-2.2u%%) %-6.6u %-10.10u%-1.1s %-10.10u (%-2.2u%%) %-6.6u\n"
struct sip_pvt *cur = __cur;
struct ast_rtp_instance_stats stats;
char durbuf[10];
int duration;
int durh, durm, durs;
struct ast_channel *c = cur->owner;
struct __show_chan_arg *arg = __arg;
int fd = arg->fd;
if (cur->subscribed != NONE) /* Subscriptions */
return 0; /* don't care, we scan all channels */
if (!cur->rtp) {
if (sipdebug)
ast_cli(fd, "%-15.15s %-11.11s (inv state: %s) -- %s\n", ast_inet_ntoa(cur->sa.sin_addr), cur->callid, invitestate2string[cur->invitestate].desc, "-- No RTP active");
return 0; /* don't care, we scan all channels */
}
ast_rtp_instance_get_stats(cur->rtp, &stats, AST_RTP_INSTANCE_STAT_ALL);
if (c && c->cdr && !ast_tvzero(c->cdr->start)) {
duration = (int)(ast_tvdiff_ms(ast_tvnow(), c->cdr->start) / 1000);
durh = duration / 3600;
durm = (duration % 3600) / 60;
durs = duration % 60;
snprintf(durbuf, sizeof(durbuf), "%02d:%02d:%02d", durh, durm, durs);
} else {
durbuf[0] = '\0';
}
ast_cli(fd, FORMAT,
ast_inet_ntoa(cur->sa.sin_addr),
cur->callid,
durbuf,
stats.rxcount > (unsigned int) 100000 ? (unsigned int) (stats.rxcount)/(unsigned int) 1000 : stats.rxcount,
stats.rxcount > (unsigned int) 100000 ? "K":" ",
stats.rxploss,
stats.rxcount > stats.rxploss ? (stats.rxploss / stats.rxcount * 100) : 0,
stats.rxjitter,
stats.txcount > (unsigned int) 100000 ? (unsigned int) (stats.txcount)/(unsigned int) 1000 : stats.txcount,
stats.txcount > (unsigned int) 100000 ? "K":" ",
stats.txploss,
stats.txcount > stats.txploss ? (stats.txploss / stats.txcount * 100) : 0,
stats.txjitter
);
arg->numchans++;
return 0; /* don't care, we scan all channels */
}
/*! \brief SIP show channelstats CLI (main function) */
static char *sip_show_channelstats(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
struct __show_chan_arg arg = { .fd = a->fd, .numchans = 0 };
switch (cmd) {
case CLI_INIT:
e->command = "sip show channelstats";
e->usage =
"Usage: sip show channelstats\n"
" Lists all currently active SIP channel's RTCP statistics.\n"
" Note that calls in the much optimized RTP P2P bridge mode will not show any packets here.";
return NULL;
case CLI_GENERATE:
return NULL;
}
if (a->argc != 3)
return CLI_SHOWUSAGE;
ast_cli(a->fd, FORMAT2, "Peer", "Call ID", "Duration", "Recv: Pack", "Lost", "%", "Jitter", "Send: Pack", "Lost", "Jitter");
/* iterate on the container and invoke the callback on each item */
ao2_t_callback(dialogs, OBJ_NODATA, show_chanstats_cb, &arg, "callback to sip show chanstats");
ast_cli(a->fd, "%d active SIP channel%s\n", arg.numchans, (arg.numchans != 1) ? "s" : "");
return CLI_SUCCESS;
}
#undef FORMAT
#undef FORMAT2
/*! \brief List global settings for the SIP channel */
static char *sip_show_settings(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
int realtimepeers;
int realtimeregs;
char codec_buf[SIPBUFSIZE];
const char *msg; /* temporary msg pointer */
switch (cmd) {
case CLI_INIT:
e->command = "sip show settings";
e->usage =
"Usage: sip show settings\n"
" Provides detailed list of the configuration of the SIP channel.\n";
return NULL;
case CLI_GENERATE:
return NULL;
}
realtimepeers = ast_check_realtime("sippeers");
realtimeregs = ast_check_realtime("sipregs");
if (a->argc != 3)
return CLI_SHOWUSAGE;
ast_cli(a->fd, "\n\nGlobal Settings:\n");
ast_cli(a->fd, "----------------\n");
ast_cli(a->fd, " UDP SIP Port: %d\n", ntohs(bindaddr.sin_port));
ast_cli(a->fd, " UDP Bindaddress: %s\n", ast_inet_ntoa(bindaddr.sin_addr));
ast_cli(a->fd, " TCP SIP Port: ");
if (sip_tcp_desc.local_address.sin_family == AF_INET) {
ast_cli(a->fd, "%d\n", ntohs(sip_tcp_desc.local_address.sin_port));
ast_cli(a->fd, " TCP Bindaddress: %s\n", ast_inet_ntoa(sip_tcp_desc.local_address.sin_addr));
} else {
ast_cli(a->fd, "Disabled\n");
}
ast_cli(a->fd, " TLS SIP Port: ");
if (default_tls_cfg.enabled != FALSE) {
ast_cli(a->fd, "%d\n", ntohs(sip_tls_desc.local_address.sin_port));
ast_cli(a->fd, " TLS Bindaddress: %s\n", ast_inet_ntoa(sip_tls_desc.local_address.sin_addr));
} else {
ast_cli(a->fd, "Disabled\n");
}
ast_cli(a->fd, " Videosupport: %s\n", cli_yesno(ast_test_flag(&global_flags[1], SIP_PAGE2_VIDEOSUPPORT)));
ast_cli(a->fd, " Textsupport: %s\n", cli_yesno(ast_test_flag(&global_flags[1], SIP_PAGE2_TEXTSUPPORT)));
ast_cli(a->fd, " Ignore SDP sess. ver.: %s\n", cli_yesno(ast_test_flag(&global_flags[1], SIP_PAGE2_IGNORESDPVERSION)));
ast_cli(a->fd, " AutoCreate Peer: %s\n", cli_yesno(sip_cfg.autocreatepeer));
ast_cli(a->fd, " Match Auth Username: %s\n", cli_yesno(global_match_auth_username));
ast_cli(a->fd, " Allow unknown access: %s\n", cli_yesno(sip_cfg.allowguest));
ast_cli(a->fd, " Allow subscriptions: %s\n", cli_yesno(ast_test_flag(&global_flags[1], SIP_PAGE2_ALLOWSUBSCRIBE)));
ast_cli(a->fd, " Allow overlap dialing: %s\n", cli_yesno(ast_test_flag(&global_flags[1], SIP_PAGE2_ALLOWOVERLAP)));
ast_cli(a->fd, " Allow promsic. redir: %s\n", cli_yesno(ast_test_flag(&global_flags[0], SIP_PROMISCREDIR)));
ast_cli(a->fd, " Enable call counters: %s\n", cli_yesno(global_callcounter));
ast_cli(a->fd, " SIP domain support: %s\n", cli_yesno(!AST_LIST_EMPTY(&domain_list)));
ast_cli(a->fd, " Realm. auth: %s\n", cli_yesno(authl != NULL));
ast_cli(a->fd, " Our auth realm %s\n", sip_cfg.realm);
ast_cli(a->fd, " Call to non-local dom.: %s\n", cli_yesno(sip_cfg.allow_external_domains));
ast_cli(a->fd, " URI user is phone no: %s\n", cli_yesno(ast_test_flag(&global_flags[0], SIP_USEREQPHONE)));
ast_cli(a->fd, " Always auth rejects: %s\n", cli_yesno(sip_cfg.alwaysauthreject));
ast_cli(a->fd, " Direct RTP setup: %s\n", cli_yesno(sip_cfg.directrtpsetup));
ast_cli(a->fd, " User Agent: %s\n", global_useragent);
ast_cli(a->fd, " SDP Session Name: %s\n", ast_strlen_zero(global_sdpsession) ? "-" : global_sdpsession);
ast_cli(a->fd, " SDP Owner Name: %s\n", ast_strlen_zero(global_sdpowner) ? "-" : global_sdpowner);
ast_cli(a->fd, " Reg. context: %s\n", S_OR(global_regcontext, "(not set)"));
ast_cli(a->fd, " Regexten on Qualify: %s\n", cli_yesno(sip_cfg.regextenonqualify));
ast_cli(a->fd, " Caller ID: %s\n", default_callerid);
ast_cli(a->fd, " From: Domain: %s\n", default_fromdomain);
ast_cli(a->fd, " Record SIP history: %s\n", recordhistory ? "On" : "Off");
ast_cli(a->fd, " Call Events: %s\n", sip_cfg.callevents ? "On" : "Off");
ast_cli(a->fd, " Auth. Failure Events: %s\n", global_authfailureevents ? "On" : "Off");
ast_cli(a->fd, " T38 fax pt UDPTL: %s\n", cli_yesno(ast_test_flag(&global_flags[1], SIP_PAGE2_T38SUPPORT_UDPTL)));
#ifdef WHEN_WE_HAVE_T38_FOR_OTHER_TRANSPORTS
ast_cli(a->fd, " T38 fax pt RTP: %s\n", cli_yesno(ast_test_flag(&global_flags[1], SIP_PAGE2_T38SUPPORT_RTP)));
ast_cli(a->fd, " T38 fax pt TCP: %s\n", cli_yesno(ast_test_flag(&global_flags[1], SIP_PAGE2_T38SUPPORT_TCP)));
#endif
if (!realtimepeers && !realtimeregs)
ast_cli(a->fd, " SIP realtime: Disabled\n" );
else
ast_cli(a->fd, " SIP realtime: Enabled\n" );
ast_cli(a->fd, " Qualify Freq : %d ms\n", global_qualifyfreq);
ast_cli(a->fd, "\nNetwork QoS Settings:\n");
ast_cli(a->fd, "---------------------------\n");
ast_cli(a->fd, " IP ToS SIP: %s\n", ast_tos2str(global_tos_sip));
ast_cli(a->fd, " IP ToS RTP audio: %s\n", ast_tos2str(global_tos_audio));
ast_cli(a->fd, " IP ToS RTP video: %s\n", ast_tos2str(global_tos_video));
ast_cli(a->fd, " IP ToS RTP text: %s\n", ast_tos2str(global_tos_text));
ast_cli(a->fd, " 802.1p CoS SIP: %d\n", global_cos_sip);
ast_cli(a->fd, " 802.1p CoS RTP audio: %d\n", global_cos_audio);
ast_cli(a->fd, " 802.1p CoS RTP video: %d\n", global_cos_video);
ast_cli(a->fd, " 802.1p CoS RTP text: %d\n", global_cos_text);
ast_cli(a->fd, " Jitterbuffer enabled: %s\n", cli_yesno(ast_test_flag(&global_jbconf, AST_JB_ENABLED)));
ast_cli(a->fd, " Jitterbuffer forced: %s\n", cli_yesno(ast_test_flag(&global_jbconf, AST_JB_FORCED)));
ast_cli(a->fd, " Jitterbuffer max size: %ld\n", global_jbconf.max_size);
ast_cli(a->fd, " Jitterbuffer resync: %ld\n", global_jbconf.resync_threshold);
ast_cli(a->fd, " Jitterbuffer impl: %s\n", global_jbconf.impl);
ast_cli(a->fd, " Jitterbuffer log: %s\n", cli_yesno(ast_test_flag(&global_jbconf, AST_JB_LOG)));
ast_cli(a->fd, "\nNetwork Settings:\n");
ast_cli(a->fd, "---------------------------\n");
/* determine if/how SIP address can be remapped */
if (localaddr == NULL)
msg = "Disabled, no localnet list";
else if (externip.sin_addr.s_addr == 0)
msg = "Disabled, externip is 0.0.0.0";
else if (stunaddr.sin_addr.s_addr != 0)
msg = "Enabled using STUN";
else if (!ast_strlen_zero(externhost))
msg = "Enabled using externhost";
else
msg = "Enabled using externip";
ast_cli(a->fd, " SIP address remapping: %s\n", msg);
ast_cli(a->fd, " Externhost: %s\n", S_OR(externhost, "<none>"));
ast_cli(a->fd, " Externip: %s:%d\n", ast_inet_ntoa(externip.sin_addr), ntohs(externip.sin_port));
ast_cli(a->fd, " Externrefresh: %d\n", externrefresh);
ast_cli(a->fd, " Internal IP: %s:%d\n", ast_inet_ntoa(internip.sin_addr), ntohs(internip.sin_port));
{
struct ast_ha *d;
const char *prefix = "Localnet:";
char buf[INET_ADDRSTRLEN]; /* need to print two addresses */
for (d = localaddr; d ; prefix = "", d = d->next) {
ast_cli(a->fd, " %-24s%s/%s\n",
prefix, ast_inet_ntoa(d->netaddr),
inet_ntop(AF_INET, &d->netmask, buf, sizeof(buf)) );
}
}
ast_cli(a->fd, " STUN server: %s:%d\n", ast_inet_ntoa(stunaddr.sin_addr), ntohs(stunaddr.sin_port));
ast_cli(a->fd, "\nGlobal Signalling Settings:\n");
ast_cli(a->fd, "---------------------------\n");
ast_cli(a->fd, " Codecs: ");
ast_getformatname_multiple(codec_buf, sizeof(codec_buf) -1, global_capability);
ast_cli(a->fd, "%s\n", codec_buf);
ast_cli(a->fd, " Codec Order: ");
print_codec_to_cli(a->fd, &default_prefs);
ast_cli(a->fd, "\n");
ast_cli(a->fd, " Relax DTMF: %s\n", cli_yesno(global_relaxdtmf));
ast_cli(a->fd, " RFC2833 Compensation: %s\n", cli_yesno(ast_test_flag(&global_flags[1], SIP_PAGE2_RFC2833_COMPENSATE)));
ast_cli(a->fd, " Compact SIP headers: %s\n", cli_yesno(sip_cfg.compactheaders));
ast_cli(a->fd, " RTP Keepalive: %d %s\n", global_rtpkeepalive, global_rtpkeepalive ? "" : "(Disabled)" );
ast_cli(a->fd, " RTP Timeout: %d %s\n", global_rtptimeout, global_rtptimeout ? "" : "(Disabled)" );
ast_cli(a->fd, " RTP Hold Timeout: %d %s\n", global_rtpholdtimeout, global_rtpholdtimeout ? "" : "(Disabled)");
ast_cli(a->fd, " MWI NOTIFY mime type: %s\n", default_notifymime);
ast_cli(a->fd, " DNS SRV lookup: %s\n", cli_yesno(sip_cfg.srvlookup));
ast_cli(a->fd, " Pedantic SIP support: %s\n", cli_yesno(sip_cfg.pedanticsipchecking));
ast_cli(a->fd, " Reg. min duration %d secs\n", min_expiry);
ast_cli(a->fd, " Reg. max duration: %d secs\n", max_expiry);
ast_cli(a->fd, " Reg. default duration: %d secs\n", default_expiry);
ast_cli(a->fd, " Outbound reg. timeout: %d secs\n", global_reg_timeout);
ast_cli(a->fd, " Outbound reg. attempts: %d\n", global_regattempts_max);
ast_cli(a->fd, " Notify ringing state: %s\n", cli_yesno(sip_cfg.notifyringing));
if (sip_cfg.notifyringing) {
ast_cli(a->fd, " Include CID: %s%s\n",
cli_yesno(sip_cfg.notifycid),
sip_cfg.notifycid == IGNORE_CONTEXT ? " (Ignoring context)" : "");
}
ast_cli(a->fd, " Notify hold state: %s\n", cli_yesno(sip_cfg.notifyhold));
ast_cli(a->fd, " SIP Transfer mode: %s\n", transfermode2str(sip_cfg.allowtransfer));
ast_cli(a->fd, " Max Call Bitrate: %d kbps\n", default_maxcallbitrate);
ast_cli(a->fd, " Auto-Framing: %s\n", cli_yesno(global_autoframing));
ast_cli(a->fd, " Outb. proxy: %s %s\n", ast_strlen_zero(sip_cfg.outboundproxy.name) ? "<not set>" : sip_cfg.outboundproxy.name,
sip_cfg.outboundproxy.force ? "(forced)" : "");
ast_cli(a->fd, " Session Timers: %s\n", stmode2str(global_st_mode));
ast_cli(a->fd, " Session Refresher: %s\n", strefresher2str (global_st_refresher));
ast_cli(a->fd, " Session Expires: %d secs\n", global_max_se);
ast_cli(a->fd, " Session Min-SE: %d secs\n", global_min_se);
ast_cli(a->fd, " Timer T1: %d\n", global_t1);
ast_cli(a->fd, " Timer T1 minimum: %d\n", global_t1min);
ast_cli(a->fd, " Timer B: %d\n", global_timer_b);
ast_cli(a->fd, "\nDefault Settings:\n");
ast_cli(a->fd, "-----------------\n");
ast_cli(a->fd, " Allowed transports: %s\n", get_transport_list(default_transports));
ast_cli(a->fd, " Outbound transport: %s\n", get_transport(default_primary_transport));
ast_cli(a->fd, " Context: %s\n", sip_cfg.default_context);
ast_cli(a->fd, " Nat: %s\n", nat2str(ast_test_flag(&global_flags[0], SIP_NAT)));
ast_cli(a->fd, " DTMF: %s\n", dtmfmode2str(ast_test_flag(&global_flags[0], SIP_DTMF)));
ast_cli(a->fd, " Qualify: %d\n", default_qualify);
ast_cli(a->fd, " Use ClientCode: %s\n", cli_yesno(ast_test_flag(&global_flags[0], SIP_USECLIENTCODE)));
ast_cli(a->fd, " Progress inband: %s\n", (ast_test_flag(&global_flags[0], SIP_PROG_INBAND) == SIP_PROG_INBAND_NEVER) ? "Never" : (ast_test_flag(&global_flags[0], SIP_PROG_INBAND) == SIP_PROG_INBAND_NO) ? "No" : "Yes" );
ast_cli(a->fd, " Language: %s\n", default_language);
ast_cli(a->fd, " MOH Interpret: %s\n", default_mohinterpret);
ast_cli(a->fd, " MOH Suggest: %s\n", default_mohsuggest);
ast_cli(a->fd, " Voice Mail Extension: %s\n", default_vmexten);
if (realtimepeers || realtimeregs) {
ast_cli(a->fd, "\nRealtime SIP Settings:\n");
ast_cli(a->fd, "----------------------\n");
ast_cli(a->fd, " Realtime Peers: %s\n", cli_yesno(realtimepeers));
ast_cli(a->fd, " Realtime Regs: %s\n", cli_yesno(realtimeregs));
ast_cli(a->fd, " Cache Friends: %s\n", cli_yesno(ast_test_flag(&global_flags[1], SIP_PAGE2_RTCACHEFRIENDS)));
ast_cli(a->fd, " Update: %s\n", cli_yesno(sip_cfg.peer_rtupdate));
ast_cli(a->fd, " Ignore Reg. Expire: %s\n", cli_yesno(sip_cfg.ignore_regexpire));
ast_cli(a->fd, " Save sys. name: %s\n", cli_yesno(sip_cfg.rtsave_sysname));
ast_cli(a->fd, " Auto Clear: %d\n", sip_cfg.rtautoclear);
}
ast_cli(a->fd, "\n----\n");
return CLI_SUCCESS;
}
static char *sip_show_mwi(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
#define FORMAT "%-30.30s %-12.12s %-10.10s %-10.10s\n"
char host[80];
switch (cmd) {
case CLI_INIT:
e->command = "sip show mwi";
e->usage =
"Usage: sip show mwi\n"
" Provides a list of MWI subscriptions and status.\n";
return NULL;
case CLI_GENERATE:
return NULL;
}
ast_cli(a->fd, FORMAT, "Host", "Username", "Mailbox", "Subscribed");
ASTOBJ_CONTAINER_TRAVERSE(&submwil, 1, do {
ASTOBJ_RDLOCK(iterator);
snprintf(host, sizeof(host), "%s:%d", iterator->hostname, iterator->portno ? iterator->portno : STANDARD_SIP_PORT);
ast_cli(a->fd, FORMAT, host, iterator->username, iterator->mailbox, iterator->subscribed ? "Yes" : "No");
ASTOBJ_UNLOCK(iterator);
} while(0));
return CLI_SUCCESS;
#undef FORMAT
}
/*! \brief Show subscription type in string format */
static const char *subscription_type2str(enum subscriptiontype subtype)
{
int i;
for (i = 1; i < ARRAY_LEN(subscription_types); i++) {
if (subscription_types[i].type == subtype) {
return subscription_types[i].text;
}
}
return subscription_types[0].text;
}
/*! \brief Find subscription type in array */
static const struct cfsubscription_types *find_subscription_type(enum subscriptiontype subtype)
{
int i;
for (i = 1; i < ARRAY_LEN(subscription_types); i++) {
if (subscription_types[i].type == subtype) {
return &subscription_types[i];
}
}
return &subscription_types[0];
}
/*
* We try to structure all functions that loop on data structures as
* a handler for individual entries, and a mainloop that iterates
* on the main data structure. This way, moving the code to containers
* that support iteration through callbacks will be a lot easier.
*/
#define FORMAT4 "%-15.15s %-10.10s %-15.15s %-15.15s %-13.13s %-15.15s %-10.10s %-6.6d\n"
#define FORMAT3 "%-15.15s %-10.10s %-15.15s %-15.15s %-13.13s %-15.15s %-10.10s %-6.6s\n"
#define FORMAT2 "%-15.15s %-10.10s %-15.15s %-15.15s %-7.7s %-15.15s %-6.6s\n"
#define FORMAT "%-15.15s %-10.10s %-15.15s %-15.15s %-3.3s %-3.3s %-15.15s %-10.10s\n"
/*! \brief callback for show channel|subscription */
static int show_channels_cb(void *__cur, void *__arg, int flags)
{
struct sip_pvt *cur = __cur;
struct __show_chan_arg *arg = __arg;
const struct sockaddr_in *dst = sip_real_dst(cur);
/* XXX indentation preserved to reduce diff. Will be fixed later */
if (cur->subscribed == NONE && !arg->subscriptions) {
/* set if SIP transfer in progress */
const char *referstatus = cur->refer ? referstatus2str(cur->refer->status) : "";
char formatbuf[SIPBUFSIZE/2];
ast_cli(arg->fd, FORMAT, ast_inet_ntoa(dst->sin_addr),
S_OR(cur->username, S_OR(cur->cid_num, "(None)")),
cur->callid,
ast_getformatname_multiple(formatbuf, sizeof(formatbuf), cur->owner ? cur->owner->nativeformats : 0),
cli_yesno(ast_test_flag(&cur->flags[1], SIP_PAGE2_CALL_ONHOLD)),
cur->needdestroy ? "(d)" : "",
cur->lastmsg ,
referstatus
);
arg->numchans++;
}
if (cur->subscribed != NONE && arg->subscriptions) {
struct ast_str *mailbox_str = ast_str_alloca(512);
if (cur->subscribed == MWI_NOTIFICATION && cur->relatedpeer)
peer_mailboxes_to_str(&mailbox_str, cur->relatedpeer);
ast_cli(arg->fd, FORMAT4, ast_inet_ntoa(dst->sin_addr),
S_OR(cur->username, S_OR(cur->cid_num, "(None)")),
cur->callid,
/* the 'complete' exten/context is hidden in the refer_to field for subscriptions */
cur->subscribed == MWI_NOTIFICATION ? "--" : cur->subscribeuri,
cur->subscribed == MWI_NOTIFICATION ? "<none>" : ast_extension_state2str(cur->laststate),
subscription_type2str(cur->subscribed),
cur->subscribed == MWI_NOTIFICATION ? S_OR(mailbox_str->str, "<none>") : "<none>",
cur->expiry
);
arg->numchans++;
}
return 0; /* don't care, we scan all channels */
}
/*! \brief CLI for show channels or subscriptions.
* This is a new-style CLI handler so a single function contains
* the prototype for the function, the 'generator' to produce multiple
* entries in case it is required, and the actual handler for the command.
*/
static char *sip_show_channels(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
struct __show_chan_arg arg = { .fd = a->fd, .numchans = 0 };
if (cmd == CLI_INIT) {
e->command = "sip show {channels|subscriptions}";
e->usage =
"Usage: sip show channels\n"
" Lists all currently active SIP calls (dialogs).\n"
"Usage: sip show subscriptions\n"
" Lists active SIP subscriptions.\n";
return NULL;
} else if (cmd == CLI_GENERATE)
return NULL;
if (a->argc != e->args)
return CLI_SHOWUSAGE;
arg.subscriptions = !strcasecmp(a->argv[e->args - 1], "subscriptions");
if (!arg.subscriptions)
ast_cli(arg.fd, FORMAT2, "Peer", "User/ANR", "Call ID", "Format", "Hold", "Last Message", "Expiry");
else
ast_cli(arg.fd, FORMAT3, "Peer", "User", "Call ID", "Extension", "Last state", "Type", "Mailbox", "Expiry");
/* iterate on the container and invoke the callback on each item */
ao2_t_callback(dialogs, OBJ_NODATA, show_channels_cb, &arg, "callback to show channels");
/* print summary information */
ast_cli(arg.fd, "%d active SIP %s%s\n", arg.numchans,
(arg.subscriptions ? "subscription" : "dialog"),
ESS(arg.numchans)); /* ESS(n) returns an "s" if n>1 */
return CLI_SUCCESS;
#undef FORMAT
#undef FORMAT2
#undef FORMAT3
}
/*! \brief Support routine for 'sip show channel' and 'sip show history' CLI
* This is in charge of generating all strings that match a prefix in the
* given position. As many functions of this kind, each invokation has
* O(state) time complexity so be careful in using it.
*/
static char *complete_sipch(const char *line, const char *word, int pos, int state)
{
int which=0;
struct sip_pvt *cur;
char *c = NULL;
int wordlen = strlen(word);
struct ao2_iterator i;
if (pos != 3) {
return NULL;
}
i = ao2_iterator_init(dialogs, 0);
while ((cur = ao2_t_iterator_next(&i, "iterate thru dialogs"))) {
sip_pvt_lock(cur);
if (!strncasecmp(word, cur->callid, wordlen) && ++which > state) {
c = ast_strdup(cur->callid);
sip_pvt_unlock(cur);
dialog_unref(cur, "drop ref in iterator loop break");
break;
}
sip_pvt_unlock(cur);
dialog_unref(cur, "drop ref in iterator loop");
}
return c;
}
/*! \brief Do completion on peer name */
static char *complete_sip_peer(const char *word, int state, int flags2)
{
char *result = NULL;
int wordlen = strlen(word);
int which = 0;
struct ao2_iterator i = ao2_iterator_init(peers, 0);
struct sip_peer *peer;
while ((peer = ao2_t_iterator_next(&i, "iterate thru peers table"))) {
/* locking of the object is not required because only the name and flags are being compared */
if (!strncasecmp(word, peer->name, wordlen) &&
(!flags2 || ast_test_flag(&peer->flags[1], flags2)) &&
++which > state)
result = ast_strdup(peer->name);
unref_peer(peer, "toss iterator peer ptr before break");
if (result) {
break;
}
}
return result;
}
/*! \brief Do completion on registered peer name */
static char *complete_sip_registered_peer(const char *word, int state, int flags2)
{
char *result = NULL;
int wordlen = strlen(word);
int which = 0;
struct ao2_iterator i;
struct sip_peer *peer;
i = ao2_iterator_init(peers, 0);
while ((peer = ao2_t_iterator_next(&i, "iterate thru peers table"))) {
if (!strncasecmp(word, peer->name, wordlen) &&
(!flags2 || ast_test_flag(&peer->flags[1], flags2)) &&
++which > state && peer->expire > 0)
result = ast_strdup(peer->name);
if (result) {
unref_peer(peer, "toss iterator peer ptr before break");
break;
}
unref_peer(peer, "toss iterator peer ptr");
}
return result;
}
/*! \brief Support routine for 'sip show history' CLI */
static char *complete_sip_show_history(const char *line, const char *word, int pos, int state)
{
if (pos == 3)
return complete_sipch(line, word, pos, state);
return NULL;
}
/*! \brief Support routine for 'sip show peer' CLI */
static char *complete_sip_show_peer(const char *line, const char *word, int pos, int state)
{
if (pos == 3) {
return complete_sip_peer(word, state, 0);
}
return NULL;
}
/*! \brief Support routine for 'sip unregister' CLI */
static char *complete_sip_unregister(const char *line, const char *word, int pos, int state)
{
if (pos == 2)
return complete_sip_registered_peer(word, state, 0);
return NULL;
}
/*! \brief Support routine for 'sip notify' CLI */
static char *complete_sipnotify(const char *line, const char *word, int pos, int state)
{
char *c = NULL;
if (pos == 2) {
int which = 0;
char *cat = NULL;
int wordlen = strlen(word);
/* do completion for notify type */
if (!notify_types)
return NULL;
while ( (cat = ast_category_browse(notify_types, cat)) ) {
if (!strncasecmp(word, cat, wordlen) && ++which > state) {
c = ast_strdup(cat);
break;
}
}
return c;
}
if (pos > 2)
return complete_sip_peer(word, state, 0);
return NULL;
}
/*! \brief Show details of one active dialog */
static char *sip_show_channel(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
struct sip_pvt *cur;
size_t len;
int found = 0;
struct ao2_iterator i;
switch (cmd) {
case CLI_INIT:
e->command = "sip show channel";
e->usage =
"Usage: sip show channel <call-id>\n"
" Provides detailed status on a given SIP dialog (identified by SIP call-id).\n";
return NULL;
case CLI_GENERATE:
return complete_sipch(a->line, a->word, a->pos, a->n);
}
if (a->argc != 4)
return CLI_SHOWUSAGE;
len = strlen(a->argv[3]);
i = ao2_iterator_init(dialogs, 0);
while ((cur = ao2_t_iterator_next(&i, "iterate thru dialogs"))) {
sip_pvt_lock(cur);
if (!strncasecmp(cur->callid, a->argv[3], len)) {
char formatbuf[SIPBUFSIZE/2];
ast_cli(a->fd, "\n");
if (cur->subscribed != NONE)
ast_cli(a->fd, " * Subscription (type: %s)\n", subscription_type2str(cur->subscribed));
else
ast_cli(a->fd, " * SIP Call\n");
ast_cli(a->fd, " Curr. trans. direction: %s\n", ast_test_flag(&cur->flags[0], SIP_OUTGOING) ? "Outgoing" : "Incoming");
ast_cli(a->fd, " Call-ID: %s\n", cur->callid);
ast_cli(a->fd, " Owner channel ID: %s\n", cur->owner ? cur->owner->name : "<none>");
ast_cli(a->fd, " Our Codec Capability: %d\n", cur->capability);
ast_cli(a->fd, " Non-Codec Capability (DTMF): %d\n", cur->noncodeccapability);
ast_cli(a->fd, " Their Codec Capability: %d\n", cur->peercapability);
ast_cli(a->fd, " Joint Codec Capability: %d\n", cur->jointcapability);
ast_cli(a->fd, " Format: %s\n", ast_getformatname_multiple(formatbuf, sizeof(formatbuf), cur->owner ? cur->owner->nativeformats : 0) );
ast_cli(a->fd, " T.38 support %s\n", cli_yesno(cur->udptl != NULL));
ast_cli(a->fd, " Video support %s\n", cli_yesno(cur->vrtp != NULL));
ast_cli(a->fd, " MaxCallBR: %d kbps\n", cur->maxcallbitrate);
ast_cli(a->fd, " Theoretical Address: %s:%d\n", ast_inet_ntoa(cur->sa.sin_addr), ntohs(cur->sa.sin_port));
ast_cli(a->fd, " Received Address: %s:%d\n", ast_inet_ntoa(cur->recv.sin_addr), ntohs(cur->recv.sin_port));
ast_cli(a->fd, " SIP Transfer mode: %s\n", transfermode2str(cur->allowtransfer));
ast_cli(a->fd, " NAT Support: %s\n", nat2str(ast_test_flag(&cur->flags[0], SIP_NAT)));
ast_cli(a->fd, " Audio IP: %s %s\n", ast_inet_ntoa(cur->redirip.sin_addr.s_addr ? cur->redirip.sin_addr : cur->ourip.sin_addr), cur->redirip.sin_addr.s_addr ? "(Outside bridge)" : "(local)" );
ast_cli(a->fd, " Our Tag: %s\n", cur->tag);
ast_cli(a->fd, " Their Tag: %s\n", cur->theirtag);
ast_cli(a->fd, " SIP User agent: %s\n", cur->useragent);
if (!ast_strlen_zero(cur->username))
ast_cli(a->fd, " Username: %s\n", cur->username);
if (!ast_strlen_zero(cur->peername))
ast_cli(a->fd, " Peername: %s\n", cur->peername);
if (!ast_strlen_zero(cur->uri))
ast_cli(a->fd, " Original uri: %s\n", cur->uri);
if (!ast_strlen_zero(cur->cid_num))
ast_cli(a->fd, " Caller-ID: %s\n", cur->cid_num);
ast_cli(a->fd, " Need Destroy: %s\n", cli_yesno(cur->needdestroy));
ast_cli(a->fd, " Last Message: %s\n", cur->lastmsg);
ast_cli(a->fd, " Promiscuous Redir: %s\n", cli_yesno(ast_test_flag(&cur->flags[0], SIP_PROMISCREDIR)));
ast_cli(a->fd, " Route: %s\n", cur->route ? cur->route->hop : "N/A");
ast_cli(a->fd, " DTMF Mode: %s\n", dtmfmode2str(ast_test_flag(&cur->flags[0], SIP_DTMF)));
ast_cli(a->fd, " SIP Options: ");
if (cur->sipoptions) {
int x;
for (x = 0 ; x < ARRAY_LEN(sip_options); x++) {
if (cur->sipoptions & sip_options[x].id)
ast_cli(a->fd, "%s ", sip_options[x].text);
}
ast_cli(a->fd, "\n");
} else
ast_cli(a->fd, "(none)\n");
if (!cur->stimer)
ast_cli(a->fd, " Session-Timer: Uninitiallized\n");
else {
ast_cli(a->fd, " Session-Timer: %s\n", cur->stimer->st_active ? "Active" : "Inactive");
if (cur->stimer->st_active == TRUE) {
ast_cli(a->fd, " S-Timer Interval: %d\n", cur->stimer->st_interval);
ast_cli(a->fd, " S-Timer Refresher: %s\n", strefresher2str(cur->stimer->st_ref));
ast_cli(a->fd, " S-Timer Expirys: %d\n", cur->stimer->st_expirys);
ast_cli(a->fd, " S-Timer Sched Id: %d\n", cur->stimer->st_schedid);
ast_cli(a->fd, " S-Timer Peer Sts: %s\n", cur->stimer->st_active_peer_ua ? "Active" : "Inactive");
ast_cli(a->fd, " S-Timer Cached Min-SE: %d\n", cur->stimer->st_cached_min_se);
ast_cli(a->fd, " S-Timer Cached SE: %d\n", cur->stimer->st_cached_max_se);
ast_cli(a->fd, " S-Timer Cached Ref: %s\n", strefresher2str(cur->stimer->st_cached_ref));
ast_cli(a->fd, " S-Timer Cached Mode: %s\n", stmode2str(cur->stimer->st_cached_mode));
}
}
ast_cli(a->fd, "\n\n");
found++;
}
sip_pvt_unlock(cur);
ao2_t_ref(cur, -1, "toss dialog ptr set by iterator_next");
}
if (!found)
ast_cli(a->fd, "No such SIP Call ID starting with '%s'\n", a->argv[3]);
return CLI_SUCCESS;
}
/*! \brief Show history details of one dialog */
static char *sip_show_history(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
struct sip_pvt *cur;
size_t len;
int found = 0;
struct ao2_iterator i;
switch (cmd) {
case CLI_INIT:
e->command = "sip show history";
e->usage =
"Usage: sip show history <call-id>\n"
" Provides detailed dialog history on a given SIP call (specified by call-id).\n";
return NULL;
case CLI_GENERATE:
return complete_sip_show_history(a->line, a->word, a->pos, a->n);
}
if (a->argc != 4)
return CLI_SHOWUSAGE;
if (!recordhistory)
ast_cli(a->fd, "\n***Note: History recording is currently DISABLED. Use 'sip set history on' to ENABLE.\n");
len = strlen(a->argv[3]);
i = ao2_iterator_init(dialogs, 0);
while ((cur = ao2_t_iterator_next(&i, "iterate thru dialogs"))) {
sip_pvt_lock(cur);
if (!strncasecmp(cur->callid, a->argv[3], len)) {
struct sip_history *hist;
int x = 0;
ast_cli(a->fd, "\n");
if (cur->subscribed != NONE)
ast_cli(a->fd, " * Subscription\n");
else
ast_cli(a->fd, " * SIP Call\n");
if (cur->history)
AST_LIST_TRAVERSE(cur->history, hist, list)
ast_cli(a->fd, "%d. %s\n", ++x, hist->event);
if (x == 0)
ast_cli(a->fd, "Call '%s' has no history\n", cur->callid);
found++;
}
sip_pvt_unlock(cur);
ao2_t_ref(cur, -1, "toss dialog ptr from iterator_next");
}
if (!found)
ast_cli(a->fd, "No such SIP Call ID starting with '%s'\n", a->argv[3]);
return CLI_SUCCESS;
}
/*! \brief Dump SIP history to debug log file at end of lifespan for SIP dialog */
static void sip_dump_history(struct sip_pvt *dialog)
{
int x = 0;
struct sip_history *hist;
static int errmsg = 0;
if (!dialog)
return;
if (!option_debug && !sipdebug) {
if (!errmsg) {
ast_log(LOG_NOTICE, "You must have debugging enabled (SIP or Asterisk) in order to dump SIP history.\n");
errmsg = 1;
}
return;
}
ast_debug(1, "\n---------- SIP HISTORY for '%s' \n", dialog->callid);
if (dialog->subscribed)
ast_debug(1, " * Subscription\n");
else
ast_debug(1, " * SIP Call\n");
if (dialog->history)
AST_LIST_TRAVERSE(dialog->history, hist, list)
ast_debug(1, " %-3.3d. %s\n", ++x, hist->event);
if (!x)
ast_debug(1, "Call '%s' has no history\n", dialog->callid);
ast_debug(1, "\n---------- END SIP HISTORY for '%s' \n", dialog->callid);
}
/*! \brief Receive SIP INFO Message */
static void handle_request_info(struct sip_pvt *p, struct sip_request *req)
{
char buf[1024];
unsigned int event;
const char *c = get_header(req, "Content-Type");
/* Need to check the media/type */
if (!strcasecmp(c, "application/dtmf-relay") ||
!strcasecmp(c, "application/vnd.nortelnetworks.digits")) {
unsigned int duration = 0;
if (!p->owner) { /* not a PBX call */
transmit_response(p, "481 Call leg/transaction does not exist", req);
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
return;
}
/* Try getting the "signal=" part */
if (ast_strlen_zero(c = get_body(req, "Signal", '=')) && ast_strlen_zero(c = get_body(req, "d", '='))) {
ast_log(LOG_WARNING, "Unable to retrieve DTMF signal from INFO message from %s\n", p->callid);
transmit_response(p, "200 OK", req); /* Should return error */
return;
} else {
ast_copy_string(buf, c, sizeof(buf));
}
if (!ast_strlen_zero((c = get_body(req, "Duration", '='))))
duration = atoi(c);
if (!duration)
duration = 100; /* 100 ms */
if (ast_strlen_zero(buf)) {
transmit_response(p, "200 OK", req);
return;
}
if (buf[0] == '*')
event = 10;
else if (buf[0] == '#')
event = 11;
else if ((buf[0] >= 'A') && (buf[0] <= 'D'))
event = 12 + buf[0] - 'A';
else if (buf[0] == '!')
event = 16;
else
event = atoi(buf);
if (event == 16) {
/* send a FLASH event */
struct ast_frame f = { AST_FRAME_CONTROL, AST_CONTROL_FLASH, };
ast_queue_frame(p->owner, &f);
if (sipdebug)
ast_verbose("* DTMF-relay event received: FLASH\n");
} else {
/* send a DTMF event */
struct ast_frame f = { AST_FRAME_DTMF, };
if (event < 10) {
f.subclass = '0' + event;
} else if (event < 11) {
f.subclass = '*';
} else if (event < 12) {
f.subclass = '#';
} else if (event < 16) {
f.subclass = 'A' + (event - 12);
}
f.len = duration;
ast_queue_frame(p->owner, &f);
if (sipdebug)
ast_verbose("* DTMF-relay event received: %c\n", f.subclass);
}
transmit_response(p, "200 OK", req);
return;
} else if (!strcasecmp(c, "application/dtmf")) {
/*! \todo Note: Doesn't read the duration of the DTMF. Should be fixed. */
unsigned int duration = 0;
if (!p->owner) { /* not a PBX call */
transmit_response(p, "481 Call leg/transaction does not exist", req);
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
return;
}
get_msg_text(buf, sizeof(buf), req, TRUE);
duration = 100; /* 100 ms */
if (ast_strlen_zero(buf)) {
transmit_response(p, "200 OK", req);
return;
}
event = atoi(buf);
if (event == 16) {
/* send a FLASH event */
struct ast_frame f = { AST_FRAME_CONTROL, AST_CONTROL_FLASH, };
ast_queue_frame(p->owner, &f);
if (sipdebug)
ast_verbose("* DTMF-relay event received: FLASH\n");
} else {
/* send a DTMF event */
struct ast_frame f = { AST_FRAME_DTMF, };
if (event < 10) {
f.subclass = '0' + event;
} else if (event < 11) {
f.subclass = '*';
} else if (event < 12) {
f.subclass = '#';
} else if (event < 16) {
f.subclass = 'A' + (event - 12);
}
f.len = duration;
ast_queue_frame(p->owner, &f);
if (sipdebug)
ast_verbose("* DTMF-relay event received: %c\n", f.subclass);
}
transmit_response(p, "200 OK", req);
return;
} else if (!strcasecmp(c, "application/media_control+xml")) {
/* Eh, we'll just assume it's a fast picture update for now */
if (p->owner)
ast_queue_control(p->owner, AST_CONTROL_VIDUPDATE);
transmit_response(p, "200 OK", req);
return;
} else if (!ast_strlen_zero(c = get_header(req, "X-ClientCode"))) {
/* Client code (from SNOM phone) */
if (ast_test_flag(&p->flags[0], SIP_USECLIENTCODE)) {
if (p->owner && p->owner->cdr)
ast_cdr_setuserfield(p->owner, c);
if (p->owner && ast_bridged_channel(p->owner) && ast_bridged_channel(p->owner)->cdr)
ast_cdr_setuserfield(ast_bridged_channel(p->owner), c);
transmit_response(p, "200 OK", req);
} else {
transmit_response(p, "403 Forbidden", req);
}
return;
} else if (!ast_strlen_zero(c = get_header(req, "Record"))) {
/* INFO messages generated by some phones to start/stop recording
on phone calls.
OEJ: I think this should be something that is enabled/disabled
per device. I don't want incoming callers to record calls in my
pbx.
*/
/* first, get the feature string, if it exists */
struct ast_call_feature *feat;
int j;
struct ast_frame f = { AST_FRAME_DTMF, };
ast_rdlock_call_features();
feat = ast_find_call_feature("automon");
if (!feat || ast_strlen_zero(feat->exten)) {
ast_log(LOG_WARNING, "Recording requested, but no One Touch Monitor registered. (See features.conf)\n");
/* 403 means that we don't support this feature, so don't request it again */
transmit_response(p, "403 Forbidden", req);
ast_unlock_call_features();
return;
}
/* Send the feature code to the PBX as DTMF, just like the handset had sent it */
f.len = 100;
for (j=0; j < strlen(feat->exten); j++) {
f.subclass = feat->exten[j];
ast_queue_frame(p->owner, &f);
if (sipdebug)
ast_verbose("* DTMF-relay event faked: %c\n", f.subclass);
}
ast_unlock_call_features();
ast_debug(1, "Got a Request to Record the channel, state %s\n", c);
transmit_response(p, "200 OK", req);
return;
} else if (ast_strlen_zero(c = get_header(req, "Content-Length")) || !strcasecmp(c, "0")) {
/* This is probably just a packet making sure the signalling is still up, just send back a 200 OK */
transmit_response(p, "200 OK", req);
return;
}
/* Other type of INFO message, not really understood by Asterisk */
/* if (get_msg_text(buf, sizeof(buf), req)) { */
ast_log(LOG_WARNING, "Unable to parse INFO message from %s. Content %s\n", p->callid, buf);
transmit_response(p, "415 Unsupported media type", req);
return;
}
/*! \brief Enable SIP Debugging for a single IP */
static char *sip_do_debug_ip(int fd, char *arg)
{
struct hostent *hp;
struct ast_hostent ahp;
int port = 0;
char *p;
p = arg;
strsep(&p, ":");
if (p)
port = atoi(p);
hp = ast_gethostbyname(arg, &ahp);
if (hp == NULL)
return CLI_SHOWUSAGE;
debugaddr.sin_family = AF_INET;
memcpy(&debugaddr.sin_addr, hp->h_addr, sizeof(debugaddr.sin_addr));
debugaddr.sin_port = htons(port);
if (port == 0)
ast_cli(fd, "SIP Debugging Enabled for IP: %s\n", ast_inet_ntoa(debugaddr.sin_addr));
else
ast_cli(fd, "SIP Debugging Enabled for IP: %s:%d\n", ast_inet_ntoa(debugaddr.sin_addr), port);
sipdebug |= sip_debug_console;
return CLI_SUCCESS;
}
/*! \brief Turn on SIP debugging for a given peer */
static char *sip_do_debug_peer(int fd, char *arg)
{
struct sip_peer *peer = find_peer(arg, NULL, TRUE, FINDPEERS, FALSE);
if (!peer)
ast_cli(fd, "No such peer '%s'\n", arg);
else if (peer->addr.sin_addr.s_addr == 0)
ast_cli(fd, "Unable to get IP address of peer '%s'\n", arg);
else {
debugaddr.sin_family = AF_INET;
debugaddr.sin_addr = peer->addr.sin_addr;
debugaddr.sin_port = peer->addr.sin_port;
ast_cli(fd, "SIP Debugging Enabled for IP: %s:%d\n",
ast_inet_ntoa(debugaddr.sin_addr), ntohs(debugaddr.sin_port));
sipdebug |= sip_debug_console;
}
if (peer)
unref_peer(peer, "sip_do_debug_peer: unref_peer, from find_peer call");
return CLI_SUCCESS;
}
/*! \brief Turn on SIP debugging (CLI command) */
static char *sip_do_debug(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
int oldsipdebug = sipdebug & sip_debug_console;
char *what;
if (cmd == CLI_INIT) {
e->command = "sip set debug {on|off|ip|peer}";
e->usage =
"Usage: sip set debug {off|on|ip addr[:port]|peer peername}\n"
" Globally disables dumping of SIP packets,\n"
" or enables it either globally or for a (single)\n"
" IP address or registered peer.\n";
return NULL;
} else if (cmd == CLI_GENERATE) {
if (a->pos == 4 && !strcasecmp(a->argv[3], "peer"))
return complete_sip_peer(a->word, a->n, 0);
return NULL;
}
what = a->argv[e->args-1]; /* guaranteed to exist */
if (a->argc == e->args) { /* on/off */
if (!strcasecmp(what, "on")) {
sipdebug |= sip_debug_console;
sipdebug_text = 1; /*! \note this can be a special debug command - "sip debug text" or something */
memset(&debugaddr, 0, sizeof(debugaddr));
ast_cli(a->fd, "SIP Debugging %senabled\n", oldsipdebug ? "re-" : "");
return CLI_SUCCESS;
} else if (!strcasecmp(what, "off")) {
sipdebug &= ~sip_debug_console;
sipdebug_text = 0;
ast_cli(a->fd, "SIP Debugging Disabled\n");
return CLI_SUCCESS;
}
} else if (a->argc == e->args +1) {/* ip/peer */
if (!strcasecmp(what, "ip"))
return sip_do_debug_ip(a->fd, a->argv[e->args]);
else if (!strcasecmp(what, "peer"))
return sip_do_debug_peer(a->fd, a->argv[e->args]);
}
return CLI_SHOWUSAGE; /* default, failure */
}
/*! \brief Cli command to send SIP notify to peer */
static char *sip_cli_notify(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
struct ast_variable *varlist;
int i;
switch (cmd) {
case CLI_INIT:
e->command = "sip notify";
e->usage =
"Usage: sip notify <type> <peer> [<peer>...]\n"
" Send a NOTIFY message to a SIP peer or peers\n"
" Message types are defined in sip_notify.conf\n";
return NULL;
case CLI_GENERATE:
return complete_sipnotify(a->line, a->word, a->pos, a->n);
}
if (a->argc < 4)
return CLI_SHOWUSAGE;
if (!notify_types) {
ast_cli(a->fd, "No %s file found, or no types listed there\n", notify_config);
return CLI_FAILURE;
}
varlist = ast_variable_browse(notify_types, a->argv[2]);
if (!varlist) {
ast_cli(a->fd, "Unable to find notify type '%s'\n", a->argv[2]);
return CLI_FAILURE;
}
for (i = 3; i < a->argc; i++) {
struct sip_pvt *p;
if (!(p = sip_alloc(NULL, NULL, 0, SIP_NOTIFY))) {
ast_log(LOG_WARNING, "Unable to build sip pvt data for notify (memory/socket error)\n");
return CLI_FAILURE;
}
if (create_addr(p, a->argv[i], NULL, 0)) {
/* Maybe they're not registered, etc. */
dialog_unlink_all(p, TRUE, TRUE);
dialog_unref(p, "unref dialog inside for loop" );
/* sip_destroy(p); */
ast_cli(a->fd, "Could not create address for '%s'\n", a->argv[i]);
continue;
}
/* Notify is outgoing call */
ast_set_flag(&p->flags[0], SIP_OUTGOING);
/* Recalculate our side, and recalculate Call ID */
ast_sip_ouraddrfor(&p->sa.sin_addr, &p->ourip);
build_via(p);
ao2_t_unlink(dialogs, p, "About to change the callid -- remove the old name");
build_callid_pvt(p);
ao2_t_link(dialogs, p, "Linking in new name");
ast_cli(a->fd, "Sending NOTIFY of type '%s' to '%s'\n", a->argv[2], a->argv[i]);
dialog_ref(p, "bump the count of p, which transmit_sip_request will decrement.");
sip_scheddestroy(p, SIP_TRANS_TIMEOUT);
transmit_notify_custom(p, varlist);
}
return CLI_SUCCESS;
}
/*! \brief Enable/Disable SIP History logging (CLI) */
static char *sip_set_history(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
switch (cmd) {
case CLI_INIT:
e->command = "sip set history {on|off}";
e->usage =
"Usage: sip set history {on|off}\n"
" Enables/Disables recording of SIP dialog history for debugging purposes.\n"
" Use 'sip show history' to view the history of a call number.\n";
return NULL;
case CLI_GENERATE:
return NULL;
}
if (a->argc != e->args)
return CLI_SHOWUSAGE;
if (!strncasecmp(a->argv[e->args - 1], "on", 2)) {
recordhistory = TRUE;
ast_cli(a->fd, "SIP History Recording Enabled (use 'sip show history')\n");
} else if (!strncasecmp(a->argv[e->args - 1], "off", 3)) {
recordhistory = FALSE;
ast_cli(a->fd, "SIP History Recording Disabled\n");
} else {
return CLI_SHOWUSAGE;
}
return CLI_SUCCESS;
}
/*! \brief Authenticate for outbound registration */
static int do_register_auth(struct sip_pvt *p, struct sip_request *req, enum sip_auth_type code)
{
char *header, *respheader;
char digest[1024];
p->authtries++;
auth_headers(code, &header, &respheader);
memset(digest, 0, sizeof(digest));
if (reply_digest(p, req, header, SIP_REGISTER, digest, sizeof(digest))) {
/* There's nothing to use for authentication */
/* No digest challenge in request */
if (sip_debug_test_pvt(p) && p->registry)
ast_verbose("No authentication challenge, sending blank registration to domain/host name %s\n", p->registry->hostname);
/* No old challenge */
return -1;
}
if (p->do_history)
append_history(p, "RegistryAuth", "Try: %d", p->authtries);
if (sip_debug_test_pvt(p) && p->registry)
ast_verbose("Responding to challenge, registration to domain/host name %s\n", p->registry->hostname);
return transmit_register(p->registry, SIP_REGISTER, digest, respheader);
}
/*! \brief Add authentication on outbound SIP packet */
static int do_proxy_auth(struct sip_pvt *p, struct sip_request *req, enum sip_auth_type code, int sipmethod, int init)
{
char *header, *respheader;
char digest[1024];
if (!p->options && !(p->options = ast_calloc(1, sizeof(*p->options))))
return -2;
p->authtries++;
auth_headers(code, &header, &respheader);
ast_debug(2, "Auth attempt %d on %s\n", p->authtries, sip_methods[sipmethod].text);
memset(digest, 0, sizeof(digest));
if (reply_digest(p, req, header, sipmethod, digest, sizeof(digest) )) {
/* No way to authenticate */
return -1;
}
/* Now we have a reply digest */
p->options->auth = digest;
p->options->authheader = respheader;
return transmit_invite(p, sipmethod, sipmethod == SIP_INVITE, init);
}
/*! \brief reply to authentication for outbound registrations
\return Returns -1 if we have no auth
\note This is used for register= servers in sip.conf, SIP proxies we register
with for receiving calls from. */
static int reply_digest(struct sip_pvt *p, struct sip_request *req, char *header, int sipmethod, char *digest, int digest_len)
{
char tmp[512];
char *c;
char oldnonce[256];
/* table of recognised keywords, and places where they should be copied */
const struct x {
const char *key;
const ast_string_field *field;
} *i, keys[] = {
{ "realm=", &p->realm },
{ "nonce=", &p->nonce },
{ "opaque=", &p->opaque },
{ "qop=", &p->qop },
{ "domain=", &p->domain },
{ NULL, 0 },
};
ast_copy_string(tmp, get_header(req, header), sizeof(tmp));
if (ast_strlen_zero(tmp))
return -1;
if (strncasecmp(tmp, "Digest ", strlen("Digest "))) {
ast_log(LOG_WARNING, "missing Digest.\n");
return -1;
}
c = tmp + strlen("Digest ");
ast_copy_string(oldnonce, p->nonce, sizeof(oldnonce));
while (c && *(c = ast_skip_blanks(c))) { /* lookup for keys */
for (i = keys; i->key != NULL; i++) {
char *src, *separator;
if (strncasecmp(c, i->key, strlen(i->key)) != 0)
continue;
/* Found. Skip keyword, take text in quotes or up to the separator. */
c += strlen(i->key);
if (*c == '"') {
src = ++c;
separator = "\"";
} else {
src = c;
separator = ",";
}
strsep(&c, separator); /* clear separator and move ptr */
ast_string_field_ptr_set(p, i->field, src);
break;
}
if (i->key == NULL) /* not found, try ',' */
strsep(&c, ",");
}
/* Reset nonce count */
if (strcmp(p->nonce, oldnonce))
p->noncecount = 0;
/* Save auth data for following registrations */
if (p->registry) {
struct sip_registry *r = p->registry;
if (strcmp(r->nonce, p->nonce)) {
ast_string_field_set(r, realm, p->realm);
ast_string_field_set(r, nonce, p->nonce);
ast_string_field_set(r, domain, p->domain);
ast_string_field_set(r, opaque, p->opaque);
ast_string_field_set(r, qop, p->qop);
r->noncecount = 0;
}
}
return build_reply_digest(p, sipmethod, digest, digest_len);
}
/*! \brief Build reply digest
\return Returns -1 if we have no auth
\note Build digest challenge for authentication of registrations and calls
Also used for authentication of BYE
*/
static int build_reply_digest(struct sip_pvt *p, int method, char* digest, int digest_len)
{
char a1[256];
char a2[256];
char a1_hash[256];
char a2_hash[256];
char resp[256];
char resp_hash[256];
char uri[256];
char opaque[256] = "";
char cnonce[80];
const char *username;
const char *secret;
const char *md5secret;
struct sip_auth *auth = NULL; /* Realm authentication */
if (!ast_strlen_zero(p->domain))
ast_copy_string(uri, p->domain, sizeof(uri));
else if (!ast_strlen_zero(p->uri))
ast_copy_string(uri, p->uri, sizeof(uri));
else
snprintf(uri, sizeof(uri), "sip:%s@%s", p->username, ast_inet_ntoa(p->sa.sin_addr));
snprintf(cnonce, sizeof(cnonce), "%08lx", ast_random());
/* Check if we have separate auth credentials */
if(!(auth = find_realm_authentication(p->peerauth, p->realm))) /* Start with peer list */
auth = find_realm_authentication(authl, p->realm); /* If not, global list */
if (auth) {
ast_debug(3, "use realm [%s] from peer [%s][%s]\n", auth->username, p->peername, p->username);
username = auth->username;
secret = auth->secret;
md5secret = auth->md5secret;
if (sipdebug)
ast_debug(1, "Using realm %s authentication for call %s\n", p->realm, p->callid);
} else {
/* No authentication, use peer or register= config */
username = p->authname;
secret = p->peersecret;
md5secret = p->peermd5secret;
}
if (ast_strlen_zero(username)) /* We have no authentication */
return -1;
/* Calculate SIP digest response */
snprintf(a1, sizeof(a1), "%s:%s:%s", username, p->realm, secret);
snprintf(a2, sizeof(a2), "%s:%s", sip_methods[method].text, uri);
if (!ast_strlen_zero(md5secret))
ast_copy_string(a1_hash, md5secret, sizeof(a1_hash));
else
ast_md5_hash(a1_hash, a1);
ast_md5_hash(a2_hash, a2);
p->noncecount++;
if (!ast_strlen_zero(p->qop))
snprintf(resp, sizeof(resp), "%s:%s:%08x:%s:%s:%s", a1_hash, p->nonce, p->noncecount, cnonce, "auth", a2_hash);
else
snprintf(resp, sizeof(resp), "%s:%s:%s", a1_hash, p->nonce, a2_hash);
ast_md5_hash(resp_hash, resp);
/* only include the opaque string if it's set */
if (!ast_strlen_zero(p->opaque)) {
snprintf(opaque, sizeof(opaque), ", opaque=\"%s\"", p->opaque);
}
/* XXX We hard code our qop to "auth" for now. XXX */
if (!ast_strlen_zero(p->qop))
snprintf(digest, digest_len, "Digest username=\"%s\", realm=\"%s\", algorithm=MD5, uri=\"%s\", nonce=\"%s\", response=\"%s\"%s, qop=auth, cnonce=\"%s\", nc=%08x", username, p->realm, uri, p->nonce, resp_hash, opaque, cnonce, p->noncecount);
else
snprintf(digest, digest_len, "Digest username=\"%s\", realm=\"%s\", algorithm=MD5, uri=\"%s\", nonce=\"%s\", response=\"%s\"%s", username, p->realm, uri, p->nonce, resp_hash, opaque);
append_history(p, "AuthResp", "Auth response sent for %s in realm %s - nc %d", username, p->realm, p->noncecount);
return 0;
}
/*! \brief Read SIP header (dialplan function) */
static int func_header_read(struct ast_channel *chan, const char *function, char *data, char *buf, size_t len)
{
struct sip_pvt *p;
const char *content = NULL;
AST_DECLARE_APP_ARGS(args,
AST_APP_ARG(header);
AST_APP_ARG(number);
);
int i, number, start = 0;
if (ast_strlen_zero(data)) {
ast_log(LOG_WARNING, "This function requires a header name.\n");
return -1;
}
ast_channel_lock(chan);
if (!IS_SIP_TECH(chan->tech)) {
ast_log(LOG_WARNING, "This function can only be used on SIP channels.\n");
ast_channel_unlock(chan);
return -1;
}
AST_STANDARD_APP_ARGS(args, data);
if (!args.number) {
number = 1;
} else {
sscanf(args.number, "%d", &number);
if (number < 1)
number = 1;
}
p = chan->tech_pvt;
/* If there is no private structure, this channel is no longer alive */
if (!p) {
ast_channel_unlock(chan);
return -1;
}
for (i = 0; i < number; i++)
content = __get_header(&p->initreq, args.header, &start);
if (ast_strlen_zero(content)) {
ast_channel_unlock(chan);
return -1;
}
ast_copy_string(buf, content, len);
ast_channel_unlock(chan);
return 0;
}
static struct ast_custom_function sip_header_function = {
.name = "SIP_HEADER",
.read = func_header_read,
};
/*! \brief Dial plan function to check if domain is local */
static int func_check_sipdomain(struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len)
{
if (ast_strlen_zero(data)) {
ast_log(LOG_WARNING, "CHECKSIPDOMAIN requires an argument - A domain name\n");
return -1;
}
if (check_sip_domain(data, NULL, 0))
ast_copy_string(buf, data, len);
else
buf[0] = '\0';
return 0;
}
static struct ast_custom_function checksipdomain_function = {
.name = "CHECKSIPDOMAIN",
.read = func_check_sipdomain,
};
/*! \brief ${SIPPEER()} Dialplan function - reads peer data */
static int function_sippeer(struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len)
{
struct sip_peer *peer;
char *colname;
if ((colname = strchr(data, ':'))) { /*! \todo Will be deprecated after 1.4 */
static int deprecation_warning = 0;
*colname++ = '\0';
if (deprecation_warning++ % 10 == 0)
ast_log(LOG_WARNING, "SIPPEER(): usage of ':' to separate arguments is deprecated. Please use ',' instead.\n");
} else if ((colname = strchr(data, ',')))
*colname++ = '\0';
else
colname = "ip";
if (!(peer = find_peer(data, NULL, TRUE, FINDPEERS, FALSE)))
return -1;
if (!strcasecmp(colname, "ip")) {
ast_copy_string(buf, peer->addr.sin_addr.s_addr ? ast_inet_ntoa(peer->addr.sin_addr) : "", len);
} else if (!strcasecmp(colname, "port")) {
snprintf(buf, len, "%d", ntohs(peer->addr.sin_port));
} else if (!strcasecmp(colname, "status")) {
peer_status(peer, buf, len);
} else if (!strcasecmp(colname, "language")) {
ast_copy_string(buf, peer->language, len);
} else if (!strcasecmp(colname, "regexten")) {
ast_copy_string(buf, peer->regexten, len);
} else if (!strcasecmp(colname, "limit")) {
snprintf(buf, len, "%d", peer->call_limit);
} else if (!strcasecmp(colname, "busylevel")) {
snprintf(buf, len, "%d", peer->busy_level);
} else if (!strcasecmp(colname, "curcalls")) {
snprintf(buf, len, "%d", peer->inUse);
} else if (!strcasecmp(colname, "accountcode")) {
ast_copy_string(buf, peer->accountcode, len);
} else if (!strcasecmp(colname, "callgroup")) {
ast_print_group(buf, len, peer->callgroup);
} else if (!strcasecmp(colname, "pickupgroup")) {
ast_print_group(buf, len, peer->pickupgroup);
} else if (!strcasecmp(colname, "useragent")) {
ast_copy_string(buf, peer->useragent, len);
} else if (!strcasecmp(colname, "mailbox")) {
struct ast_str *mailbox_str = ast_str_alloca(512);
peer_mailboxes_to_str(&mailbox_str, peer);
ast_copy_string(buf, mailbox_str->str, len);
} else if (!strcasecmp(colname, "context")) {
ast_copy_string(buf, peer->context, len);
} else if (!strcasecmp(colname, "expire")) {
snprintf(buf, len, "%d", peer->expire);
} else if (!strcasecmp(colname, "dynamic")) {
ast_copy_string(buf, peer->host_dynamic ? "yes" : "no", len);
} else if (!strcasecmp(colname, "callerid_name")) {
ast_copy_string(buf, peer->cid_name, len);
} else if (!strcasecmp(colname, "callerid_num")) {
ast_copy_string(buf, peer->cid_num, len);
} else if (!strcasecmp(colname, "codecs")) {
ast_getformatname_multiple(buf, len -1, peer->capability);
} else if (!strncasecmp(colname, "chanvar[", 8)) {
char *chanvar=colname + 8;
struct ast_variable *v;
chanvar = strsep(&chanvar, "]");
for (v = peer->chanvars ; v ; v = v->next) {
if (!strcasecmp(v->name, chanvar)) {
ast_copy_string(buf, v->value, len);
}
}
} else if (!strncasecmp(colname, "codec[", 6)) {
char *codecnum;
int codec = 0;
codecnum = colname + 6; /* move past the '[' */
codecnum = strsep(&codecnum, "]"); /* trim trailing ']' if any */
if((codec = ast_codec_pref_index(&peer->prefs, atoi(codecnum)))) {
ast_copy_string(buf, ast_getformatname(codec), len);
} else {
buf[0] = '\0';
}
} else {
buf[0] = '\0';
}
unref_peer(peer, "unref_peer from function_sippeer, just before return");
return 0;
}
/*! \brief Structure to declare a dialplan function: SIPPEER */
static struct ast_custom_function sippeer_function = {
.name = "SIPPEER",
.read = function_sippeer,
};
/*! \brief ${SIPCHANINFO()} Dialplan function - reads sip channel data */
static int function_sipchaninfo_read(struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len)
{
struct sip_pvt *p;
static int deprecated = 0;
*buf = 0;
if (!data) {
ast_log(LOG_WARNING, "This function requires a parameter name.\n");
return -1;
}
ast_channel_lock(chan);
if (!IS_SIP_TECH(chan->tech)) {
ast_log(LOG_WARNING, "This function can only be used on SIP channels.\n");
ast_channel_unlock(chan);
return -1;
}
if (deprecated++ % 20 == 0) {
/* Deprecated in 1.6.1 */
ast_log(LOG_WARNING, "SIPCHANINFO() is deprecated. Please transition to using CHANNEL().\n");
}
p = chan->tech_pvt;
/* If there is no private structure, this channel is no longer alive */
if (!p) {
ast_channel_unlock(chan);
return -1;
}
if (!strcasecmp(data, "peerip")) {
ast_copy_string(buf, p->sa.sin_addr.s_addr ? ast_inet_ntoa(p->sa.sin_addr) : "", len);
} else if (!strcasecmp(data, "recvip")) {
ast_copy_string(buf, p->recv.sin_addr.s_addr ? ast_inet_ntoa(p->recv.sin_addr) : "", len);
} else if (!strcasecmp(data, "from")) {
ast_copy_string(buf, p->from, len);
} else if (!strcasecmp(data, "uri")) {
ast_copy_string(buf, p->uri, len);
} else if (!strcasecmp(data, "useragent")) {
ast_copy_string(buf, p->useragent, len);
} else if (!strcasecmp(data, "peername")) {
ast_copy_string(buf, p->peername, len);
} else if (!strcasecmp(data, "t38passthrough")) {
if (p->t38.state == T38_DISABLED)
ast_copy_string(buf, "0", sizeof("0"));
else /* T38 is offered or enabled in this call */
ast_copy_string(buf, "1", sizeof("1"));
} else {
ast_channel_unlock(chan);
return -1;
}
ast_channel_unlock(chan);
return 0;
}
/*! \brief Structure to declare a dialplan function: SIPCHANINFO */
static struct ast_custom_function sipchaninfo_function = {
.name = "SIPCHANINFO",
.read = function_sipchaninfo_read,
};
static int read_to_parts(struct sip_pvt *p, struct sip_request *req, char **name, char **number)
{
char to_header[256];
char *to_name = NULL;
char *to_number = NULL;
char *separator;
ast_copy_string(to_header, get_header(req, "To"), sizeof(to_header));
/* Let's get that number first! */
to_number = get_in_brackets(to_header);
if (!strncasecmp(to_number, "sip:", 4)) {
to_number += 4;
} else if (!strncasecmp(to_number, "sips:", 5)) {
to_number += 5;
} else {
ast_log(LOG_WARNING, "Not a SIP URI? (%s)!\n", to_number);
return -1;
}
/* Remove the host and such since we just want the number */
if ((separator = strchr(to_number, '@'))) {
*separator = '\0';
}
/* We have the number. Let's get the name now. */
if (*to_header == '\"') {
to_name = to_header + 1;
if (!(separator = (char *)find_closing_quote(to_name, NULL))) {
ast_log(LOG_NOTICE, "No closing quote in name section of To: header (%s)\n", to_header);
return -1;
}
*separator = '\0';
}
if (number) {
*number = ast_strdup(to_number);
}
if (name && !ast_strlen_zero(to_name)) {
*name = ast_strdup(to_name);
}
return 0;
}
/*! \brief update redirecting information for a channel based on headers
*
*/
static void change_redirecting_information(struct sip_pvt *p, struct sip_request *req, struct ast_party_redirecting *redirecting, int set_call_forward)
{
char *redirecting_from_name = NULL;
char *redirecting_from_number = NULL;
char *redirecting_to_name = NULL;
char *redirecting_to_number = NULL;
int reason = AST_REDIRECTING_REASON_UNCONDITIONAL;
int is_response = req->method == SIP_RESPONSE;
int res = 0;
res = get_rdnis(p, req, &redirecting_from_name, &redirecting_from_number, &reason);
if (res == -1) {
if (is_response) {
read_to_parts(p, req, &redirecting_from_name, &redirecting_from_number);
} else {
return;
}
}
/* At this point, all redirecting "from" info should be filled in appropriately
* on to the "to" info
*/
if (is_response) {
parse_moved_contact(p, req, &redirecting_to_name, &redirecting_to_number, set_call_forward);
} else {
read_to_parts(p, req, &redirecting_to_name, &redirecting_to_number);
}
if (!ast_strlen_zero(redirecting_from_number)) {
if (redirecting->from.number) {
ast_free(redirecting->from.number);
}
ast_debug(3, "Got redirecting from number %s\n", redirecting_from_number);
redirecting->from.number = redirecting_from_number;
}
if (!ast_strlen_zero(redirecting_from_name)) {
if (redirecting->from.name) {
ast_free(redirecting->from.name);
}
ast_debug(3, "Got redirecting from name %s\n", redirecting_from_name);
redirecting->from.name = redirecting_from_name;
}
if (!ast_strlen_zero(redirecting_to_number)) {
if (redirecting->to.number) {
ast_free(redirecting->to.number);
}
ast_debug(3, "Got redirecting to number %s\n", redirecting_to_number);
redirecting->to.number = redirecting_to_number;
}
if (!ast_strlen_zero(redirecting_to_name)) {
if (redirecting->to.name) {
ast_free(redirecting->to.name);
}
ast_debug(3, "Got redirecting to name %s\n", redirecting_from_number);
redirecting->to.name = redirecting_to_name;
}
redirecting->reason = reason;
}
/*! \brief Parse 302 Moved temporalily response
\todo XXX Doesn't redirect over TLS on sips: uri's.
If we get a redirect to a SIPS: uri, this needs to be going back to the
dialplan (this is a request for a secure signalling path).
Note that transport=tls is deprecated, but we need to support it on incoming requests.
*/
static void parse_moved_contact(struct sip_pvt *p, struct sip_request *req, char **name, char **number, int set_call_forward)
{
char contact[SIPBUFSIZE];
char *contact_name = NULL;
char *contact_number = NULL;
char *separator, *trans;
char *domain;
enum sip_transport transport = SIP_TRANSPORT_UDP;
ast_copy_string(contact, get_header(req, "Contact"), sizeof(contact));
if ((separator = strchr(contact, ',')))
*separator = '\0';
/* ooh, a name */
if (*contact == '"') {
contact_name = contact + 1;
if ((separator = strchr(contact_name, '"'))) {
*separator++ = '\0';
}
}
contact_number = get_in_brackets(contact);
if ((trans = strcasestr(contact_number, ";transport="))) {
trans += 11;
if ((separator = strchr(trans, ';')))
*separator = '\0';
if (!strncasecmp(trans, "tcp", 3))
transport = SIP_TRANSPORT_TCP;
else if (!strncasecmp(trans, "tls", 3))
transport = SIP_TRANSPORT_TLS;
else {
if (strncasecmp(trans, "udp", 3))
ast_debug(1, "received contact with an invalid transport, '%s'\n", contact_number);
/* This will assume UDP for all unknown transports */
transport = SIP_TRANSPORT_UDP;
}
}
contact_number = remove_uri_parameters(contact_number);
if (p->socket.tcptls_session) {
ao2_ref(p->socket.tcptls_session, -1);
p->socket.tcptls_session = NULL;
}
p->socket.fd = -1;
p->socket.type = transport;
if (set_call_forward && ast_test_flag(&p->flags[0], SIP_PROMISCREDIR)) {
char *host = NULL;
if (!strncasecmp(contact_number, "sip:", 4))
contact_number += 4;
else if (!strncasecmp(contact_number, "sips:", 5))
contact_number += 5;
separator = strchr(contact_number, '/');
if (separator)
*separator = '\0';
if ((host = strchr(contact_number, '@'))) {
*host++ = '\0';
ast_debug(2, "Found promiscuous redirection to 'SIP/%s::::%s@%s'\n", contact_number, get_transport(transport), host);
if (p->owner)
ast_string_field_build(p->owner, call_forward, "SIP/%s::::%s@%s", contact_number, get_transport(transport), host);
} else {
ast_debug(2, "Found promiscuous redirection to 'SIP/::::%s@%s'\n", get_transport(transport), contact_number);
if (p->owner)
ast_string_field_build(p->owner, call_forward, "SIP/::::%s@%s", get_transport(transport), contact_number);
}
} else {
separator = strchr(contact, '@');
if (separator) {
*separator++ = '\0';
domain = separator;
} else {
/* No username part */
domain = contact;
}
separator = strchr(contact, '/'); /* WHEN do we hae a forward slash in the URI? */
if (separator)
*separator = '\0';
if (!strncasecmp(contact_number, "sip:", 4))
contact_number += 4;
else if (!strncasecmp(contact_number, "sips:", 5))
contact_number += 5;
separator = strchr(contact_number, ';'); /* And username ; parameters? */
if (separator)
*separator = '\0';
if (set_call_forward) {
ast_debug(2, "Received 302 Redirect to extension '%s' (domain %s)\n", contact_number, domain);
if (p->owner) {
pbx_builtin_setvar_helper(p->owner, "SIPDOMAIN", domain);
ast_string_field_set(p->owner, call_forward, contact_number);
}
}
}
/* We've gotten the number for the contact, now get the name */
if (*contact == '\"') {
contact_name = contact + 1;
if (!(separator = (char *)find_closing_quote(contact_name, NULL))) {
ast_log(LOG_NOTICE, "No closing quote on name in Contact header? %s\n", contact);
}
*separator = '\0';
}
if (name && !ast_strlen_zero(contact_name)) {
*name = ast_strdup(contact_name);
}
if (number) {
*number = ast_strdup(contact_number);
}
}
/*! \brief Check pending actions on SIP call */
static void check_pendings(struct sip_pvt *p)
{
if (ast_test_flag(&p->flags[0], SIP_PENDINGBYE)) {
/* if we can't BYE, then this is really a pending CANCEL */
if (p->invitestate == INV_PROCEEDING || p->invitestate == INV_EARLY_MEDIA)
transmit_request(p, SIP_CANCEL, p->lastinvite, XMIT_RELIABLE, FALSE);
/* Actually don't destroy us yet, wait for the 487 on our original
INVITE, but do set an autodestruct just in case we never get it. */
else {
/* We have a pending outbound invite, don't send something
new in-transaction */
if (p->pendinginvite)
return;
/* Perhaps there is an SD change INVITE outstanding */
transmit_request_with_auth(p, SIP_BYE, 0, XMIT_RELIABLE, TRUE);
}
ast_clear_flag(&p->flags[0], SIP_PENDINGBYE);
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
} else if (ast_test_flag(&p->flags[0], SIP_NEEDREINVITE)) {
/* if we can't REINVITE, hold it for later */
if (p->pendinginvite || p->invitestate == INV_CALLING || p->invitestate == INV_PROCEEDING || p->invitestate == INV_EARLY_MEDIA || p->waitid > 0) {
ast_debug(2, "NOT Sending pending reinvite (yet) on '%s'\n", p->callid);
} else {
ast_debug(2, "Sending pending reinvite on '%s'\n", p->callid);
/* Didn't get to reinvite yet, so do it now */
transmit_reinvite_with_sdp(p, (p->t38.state == T38_LOCAL_REINVITE ? TRUE : FALSE), FALSE);
ast_clear_flag(&p->flags[0], SIP_NEEDREINVITE);
}
}
}
/*! \brief Reset the NEEDREINVITE flag after waiting when we get 491 on a Re-invite
to avoid race conditions between asterisk servers.
Called from the scheduler.
*/
static int sip_reinvite_retry(const void *data)
{
struct sip_pvt *p = (struct sip_pvt *) data;
sip_pvt_lock(p); /* called from schedule thread which requires a lock */
ast_set_flag(&p->flags[0], SIP_NEEDREINVITE);
p->waitid = -1;
check_pendings(p);
sip_pvt_unlock(p);
dialog_unref(p, "unref the dialog ptr from sip_reinvite_retry, because it held a dialog ptr");
return 0;
}
/*! \brief Handle SIP response to INVITE dialogue */
static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int seqno)
{
int outgoing = ast_test_flag(&p->flags[0], SIP_OUTGOING);
int res = 0;
int xmitres = 0;
int reinvite = (p->owner && p->owner->_state == AST_STATE_UP);
char *p_hdrval;
int rtn;
struct ast_party_connected_line connected;
if (reinvite)
ast_debug(4, "SIP response %d to RE-invite on %s call %s\n", resp, outgoing ? "outgoing" : "incoming", p->callid);
else
ast_debug(4, "SIP response %d to standard invite\n", resp);
if (p->alreadygone) { /* This call is already gone */
ast_debug(1, "Got response on call that is already terminated: %s (ignoring)\n", p->callid);
return;
}
/* Acknowledge sequence number - This only happens on INVITE from SIP-call */
/* Don't auto congest anymore since we've gotten something useful back */
AST_SCHED_DEL_UNREF(sched, p->initid, dialog_unref(p, "when you delete the initid sched, you should dec the refcount for the stored dialog ptr"));
/* RFC3261 says we must treat every 1xx response (but not 100)
that we don't recognize as if it was 183.
*/
if (resp > 100 && resp < 200 && resp!=101 && resp != 180 && resp != 181 && resp != 182 && resp != 183)
resp = 183;
/* Any response between 100 and 199 is PROCEEDING */
if (resp >= 100 && resp < 200 && p->invitestate == INV_CALLING)
p->invitestate = INV_PROCEEDING;
/* Final response, not 200 ? */
if (resp >= 300 && (p->invitestate == INV_CALLING || p->invitestate == INV_PROCEEDING || p->invitestate == INV_EARLY_MEDIA ))
p->invitestate = INV_COMPLETED;
/* Final response, clear out pending invite */
if ((resp == 200 || resp >= 300) && p->pendinginvite && seqno == p->pendinginvite)
p->pendinginvite = 0;
switch (resp) {
case 100: /* Trying */
case 101: /* Dialog establishment */
if (!req->ignore && p->invitestate != INV_CANCELLED && sip_cancel_destroy(p))
ast_log(LOG_WARNING, "Unable to cancel SIP destruction. Expect bad things.\n");
check_pendings(p);
break;
case 180: /* 180 Ringing */
case 182: /* 182 Queued */
if (!req->ignore && p->invitestate != INV_CANCELLED && sip_cancel_destroy(p))
ast_log(LOG_WARNING, "Unable to cancel SIP destruction. Expect bad things.\n");
if (!req->ignore && p->owner) {
if (get_rpid(p, req)) {
ast_party_connected_line_init(&connected);
connected.id.number = (char *) p->cid_num;
connected.id.name = (char *) p->cid_name;
connected.id.number_presentation = p->callingpres;
connected.source = AST_CONNECTED_LINE_UPDATE_SOURCE_ANSWER;
ast_channel_queue_connected_line_update(p->owner, &connected);
}
ast_queue_control(p->owner, AST_CONTROL_RINGING);
if (p->owner->_state != AST_STATE_UP) {
ast_setstate(p->owner, AST_STATE_RINGING);
}
}
if (find_sdp(req)) {
if (p->invitestate != INV_CANCELLED)
p->invitestate = INV_EARLY_MEDIA;
res = process_sdp(p, req, SDP_T38_NONE);
if (!req->ignore && p->owner) {
/* Queue a progress frame only if we have SDP in 180 or 182 */
ast_queue_control(p->owner, AST_CONTROL_PROGRESS);
}
}
check_pendings(p);
break;
case 181: /* Call Is Being Forwarded */
if (!req->ignore && (p->invitestate != INV_CANCELLED) && sip_cancel_destroy(p))
ast_log(LOG_WARNING, "Unable to cancel SIP destruction. Expect bad things.\n");
if (!req->ignore && p->owner) {
struct ast_party_redirecting redirecting = {{0,},};
change_redirecting_information(p, req, &redirecting, FALSE);
ast_channel_queue_redirecting_update(p->owner, &redirecting);
}
check_pendings(p);
break;
case 183: /* Session progress */
if (!req->ignore && (p->invitestate != INV_CANCELLED) && sip_cancel_destroy(p))
ast_log(LOG_WARNING, "Unable to cancel SIP destruction. Expect bad things.\n");
/* Ignore 183 Session progress without SDP */
if (!req->ignore && p->owner) {
if (get_rpid(p, req)) {
/* Queue a connected line update */
ast_party_connected_line_init(&connected);
connected.id.number = (char *) p->cid_num;
connected.id.name = (char *) p->cid_name;
connected.id.number_presentation = p->callingpres;
connected.source = AST_CONNECTED_LINE_UPDATE_SOURCE_ANSWER;
ast_channel_queue_connected_line_update(p->owner, &connected);
}
}
if (find_sdp(req)) {
if (p->invitestate != INV_CANCELLED)
p->invitestate = INV_EARLY_MEDIA;
res = process_sdp(p, req, SDP_T38_NONE);
if (!req->ignore && p->owner) {
/* Queue a progress frame */
ast_queue_control(p->owner, AST_CONTROL_PROGRESS);
}
}
check_pendings(p);
break;
case 200: /* 200 OK on invite - someone's answering our call */
if (!req->ignore && (p->invitestate != INV_CANCELLED) && sip_cancel_destroy(p))
ast_log(LOG_WARNING, "Unable to cancel SIP destruction. Expect bad things.\n");
p->authtries = 0;
if (find_sdp(req)) {
if ((res = process_sdp(p, req, SDP_T38_ACCEPT)) && !req->ignore)
if (!reinvite)
/* This 200 OK's SDP is not acceptable, so we need to ack, then hangup */
/* For re-invites, we try to recover */
ast_set_flag(&p->flags[0], SIP_PENDINGBYE);
}
if (!req->ignore && p->owner && get_rpid(p, req)) {
/* Queue a connected line update */
ast_party_connected_line_init(&connected);
connected.id.number = (char *) p->cid_num;
connected.id.name = (char *) p->cid_name;
connected.id.number_presentation = p->callingpres;
connected.source = AST_CONNECTED_LINE_UPDATE_SOURCE_ANSWER;
ast_channel_queue_connected_line_update(p->owner, &connected);
}
/* Parse contact header for continued conversation */
/* When we get 200 OK, we know which device (and IP) to contact for this call */
/* This is important when we have a SIP proxy between us and the phone */
if (outgoing) {
update_call_counter(p, DEC_CALL_RINGING);
parse_ok_contact(p, req);
/* Save Record-Route for any later requests we make on this dialogue */
if (!reinvite)
build_route(p, req, 1);
if(set_address_from_contact(p)) {
/* Bad contact - we don't know how to reach this device */
/* We need to ACK, but then send a bye */
if (!p->route && !req->ignore)
ast_set_flag(&p->flags[0], SIP_PENDINGBYE);
}
}
if (!req->ignore && p->owner) {
if (!reinvite) {
struct ast_party_connected_line connected;
ast_party_connected_line_collect_caller(&connected, &p->owner->cid);
ast_channel_queue_connected_line_update(p->owner, &connected);
ast_queue_control(p->owner, AST_CONTROL_ANSWER);
if (sip_cfg.callevents)
manager_event(EVENT_FLAG_SYSTEM, "ChannelUpdate",
"Channel: %s\r\nChanneltype: %s\r\nUniqueid: %s\r\nSIPcallid: %s\r\nSIPfullcontact: %s\r\nPeername: %s\r\n",
p->owner->name, "SIP", p->owner->uniqueid, p->callid, p->fullcontact, p->peername);
} else { /* RE-invite */
ast_queue_frame(p->owner, &ast_null_frame);
}
} else {
/* It's possible we're getting an 200 OK after we've tried to disconnect
by sending CANCEL */
/* First send ACK, then send bye */
if (!req->ignore)
ast_set_flag(&p->flags[0], SIP_PENDINGBYE);
}
/* Check for Session-Timers related headers */
if (st_get_mode(p) != SESSION_TIMER_MODE_REFUSE && p->outgoing_call == TRUE && !reinvite) {
p_hdrval = (char*)get_header(req, "Session-Expires");
if (!ast_strlen_zero(p_hdrval)) {
/* UAS supports Session-Timers */
enum st_refresher tmp_st_ref = SESSION_TIMER_REFRESHER_AUTO;
int tmp_st_interval = 0;
rtn = parse_session_expires(p_hdrval, &tmp_st_interval, &tmp_st_ref);
if (rtn != 0) {
ast_set_flag(&p->flags[0], SIP_PENDINGBYE);
}
if (tmp_st_ref == SESSION_TIMER_REFRESHER_UAC ||
tmp_st_ref == SESSION_TIMER_REFRESHER_UAS) {
p->stimer->st_ref = tmp_st_ref;
}
if (tmp_st_interval) {
p->stimer->st_interval = tmp_st_interval;
}
p->stimer->st_active = TRUE;
p->stimer->st_active_peer_ua = TRUE;
start_session_timer(p);
} else {
/* UAS doesn't support Session-Timers */
if (st_get_mode(p) == SESSION_TIMER_MODE_ORIGINATE) {
p->stimer->st_ref = SESSION_TIMER_REFRESHER_UAC;
p->stimer->st_active_peer_ua = FALSE;
start_session_timer(p);
}
}
}
/* If I understand this right, the branch is different for a non-200 ACK only */
p->invitestate = INV_TERMINATED;
ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, TRUE);
check_pendings(p);
break;
case 407: /* Proxy authentication */
case 401: /* Www auth */
/* First we ACK */
xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
if (p->options)
p->options->auth_type = resp;
/* Then we AUTH */
ast_string_field_set(p, theirtag, NULL); /* forget their old tag, so we don't match tags when getting response */
if (!req->ignore) {
if (p->authtries < MAX_AUTHTRIES)
p->invitestate = INV_CALLING;
if (p->authtries == MAX_AUTHTRIES || do_proxy_auth(p, req, resp, SIP_INVITE, 1)) {
ast_log(LOG_NOTICE, "Failed to authenticate on INVITE to '%s'\n", get_header(&p->initreq, "From"));
pvt_set_needdestroy(p, "failed to authenticate on INVITE");
sip_alreadygone(p);
if (p->owner)
ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
}
}
break;
case 403: /* Forbidden */
/* First we ACK */
xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
ast_log(LOG_WARNING, "Received response: \"Forbidden\" from '%s'\n", get_header(&p->initreq, "From"));
if (!req->ignore && p->owner)
ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
pvt_set_needdestroy(p, "received 403 response");
sip_alreadygone(p);
break;
case 404: /* Not found */
xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
if (p->owner && !req->ignore)
ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
sip_alreadygone(p);
break;
case 408: /* Request timeout */
case 481: /* Call leg does not exist */
/* Could be REFER caused INVITE with replaces */
ast_log(LOG_WARNING, "Re-invite to non-existing call leg on other UA. SIP dialog '%s'. Giving up.\n", p->callid);
xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
if (p->owner)
ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
break;
case 422: /* Session-Timers: Session interval too small */
xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
ast_string_field_set(p, theirtag, NULL);
proc_422_rsp(p, req);
break;
case 428: /* Use identity header - rfc 4474 - not supported by Asterisk yet */
xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
append_history(p, "Identity", "SIP identity is required. Not supported by Asterisk.");
ast_log(LOG_WARNING, "SIP identity required by proxy. SIP dialog '%s'. Giving up.\n", p->callid);
if (p->owner)
ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
break;
case 487: /* Cancelled transaction */
/* We have sent CANCEL on an outbound INVITE
This transaction is already scheduled to be killed by sip_hangup().
*/
xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
if (p->owner && !req->ignore) {
ast_queue_hangup_with_cause(p->owner, AST_CAUSE_NORMAL_CLEARING);
append_history(p, "Hangup", "Got 487 on CANCEL request from us. Queued AST hangup request");
} else if (!req->ignore) {
update_call_counter(p, DEC_CALL_LIMIT);
append_history(p, "Hangup", "Got 487 on CANCEL request from us on call without owner. Killing this dialog.");
pvt_set_needdestroy(p, "received 487 response");
sip_alreadygone(p);
}
break;
case 488: /* Not acceptable here */
xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
if (p->udptl && p->t38.state == T38_LOCAL_REINVITE) {
change_t38_state(p, T38_DISABLED);
/* Try to reset RTP timers */
//ast_rtp_set_rtptimers_onhold(p->rtp);
/* Trigger a reinvite back to audio */
transmit_reinvite_with_sdp(p, FALSE, FALSE);
} else {
/* We can't set up this call, so give up */
if (p->owner && !req->ignore)
ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
pvt_set_needdestroy(p, "received 488 response");
/* If there's no dialog to end, then mark p as already gone */
if (!reinvite)
sip_alreadygone(p);
}
break;
case 491: /* Pending */
xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
if (p->owner && !req->ignore) {
if (p->owner->_state != AST_STATE_UP) {
ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
pvt_set_needdestroy(p, "received 491 response");
} else {
/* This is a re-invite that failed. */
/* Reset the flag after a while
*/
int wait;
/* RFC 3261, if owner of call, wait between 2.1 to 4 seconds,
* if not owner of call, wait 0 to 2 seconds */
if (p->outgoing_call) {
wait = 2100 + ast_random() % 2000;
} else {
wait = ast_random() % 2000;
}
p->waitid = ast_sched_add(sched, wait, sip_reinvite_retry, dialog_ref(p, "passing dialog ptr into sched structure based on waitid for sip_reinvite_retry."));
ast_log(LOG_WARNING, "just did sched_add waitid(%d) for sip_reinvite_retry for dialog %s in handle_response_invite\n", p->waitid, p->callid);
ast_debug(2, "Reinvite race. Waiting %d secs before retry\n", wait);
}
}
break;
case 501: /* Not implemented */
xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
if (p->owner)
ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
break;
}
if (xmitres == XMIT_ERROR)
ast_log(LOG_WARNING, "Could not transmit message in dialog %s\n", p->callid);
}
/* \brief Handle SIP response in NOTIFY transaction
We've sent a NOTIFY, now handle responses to it
*/
static void handle_response_notify(struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int seqno)
{
switch (resp) {
case 200: /* Notify accepted */
/* They got the notify, this is the end */
if (p->owner) {
if (!p->refer) {
ast_log(LOG_WARNING, "Notify answer on an owned channel? - %s\n", p->owner->name);
ast_queue_hangup_with_cause(p->owner, AST_CAUSE_NORMAL_UNSPECIFIED);
} else {
ast_debug(4, "Got OK on REFER Notify message\n");
}
} else {
if (p->subscribed == NONE) {
ast_debug(4, "Got 200 accepted on NOTIFY\n");
pvt_set_needdestroy(p, "received 200 response");
}
if (ast_test_flag(&p->flags[1], SIP_PAGE2_STATECHANGEQUEUE)) {
/* Ready to send the next state we have on queue */
ast_clear_flag(&p->flags[1], SIP_PAGE2_STATECHANGEQUEUE);
cb_extensionstate((char *)p->context, (char *)p->exten, p->laststate, (void *) p);
}
}
break;
case 401: /* Not www-authorized on SIP method */
case 407: /* Proxy auth */
if (!p->notify_headers) {
break; /* Only device notify can use NOTIFY auth */
}
ast_string_field_set(p, theirtag, NULL);
if (ast_strlen_zero(p->authname)) {
ast_log(LOG_WARNING, "Asked to authenticate NOTIFY to %s:%d but we have no matching peer or realm auth!\n", ast_inet_ntoa(p->recv.sin_addr), ntohs(p->recv.sin_port));
pvt_set_needdestroy(p, "unable to authenticate NOTIFY");
}
if (p->authtries > 1 || do_proxy_auth(p, req, resp, SIP_NOTIFY, 0)) {
ast_log(LOG_NOTICE, "Failed to authenticate on NOTIFY to '%s'\n", get_header(&p->initreq, "From"));
pvt_set_needdestroy(p, "failed to authenticate NOTIFY");
}
break;
}
}
/* \brief Handle SIP response in SUBSCRIBE transaction */
static void handle_response_subscribe(struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int seqno)
{
if (!p->mwi) {
return;
}
switch (resp) {
case 200: /* Subscription accepted */
ast_debug(3, "Got 200 OK on subscription for MWI\n");
if (p->options) {
ast_free(p->options);
p->options = NULL;
}
p->mwi->subscribed = 1;
if ((p->mwi->resub = ast_sched_add(sched, mwi_expiry * 1000, sip_subscribe_mwi_do, ASTOBJ_REF(p->mwi))) < 0) {
ASTOBJ_UNREF(p->mwi, sip_subscribe_mwi_destroy);
}
break;
case 401:
case 407:
ast_string_field_set(p, theirtag, NULL);
if (p->authtries > 1 || do_proxy_auth(p, req, resp, SIP_SUBSCRIBE, 0)) {
ast_log(LOG_NOTICE, "Failed to authenticate on SUBSCRIBE to '%s'\n", get_header(&p->initreq, "From"));
p->mwi->call = NULL;
ASTOBJ_UNREF(p->mwi, sip_subscribe_mwi_destroy);
pvt_set_needdestroy(p, "failed to authenticate SUBSCRIBE");
}
break;
case 403:
transmit_response_with_date(p, "200 OK", req);
ast_log(LOG_WARNING, "Authentication failed while trying to subscribe for MWI.\n");
p->mwi->call = NULL;
ASTOBJ_UNREF(p->mwi, sip_subscribe_mwi_destroy);
pvt_set_needdestroy(p, "received 403 response");
sip_alreadygone(p);
break;
case 404:
ast_log(LOG_WARNING, "Subscription failed for MWI. The remote side said that a mailbox may not have been configured.\n");
p->mwi->call = NULL;
ASTOBJ_UNREF(p->mwi, sip_subscribe_mwi_destroy);
pvt_set_needdestroy(p, "received 404 response");
break;
case 481:
ast_log(LOG_WARNING, "Subscription failed for MWI. The remote side said that our dialog did not exist.\n");
p->mwi->call = NULL;
ASTOBJ_UNREF(p->mwi, sip_subscribe_mwi_destroy);
pvt_set_needdestroy(p, "received 481 response");
break;
case 500:
case 501:
ast_log(LOG_WARNING, "Subscription failed for MWI. The remote side may have suffered a heart attack.\n");
p->mwi->call = NULL;
ASTOBJ_UNREF(p->mwi, sip_subscribe_mwi_destroy);
pvt_set_needdestroy(p, "received 500/501 response");
break;
}
}
/* \brief Handle SIP response in REFER transaction
We've sent a REFER, now handle responses to it
*/
static void handle_response_refer(struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int seqno)
{
enum ast_control_transfer message = AST_TRANSFER_FAILED;
/* If no refer structure exists, then do nothing */
if (!p->refer)
return;
switch (resp) {
case 202: /* Transfer accepted */
/* We need to do something here */
/* The transferee is now sending INVITE to target */
p->refer->status = REFER_ACCEPTED;
/* Now wait for next message */
ast_debug(3, "Got 202 accepted on transfer\n");
/* We should hang along, waiting for NOTIFY's here */
break;
case 401: /* Not www-authorized on SIP method */
case 407: /* Proxy auth */
if (ast_strlen_zero(p->authname)) {
ast_log(LOG_WARNING, "Asked to authenticate REFER to %s:%d but we have no matching peer or realm auth!\n",
ast_inet_ntoa(p->recv.sin_addr), ntohs(p->recv.sin_port));
if (p->owner) {
ast_queue_control_data(p->owner, AST_CONTROL_TRANSFER, &message, sizeof(message));
}
pvt_set_needdestroy(p, "unable to authenticate REFER");
}
if (p->authtries > 1 || do_proxy_auth(p, req, resp, SIP_REFER, 0)) {
ast_log(LOG_NOTICE, "Failed to authenticate on REFER to '%s'\n", get_header(&p->initreq, "From"));
p->refer->status = REFER_NOAUTH;
if (p->owner) {
ast_queue_control_data(p->owner, AST_CONTROL_TRANSFER, &message, sizeof(message));
}
pvt_set_needdestroy(p, "failed to authenticate REFER");
}
break;
case 481: /* Call leg does not exist */
/* A transfer with Replaces did not work */
/* OEJ: We should Set flag, cancel the REFER, go back
to original call - but right now we can't */
ast_log(LOG_WARNING, "Remote host can't match REFER request to call '%s'. Giving up.\n", p->callid);
if (p->owner)
ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
pvt_set_needdestroy(p, "received 481 response");
break;
case 500: /* Server error */
case 501: /* Method not implemented */
/* Return to the current call onhold */
/* Status flag needed to be reset */
ast_log(LOG_NOTICE, "SIP transfer to %s failed, call miserably fails. \n", p->refer->refer_to);
pvt_set_needdestroy(p, "received 500/501 response");
p->refer->status = REFER_FAILED;
if (p->owner) {
ast_queue_control_data(p->owner, AST_CONTROL_TRANSFER, &message, sizeof(message));
}
break;
case 603: /* Transfer declined */
ast_log(LOG_NOTICE, "SIP transfer to %s declined, call miserably fails. \n", p->refer->refer_to);
p->refer->status = REFER_FAILED;
pvt_set_needdestroy(p, "received 603 response");
if (p->owner) {
ast_queue_control_data(p->owner, AST_CONTROL_TRANSFER, &message, sizeof(message));
}
break;
}
}
/*! \brief Handle responses on REGISTER to services */
static int handle_response_register(struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int seqno)
{
int expires, expires_ms;
struct sip_registry *r;
r=p->registry;
switch (resp) {
case 401: /* Unauthorized */
if (p->authtries == MAX_AUTHTRIES || do_register_auth(p, req, resp)) {
ast_log(LOG_NOTICE, "Failed to authenticate on REGISTER to '%s@%s' (Tries %d)\n", p->registry->username, p->registry->hostname, p->authtries);
pvt_set_needdestroy(p, "failed to authenticate REGISTER");
}
break;
case 403: /* Forbidden */
ast_log(LOG_WARNING, "Forbidden - wrong password on authentication for REGISTER for '%s' to '%s'\n", p->registry->username, p->registry->hostname);
AST_SCHED_DEL(sched, r->timeout);
r->regstate = REG_STATE_NOAUTH;
pvt_set_needdestroy(p, "received 403 response");
break;
case 404: /* Not found */
ast_log(LOG_WARNING, "Got 404 Not found on SIP register to service %s@%s, giving up\n", p->registry->username, p->registry->hostname);
pvt_set_needdestroy(p, "received 404 response");
if (r->call)
r->call = dialog_unref(r->call, "unsetting registry->call pointer-- case 404");
r->regstate = REG_STATE_REJECTED;
AST_SCHED_DEL(sched, r->timeout);
break;
case 407: /* Proxy auth */
if (p->authtries == MAX_AUTHTRIES || do_register_auth(p, req, resp)) {
ast_log(LOG_NOTICE, "Failed to authenticate on REGISTER to '%s' (tries '%d')\n", get_header(&p->initreq, "From"), p->authtries);
pvt_set_needdestroy(p, "failed to authenticate REGISTER");
}
break;
case 408: /* Request timeout */
/* Got a timeout response, so reset the counter of failed responses */
if (r) {
r->regattempts = 0;
} else {
ast_log(LOG_WARNING, "Got a 408 response to our REGISTER on call %s after we had destroyed the registry object\n", p->callid);
}
break;
case 423: /* Interval too brief */
r->expiry = atoi(get_header(req, "Min-Expires"));
ast_log(LOG_WARNING, "Got 423 Interval too brief for service %s@%s, minimum is %d seconds\n", p->registry->username, p->registry->hostname, r->expiry);
AST_SCHED_DEL(sched, r->timeout);
r->timeout = -1;
if (r->call) {
r->call = dialog_unref(r->call, "unsetting registry->call pointer-- case 423");
pvt_set_needdestroy(p, "received 423 response");
}
if (r->expiry > max_expiry) {
ast_log(LOG_WARNING, "Required expiration time from %s@%s is too high, giving up\n", p->registry->username, p->registry->hostname);
r->expiry = default_expiry;
r->regstate = REG_STATE_REJECTED;
} else {
r->regstate = REG_STATE_UNREGISTERED;
transmit_register(r, SIP_REGISTER, NULL, NULL);
}
manager_event(EVENT_FLAG_SYSTEM, "Registry", "ChannelType: SIP\r\nUsername: %s\r\nDomain: %s\r\nStatus: %s\r\n", r->username, r->hostname, regstate2str(r->regstate));
break;
case 479: /* SER: Not able to process the URI - address is wrong in register*/
ast_log(LOG_WARNING, "Got error 479 on register to %s@%s, giving up (check config)\n", p->registry->username, p->registry->hostname);
pvt_set_needdestroy(p, "received 479 response");
if (r->call)
r->call = dialog_unref(r->call, "unsetting registry->call pointer-- case 479");
r->regstate = REG_STATE_REJECTED;
AST_SCHED_DEL(sched, r->timeout);
break;
case 200: /* 200 OK */
if (!r) {
ast_log(LOG_WARNING, "Got 200 OK on REGISTER, but there isn't a registry entry for '%s' (we probably already got the OK)\n", S_OR(p->peername, p->username));
pvt_set_needdestroy(p, "received erroneous 200 response");
return 0;
}
r->regstate = REG_STATE_REGISTERED;
r->regtime = ast_tvnow(); /* Reset time of last succesful registration */
manager_event(EVENT_FLAG_SYSTEM, "Registry", "ChannelType: SIP\r\nDomain: %s\r\nStatus: %s\r\n", r->hostname, regstate2str(r->regstate));
r->regattempts = 0;
ast_debug(1, "Registration successful\n");
if (r->timeout > -1) {
ast_debug(1, "Cancelling timeout %d\n", r->timeout);
}
AST_SCHED_DEL(sched, r->timeout);
if (r->call)
r->call = dialog_unref(r->call, "unsetting registry->call pointer-- case 200");
p->registry = registry_unref(p->registry, "unref registry entry p->registry");
/* Let this one hang around until we have all the responses */
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
/* p->needdestroy = 1; */
/* set us up for re-registering */
/* figure out how long we got registered for */
AST_SCHED_DEL(sched, r->expire);
/* according to section 6.13 of RFC, contact headers override
expires headers, so check those first */
expires = 0;
/* XXX todo: try to save the extra call */
if (!ast_strlen_zero(get_header(req, "Contact"))) {
const char *contact = NULL;
const char *tmptmp = NULL;
int start = 0;
for(;;) {
contact = __get_header(req, "Contact", &start);
/* this loop ensures we get a contact header about our register request */
if(!ast_strlen_zero(contact)) {
if( (tmptmp=strstr(contact, p->our_contact))) {
contact=tmptmp;
break;
}
} else
break;
}
tmptmp = strcasestr(contact, "expires=");
if (tmptmp) {
if (sscanf(tmptmp + 8, "%d;", &expires) != 1)
expires = 0;
}
}
if (!expires)
expires=atoi(get_header(req, "expires"));
if (!expires)
expires=default_expiry;
expires_ms = expires * 1000;
if (expires <= EXPIRY_GUARD_LIMIT)
expires_ms -= MAX((expires_ms * EXPIRY_GUARD_PCT), EXPIRY_GUARD_MIN);
else
expires_ms -= EXPIRY_GUARD_SECS * 1000;
if (sipdebug)
ast_log(LOG_NOTICE, "Outbound Registration: Expiry for %s is %d sec (Scheduling reregistration in %d s)\n", r->hostname, expires, expires_ms/1000);
r->refresh= (int) expires_ms / 1000;
/* Schedule re-registration before we expire */
AST_SCHED_REPLACE_UNREF(r->expire, sched, expires_ms, sip_reregister, r,
registry_unref(_data,"unref in REPLACE del fail"),
registry_unref(r,"unref in REPLACE add fail"),
registry_addref(r,"The Addition side of REPLACE"));
/* it is clear that we would not want to destroy the registry entry if we just
scheduled a callback and recorded it in there! */
/* since we never bumped the count, we shouldn't decrement it! registry_unref(r, "unref registry ptr r"); if this gets deleted, p->registry will be a bad pointer! */
}
return 1;
}
/*! \brief Handle qualification responses (OPTIONS) */
static void handle_response_peerpoke(struct sip_pvt *p, int resp, struct sip_request *req)
{
struct sip_peer *peer = /* ref_peer( */ p->relatedpeer /* , "bump refcount on p, as it is being used in this function(handle_response_peerpoke)")*/ ; /* hope this is already refcounted! */
int statechanged, is_reachable, was_reachable;
int pingtime = ast_tvdiff_ms(ast_tvnow(), peer->ps);
/*
* Compute the response time to a ping (goes in peer->lastms.)
* -1 means did not respond, 0 means unknown,
* 1..maxms is a valid response, >maxms means late response.
*/
if (pingtime < 1) /* zero = unknown, so round up to 1 */
pingtime = 1;
/* Now determine new state and whether it has changed.
* Use some helper variables to simplify the writing
* of the expressions.
*/
was_reachable = peer->lastms > 0 && peer->lastms <= peer->maxms;
is_reachable = pingtime <= peer->maxms;
statechanged = peer->lastms == 0 /* yes, unknown before */
|| was_reachable != is_reachable;
peer->lastms = pingtime;
peer->call = dialog_unref(peer->call, "unref dialog peer->call");
if (statechanged) {
const char *s = is_reachable ? "Reachable" : "Lagged";
char str_lastms[20];
snprintf(str_lastms, sizeof(str_lastms), "%d", pingtime);
ast_log(LOG_NOTICE, "Peer '%s' is now %s. (%dms / %dms)\n",
peer->name, s, pingtime, peer->maxms);
ast_devstate_changed(AST_DEVICE_UNKNOWN, "SIP/%s", peer->name);
if (sip_cfg.peer_rtupdate) {
ast_update_realtime(ast_check_realtime("sipregs") ? "sipregs" : "sippeers", "name", peer->name, "lastms", str_lastms, SENTINEL);
}
manager_event(EVENT_FLAG_SYSTEM, "PeerStatus",
"ChannelType: SIP\r\nPeer: SIP/%s\r\nPeerStatus: %s\r\nTime: %d\r\n",
peer->name, s, pingtime);
if (is_reachable && sip_cfg.regextenonqualify)
register_peer_exten(peer, TRUE);
}
pvt_set_needdestroy(p, "got OPTIONS response");
/* Try again eventually */
AST_SCHED_REPLACE_UNREF(peer->pokeexpire, sched,
is_reachable ? peer->qualifyfreq : DEFAULT_FREQ_NOTOK,
sip_poke_peer_s, peer,
unref_peer(_data, "removing poke peer ref"),
unref_peer(peer, "removing poke peer ref"),
ref_peer(peer, "adding poke peer ref"));
}
/*! \brief Immediately stop RTP, VRTP and UDPTL as applicable */
static void stop_media_flows(struct sip_pvt *p)
{
/* Immediately stop RTP, VRTP and UDPTL as applicable */
if (p->rtp)
ast_rtp_instance_stop(p->rtp);
if (p->vrtp)
ast_rtp_instance_stop(p->vrtp);
if (p->trtp)
ast_rtp_instance_stop(p->trtp);
if (p->udptl)
ast_udptl_stop(p->udptl);
}
/*! \brief Handle SIP response in dialogue
\note only called by handle_incoming */
static void handle_response(struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int seqno)
{
struct ast_channel *owner;
int sipmethod;
int res = 1;
const char *c = get_header(req, "Cseq");
/* GCC 4.2 complains if I try to cast c as a char * when passing it to ast_skip_nonblanks, so make a copy of it */
char *c_copy = ast_strdupa(c);
/* Skip the Cseq and its subsequent spaces */
const char *msg = ast_skip_blanks(ast_skip_nonblanks(c_copy));
if (!msg)
msg = "";
sipmethod = find_sip_method(msg);
owner = p->owner;
if (owner)
owner->hangupcause = hangup_sip2cause(resp);
/* Acknowledge whatever it is destined for */
if ((resp >= 100) && (resp <= 199))
__sip_semi_ack(p, seqno, 0, sipmethod);
else
__sip_ack(p, seqno, 0, sipmethod);
/* If this is a NOTIFY for a subscription clear the flag that indicates that we have a NOTIFY pending */
if (!p->owner && sipmethod == SIP_NOTIFY && p->pendinginvite)
p->pendinginvite = 0;
/* Get their tag if we haven't already */
if (ast_strlen_zero(p->theirtag) || (resp >= 200)) {
char tag[128];
gettag(req, "To", tag, sizeof(tag));
ast_string_field_set(p, theirtag, tag);
}
/* This needs to be configurable on a channel/peer level,
not mandatory for all communication. Sadly enough, NAT implementations
are not so stable so we can always rely on these headers.
Temporarily disabled, while waiting for fix.
Fix assigned to Rizzo :-)
*/
/* check_via_response(p, req); */
/* RFC 3261 Section 15 specifies that if we receive a 408 or 481
* in response to a BYE, then we should end the current dialog
* and session. It is known that at least one phone manufacturer
* potentially will send a 404 in response to a BYE, so we'll be
* liberal in what we accept and end the dialog and session if we
* receive any of those responses to a BYE.
*/
if ((resp == 404 || resp == 408 || resp == 481) && sipmethod == SIP_BYE) {
pvt_set_needdestroy(p, "received 4XX response to a BYE");
return;
}
if (p->relatedpeer && p->method == SIP_OPTIONS) {
/* We don't really care what the response is, just that it replied back.
Well, as long as it's not a 100 response... since we might
need to hang around for something more "definitive" */
if (resp != 100)
handle_response_peerpoke(p, resp, req);
} else if (ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
switch(resp) {
case 100: /* 100 Trying */
case 101: /* 101 Dialog establishment */
case 183: /* 183 Session Progress */
case 180: /* 180 Ringing */
case 182: /* 182 Queued */
case 181: /* 181 Call Is Being Forwarded */
if (sipmethod == SIP_INVITE)
handle_response_invite(p, resp, rest, req, seqno);
break;
case 200: /* 200 OK */
p->authtries = 0; /* Reset authentication counter */
if (sipmethod == SIP_MESSAGE || sipmethod == SIP_INFO) {
/* We successfully transmitted a message
or a video update request in INFO */
/* Nothing happens here - the message is inside a dialog */
} else if (sipmethod == SIP_INVITE) {
handle_response_invite(p, resp, rest, req, seqno);
} else if (sipmethod == SIP_NOTIFY) {
handle_response_notify(p, resp, rest, req, seqno);
} else if (sipmethod == SIP_REGISTER) {
res = handle_response_register(p, resp, rest, req, seqno);
} else if (sipmethod == SIP_SUBSCRIBE) {
ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
handle_response_subscribe(p, resp, rest, req, seqno);
} else if (sipmethod == SIP_BYE) { /* Ok, we're ready to go */
pvt_set_needdestroy(p, "received 200 response");
ast_clear_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
}
break;
case 202: /* Transfer accepted */
if (sipmethod == SIP_REFER)
handle_response_refer(p, resp, rest, req, seqno);
break;
case 401: /* Not www-authorized on SIP method */
case 407: /* Proxy auth required */
if (sipmethod == SIP_INVITE)
handle_response_invite(p, resp, rest, req, seqno);
else if (sipmethod == SIP_NOTIFY)
handle_response_notify(p, resp, rest, req, seqno);
else if (sipmethod == SIP_REFER)
handle_response_refer(p, resp, rest, req, seqno);
else if (sipmethod == SIP_SUBSCRIBE)
handle_response_subscribe(p, resp, rest, req, seqno);
else if (p->registry && sipmethod == SIP_REGISTER)
res = handle_response_register(p, resp, rest, req, seqno);
else if (sipmethod == SIP_BYE) {
if (p->options)
p->options->auth_type = resp;
if (ast_strlen_zero(p->authname)) {
ast_log(LOG_WARNING, "Asked to authenticate %s, to %s:%d but we have no matching peer!\n",
msg, ast_inet_ntoa(p->recv.sin_addr), ntohs(p->recv.sin_port));
pvt_set_needdestroy(p, "unable to authenticate BYE");
} else if ((p->authtries == MAX_AUTHTRIES) || do_proxy_auth(p, req, resp, sipmethod, 0)) {
ast_log(LOG_NOTICE, "Failed to authenticate on %s to '%s'\n", msg, get_header(&p->initreq, "From"));
pvt_set_needdestroy(p, "failed to authenticate BYE");
}
} else {
ast_log(LOG_WARNING, "Got authentication request (%d) on %s to '%s'\n", resp, sip_methods[sipmethod].text, get_header(req, "To"));
pvt_set_needdestroy(p, "received 407 response");
}
break;
case 403: /* Forbidden - we failed authentication */
if (sipmethod == SIP_INVITE)
handle_response_invite(p, resp, rest, req, seqno);
else if (sipmethod == SIP_SUBSCRIBE)
handle_response_subscribe(p, resp, rest, req, seqno);
else if (p->registry && sipmethod == SIP_REGISTER)
res = handle_response_register(p, resp, rest, req, seqno);
else {
ast_log(LOG_WARNING, "Forbidden - maybe wrong password on authentication for %s\n", msg);
pvt_set_needdestroy(p, "received 403 response");
}
break;
case 404: /* Not found */
if (p->registry && sipmethod == SIP_REGISTER)
res = handle_response_register(p, resp, rest, req, seqno);
else if (sipmethod == SIP_INVITE)
handle_response_invite(p, resp, rest, req, seqno);
else if (sipmethod == SIP_SUBSCRIBE)
handle_response_subscribe(p, resp, rest, req, seqno);
else if (owner)
ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
break;
case 423: /* Interval too brief */
if (sipmethod == SIP_REGISTER)
res = handle_response_register(p, resp, rest, req, seqno);
break;
case 408: /* Request timeout - terminate dialog */
if (sipmethod == SIP_INVITE)
handle_response_invite(p, resp, rest, req, seqno);
else if (sipmethod == SIP_REGISTER)
res = handle_response_register(p, resp, rest, req, seqno);
else if (sipmethod == SIP_BYE) {
pvt_set_needdestroy(p, "received 408 response");
ast_debug(4, "Got timeout on bye. Thanks for the answer. Now, kill this call\n");
} else {
if (owner)
ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
pvt_set_needdestroy(p, "received 408 response");
}
break;
case 422: /* Session-Timers: Session Interval Too Small */
if (sipmethod == SIP_INVITE) {
handle_response_invite(p, resp, rest, req, seqno);
}
break;
case 481: /* Call leg does not exist */
if (sipmethod == SIP_INVITE) {
handle_response_invite(p, resp, rest, req, seqno);
} else if (sipmethod == SIP_REFER) {
handle_response_refer(p, resp, rest, req, seqno);
} else if (sipmethod == SIP_SUBSCRIBE) {
handle_response_subscribe(p, resp, rest, req, seqno);
} else if (sipmethod == SIP_BYE) {
/* The other side has no transaction to bye,
just assume it's all right then */
ast_log(LOG_WARNING, "Remote host can't match request %s to call '%s'. Giving up.\n", sip_methods[sipmethod].text, p->callid);
} else if (sipmethod == SIP_CANCEL) {
/* The other side has no transaction to cancel,
just assume it's all right then */
ast_log(LOG_WARNING, "Remote host can't match request %s to call '%s'. Giving up.\n", sip_methods[sipmethod].text, p->callid);
} else {
ast_log(LOG_WARNING, "Remote host can't match request %s to call '%s'. Giving up.\n", sip_methods[sipmethod].text, p->callid);
/* Guessing that this is not an important request */
}
break;
case 487:
if (sipmethod == SIP_INVITE)
handle_response_invite(p, resp, rest, req, seqno);
break;
case 488: /* Not acceptable here - codec error */
if (sipmethod == SIP_INVITE)
handle_response_invite(p, resp, rest, req, seqno);
break;
case 491: /* Pending */
if (sipmethod == SIP_INVITE)
handle_response_invite(p, resp, rest, req, seqno);
else {
ast_debug(1, "Got 491 on %s, unsupported. Call ID %s\n", sip_methods[sipmethod].text, p->callid);
pvt_set_needdestroy(p, "received 491 response");
}
break;
case 501: /* Not Implemented */
if (sipmethod == SIP_INVITE)
handle_response_invite(p, resp, rest, req, seqno);
else if (sipmethod == SIP_REFER)
handle_response_refer(p, resp, rest, req, seqno);
else
ast_log(LOG_WARNING, "Host '%s' does not implement '%s'\n", ast_inet_ntoa(p->sa.sin_addr), msg);
break;
case 603: /* Declined transfer */
if (sipmethod == SIP_REFER) {
handle_response_refer(p, resp, rest, req, seqno);
break;
}
/* Fallthrough */
default:
if ((resp >= 300) && (resp < 700)) {
/* Fatal response */
if ((resp != 487))
ast_verb(3, "Got SIP response %d \"%s\" back from %s\n", resp, rest, ast_inet_ntoa(p->sa.sin_addr));
if (sipmethod == SIP_INVITE)
stop_media_flows(p); /* Immediately stop RTP, VRTP and UDPTL as applicable */
/* XXX Locking issues?? XXX */
switch(resp) {
case 300: /* Multiple Choices */
case 301: /* Moved permanently */
case 302: /* Moved temporarily */
case 305: /* Use Proxy */
{
struct ast_party_redirecting redirecting = {{0,},};
change_redirecting_information(p, req, &redirecting, TRUE);
ast_channel_set_redirecting(p->owner, &redirecting);
}
/* Fall through */
case 486: /* Busy here */
case 600: /* Busy everywhere */
case 603: /* Decline */
if (p->owner)
ast_queue_control(p->owner, AST_CONTROL_BUSY);
break;
case 482: /*!
\note SIP is incapable of performing a hairpin call, which
is yet another failure of not having a layer 2 (again, YAY
IETF for thinking ahead). So we treat this as a call
forward and hope we end up at the right place... */
ast_debug(1, "Hairpin detected, setting up call forward for what it's worth\n");
if (p->owner)
ast_string_field_build(p->owner, call_forward,
"Local/%s@%s", p->username, p->context);
/* Fall through */
case 480: /* Temporarily Unavailable */
case 404: /* Not Found */
case 410: /* Gone */
case 400: /* Bad Request */
case 500: /* Server error */
if (sipmethod == SIP_REFER) {
handle_response_refer(p, resp, rest, req, seqno);
break;
} else if (sipmethod == SIP_SUBSCRIBE) {
handle_response_subscribe(p, resp, rest, req, seqno);
break;
}
/* Fall through */
case 502: /* Bad gateway */
case 503: /* Service Unavailable */
case 504: /* Server Timeout */
if (owner)
ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
break;
default:
/* Send hangup */
if (owner && sipmethod != SIP_MESSAGE && sipmethod != SIP_INFO && sipmethod != SIP_BYE)
ast_queue_hangup_with_cause(p->owner, AST_CAUSE_PROTOCOL_ERROR);
break;
}
/* ACK on invite */
if (sipmethod == SIP_INVITE)
transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
if (sipmethod != SIP_MESSAGE && sipmethod != SIP_INFO)
sip_alreadygone(p);
if (!p->owner) {
pvt_set_needdestroy(p, "transaction completed");
}
} else if ((resp >= 100) && (resp < 200)) {
if (sipmethod == SIP_INVITE) {
if (!req->ignore && sip_cancel_destroy(p))
ast_log(LOG_WARNING, "Unable to cancel SIP destruction. Expect bad things.\n");
if (find_sdp(req))
process_sdp(p, req, SDP_T38_NONE);
if (p->owner) {
/* Queue a progress frame */
ast_queue_control(p->owner, AST_CONTROL_PROGRESS);
}
}
} else
ast_log(LOG_NOTICE, "Dont know how to handle a %d %s response from %s\n", resp, rest, p->owner ? p->owner->name : ast_inet_ntoa(p->sa.sin_addr));
}
} else {
/* Responses to OUTGOING SIP requests on INCOMING calls
get handled here. As well as out-of-call message responses */
if (req->debug)
ast_verbose("SIP Response message for INCOMING dialog %s arrived\n", msg);
if (sipmethod == SIP_INVITE && resp == 200) {
/* Tags in early session is replaced by the tag in 200 OK, which is
the final reply to our INVITE */
char tag[128];
gettag(req, "To", tag, sizeof(tag));
ast_string_field_set(p, theirtag, tag);
}
switch(resp) {
case 200:
if (sipmethod == SIP_INVITE) {
handle_response_invite(p, resp, rest, req, seqno);
} else if (sipmethod == SIP_CANCEL) {
ast_debug(1, "Got 200 OK on CANCEL\n");
/* Wait for 487, then destroy */
} else if (sipmethod == SIP_NOTIFY) {
/* They got the notify, this is the end */
if (p->owner) {
if (p->refer) {
ast_debug(1, "Got 200 OK on NOTIFY for transfer\n");
} else
ast_log(LOG_WARNING, "Notify answer on an owned channel?\n");
/* ast_queue_hangup(p->owner); Disabled */
} else {
if (!p->subscribed && !p->refer) {
pvt_set_needdestroy(p, "transaction completed");
}
if (ast_test_flag(&p->flags[1], SIP_PAGE2_STATECHANGEQUEUE)) {
/* Ready to send the next state we have on queue */
ast_clear_flag(&p->flags[1], SIP_PAGE2_STATECHANGEQUEUE);
cb_extensionstate((char *)p->context, (char *)p->exten, p->laststate, (void *) p);
}
}
} else if (sipmethod == SIP_BYE) {
pvt_set_needdestroy(p, "transaction completed");
} else if (sipmethod == SIP_MESSAGE || sipmethod == SIP_INFO) {
/* We successfully transmitted a message or
a video update request in INFO */
;
}
break;
case 202: /* Transfer accepted */
if (sipmethod == SIP_REFER)
handle_response_refer(p, resp, rest, req, seqno);
break;
case 401: /* www-auth */
case 407:
if (sipmethod == SIP_REFER)
handle_response_refer(p, resp, rest, req, seqno);
else if (sipmethod == SIP_INVITE)
handle_response_invite(p, resp, rest, req, seqno);
else if (sipmethod == SIP_BYE) {
if (p->authtries == MAX_AUTHTRIES || do_proxy_auth(p, req, resp, sipmethod, 0)) {
ast_log(LOG_NOTICE, "Failed to authenticate on %s to '%s'\n", msg, get_header(&p->initreq, "From"));
pvt_set_needdestroy(p, "failed to authenticate BYE");
}
}
break;
case 481: /* Call leg does not exist */
if (sipmethod == SIP_INVITE) {
/* Re-invite failed */
handle_response_invite(p, resp, rest, req, seqno);
} else if (sipmethod == SIP_BYE) {
pvt_set_needdestroy(p, "received 481 response");
} else if (sipdebug) {
ast_debug(1, "Remote host can't match request %s to call '%s'. Giving up\n", sip_methods[sipmethod].text, p->callid);
}
break;
case 501: /* Not Implemented */
if (sipmethod == SIP_INVITE)
handle_response_invite(p, resp, rest, req, seqno);
else if (sipmethod == SIP_REFER)
handle_response_refer(p, resp, rest, req, seqno);
break;
case 603: /* Declined transfer */
if (sipmethod == SIP_REFER) {
handle_response_refer(p, resp, rest, req, seqno);
break;
}
/* Fallthrough */
default: /* Errors without handlers */
if ((resp >= 100) && (resp < 200)) {
if (sipmethod == SIP_INVITE) { /* re-invite */
if (!req->ignore && sip_cancel_destroy(p))
ast_log(LOG_WARNING, "Unable to cancel SIP destruction. Expect bad things.\n");
}
}
if ((resp >= 300) && (resp < 700)) {
if ((resp != 487))
ast_verb(3, "Incoming call: Got SIP response %d \"%s\" back from %s\n", resp, rest, ast_inet_ntoa(p->sa.sin_addr));
switch(resp) {
case 488: /* Not acceptable here - codec error */
case 603: /* Decline */
case 500: /* Server error */
case 502: /* Bad gateway */
case 503: /* Service Unavailable */
case 504: /* Server timeout */
/* re-invite failed */
if (sipmethod == SIP_INVITE && sip_cancel_destroy(p))
ast_log(LOG_WARNING, "Unable to cancel SIP destruction. Expect bad things.\n");
break;
}
}
break;
}
}
}
/*! \brief Park SIP call support function
Starts in a new thread, then parks the call
XXX Should we add a wait period after streaming audio and before hangup?? Sometimes the
audio can't be heard before hangup
*/
static void *sip_park_thread(void *stuff)
{
struct ast_channel *transferee, *transferer; /* Chan1: The transferee, Chan2: The transferer */
struct sip_dual *d;
struct sip_request req = {0,};
int ext;
int res;
d = stuff;
transferee = d->chan1;
transferer = d->chan2;
copy_request(&req, &d->req);
if (!transferee || !transferer) {
ast_log(LOG_ERROR, "Missing channels for parking! Transferer %s Transferee %s\n", transferer ? "<available>" : "<missing>", transferee ? "<available>" : "<missing>" );
if (d->req.data)
ast_free(d->req.data);
free(d);
return NULL;
}
ast_debug(4, "SIP Park: Transferer channel %s, Transferee %s\n", transferer->name, transferee->name);
ast_channel_lock(transferee);
if (ast_do_masquerade(transferee)) {
ast_log(LOG_WARNING, "Masquerade failed.\n");
transmit_response(transferer->tech_pvt, "503 Internal error", &req);
ast_channel_unlock(transferee);
if (d->req.data)
ast_free(d->req.data);
free(d);
return NULL;
}
ast_channel_unlock(transferee);
res = ast_park_call(transferee, transferer, 0, &ext);
#ifdef WHEN_WE_KNOW_THAT_THE_CLIENT_SUPPORTS_MESSAGE
if (!res) {
transmit_message_with_text(transferer->tech_pvt, "Unable to park call.\n");
} else {
/* Then tell the transferer what happened */
sprintf(buf, "Call parked on extension '%d'", ext);
transmit_message_with_text(transferer->tech_pvt, buf);
}
#endif
/* Any way back to the current call??? */
/* Transmit response to the REFER request */
transmit_response(transferer->tech_pvt, "202 Accepted", &req);
if (!res) {
/* Transfer succeeded */
append_history(transferer->tech_pvt, "SIPpark", "Parked call on %d", ext);
transmit_notify_with_sipfrag(transferer->tech_pvt, d->seqno, "200 OK", TRUE);
transferer->hangupcause = AST_CAUSE_NORMAL_CLEARING;
ast_hangup(transferer); /* This will cause a BYE */
ast_debug(1, "SIP Call parked on extension '%d'\n", ext);
} else {
transmit_notify_with_sipfrag(transferer->tech_pvt, d->seqno, "503 Service Unavailable", TRUE);
append_history(transferer->tech_pvt, "SIPpark", "Parking failed\n");
ast_debug(1, "SIP Call parked failed \n");
/* Do not hangup call */
}
if (d->req.data)
ast_free(d->req.data);
free(d);
return NULL;
}
/*! \brief Park a call using the subsystem in res_features.c
This is executed in a separate thread
*/
static int sip_park(struct ast_channel *chan1, struct ast_channel *chan2, struct sip_request *req, int seqno)
{
struct sip_dual *d;
struct ast_channel *transferee, *transferer;
/* Chan2m: The transferer, chan1m: The transferee */
pthread_t th;
transferee = ast_channel_alloc(0, AST_STATE_DOWN, 0, 0, chan1->accountcode, chan1->exten, chan1->context, chan1->amaflags, "Parking/%s", chan1->name);
transferer = ast_channel_alloc(0, AST_STATE_DOWN, 0, 0, chan2->accountcode, chan2->exten, chan2->context, chan2->amaflags, "SIPPeer/%s", chan2->name);
if ((!transferer) || (!transferee)) {
if (transferee) {
transferee->hangupcause = AST_CAUSE_SWITCH_CONGESTION;
ast_hangup(transferee);
}
if (transferer) {
transferer->hangupcause = AST_CAUSE_SWITCH_CONGESTION;
ast_hangup(transferer);
}
return -1;
}
/* Make formats okay */
transferee->readformat = chan1->readformat;
transferee->writeformat = chan1->writeformat;
/* Prepare for taking over the channel */
ast_channel_masquerade(transferee, chan1);
/* Setup the extensions and such */
ast_copy_string(transferee->context, chan1->context, sizeof(transferee->context));
ast_copy_string(transferee->exten, chan1->exten, sizeof(transferee->exten));
transferee->priority = chan1->priority;
/* We make a clone of the peer channel too, so we can play
back the announcement */
/* Make formats okay */
transferer->readformat = chan2->readformat;
transferer->writeformat = chan2->writeformat;
/* Prepare for taking over the channel. Go ahead and grab this channel
* lock here to avoid a deadlock with callbacks into the channel driver
* that hold the channel lock and want the pvt lock. */
while (ast_channel_trylock(chan2)) {
struct sip_pvt *pvt = chan2->tech_pvt;
sip_pvt_unlock(pvt);
usleep(1);
sip_pvt_lock(pvt);
}
ast_channel_masquerade(transferer, chan2);
ast_channel_unlock(chan2);
/* Setup the extensions and such */
ast_copy_string(transferer->context, chan2->context, sizeof(transferer->context));
ast_copy_string(transferer->exten, chan2->exten, sizeof(transferer->exten));
transferer->priority = chan2->priority;
ast_channel_lock(transferer);
if (ast_do_masquerade(transferer)) {
ast_log(LOG_WARNING, "Masquerade failed :(\n");
ast_channel_unlock(transferer);
transferer->hangupcause = AST_CAUSE_SWITCH_CONGESTION;
ast_hangup(transferer);
return -1;
}
ast_channel_unlock(transferer);
if (!transferer || !transferee) {
if (!transferer) {
ast_debug(1, "No transferer channel, giving up parking\n");
}
if (!transferee) {
ast_debug(1, "No transferee channel, giving up parking\n");
}
return -1;
}
if ((d = ast_calloc(1, sizeof(*d)))) {
/* Save original request for followup */
copy_request(&d->req, req);
d->chan1 = transferee; /* Transferee */
d->chan2 = transferer; /* Transferer */
d->seqno = seqno;
if (ast_pthread_create_detached_background(&th, NULL, sip_park_thread, d) < 0) {
/* Could not start thread */
if (d->req.data)
ast_free(d->req.data);
ast_free(d); /* We don't need it anymore. If thread is created, d will be free'd
by sip_park_thread() */
return 0;
}
}
return -1;
}
/*! \brief Turn off generator data
XXX Does this function belong in the SIP channel?
*/
static void ast_quiet_chan(struct ast_channel *chan)
{
if (chan && chan->_state == AST_STATE_UP) {
if (ast_test_flag(chan, AST_FLAG_MOH))
ast_moh_stop(chan);
else if (chan->generatordata)
ast_deactivate_generator(chan);
}
}
/*! \brief Attempt transfer of SIP call
This fix for attended transfers on a local PBX */
static int attempt_transfer(struct sip_dual *transferer, struct sip_dual *target)
{
int res = 0;
struct ast_channel *peera = NULL,
*peerb = NULL,
*peerc = NULL,
*peerd = NULL;
/* We will try to connect the transferee with the target and hangup
all channels to the transferer */
ast_debug(4, "Sip transfer:--------------------\n");
if (transferer->chan1)
ast_debug(4, "-- Transferer to PBX channel: %s State %s\n", transferer->chan1->name, ast_state2str(transferer->chan1->_state));
else
ast_debug(4, "-- No transferer first channel - odd??? \n");
if (target->chan1)
ast_debug(4, "-- Transferer to PBX second channel (target): %s State %s\n", target->chan1->name, ast_state2str(target->chan1->_state));
else
ast_debug(4, "-- No target first channel ---\n");
if (transferer->chan2)
ast_debug(4, "-- Bridged call to transferee: %s State %s\n", transferer->chan2->name, ast_state2str(transferer->chan2->_state));
else
ast_debug(4, "-- No bridged call to transferee\n");
if (target->chan2)
ast_debug(4, "-- Bridged call to transfer target: %s State %s\n", target->chan2 ? target->chan2->name : "<none>", target->chan2 ? ast_state2str(target->chan2->_state) : "(none)");
else
ast_debug(4, "-- No target second channel ---\n");
ast_debug(4, "-- END Sip transfer:--------------------\n");
if (transferer->chan2) { /* We have a bridge on the transferer's channel */
peera = transferer->chan1; /* Transferer - PBX -> transferee channel * the one we hangup */
peerb = target->chan1; /* Transferer - PBX -> target channel - This will get lost in masq */
peerc = transferer->chan2; /* Asterisk to Transferee */
peerd = target->chan2; /* Asterisk to Target */
ast_debug(3, "SIP transfer: Four channels to handle\n");
} else if (target->chan2) { /* Transferer has no bridge (IVR), but transferee */
peera = target->chan1; /* Transferer to PBX -> target channel */
peerb = transferer->chan1; /* Transferer to IVR*/
peerc = target->chan2; /* Asterisk to Target */
peerd = transferer->chan2; /* Nothing */
ast_debug(3, "SIP transfer: Three channels to handle\n");
}
if (peera && peerb && peerc && (peerb != peerc)) {
ast_quiet_chan(peera); /* Stop generators */
ast_quiet_chan(peerb);
ast_quiet_chan(peerc);
if (peerd)
ast_quiet_chan(peerd);
ast_debug(4, "SIP transfer: trying to masquerade %s into %s\n", peerc->name, peerb->name);
if (ast_channel_masquerade(peerb, peerc)) {
ast_log(LOG_WARNING, "Failed to masquerade %s into %s\n", peerb->name, peerc->name);
res = -1;
} else
ast_debug(4, "SIP transfer: Succeeded to masquerade channels.\n");
return res;
} else {
ast_log(LOG_NOTICE, "SIP Transfer attempted with no appropriate bridged calls to transfer\n");
if (transferer->chan1)
ast_softhangup_nolock(transferer->chan1, AST_SOFTHANGUP_DEV);
if (target->chan1)
ast_softhangup_nolock(target->chan1, AST_SOFTHANGUP_DEV);
return -1;
}
return 0;
}
/*! \brief Get tag from packet
*
* \return Returns the pointer to the provided tag buffer,
* or NULL if the tag was not found.
*/
static const char *gettag(const struct sip_request *req, const char *header, char *tagbuf, int tagbufsize)
{
const char *thetag;
if (!tagbuf)
return NULL;
tagbuf[0] = '\0'; /* reset the buffer */
thetag = get_header(req, header);
thetag = strcasestr(thetag, ";tag=");
if (thetag) {
thetag += 5;
ast_copy_string(tagbuf, thetag, tagbufsize);
return strsep(&tagbuf, ";");
}
return NULL;
}
/*! \brief Handle incoming notifications */
static int handle_request_notify(struct sip_pvt *p, struct sip_request *req, struct sockaddr_in *sin, int seqno, char *e)
{
/* This is mostly a skeleton for future improvements */
/* Mostly created to return proper answers on notifications on outbound REFER's */
int res = 0;
const char *event = get_header(req, "Event");
char *eventid = NULL;
char *sep;
if( (sep = strchr(event, ';')) ) { /* XXX bug here - overwriting string ? */
*sep++ = '\0';
eventid = sep;
}
if (sipdebug)
ast_debug(2, "Got NOTIFY Event: %s\n", event);
if (!strcmp(event, "refer")) {
/* Save nesting depth for now, since there might be other events we will
support in the future */
/* Handle REFER notifications */
char buf[1024];
char *cmd, *code;
int respcode;
int success = TRUE;
/* EventID for each transfer... EventID is basically the REFER cseq
We are getting notifications on a call that we transfered
We should hangup when we are getting a 200 OK in a sipfrag
Check if we have an owner of this event */
/* Check the content type */
if (strncasecmp(get_header(req, "Content-Type"), "message/sipfrag", strlen("message/sipfrag"))) {
/* We need a sipfrag */
transmit_response(p, "400 Bad request", req);
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
return -1;
}
/* Get the text of the attachment */
if (get_msg_text(buf, sizeof(buf), req, TRUE)) {
ast_log(LOG_WARNING, "Unable to retrieve attachment from NOTIFY %s\n", p->callid);
transmit_response(p, "400 Bad request", req);
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
return -1;
}
/*
From the RFC...
A minimal, but complete, implementation can respond with a single
NOTIFY containing either the body:
SIP/2.0 100 Trying
if the subscription is pending, the body:
SIP/2.0 200 OK
if the reference was successful, the body:
SIP/2.0 503 Service Unavailable
if the reference failed, or the body:
SIP/2.0 603 Declined
if the REFER request was accepted before approval to follow the
reference could be obtained and that approval was subsequently denied
(see Section 2.4.7).
If there are several REFERs in the same dialog, we need to
match the ID of the event header...
*/
ast_debug(3, "* SIP Transfer NOTIFY Attachment: \n---%s\n---\n", buf);
cmd = ast_skip_blanks(buf);
code = cmd;
/* We are at SIP/2.0 */
while(*code && (*code > 32)) { /* Search white space */
code++;
}
*code++ = '\0';
code = ast_skip_blanks(code);
sep = code;
sep++;
while(*sep && (*sep > 32)) { /* Search white space */
sep++;
}
*sep++ = '\0'; /* Response string */
respcode = atoi(code);
switch (respcode) {
case 100: /* Trying: */
case 101: /* dialog establishment */
/* Don't do anything yet */
break;
case 183: /* Ringing: */
/* Don't do anything yet */
break;
case 200: /* OK: The new call is up, hangup this call */
/* Hangup the call that we are replacing */
break;
case 301: /* Moved permenantly */
case 302: /* Moved temporarily */
/* Do we get the header in the packet in this case? */
success = FALSE;
break;
case 503: /* Service Unavailable: The new call failed */
/* Cancel transfer, continue the call */
success = FALSE;
break;
case 603: /* Declined: Not accepted */
/* Cancel transfer, continue the current call */
success = FALSE;
break;
}
if (!success) {
ast_log(LOG_NOTICE, "Transfer failed. Sorry. Nothing further to do with this call\n");
}
if (p->owner) {
enum ast_control_transfer message = success ? AST_TRANSFER_SUCCESS : AST_TRANSFER_FAILED;
ast_queue_control_data(p->owner, AST_CONTROL_TRANSFER, &message, sizeof(message));
}
/* Confirm that we received this packet */
transmit_response(p, "200 OK", req);
} else if (p->mwi && !strcmp(event, "message-summary")) {
char *c = ast_strdupa(get_body(req, "Voice-Message", ':'));
if (!ast_strlen_zero(c)) {
char *old = strsep(&c, " ");
char *new = strsep(&old, "/");
struct ast_event *event;
if ((event = ast_event_new(AST_EVENT_MWI,
AST_EVENT_IE_MAILBOX, AST_EVENT_IE_PLTYPE_STR, p->mwi->mailbox,
AST_EVENT_IE_CONTEXT, AST_EVENT_IE_PLTYPE_STR, "SIP_Remote",
AST_EVENT_IE_NEWMSGS, AST_EVENT_IE_PLTYPE_UINT, atoi(new),
AST_EVENT_IE_OLDMSGS, AST_EVENT_IE_PLTYPE_UINT, atoi(old),
AST_EVENT_IE_END))) {
ast_event_queue_and_cache(event);
}
}
transmit_response(p, "200 OK", req);
} else {
/* We don't understand this event. */
transmit_response(p, "489 Bad event", req);
res = -1;
}
if (!p->lastinvite)
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
return res;
}
/*! \brief Handle incoming OPTIONS request
An OPTIONS request should be answered like an INVITE from the same UA, including SDP
*/
static int handle_request_options(struct sip_pvt *p, struct sip_request *req)
{
int res;
/*! XXX get_destination assumes we're already authenticated. This means that a request from
a known device (peer) will end up in the wrong context if this is out-of-dialog.
However, we want to handle OPTIONS as light as possible, so we might want to have
a configuration option whether we care or not. Some devices use this for testing
capabilities, which means that we need to match device to answer with proper
capabilities (including SDP).
\todo Fix handle_request_options device handling with optional authentication
(this needs to be fixed in 1.4 as well)
*/
if (p->lastinvite) {
/* if this is a request in an active dialog, just confirm that the dialog exists. */
transmit_response_with_allow(p, "200 OK", req, 0);
return 0;
}
res = get_destination(p, req);
build_contact(p);
if (ast_strlen_zero(p->context))
ast_string_field_set(p, context, sip_cfg.default_context);
if (ast_shutting_down())
transmit_response_with_allow(p, "503 Unavailable", req, 0);
else if (res < 0)
transmit_response_with_allow(p, "404 Not Found", req, 0);
else
transmit_response_with_allow(p, "200 OK", req, 0);
/* Destroy if this OPTIONS was the opening request, but not if
it's in the middle of a normal call flow. */
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
return res;
}
/*! \brief Handle the transfer part of INVITE with a replaces: header,
meaning a target pickup or an attended transfer.
Used only once.
XXX 'ignore' is unused.
*/
static int handle_invite_replaces(struct sip_pvt *p, struct sip_request *req, int debug, int seqno, struct sockaddr_in *sin)
{
struct ast_frame *f;
int earlyreplace = 0;
int oneleggedreplace = 0; /* Call with no bridge, propably IVR or voice message */
struct ast_channel *c = p->owner; /* Our incoming call */
struct ast_channel *replacecall = p->refer->refer_call->owner; /* The channel we're about to take over */
struct ast_channel *targetcall; /* The bridge to the take-over target */
struct ast_channel *test;
/* Check if we're in ring state */
if (replacecall->_state == AST_STATE_RING)
earlyreplace = 1;
/* Check if we have a bridge */
if (!(targetcall = ast_bridged_channel(replacecall))) {
/* We have no bridge */
if (!earlyreplace) {
ast_debug(2, " Attended transfer attempted to replace call with no bridge (maybe ringing). Channel %s!\n", replacecall->name);
oneleggedreplace = 1;
}
}
if (targetcall && targetcall->_state == AST_STATE_RINGING)
ast_debug(4, "SIP transfer: Target channel is in ringing state\n");
if (targetcall)
ast_debug(4, "SIP transfer: Invite Replace incoming channel should bridge to channel %s while hanging up channel %s\n", targetcall->name, replacecall->name);
else
ast_debug(4, "SIP transfer: Invite Replace incoming channel should replace and hang up channel %s (one call leg)\n", replacecall->name);
if (req->ignore) {
ast_log(LOG_NOTICE, "Ignoring this INVITE with replaces in a stupid way.\n");
/* We should answer something here. If we are here, the
call we are replacing exists, so an accepted
can't harm */
transmit_response_with_sdp(p, "200 OK", req, XMIT_RELIABLE, FALSE, FALSE);
/* Do something more clever here */
ast_channel_unlock(c);
sip_pvt_unlock(p->refer->refer_call);
return 1;
}
if (!c) {
/* What to do if no channel ??? */
ast_log(LOG_ERROR, "Unable to create new channel. Invite/replace failed.\n");
transmit_response_reliable(p, "503 Service Unavailable", req);
append_history(p, "Xfer", "INVITE/Replace Failed. No new channel.");
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
sip_pvt_unlock(p->refer->refer_call);
return 1;
}
append_history(p, "Xfer", "INVITE/Replace received");
/* We have three channels to play with
channel c: New incoming call
targetcall: Call from PBX to target
p->refer->refer_call: SIP pvt dialog from transferer to pbx.
replacecall: The owner of the previous
We need to masq C into refer_call to connect to
targetcall;
If we are talking to internal audio stream, target call is null.
*/
/* Fake call progress */
transmit_response(p, "100 Trying", req);
ast_setstate(c, AST_STATE_RING);
/* Masquerade the new call into the referred call to connect to target call
Targetcall is not touched by the masq */
/* Answer the incoming call and set channel to UP state */
transmit_response_with_sdp(p, "200 OK", req, XMIT_RELIABLE, FALSE, FALSE);
ast_setstate(c, AST_STATE_UP);
/* Stop music on hold and other generators */
ast_quiet_chan(replacecall);
ast_quiet_chan(targetcall);
ast_debug(4, "Invite/Replaces: preparing to masquerade %s into %s\n", c->name, replacecall->name);
/* Unlock clone, but not original (replacecall) */
if (!oneleggedreplace)
ast_channel_unlock(c);
/* Unlock PVT */
sip_pvt_unlock(p->refer->refer_call);
/* Make sure that the masq does not free our PVT for the old call */
if (! earlyreplace && ! oneleggedreplace )
ast_set_flag(&p->refer->refer_call->flags[0], SIP_DEFER_BYE_ON_TRANSFER); /* Delay hangup */
/* Prepare the masquerade - if this does not happen, we will be gone */
if(ast_channel_masquerade(replacecall, c))
ast_log(LOG_ERROR, "Failed to masquerade C into Replacecall\n");
else
ast_debug(4, "Invite/Replaces: Going to masquerade %s into %s\n", c->name, replacecall->name);
/* The masquerade will happen as soon as someone reads a frame from the channel */
/* C should now be in place of replacecall */
/* ast_read needs to lock channel */
ast_channel_unlock(c);
if (earlyreplace || oneleggedreplace ) {
/* Force the masq to happen */
if ((f = ast_read(replacecall))) { /* Force the masq to happen */
ast_frfree(f);
f = NULL;
ast_debug(4, "Invite/Replace: Could successfully read frame from RING channel!\n");
} else {
ast_log(LOG_WARNING, "Invite/Replace: Could not read frame from RING channel \n");
}
c->hangupcause = AST_CAUSE_SWITCH_CONGESTION;
if (!oneleggedreplace)
ast_channel_unlock(replacecall);
} else { /* Bridged call, UP channel */
if ((f = ast_read(replacecall))) { /* Force the masq to happen */
/* Masq ok */
ast_frfree(f);
f = NULL;
ast_debug(3, "Invite/Replace: Could successfully read frame from channel! Masq done.\n");
} else {
ast_log(LOG_WARNING, "Invite/Replace: Could not read frame from channel. Transfer failed\n");
}
ast_channel_unlock(replacecall);
}
sip_pvt_unlock(p->refer->refer_call);
ast_setstate(c, AST_STATE_DOWN);
ast_debug(4, "After transfer:----------------------------\n");
ast_debug(4, " -- C: %s State %s\n", c->name, ast_state2str(c->_state));
if (replacecall)
ast_debug(4, " -- replacecall: %s State %s\n", replacecall->name, ast_state2str(replacecall->_state));
if (p->owner) {
ast_debug(4, " -- P->owner: %s State %s\n", p->owner->name, ast_state2str(p->owner->_state));
test = ast_bridged_channel(p->owner);
if (test)
ast_debug(4, " -- Call bridged to P->owner: %s State %s\n", test->name, ast_state2str(test->_state));
else
ast_debug(4, " -- No call bridged to C->owner \n");
} else
ast_debug(4, " -- No channel yet \n");
ast_debug(4, "End After transfer:----------------------------\n");
ast_channel_unlock(p->owner); /* Unlock new owner */
if (!oneleggedreplace)
sip_pvt_unlock(p); /* Unlock SIP structure */
/* The call should be down with no ast_channel, so hang it up */
c->tech_pvt = dialog_unref(c->tech_pvt, "unref dialog c->tech_pvt");
ast_hangup(c);
return 0;
}
/*! \brief helper routine for sip_uri_cmp
*
* This takes the parameters from two SIP URIs and determines
* if the URIs match. The rules for parameters *suck*. Here's a breakdown
* 1. If a parameter appears in both URIs, then they must have the same value
* in order for the URIs to match
* 2. If one URI has a user, maddr, ttl, or method parameter, then the other
* URI must also have that parameter and must have the same value
* in order for the URIs to match
* 3. All other headers appearing in only one URI are not considered when
* determining if URIs match
*
* \param input1 Parameters from URI 1
* \param input2 Parameters from URI 2
* \return Return 0 if the URIs' parameters match, 1 if they do not
*/
static int sip_uri_params_cmp(const char *input1, const char *input2)
{
char *params1 = NULL;
char *params2 = NULL;
char *pos1;
char *pos2;
int zerolength1 = 0;
int zerolength2 = 0;
int maddrmatch = 0;
int ttlmatch = 0;
int usermatch = 0;
int methodmatch = 0;
if (ast_strlen_zero(input1)) {
zerolength1 = 1;
} else {
params1 = ast_strdupa(input1);
}
if (ast_strlen_zero(input2)) {
zerolength2 = 1;
} else {
params2 = ast_strdupa(input2);
}
/*Quick optimization. If both params are zero-length, then
* they match
*/
if (zerolength1 && zerolength2) {
return 0;
}
pos1 = params1;
while (!ast_strlen_zero(pos1)) {
char *name1 = pos1;
char *value1 = strchr(pos1, '=');
char *semicolon1 = strchr(pos1, ';');
int matched = 0;
if (semicolon1) {
*semicolon1++ = '\0';
}
if (!value1) {
goto fail;
}
*value1++ = '\0';
/* Checkpoint reached. We have the name and value parsed for param1
* We have to duplicate params2 each time through the second loop
* or else we can't search and replace the semicolons with \0 each
* time
*/
pos2 = ast_strdupa(params2);
while (!ast_strlen_zero(pos2)) {
char *name2 = pos2;
char *value2 = strchr(pos2, '=');
char *semicolon2 = strchr(pos2, ';');
if (semicolon2) {
*semicolon2++ = '\0';
}
if (!value2) {
goto fail;
}
*value2++ = '\0';
if (!strcasecmp(name1, name2)) {
if (strcasecmp(value1, value2)) {
goto fail;
} else {
matched = 1;
break;
}
}
pos2 = semicolon2;
}
/* Need to see if the parameter we're looking at is one of the 'must-match' parameters */
if (!strcasecmp(name1, "maddr")) {
if (matched) {
maddrmatch = 1;
} else {
goto fail;
}
} else if (!strcasecmp(name1, "ttl")) {
if (matched) {
ttlmatch = 1;
} else {
goto fail;
}
} else if (!strcasecmp(name1, "user")) {
if (matched) {
usermatch = 1;
} else {
goto fail;
}
} else if (!strcasecmp(name1, "method")) {
if (matched) {
methodmatch = 1;
} else {
goto fail;
}
}
pos1 = semicolon1;
}
/* We've made it out of that horrible O(m*n) construct and there are no
* failures yet. We're not done yet, though, because params2 could have
* an maddr, ttl, user, or method header and params1 did not.
*/
pos2 = params2;
while (!ast_strlen_zero(pos2)) {
char *name2 = pos2;
char *value2 = strchr(pos2, '=');
char *semicolon2 = strchr(pos2, ';');
if (semicolon2) {
*semicolon2++ = '\0';
}
if (!value2) {
goto fail;
}
*value2++ = '\0';
if ((!strcasecmp(name2, "maddr") && !maddrmatch) ||
(!strcasecmp(name2, "ttl") && !ttlmatch) ||
(!strcasecmp(name2, "user") && !usermatch) ||
(!strcasecmp(name2, "method") && !methodmatch)) {
goto fail;
}
}
return 0;
fail:
return 1;
}
/*! \brief helper routine for sip_uri_cmp
*
* This takes the "headers" from two SIP URIs and determines
* if the URIs match. The rules for headers is simple. If a header
* appears in one URI, then it must also appear in the other URI. The
* order in which the headers appear does not matter.
*
* \param input1 Headers from URI 1
* \param input2 Headers from URI 2
* \return Return 0 if the URIs' headers match, 1 if they do not
*/
static int sip_uri_headers_cmp(const char *input1, const char *input2)
{
char *headers1 = NULL;
char *headers2 = NULL;
int zerolength1 = 0;
int zerolength2 = 0;
int different = 0;
char *header1;
if (ast_strlen_zero(input1)) {
zerolength1 = 1;
} else {
headers1 = ast_strdupa(input1);
}
if (ast_strlen_zero(input2)) {
zerolength2 = 1;
} else {
headers2 = ast_strdupa(input2);
}
if ((zerolength1 && !zerolength2) ||
(zerolength2 && !zerolength1))
return 1;
if (zerolength1 && zerolength2)
return 0;
/* At this point, we can definitively state that both inputs are
* not zero-length. First, one more optimization. If the length
* of the headers is not equal, then we definitely have no match
*/
if (strlen(headers1) != strlen(headers2)) {
return 1;
}
for (header1 = strsep(&headers1, "&"); header1; header1 = strsep(&headers1, "&")) {
if (!strcasestr(headers2, header1)) {
different = 1;
break;
}
}
return different;
}
static int sip_uri_cmp(const char *input1, const char *input2)
{
char *uri1 = ast_strdupa(input1);
char *uri2 = ast_strdupa(input2);
char *host1;
char *host2;
char *params1;
char *params2;
char *headers1;
char *headers2;
/* Strip off "sip:" from the URI. We know this is present
* because it was checked back in parse_request()
*/
strsep(&uri1, ":");
strsep(&uri2, ":");
if ((host1 = strchr(uri1, '@'))) {
*host1++ = '\0';
}
if ((host2 = strchr(uri2, '@'))) {
*host2++ = '\0';
}
/* Check for mismatched username and passwords. This is the
* only case-sensitive comparison of a SIP URI
*/
if ((host1 && !host2) ||
(host2 && !host1) ||
(host1 && host2 && strcmp(uri1, uri2))) {
return 1;
}
if (!host1)
host1 = uri1;
if (!host2)
host2 = uri2;
/* Strip off the parameters and headers so we can compare
* host and port
*/
if ((params1 = strchr(host1, ';'))) {
*params1++ = '\0';
}
if ((params2 = strchr(host2, ';'))) {
*params2++ = '\0';
}
/* Headers come after parameters, but there may be headers without
* parameters, thus the S_OR
*/
if ((headers1 = strchr(S_OR(params1, host1), '?'))) {
*headers1++ = '\0';
}
if ((headers2 = strchr(S_OR(params2, host2), '?'))) {
*headers2++ = '\0';
}
/* Now the host/port are properly isolated. We can get by with a string comparison
* because the SIP URI checking rules have some interesting exceptions that make
* this possible. I will note 2 in particular
* 1. hostnames which resolve to the same IP address as well as a hostname and its
* IP address are not considered a match with SIP URI's.
* 2. If one URI specifies a port and the other does not, then the URIs do not match.
* This includes if one URI explicitly contains port 5060 and the other implies it
* by not having a port specified.
*/
if (strcasecmp(host1, host2)) {
return 1;
}
/* Headers have easier rules to follow, so do those first */
if (sip_uri_headers_cmp(headers1, headers2)) {
return 1;
}
/* And now the parameters. Ugh */
return sip_uri_params_cmp(params1, params2);
}
static int do_magic_pickup(struct ast_channel *channel, const char *extension, const char *context)
{
struct ast_str *str = ast_str_alloca(AST_MAX_EXTENSION + AST_MAX_CONTEXT + 2);
struct ast_app *pickup = pbx_findapp("Pickup");
if (!pickup) {
ast_log(LOG_ERROR, "Unable to perform pickup: Application 'Pickup' not loaded (app_directed_pickup.so).\n");
return -1;
}
ast_str_set(&str, 0, "%s@%s", extension, sip_cfg.notifycid == IGNORE_CONTEXT ? "PICKUPMARK" : context);
ast_debug(2, "About to call Pickup(%s)\n", str->str);
/* There is no point in capturing the return value since pickup_exec
doesn't return anything meaningful unless the passed data is an empty
string (which in our case it will not be) */
pbx_exec(channel, pickup, str->str);
return 0;
}
static int sip_t38_abort(const void *data)
{
struct sip_pvt *p = (struct sip_pvt *) data;
change_t38_state(p, T38_DISABLED);
transmit_response_reliable(p, "488 Not acceptable here", &p->initreq);
p->t38id = -1;
dialog_unref(p, "unref the dialog ptr from sip_t38_abort, because it held a dialog ptr");
return 0;
}
/*!
* \brief Handle incoming INVITE request
* \note If the INVITE has a Replaces header, it is part of an
* attended transfer. If so, we do not go through the dial
* plan but try to find the active call and masquerade
* into it
*/
static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int debug, int seqno, struct sockaddr_in *sin, int *recount, char *e, int *nounlock)
{
int res = 1;
int gotdest;
const char *p_replaces;
char *replace_id = NULL;
const char *required;
unsigned int required_profile = 0;
struct ast_channel *c = NULL; /* New channel */
int reinvite = 0;
int rtn;
const char *p_uac_se_hdr; /* UAC's Session-Expires header string */
const char *p_uac_min_se; /* UAC's requested Min-SE interval (char string) */
int uac_max_se = -1; /* UAC's Session-Expires in integer format */
int uac_min_se = -1; /* UAC's Min-SE in integer format */
int st_active = FALSE; /* Session-Timer on/off boolean */
int st_interval = 0; /* Session-Timer negotiated refresh interval */
enum st_refresher st_ref; /* Session-Timer session refresher */
int dlg_min_se = -1;
struct {
char exten[AST_MAX_EXTENSION];
char context[AST_MAX_CONTEXT];
} pickup = {
.exten = "",
};
st_ref = SESSION_TIMER_REFRESHER_AUTO;
/* Find out what they support */
if (!p->sipoptions) {
const char *supported = get_header(req, "Supported");
if (!ast_strlen_zero(supported))
parse_sip_options(p, supported);
}
/* Find out what they require */
required = get_header(req, "Require");
if (!ast_strlen_zero(required)) {
required_profile = parse_sip_options(NULL, required);
if (required_profile && required_profile != SIP_OPT_REPLACES && required_profile != SIP_OPT_TIMER) {
/* At this point we only support REPLACES and Session-Timer */
transmit_response_with_unsupported(p, "420 Bad extension (unsupported)", req, required);
ast_log(LOG_WARNING, "Received SIP INVITE with unsupported required extension: %s\n", required);
p->invitestate = INV_COMPLETED;
if (!p->lastinvite)
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
return -1;
}
}
/* The option tags may be present in Supported: or Require: headers.
Include the Require: option tags for further processing as well */
p->sipoptions |= required_profile;
p->reqsipoptions = required_profile;
/* Check if this is a loop */
if (ast_test_flag(&p->flags[0], SIP_OUTGOING) && p->owner && (p->owner->_state != AST_STATE_UP)) {
/* This is a call to ourself. Send ourselves an error code and stop
processing immediately, as SIP really has no good mechanism for
being able to call yourself */
/* If pedantic is on, we need to check the tags. If they're different, this is
in fact a forked call through a SIP proxy somewhere. */
int different;
char *initial_rlPart2 = REQ_OFFSET_TO_STR(&p->initreq, rlPart2);
char *this_rlPart2 = REQ_OFFSET_TO_STR(req, rlPart2);
if (sip_cfg.pedanticsipchecking)
different = sip_uri_cmp(initial_rlPart2, this_rlPart2);
else
different = strcmp(initial_rlPart2, this_rlPart2);
if (!different) {
transmit_response(p, "482 Loop Detected", req);
p->invitestate = INV_COMPLETED;
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
return 0;
} else {
/*! This is a spiral. What we need to do is to just change the outgoing INVITE
* so that it now routes to the new Request URI. Since we created the INVITE ourselves
* that should be all we need to do.
*
* \todo XXX This needs to be reviewed. YOu don't change the request URI really, you route the packet
* correctly instead...
*/
char *uri = ast_strdupa(this_rlPart2);
char *at = strchr(uri, '@');
char *peerorhost;
struct sip_pkt *pkt = NULL;
if (option_debug > 2) {
ast_log(LOG_DEBUG, "Potential spiral detected. Original RURI was %s, new RURI is %s\n", initial_rlPart2, this_rlPart2);
}
if (at) {
*at = '\0';
}
/* Parse out "sip:" */
if ((peerorhost = strchr(uri, ':'))) {
*peerorhost++ = '\0';
}
create_addr(p, peerorhost, NULL, 0);
ast_string_field_set(p, theirtag, NULL);
for (pkt = p->packets; pkt; pkt = pkt->next) {
if (pkt->seqno == p->icseq && pkt->method == SIP_INVITE) {
AST_SCHED_DEL(sched, pkt->retransid);
}
}
return transmit_invite(p, SIP_INVITE, 1, 3);
}
}
if (!req->ignore && p->pendinginvite) {
if (!ast_test_flag(&p->flags[0], SIP_OUTGOING) && ast_test_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED)) {
/* We have received a reINVITE on an incoming call to which we have sent a 200 OK but not yet received
* an ACK. According to RFC 5407, Section 3.1.4, the proper way to handle this race condition is to accept
* the reINVITE since we have established a dialog.
*/
/* Note that this will both clear the pendinginvite flag and cancel the
* retransmission of the 200 OK. Basically, we're accepting this reINVITE as both an ACK
* and a reINVITE in one request.
*/
__sip_ack(p, p->lastinvite, 1, 0);
} else {
/* We already have a pending invite. Sorry. You are on hold. */
p->glareinvite = seqno;
transmit_response_reliable(p, "491 Request Pending", req);
ast_debug(1, "Got INVITE on call where we already have pending INVITE, deferring that - %s\n", p->callid);
/* Don't destroy dialog here */
return 0;
}
}
p_replaces = get_header(req, "Replaces");
if (!ast_strlen_zero(p_replaces)) {
/* We have a replaces header */
char *ptr;
char *fromtag = NULL;
char *totag = NULL;
char *start, *to;
int error = 0;
if (p->owner) {
ast_debug(3, "INVITE w Replaces on existing call? Refusing action. [%s]\n", p->callid);
transmit_response_reliable(p, "400 Bad request", req); /* The best way to not not accept the transfer */
/* Do not destroy existing call */
return -1;
}
if (sipdebug)
ast_debug(3, "INVITE part of call transfer. Replaces [%s]\n", p_replaces);
/* Create a buffer we can manipulate */
replace_id = ast_strdupa(p_replaces);
ast_uri_decode(replace_id);
if (!p->refer && !sip_refer_allocate(p)) {
transmit_response_reliable(p, "500 Server Internal Error", req);
append_history(p, "Xfer", "INVITE/Replace Failed. Out of memory.");
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
p->invitestate = INV_COMPLETED;
return -1;
}
/* Todo: (When we find phones that support this)
if the replaces header contains ";early-only"
we can only replace the call in early
stage, not after it's up.
If it's not in early mode, 486 Busy.
*/
/* Skip leading whitespace */
replace_id = ast_skip_blanks(replace_id);
start = replace_id;
while ( (ptr = strsep(&start, ";")) ) {
ptr = ast_skip_blanks(ptr); /* XXX maybe unnecessary ? */
if ( (to = strcasestr(ptr, "to-tag=") ) )
totag = to + 7; /* skip the keyword */
else if ( (to = strcasestr(ptr, "from-tag=") ) ) {
fromtag = to + 9; /* skip the keyword */
fromtag = strsep(&fromtag, "&"); /* trim what ? */
}
}
if (sipdebug)
ast_debug(4, "Invite/replaces: Will use Replace-Call-ID : %s Fromtag: %s Totag: %s\n",
replace_id,
fromtag ? fromtag : "<no from tag>",
totag ? totag : "<no to tag>");
/* Try to find call that we are replacing.
If we have a Replaces header, we need to cancel that call if we succeed with this call.
First we cheat a little and look for a magic call-id from phones that support
dialog-info+xml so we can do technology independent pickup... */
if (strncmp(replace_id, "pickup-", 7) == 0) {
struct sip_pvt *subscription = NULL;
replace_id += 7; /* Worst case we are looking at \0 */
if ((subscription = get_sip_pvt_byid_locked(replace_id, NULL, NULL)) == NULL) {
ast_log(LOG_NOTICE, "Unable to find subscription with call-id: %s\n", replace_id);
transmit_response_reliable(p, "481 Call Leg Does Not Exist (Replaces)", req);
error = 1;
} else {
ast_log(LOG_NOTICE, "Trying to pick up %s@%s\n", subscription->exten, subscription->context);
ast_copy_string(pickup.exten, subscription->exten, sizeof(pickup.exten));
ast_copy_string(pickup.context, subscription->context, sizeof(pickup.context));
sip_pvt_unlock(subscription);
if (subscription->owner) {
ast_channel_unlock(subscription->owner);
}
}
}
if (!error && ast_strlen_zero(pickup.exten) && (p->refer->refer_call = get_sip_pvt_byid_locked(replace_id, totag, fromtag)) == NULL) {
ast_log(LOG_NOTICE, "Supervised transfer attempted to replace non-existent call id (%s)!\n", replace_id);
transmit_response_reliable(p, "481 Call Leg Does Not Exist (Replaces)", req);
error = 1;
}
/* At this point, bot the pvt and the owner of the call to be replaced is locked */
/* The matched call is the call from the transferer to Asterisk .
We want to bridge the bridged part of the call to the
incoming invite, thus taking over the refered call */
if (p->refer->refer_call == p) {
ast_log(LOG_NOTICE, "INVITE with replaces into it's own call id (%s == %s)!\n", replace_id, p->callid);
p->refer->refer_call = dialog_unref(p->refer->refer_call, "unref dialog p->refer->refer_call");
transmit_response_reliable(p, "400 Bad request", req); /* The best way to not not accept the transfer */
error = 1;
}
if (!error && ast_strlen_zero(pickup.exten) && !p->refer->refer_call->owner) {
/* Oops, someting wrong anyway, no owner, no call */
ast_log(LOG_NOTICE, "Supervised transfer attempted to replace non-existing call id (%s)!\n", replace_id);
/* Check for better return code */
transmit_response_reliable(p, "481 Call Leg Does Not Exist (Replace)", req);
error = 1;
}
if (!error && ast_strlen_zero(pickup.exten) && p->refer->refer_call->owner->_state != AST_STATE_RINGING && p->refer->refer_call->owner->_state != AST_STATE_RING && p->refer->refer_call->owner->_state != AST_STATE_UP) {
ast_log(LOG_NOTICE, "Supervised transfer attempted to replace non-ringing or active call id (%s)!\n", replace_id);
transmit_response_reliable(p, "603 Declined (Replaces)", req);
error = 1;
}
if (error) { /* Give up this dialog */
append_history(p, "Xfer", "INVITE/Replace Failed.");
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
sip_pvt_unlock(p);
if (p->refer->refer_call) {
sip_pvt_unlock(p->refer->refer_call);
if (p->refer->refer_call->owner) {
ast_channel_unlock(p->refer->refer_call->owner);
}
}
p->invitestate = INV_COMPLETED;
return -1;
}
}
/* Check if this is an INVITE that sets up a new dialog or
a re-invite in an existing dialog */
if (!req->ignore) {
int newcall = (p->initreq.headers ? TRUE : FALSE);
if (sip_cancel_destroy(p))
ast_log(LOG_WARNING, "Unable to cancel SIP destruction. Expect bad things.\n");
/* This also counts as a pending invite */
p->pendinginvite = seqno;
check_via(p, req);
copy_request(&p->initreq, req); /* Save this INVITE as the transaction basis */
if (sipdebug)
ast_debug(1, "Initializing initreq for method %s - callid %s\n", sip_methods[req->method].text, p->callid);
if (!p->owner) { /* Not a re-invite */
if (debug)
ast_verbose("Using INVITE request as basis request - %s\n", p->callid);
if (newcall)
append_history(p, "Invite", "New call: %s", p->callid);
parse_ok_contact(p, req);
} else { /* Re-invite on existing call */
ast_clear_flag(&p->flags[0], SIP_OUTGOING); /* This is now an inbound dialog */
if (get_rpid(p, req)) {
struct ast_party_connected_line connected;
ast_party_connected_line_init(&connected);
connected.id.number = (char *) p->cid_num;
connected.id.name = (char *) p->cid_name;
connected.id.number_presentation = p->callingpres;
connected.source = AST_CONNECTED_LINE_UPDATE_SOURCE_TRANSFER;
ast_channel_queue_connected_line_update(p->owner, &connected);
}
/* Handle SDP here if we already have an owner */
if (find_sdp(req)) {
if (process_sdp(p, req, SDP_T38_INITIATE)) {
transmit_response_reliable(p, "488 Not acceptable here", req);
if (!p->lastinvite)
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
return -1;
}
} else {
p->jointcapability = p->capability;
ast_debug(1, "Hm.... No sdp for the moment\n");
}
if (p->do_history) /* This is a response, note what it was for */
append_history(p, "ReInv", "Re-invite received");
}
} else if (debug)
ast_verbose("Ignoring this INVITE request\n");
if (!p->lastinvite && !req->ignore && !p->owner) {
/* This is a new invite */
/* Handle authentication if this is our first invite */
struct ast_party_redirecting redirecting = {{0,},};
res = check_user(p, req, SIP_INVITE, e, XMIT_RELIABLE, sin);
if (res == AUTH_CHALLENGE_SENT) {
p->invitestate = INV_COMPLETED; /* Needs to restart in another INVITE transaction */
return 0;
}
if (res < 0) { /* Something failed in authentication */
if (res == AUTH_FAKE_AUTH) {
ast_log(LOG_NOTICE, "Sending fake auth rejection for device %s\n", get_header(req, "From"));
transmit_fake_auth_response(p, SIP_INVITE, req, XMIT_RELIABLE);
} else {
ast_log(LOG_NOTICE, "Failed to authenticate device %s\n", get_header(req, "From"));
transmit_response_reliable(p, "403 Forbidden", req);
}
p->invitestate = INV_COMPLETED;
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
ast_string_field_set(p, theirtag, NULL);
return 0;
}
/* If T38 is needed but not present, then make it magically appear */
if (ast_test_flag(&p->flags[1], SIP_PAGE2_T38SUPPORT) && !p->udptl) {
p->udptl = ast_udptl_new_with_bindaddr(sched, io, 0, bindaddr.sin_addr);
}
/* We have a succesful authentication, process the SDP portion if there is one */
if (find_sdp(req)) {
if (process_sdp(p, req, SDP_T38_INITIATE)) {
/* Unacceptable codecs */
transmit_response_reliable(p, "488 Not acceptable here", req);
p->invitestate = INV_COMPLETED;
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
ast_debug(1, "No compatible codecs for this SIP call.\n");
return -1;
}
} else { /* No SDP in invite, call control session */
p->jointcapability = p->capability;
ast_debug(2, "No SDP in Invite, third party call control\n");
}
/* Queue NULL frame to prod ast_rtp_bridge if appropriate */
/* This seems redundant ... see !p-owner above */
if (p->owner)
ast_queue_frame(p->owner, &ast_null_frame);
/* Initialize the context if it hasn't been already */
if (ast_strlen_zero(p->context))
ast_string_field_set(p, context, sip_cfg.default_context);
/* Check number of concurrent calls -vs- incoming limit HERE */
ast_debug(1, "Checking SIP call limits for device %s\n", p->username);
if ((res = update_call_counter(p, INC_CALL_LIMIT))) {
if (res < 0) {
ast_log(LOG_NOTICE, "Failed to place call for device %s, too many calls\n", p->username);
transmit_response_reliable(p, "480 Temporarily Unavailable (Call limit) ", req);
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
p->invitestate = INV_COMPLETED;
}
return 0;
}
gotdest = get_destination(p, NULL); /* Get destination right away */
change_redirecting_information(p, req, &redirecting, FALSE); /*Will return immediately if no Diversion header is present */
extract_uri(p, req); /* Get the Contact URI */
build_contact(p); /* Build our contact header */
if (p->rtp) {
ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_DTMF, ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833);
ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_DTMF_COMPENSATE, ast_test_flag(&p->flags[1], SIP_PAGE2_RFC2833_COMPENSATE));
}
if (!replace_id && gotdest) { /* No matching extension found */
if (gotdest == 1 && ast_test_flag(&p->flags[1], SIP_PAGE2_ALLOWOVERLAP))
transmit_response_reliable(p, "484 Address Incomplete", req);
else {
char *decoded_exten = ast_strdupa(p->exten);
transmit_response_reliable(p, "404 Not Found", req);
ast_uri_decode(decoded_exten);
ast_log(LOG_NOTICE, "Call from '%s' to extension"
" '%s' rejected because extension not found.\n",
S_OR(p->username, p->peername), decoded_exten);
}
p->invitestate = INV_COMPLETED;
update_call_counter(p, DEC_CALL_LIMIT);
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
return 0;
} else {
/* If no extension was specified, use the s one */
/* Basically for calling to IP/Host name only */
if (ast_strlen_zero(p->exten))
ast_string_field_set(p, exten, "s");
/* Initialize our tag */
make_our_tag(p->tag, sizeof(p->tag));
/* First invitation - create the channel */
c = sip_new(p, AST_STATE_DOWN, S_OR(p->peername, NULL));
*recount = 1;
/* Save Record-Route for any later requests we make on this dialogue */
build_route(p, req, 0);
if (c) {
/* Pre-lock the call */
ast_channel_lock(c);
ast_channel_set_redirecting(c, &redirecting);
}
}
} else {
struct ast_party_redirecting redirecting = {{0,},};
if (sipdebug) {
if (!req->ignore)
ast_debug(2, "Got a SIP re-invite for call %s\n", p->callid);
else
ast_debug(2, "Got a SIP re-transmit of INVITE for call %s\n", p->callid);
}
if (!req->ignore)
reinvite = 1;
c = p->owner;
change_redirecting_information(p, req, &redirecting, FALSE); /*Will return immediately if no Diversion header is present */
if (c) {
ast_channel_set_redirecting(c, &redirecting);
}
}
/* Session-Timers */
if (p->sipoptions == SIP_OPT_TIMER) {
/* The UAC has requested session-timers for this session. Negotiate
the session refresh interval and who will be the refresher */
ast_debug(2, "Incoming INVITE with 'timer' option enabled\n");
/* Allocate Session-Timers struct w/in the dialog */
if (!p->stimer)
sip_st_alloc(p);
/* Parse the Session-Expires header */
p_uac_se_hdr = get_header(req, "Session-Expires");
if (!ast_strlen_zero(p_uac_se_hdr)) {
rtn = parse_session_expires(p_uac_se_hdr, &uac_max_se, &st_ref);
if (rtn != 0) {
transmit_response_reliable(p, "400 Session-Expires Invalid Syntax", req);
p->invitestate = INV_COMPLETED;
if (!p->lastinvite) {
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
}
return -1;
}
}
/* Parse the Min-SE header */
p_uac_min_se = get_header(req, "Min-SE");
if (!ast_strlen_zero(p_uac_min_se)) {
rtn = parse_minse(p_uac_min_se, &uac_min_se);
if (rtn != 0) {
transmit_response_reliable(p, "400 Min-SE Invalid Syntax", req);
p->invitestate = INV_COMPLETED;
if (!p->lastinvite) {
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
}
return -1;
}
}
dlg_min_se = st_get_se(p, FALSE);
switch (st_get_mode(p)) {
case SESSION_TIMER_MODE_ACCEPT:
case SESSION_TIMER_MODE_ORIGINATE:
if (uac_max_se > 0 && uac_max_se < dlg_min_se) {
transmit_response_with_minse(p, "422 Session Interval Too Small", req, dlg_min_se);
p->invitestate = INV_COMPLETED;
if (!p->lastinvite) {
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
}
return -1;
}
p->stimer->st_active_peer_ua = TRUE;
st_active = TRUE;
if (st_ref == SESSION_TIMER_REFRESHER_AUTO) {
st_ref = st_get_refresher(p);
}
if (uac_max_se > 0) {
int dlg_max_se = st_get_se(p, TRUE);
if (dlg_max_se >= uac_min_se) {
st_interval = (uac_max_se < dlg_max_se) ? uac_max_se : dlg_max_se;
} else {
st_interval = uac_max_se;
}
} else {
st_interval = uac_min_se;
}
break;
case SESSION_TIMER_MODE_REFUSE:
if (p->reqsipoptions == SIP_OPT_TIMER) {
transmit_response_with_unsupported(p, "420 Option Disabled", req, required);
ast_log(LOG_WARNING, "Received SIP INVITE with supported but disabled option: %s\n", required);
p->invitestate = INV_COMPLETED;
if (!p->lastinvite) {
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
}
return -1;
}
break;
default:
ast_log(LOG_ERROR, "Internal Error %d at %s:%d\n", st_get_mode(p), __FILE__, __LINE__);
break;
}
} else {
/* The UAC did not request session-timers. Asterisk (UAS), will now decide
(based on session-timer-mode in sip.conf) whether to run session-timers for
this session or not. */
switch (st_get_mode(p)) {
case SESSION_TIMER_MODE_ORIGINATE:
st_active = TRUE;
st_interval = st_get_se(p, TRUE);
st_ref = SESSION_TIMER_REFRESHER_UAS;
p->stimer->st_active_peer_ua = FALSE;
break;
default:
break;
}
}
if (reinvite == 0) {
/* Session-Timers: Start session refresh timer based on negotiation/config */
if (st_active == TRUE) {
p->stimer->st_active = TRUE;
p->stimer->st_interval = st_interval;
p->stimer->st_ref = st_ref;
start_session_timer(p);
}
} else {
if (p->stimer->st_active == TRUE) {
/* Session-Timers: A re-invite request sent within a dialog will serve as
a refresh request, no matter whether the re-invite was sent for refreshing
the session or modifying it.*/
ast_debug (2, "Restarting session-timers on a refresh - %s\n", p->callid);
/* The UAC may be adjusting the session-timers mid-session */
if (st_interval > 0) {
p->stimer->st_interval = st_interval;
p->stimer->st_ref = st_ref;
}
restart_session_timer(p);
if (p->stimer->st_expirys > 0) {
p->stimer->st_expirys--;
}
}
}
if (!req->ignore && p)
p->lastinvite = seqno;
if (replace_id) { /* Attended transfer or call pickup - we're the target */
if (!ast_strlen_zero(pickup.exten)) {
append_history(p, "Xfer", "INVITE/Replace received");
/* Let the caller know we're giving it a shot */
transmit_response(p, "100 Trying", req);
ast_setstate(c, AST_STATE_RING);
/* Do the pickup itself */
ast_channel_unlock(c);
*nounlock = 1;
do_magic_pickup(c, pickup.exten, pickup.context);
/* Now we're either masqueraded or we failed to pickup, in either case we... */
ast_hangup(c);
return 0;
} else {
/* Go and take over the target call */
if (sipdebug)
ast_debug(4, "Sending this call to the invite/replcaes handler %s\n", p->callid);
return handle_invite_replaces(p, req, debug, seqno, sin);
}
}
if (c) { /* We have a call -either a new call or an old one (RE-INVITE) */
switch(c->_state) {
case AST_STATE_DOWN:
ast_debug(2, "%s: New call is still down.... Trying... \n", c->name);
transmit_response(p, "100 Trying", req);
p->invitestate = INV_PROCEEDING;
ast_setstate(c, AST_STATE_RING);
if (strcmp(p->exten, ast_pickup_ext())) { /* Call to extension -start pbx on this call */
enum ast_pbx_result result;
result = ast_pbx_start(c);
switch(result) {
case AST_PBX_FAILED:
ast_log(LOG_WARNING, "Failed to start PBX :(\n");
p->invitestate = INV_COMPLETED;
transmit_response_reliable(p, "503 Unavailable", req);
break;
case AST_PBX_CALL_LIMIT:
ast_log(LOG_WARNING, "Failed to start PBX (call limit reached) \n");
p->invitestate = INV_COMPLETED;
transmit_response_reliable(p, "480 Temporarily Unavailable", req);
break;
case AST_PBX_SUCCESS:
/* nothing to do */
break;
}
if (result) {
/* Unlock locks so ast_hangup can do its magic */
ast_channel_unlock(c);
sip_pvt_unlock(p);
ast_hangup(c);
sip_pvt_lock(p);
c = NULL;
}
} else { /* Pickup call in call group */
ast_channel_unlock(c);
*nounlock = 1;
if (ast_pickup_call(c)) {
ast_log(LOG_NOTICE, "Nothing to pick up for %s\n", p->callid);
transmit_response_reliable(p, "503 Unavailable", req);
sip_alreadygone(p);
/* Unlock locks so ast_hangup can do its magic */
sip_pvt_unlock(p);
c->hangupcause = AST_CAUSE_CALL_REJECTED;
} else {
sip_pvt_unlock(p);
c->hangupcause = AST_CAUSE_NORMAL_CLEARING;
}
p->invitestate = INV_COMPLETED;
ast_hangup(c);
sip_pvt_lock(p);
c = NULL;
}
break;
case AST_STATE_RING:
transmit_response(p, "100 Trying", req);
p->invitestate = INV_PROCEEDING;
break;
case AST_STATE_RINGING:
transmit_response(p, "180 Ringing", req);
p->invitestate = INV_PROCEEDING;
break;
case AST_STATE_UP:
ast_debug(2, "%s: This call is UP.... \n", c->name);
transmit_response(p, "100 Trying", req);
if (p->t38.state == T38_PEER_REINVITE) {
p->t38id = ast_sched_add(sched, 5000, sip_t38_abort, dialog_ref(p, "passing dialog ptr into sched structure based on t38id for sip_t38_abort."));
} else if (p->t38.state == T38_ENABLED) {
ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
transmit_response_with_t38_sdp(p, "200 OK", req, (reinvite ? XMIT_RELIABLE : (req->ignore ? XMIT_UNRELIABLE : XMIT_CRITICAL)));
} else if (p->t38.state == T38_DISABLED) {
/* If this is not a re-invite or something to ignore - it's critical */
ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
transmit_response_with_sdp(p, "200 OK", req, (reinvite ? XMIT_RELIABLE : (req->ignore ? XMIT_UNRELIABLE : XMIT_CRITICAL)), p->session_modify == TRUE ? FALSE : TRUE, FALSE);
}
p->invitestate = INV_TERMINATED;
break;
default:
ast_log(LOG_WARNING, "Don't know how to handle INVITE in state %d\n", c->_state);
transmit_response(p, "100 Trying", req);
break;
}
} else {
if (p && (p->autokillid == -1)) {
const char *msg;
if (!p->jointcapability)
msg = "488 Not Acceptable Here (codec error)";
else {
ast_log(LOG_NOTICE, "Unable to create/find SIP channel for this INVITE\n");
msg = "503 Unavailable";
}
transmit_response_reliable(p, msg, req);
p->invitestate = INV_COMPLETED;
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
}
}
return res;
}
/*! \brief Find all call legs and bridge transferee with target
* called from handle_request_refer */
static int local_attended_transfer(struct sip_pvt *transferer, struct sip_dual *current, struct sip_request *req, int seqno)
{
struct sip_dual target; /* Chan 1: Call from tranferer to Asterisk */
/* Chan 2: Call from Asterisk to target */
int res = 0;
struct sip_pvt *targetcall_pvt;
/* Check if the call ID of the replaces header does exist locally */
if (!(targetcall_pvt = get_sip_pvt_byid_locked(transferer->refer->replaces_callid, transferer->refer->replaces_callid_totag,
transferer->refer->replaces_callid_fromtag))) {
if (transferer->refer->localtransfer) {
/* We did not find the refered call. Sorry, can't accept then */
transmit_response(transferer, "202 Accepted", req);
/* Let's fake a response from someone else in order
to follow the standard */
transmit_notify_with_sipfrag(transferer, seqno, "481 Call leg/transaction does not exist", TRUE);
append_history(transferer, "Xfer", "Refer failed");
ast_clear_flag(&transferer->flags[0], SIP_GOTREFER);
transferer->refer->status = REFER_FAILED;
return -1;
}
/* Fall through for remote transfers that we did not find locally */
ast_debug(3, "SIP attended transfer: Not our call - generating INVITE with replaces\n");
return 0;
}
/* Ok, we can accept this transfer */
transmit_response(transferer, "202 Accepted", req);
append_history(transferer, "Xfer", "Refer accepted");
if (!targetcall_pvt->owner) { /* No active channel */
ast_debug(4, "SIP attended transfer: Error: No owner of target call\n");
/* Cancel transfer */
transmit_notify_with_sipfrag(transferer, seqno, "503 Service Unavailable", TRUE);
append_history(transferer, "Xfer", "Refer failed");
ast_clear_flag(&transferer->flags[0], SIP_GOTREFER);
transferer->refer->status = REFER_FAILED;
sip_pvt_unlock(targetcall_pvt);
if (targetcall_pvt)
ao2_t_ref(targetcall_pvt, -1, "Drop targetcall_pvt pointer");
return -1;
}
/* We have a channel, find the bridge */
target.chan1 = targetcall_pvt->owner; /* Transferer to Asterisk */
target.chan2 = ast_bridged_channel(targetcall_pvt->owner); /* Asterisk to target */
if (!target.chan2 || !(target.chan2->_state == AST_STATE_UP || target.chan2->_state == AST_STATE_RINGING) ) {
/* Wrong state of new channel */
if (target.chan2)
ast_debug(4, "SIP attended transfer: Error: Wrong state of target call: %s\n", ast_state2str(target.chan2->_state));
else if (target.chan1->_state != AST_STATE_RING)
ast_debug(4, "SIP attended transfer: Error: No target channel\n");
else
ast_debug(4, "SIP attended transfer: Attempting transfer in ringing state\n");
}
/* Transfer */
if (sipdebug) {
if (current->chan2) /* We have two bridges */
ast_debug(4, "SIP attended transfer: trying to bridge %s and %s\n", target.chan1->name, current->chan2->name);
else /* One bridge, propably transfer of IVR/voicemail etc */
ast_debug(4, "SIP attended transfer: trying to make %s take over (masq) %s\n", target.chan1->name, current->chan1->name);
}
ast_set_flag(&transferer->flags[0], SIP_DEFER_BYE_ON_TRANSFER); /* Delay hangup */
/* Perform the transfer */
manager_event(EVENT_FLAG_CALL, "Transfer", "TransferMethod: SIP\r\nTransferType: Attended\r\nChannel: %s\r\nUniqueid: %s\r\nSIP-Callid: %s\r\nTargetChannel: %s\r\nTargetUniqueid: %s\r\n",
transferer->owner->name,
transferer->owner->uniqueid,
transferer->callid,
target.chan1->name,
target.chan1->uniqueid);
res = attempt_transfer(current, &target);
sip_pvt_unlock(targetcall_pvt);
if (res) {
/* Failed transfer */
transmit_notify_with_sipfrag(transferer, seqno, "486 Busy Here", TRUE);
append_history(transferer, "Xfer", "Refer failed");
if (targetcall_pvt->owner)
ast_channel_unlock(targetcall_pvt->owner);
ast_clear_flag(&transferer->flags[0], SIP_DEFER_BYE_ON_TRANSFER);
} else {
struct ast_party_connected_line connected_caller;
/* Transfer succeeded! */
const char *xfersound = pbx_builtin_getvar_helper(target.chan1, "ATTENDED_TRANSFER_COMPLETE_SOUND");
/* Tell transferer that we're done. */
transmit_notify_with_sipfrag(transferer, seqno, "200 OK", TRUE);
append_history(transferer, "Xfer", "Refer succeeded");
transferer->refer->status = REFER_200OK;
if (target.chan2 && !ast_strlen_zero(xfersound) && ast_streamfile(target.chan2, xfersound, target.chan2->language) >= 0) {
ast_waitstream(target.chan2, "");
}
if (targetcall_pvt->owner) {
ast_debug(1, "SIP attended transfer: Unlocking channel %s\n", targetcall_pvt->owner->name);
ast_channel_unlock(targetcall_pvt->owner);
}
ast_party_connected_line_init(&connected_caller);
if (target.chan2) {
if (current->chan2) {
/* Tell each of the other channels to whom they are now connected */
ast_channel_lock(current->chan2);
ast_connected_line_copy_from_caller(&connected_caller, &current->chan2->cid);
ast_channel_unlock(current->chan2);
connected_caller.source = AST_CONNECTED_LINE_UPDATE_SOURCE_TRANSFER;
ast_channel_update_connected_line(target.chan2, &connected_caller);
ast_channel_lock(target.chan2);
ast_connected_line_copy_from_caller(&connected_caller, &target.chan2->cid);
ast_channel_unlock(target.chan2);
connected_caller.source = AST_CONNECTED_LINE_UPDATE_SOURCE_TRANSFER;
ast_channel_update_connected_line(current->chan2, &connected_caller);
ast_party_connected_line_free(&connected_caller);
}
} else {
/* Notify the first other party that they are connected to someone else assuming that target.chan1
has progressed far enough through the dialplan to have it's called party information set. */
if (current->chan2) {
ast_channel_lock(target.chan1);
ast_party_connected_line_copy(&connected_caller, &target.chan1->connected);
ast_channel_unlock(target.chan1);
connected_caller = target.chan1->connected;
connected_caller.source = AST_CONNECTED_LINE_UPDATE_SOURCE_TRANSFER;
ast_channel_update_connected_line(current->chan2, &connected_caller);
ast_party_connected_line_free(&connected_caller);
}
/* We can't indicate to the called channel directly so we force the masquerade to complete
and queue and update to be read and passed-through */
ast_channel_lock(target.chan1);
ast_do_masquerade(target.chan1);
ast_channel_unlock(target.chan1);
ast_party_connected_line_collect_caller(&connected_caller, &target.chan1->cid);
connected_caller.source = AST_CONNECTED_LINE_UPDATE_SOURCE_TRANSFER;
ast_channel_queue_connected_line_update(target.chan1, &connected_caller);
}
}
if (targetcall_pvt)
ao2_t_ref(targetcall_pvt, -1, "drop targetcall_pvt");
return 1;
}
/*! \brief Handle incoming REFER request */
/*! \page SIP_REFER SIP transfer Support (REFER)
REFER is used for call transfer in SIP. We get a REFER
to place a new call with an INVITE somwhere and then
keep the transferor up-to-date of the transfer. If the
transfer fails, get back on line with the orginal call.
- REFER can be sent outside or inside of a dialog.
Asterisk only accepts REFER inside of a dialog.
- If we get a replaces header, it is an attended transfer
\par Blind transfers
The transferor provides the transferee
with the transfer targets contact. The signalling between
transferer or transferee should not be cancelled, so the
call is recoverable if the transfer target can not be reached
by the transferee.
In this case, Asterisk receives a TRANSFER from
the transferor, thus is the transferee. We should
try to set up a call to the contact provided
and if that fails, re-connect the current session.
If the new call is set up, we issue a hangup.
In this scenario, we are following section 5.2
in the SIP CC Transfer draft. (Transfer without
a GRUU)
\par Transfer with consultation hold
In this case, the transferor
talks to the transfer target before the transfer takes place.
This is implemented with SIP hold and transfer.
Note: The invite From: string could indicate a transfer.
(Section 6. Transfer with consultation hold)
The transferor places the transferee on hold, starts a call
with the transfer target to alert them to the impending
transfer, terminates the connection with the target, then
proceeds with the transfer (as in Blind transfer above)
\par Attended transfer
The transferor places the transferee
on hold, calls the transfer target to alert them,
places the target on hold, then proceeds with the transfer
using a Replaces header field in the Refer-to header. This
will force the transfee to send an Invite to the target,
with a replaces header that instructs the target to
hangup the call between the transferor and the target.
In this case, the Refer/to: uses the AOR address. (The same
URI that the transferee used to establish the session with
the transfer target (To: ). The Require: replaces header should
be in the INVITE to avoid the wrong UA in a forked SIP proxy
scenario to answer and have no call to replace with.
The referred-by header is *NOT* required, but if we get it,
can be copied into the INVITE to the transfer target to
inform the target about the transferor
"Any REFER request has to be appropriately authenticated.".
We can't destroy dialogs, since we want the call to continue.
*/
static int handle_request_refer(struct sip_pvt *p, struct sip_request *req, int debug, int seqno, int *nounlock)
{
struct sip_dual current; /* Chan1: Call between asterisk and transferer */
/* Chan2: Call between asterisk and transferee */
int res = 0;
current.req.data = NULL;
if (req->debug)
ast_verbose("Call %s got a SIP call transfer from %s: (REFER)!\n", p->callid, ast_test_flag(&p->flags[0], SIP_OUTGOING) ? "callee" : "caller");
if (!p->owner) {
/* This is a REFER outside of an existing SIP dialog */
/* We can't handle that, so decline it */
ast_debug(3, "Call %s: Declined REFER, outside of dialog...\n", p->callid);
transmit_response(p, "603 Declined (No dialog)", req);
if (!req->ignore) {
append_history(p, "Xfer", "Refer failed. Outside of dialog.");
sip_alreadygone(p);
pvt_set_needdestroy(p, "outside of dialog");
}
return 0;
}
/* Check if transfer is allowed from this device */
if (p->allowtransfer == TRANSFER_CLOSED ) {
/* Transfer not allowed, decline */
transmit_response(p, "603 Declined (policy)", req);
append_history(p, "Xfer", "Refer failed. Allowtransfer == closed.");
/* Do not destroy SIP session */
return 0;
}
if (!req->ignore && ast_test_flag(&p->flags[0], SIP_GOTREFER)) {
/* Already have a pending REFER */
transmit_response(p, "491 Request pending", req);
append_history(p, "Xfer", "Refer failed. Request pending.");
return 0;
}
/* Allocate memory for call transfer data */
if (!p->refer && !sip_refer_allocate(p)) {
transmit_response(p, "500 Internal Server Error", req);
append_history(p, "Xfer", "Refer failed. Memory allocation error.");
return -3;
}
res = get_refer_info(p, req); /* Extract headers */
p->refer->status = REFER_SENT;
if (res != 0) {
switch (res) {
case -2: /* Syntax error */
transmit_response(p, "400 Bad Request (Refer-to missing)", req);
append_history(p, "Xfer", "Refer failed. Refer-to missing.");
if (req->debug)
ast_debug(1, "SIP transfer to black hole can't be handled (no refer-to: )\n");
break;
case -3:
transmit_response(p, "603 Declined (Non sip: uri)", req);
append_history(p, "Xfer", "Refer failed. Non SIP uri");
if (req->debug)
ast_debug(1, "SIP transfer to non-SIP uri denied\n");
break;
default:
/* Refer-to extension not found, fake a failed transfer */
transmit_response(p, "202 Accepted", req);
append_history(p, "Xfer", "Refer failed. Bad extension.");
transmit_notify_with_sipfrag(p, seqno, "404 Not found", TRUE);
ast_clear_flag(&p->flags[0], SIP_GOTREFER);
if (req->debug)
ast_debug(1, "SIP transfer to bad extension: %s\n", p->refer->refer_to);
break;
}
return 0;
}
if (ast_strlen_zero(p->context))
ast_string_field_set(p, context, sip_cfg.default_context);
/* If we do not support SIP domains, all transfers are local */
if (sip_cfg.allow_external_domains && check_sip_domain(p->refer->refer_to_domain, NULL, 0)) {
p->refer->localtransfer = 1;
if (sipdebug)
ast_debug(3, "This SIP transfer is local : %s\n", p->refer->refer_to_domain);
} else if (AST_LIST_EMPTY(&domain_list) || check_sip_domain(p->refer->refer_to_domain, NULL, 0)) {
/* This PBX doesn't bother with SIP domains or domain is local, so this transfer is local */
p->refer->localtransfer = 1;
} else if (sipdebug)
ast_debug(3, "This SIP transfer is to a remote SIP extension (remote domain %s)\n", p->refer->refer_to_domain);
/* Is this a repeat of a current request? Ignore it */
/* Don't know what else to do right now. */
if (req->ignore)
return res;
/* If this is a blind transfer, we have the following
channels to work with:
- chan1, chan2: The current call between transferer and transferee (2 channels)
- target_channel: A new call from the transferee to the target (1 channel)
We need to stay tuned to what happens in order to be able
to bring back the call to the transferer */
/* If this is a attended transfer, we should have all call legs within reach:
- chan1, chan2: The call between the transferer and transferee (2 channels)
- target_channel, targetcall_pvt: The call between the transferer and the target (2 channels)
We want to bridge chan2 with targetcall_pvt!
The replaces call id in the refer message points
to the call leg between Asterisk and the transferer.
So we need to connect the target and the transferee channel
and hangup the two other channels silently
If the target is non-local, the call ID could be on a remote
machine and we need to send an INVITE with replaces to the
target. We basically handle this as a blind transfer
and let the sip_call function catch that we need replaces
header in the INVITE.
*/
/* Get the transferer's channel */
current.chan1 = p->owner;
/* Find the other part of the bridge (2) - transferee */
current.chan2 = ast_bridged_channel(current.chan1);
if (sipdebug)
ast_debug(3, "SIP %s transfer: Transferer channel %s, transferee channel %s\n", p->refer->attendedtransfer ? "attended" : "blind", current.chan1->name, current.chan2 ? current.chan2->name : "<none>");
if (!current.chan2 && !p->refer->attendedtransfer) {
/* No bridged channel, propably IVR or echo or similar... */
/* Guess we should masquerade or something here */
/* Until we figure it out, refuse transfer of such calls */
if (sipdebug)
ast_debug(3, "Refused SIP transfer on non-bridged channel.\n");
p->refer->status = REFER_FAILED;
append_history(p, "Xfer", "Refer failed. Non-bridged channel.");
transmit_response(p, "603 Declined", req);
return -1;
}
if (current.chan2) {
if (sipdebug)
ast_debug(4, "Got SIP transfer, applying to bridged peer '%s'\n", current.chan2->name);
ast_queue_control(current.chan1, AST_CONTROL_UNHOLD);
}
ast_set_flag(&p->flags[0], SIP_GOTREFER);
/* Attended transfer: Find all call legs and bridge transferee with target*/
if (p->refer->attendedtransfer) {
if ((res = local_attended_transfer(p, &current, req, seqno)))
return res; /* We're done with the transfer */
/* Fall through for remote transfers that we did not find locally */
if (sipdebug)
ast_debug(4, "SIP attended transfer: Still not our call - generating INVITE with replaces\n");
/* Fallthrough if we can't find the call leg internally */
}
/* Parking a call */
if (p->refer->localtransfer && !strcmp(p->refer->refer_to, ast_parking_ext())) {
/* Must release c's lock now, because it will not longer be accessible after the transfer! */
*nounlock = 1;
ast_channel_unlock(current.chan1);
copy_request(&current.req, req);
ast_clear_flag(&p->flags[0], SIP_GOTREFER);
p->refer->status = REFER_200OK;
append_history(p, "Xfer", "REFER to call parking.");
manager_event(EVENT_FLAG_CALL, "Transfer", "TransferMethod: SIP\r\nTransferType: Blind\r\nChannel: %s\r\nUniqueid: %s\r\nSIP-Callid: %s\r\nTargetChannel: %s\r\nTargetUniqueid: %s\r\nTransferExten: %s\r\nTransfer2Parking: Yes\r\n",
current.chan1->name,
current.chan1->uniqueid,
p->callid,
current.chan2->name,
current.chan2->uniqueid,
p->refer->refer_to);
if (sipdebug)
ast_debug(4, "SIP transfer to parking: trying to park %s. Parked by %s\n", current.chan2->name, current.chan1->name);
sip_park(current.chan2, current.chan1, req, seqno);
return res;
}
/* Blind transfers and remote attended xfers */
transmit_response(p, "202 Accepted", req);
if (current.chan1 && current.chan2) {
ast_debug(3, "chan1->name: %s\n", current.chan1->name);
pbx_builtin_setvar_helper(current.chan1, "BLINDTRANSFER", current.chan2->name);
}
if (current.chan2) {
pbx_builtin_setvar_helper(current.chan2, "BLINDTRANSFER", current.chan1->name);
pbx_builtin_setvar_helper(current.chan2, "SIPDOMAIN", p->refer->refer_to_domain);
pbx_builtin_setvar_helper(current.chan2, "SIPTRANSFER", "yes");
/* One for the new channel */
pbx_builtin_setvar_helper(current.chan2, "_SIPTRANSFER", "yes");
/* Attended transfer to remote host, prepare headers for the INVITE */
if (p->refer->referred_by)
pbx_builtin_setvar_helper(current.chan2, "_SIPTRANSFER_REFERER", p->refer->referred_by);
}
/* Generate a Replaces string to be used in the INVITE during attended transfer */
if (!ast_strlen_zero(p->refer->replaces_callid)) {
char tempheader[SIPBUFSIZE];
snprintf(tempheader, sizeof(tempheader), "%s%s%s%s%s", p->refer->replaces_callid,
p->refer->replaces_callid_totag ? ";to-tag=" : "",
p->refer->replaces_callid_totag,
p->refer->replaces_callid_fromtag ? ";from-tag=" : "",
p->refer->replaces_callid_fromtag);
if (current.chan2)
pbx_builtin_setvar_helper(current.chan2, "_SIPTRANSFER_REPLACES", tempheader);
}
/* Must release lock now, because it will not longer
be accessible after the transfer! */
*nounlock = 1;
ast_channel_unlock(current.chan1);
/* Connect the call */
/* FAKE ringing if not attended transfer */
if (!p->refer->attendedtransfer)
transmit_notify_with_sipfrag(p, seqno, "183 Ringing", FALSE);
/* For blind transfer, this will lead to a new call */
/* For attended transfer to remote host, this will lead to
a new SIP call with a replaces header, if the dial plan allows it
*/
if (!current.chan2) {
/* We have no bridge, so we're talking with Asterisk somehow */
/* We need to masquerade this call */
/* What to do to fix this situation:
* Set up the new call in a new channel
* Let the new channel masq into this channel
Please add that code here :-)
*/
p->refer->status = REFER_FAILED;
transmit_notify_with_sipfrag(p, seqno, "503 Service Unavailable (can't handle one-legged xfers)", TRUE);
ast_clear_flag(&p->flags[0], SIP_GOTREFER);
append_history(p, "Xfer", "Refer failed (only bridged calls).");
return -1;
}
ast_set_flag(&p->flags[0], SIP_DEFER_BYE_ON_TRANSFER); /* Delay hangup */
/* For blind transfers, move the call to the new extensions. For attended transfers on multiple
servers - generate an INVITE with Replaces. Either way, let the dial plan decided */
res = ast_async_goto(current.chan2, p->refer->refer_to_context, p->refer->refer_to, 1);
if (!res) {
manager_event(EVENT_FLAG_CALL, "Transfer", "TransferMethod: SIP\r\nTransferType: Blind\r\nChannel: %s\r\nUniqueid: %s\r\nSIP-Callid: %s\r\nTargetChannel: %s\r\nTargetUniqueid: %s\r\nTransferExten: %s\r\nTransferContext: %s\r\n",
current.chan1->name,
current.chan1->uniqueid,
p->callid,
current.chan2->name,
current.chan2->uniqueid,
p->refer->refer_to, p->refer->refer_to_context);
/* Success - we have a new channel */
ast_debug(3, "%s transfer succeeded. Telling transferer.\n", p->refer->attendedtransfer? "Attended" : "Blind");
transmit_notify_with_sipfrag(p, seqno, "200 Ok", TRUE);
if (p->refer->localtransfer)
p->refer->status = REFER_200OK;
if (p->owner)
p->owner->hangupcause = AST_CAUSE_NORMAL_CLEARING;
append_history(p, "Xfer", "Refer succeeded.");
ast_clear_flag(&p->flags[0], SIP_GOTREFER);
/* Do not hangup call, the other side do that when we say 200 OK */
/* We could possibly implement a timer here, auto congestion */
res = 0;
} else {
ast_clear_flag(&p->flags[0], SIP_DEFER_BYE_ON_TRANSFER); /* Don't delay hangup */
ast_debug(3, "%s transfer failed. Resuming original call.\n", p->refer->attendedtransfer? "Attended" : "Blind");
append_history(p, "Xfer", "Refer failed.");
/* Failure of some kind */
p->refer->status = REFER_FAILED;
transmit_notify_with_sipfrag(p, seqno, "503 Service Unavailable", TRUE);
ast_clear_flag(&p->flags[0], SIP_GOTREFER);
res = -1;
}
return res;
}
/*! \brief Handle incoming CANCEL request */
static int handle_request_cancel(struct sip_pvt *p, struct sip_request *req)
{
check_via(p, req);
sip_alreadygone(p);
/* At this point, we could have cancelled the invite at the same time
as the other side sends a CANCEL. Our final reply with error code
might not have been received by the other side before the CANCEL
was sent, so let's just give up retransmissions and waiting for
ACK on our error code. The call is hanging up any way. */
if (p->invitestate == INV_TERMINATED)
__sip_pretend_ack(p);
else
p->invitestate = INV_CANCELLED;
if (p->owner && p->owner->_state == AST_STATE_UP) {
/* This call is up, cancel is ignored, we need a bye */
transmit_response(p, "200 OK", req);
ast_debug(1, "Got CANCEL on an answered call. Ignoring... \n");
return 0;
}
if (ast_test_flag(&p->flags[0], SIP_INC_COUNT) || ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD))
update_call_counter(p, DEC_CALL_LIMIT);
stop_media_flows(p); /* Immediately stop RTP, VRTP and UDPTL as applicable */
if (p->owner)
ast_queue_hangup(p->owner);
else
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
if (p->initreq.len > 0) {
transmit_response_reliable(p, "487 Request Terminated", &p->initreq);
transmit_response(p, "200 OK", req);
return 1;
} else {
transmit_response(p, "481 Call Leg Does Not Exist", req);
return 0;
}
}
static int acf_channel_read(struct ast_channel *chan, const char *funcname, char *preparse, char *buf, size_t buflen)
{
struct sip_pvt *p = chan->tech_pvt;
char *parse = ast_strdupa(preparse);
int res = 0;
AST_DECLARE_APP_ARGS(args,
AST_APP_ARG(param);
AST_APP_ARG(type);
AST_APP_ARG(field);
);
AST_STANDARD_APP_ARGS(args, parse);
/* Sanity check */
if (!IS_SIP_TECH(chan->tech)) {
ast_log(LOG_ERROR, "Cannot call %s on a non-SIP channel\n", funcname);
return 0;
}
memset(buf, 0, buflen);
if (!strcasecmp(args.param, "peerip")) {
ast_copy_string(buf, p->sa.sin_addr.s_addr ? ast_inet_ntoa(p->sa.sin_addr) : "", buflen);
} else if (!strcasecmp(args.param, "recvip")) {
ast_copy_string(buf, p->recv.sin_addr.s_addr ? ast_inet_ntoa(p->recv.sin_addr) : "", buflen);
} else if (!strcasecmp(args.param, "from")) {
ast_copy_string(buf, p->from, buflen);
} else if (!strcasecmp(args.param, "uri")) {
ast_copy_string(buf, p->uri, buflen);
} else if (!strcasecmp(args.param, "useragent")) {
ast_copy_string(buf, p->useragent, buflen);
} else if (!strcasecmp(args.param, "peername")) {
ast_copy_string(buf, p->peername, buflen);
} else if (!strcasecmp(args.param, "t38passthrough")) {
ast_copy_string(buf, (p->t38.state == T38_DISABLED) ? "0" : "1", buflen);
} else if (!strcasecmp(args.param, "rtpdest")) {
struct sockaddr_in sin;
if (ast_strlen_zero(args.type))
args.type = "audio";
if (!strcasecmp(args.type, "audio"))
ast_rtp_instance_get_remote_address(p->rtp, &sin);
else if (!strcasecmp(args.type, "video"))
ast_rtp_instance_get_remote_address(p->vrtp, &sin);
else if (!strcasecmp(args.type, "text"))
ast_rtp_instance_get_remote_address(p->trtp, &sin);
else
return -1;
snprintf(buf, buflen, "%s:%d", ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port));
} else if (!strcasecmp(args.param, "rtpqos")) {
struct ast_rtp_instance *rtp = NULL;
if (ast_strlen_zero(args.type)) {
args.type = "audio";
}
if (!strcasecmp(args.type, "audio")) {
rtp = p->rtp;
} else if (!strcasecmp(args.type, "video")) {
rtp = p->vrtp;
} else if (!strcasecmp(args.type, "text")) {
rtp = p->trtp;
} else {
return -1;
}
if (ast_strlen_zero(args.field) || !strcasecmp(args.field, "all")) {
char quality_buf[AST_MAX_USER_FIELD], *quality;
if (!(quality = ast_rtp_instance_get_quality(rtp, AST_RTP_INSTANCE_STAT_FIELD_QUALITY, quality_buf, sizeof(quality_buf)))) {
return -1;
}
ast_copy_string(buf, quality_buf, buflen);
return res;
} else {
struct ast_rtp_instance_stats stats;
if (ast_rtp_instance_get_stats(rtp, &stats, AST_RTP_INSTANCE_STAT_ALL)) {
return -1;
}
if (!strcasecmp(args.field, "local_ssrc")) {
snprintf(buf, buflen, "%u", stats.local_ssrc);
} else if (!strcasecmp(args.field, "local_lostpackets")) {
snprintf(buf, buflen, "%u", stats.rxploss);
} else if (!strcasecmp(args.field, "local_jitter")) {
snprintf(buf, buflen, "%u", stats.rxjitter);
} else if (!strcasecmp(args.field, "local_count")) {
snprintf(buf, buflen, "%u", stats.rxcount);
} else if (!strcasecmp(args.field, "remote_ssrc")) {
snprintf(buf, buflen, "%u", stats.remote_ssrc);
} else if (!strcasecmp(args.field, "remote_lostpackets")) {
snprintf(buf, buflen, "%u", stats.txploss);
} else if (!strcasecmp(args.field, "remote_jitter")) {
snprintf(buf, buflen, "%u", stats.txjitter);
} else if (!strcasecmp(args.field, "remote_count")) {
snprintf(buf, buflen, "%u", stats.txcount);
} else if (!strcasecmp(args.field, "rtt")) {
snprintf(buf, buflen, "%u", stats.rtt);
} else {
ast_log(LOG_WARNING, "Unrecognized argument '%s' to %s\n", preparse, funcname);
return -1;
}
}
} else {
res = -1;
}
return res;
}
/*! \brief Handle incoming BYE request */
static int handle_request_bye(struct sip_pvt *p, struct sip_request *req)
{
struct ast_channel *c=NULL;
int res;
struct ast_channel *bridged_to;
/* If we have an INCOMING invite that we haven't answered, terminate that transaction */
if (p->pendinginvite && !ast_test_flag(&p->flags[0], SIP_OUTGOING) && !req->ignore) {
transmit_response_reliable(p, "487 Request Terminated", &p->initreq);
}
__sip_pretend_ack(p);
p->invitestate = INV_TERMINATED;
copy_request(&p->initreq, req);
if (sipdebug)
ast_debug(1, "Initializing initreq for method %s - callid %s\n", sip_methods[req->method].text, p->callid);
check_via(p, req);
sip_alreadygone(p);
/* Get RTCP quality before end of call */
if (p->do_history || p->owner) {
char quality_buf[AST_MAX_USER_FIELD], *quality;
struct ast_channel *bridge = p->owner ? ast_bridged_channel(p->owner) : NULL;
if (p->rtp && (quality = ast_rtp_instance_get_quality(p->rtp, AST_RTP_INSTANCE_STAT_FIELD_QUALITY, quality_buf, sizeof(quality_buf)))) {
if (p->do_history) {
append_history(p, "RTCPaudio", "Quality:%s", quality);
if ((quality = ast_rtp_instance_get_quality(p->rtp, AST_RTP_INSTANCE_STAT_FIELD_QUALITY_JITTER, quality_buf, sizeof(quality_buf)))) {
append_history(p, "RTCPaudioJitter", "Quality:%s", quality);
}
if ((quality = ast_rtp_instance_get_quality(p->rtp, AST_RTP_INSTANCE_STAT_FIELD_QUALITY_LOSS, quality_buf, sizeof(quality_buf)))) {
append_history(p, "RTCPaudioLoss", "Quality:%s", quality);
}
if ((quality = ast_rtp_instance_get_quality(p->rtp, AST_RTP_INSTANCE_STAT_FIELD_QUALITY_RTT, quality_buf, sizeof(quality_buf)))) {
append_history(p, "RTCPaudioRTT", "Quality:%s", quality);
}
}
if (p->owner) {
ast_rtp_instance_set_stats_vars(p->owner, p->rtp);
}
}
if (bridge) {
struct sip_pvt *q = bridge->tech_pvt;
if (IS_SIP_TECH(bridge->tech) && q && q->rtp) {
ast_rtp_instance_set_stats_vars(bridge, q->rtp);
}
}
if (p->vrtp && (quality = ast_rtp_instance_get_quality(p->vrtp, AST_RTP_INSTANCE_STAT_FIELD_QUALITY, quality_buf, sizeof(quality_buf)))) {
if (p->do_history) {
append_history(p, "RTCPvideo", "Quality:%s", quality);
}
if (p->owner) {
pbx_builtin_setvar_helper(p->owner, "RTPVIDEOQOS", quality);
}
}
if (p->trtp && (quality = ast_rtp_instance_get_quality(p->trtp, AST_RTP_INSTANCE_STAT_FIELD_QUALITY, quality_buf, sizeof(quality_buf)))) {
if (p->do_history) {
append_history(p, "RTCPtext", "Quality:%s", quality);
}
if (p->owner) {
pbx_builtin_setvar_helper(p->owner, "RTPTEXTQOS", quality);
}
}
}
stop_media_flows(p); /* Immediately stop RTP, VRTP and UDPTL as applicable */
stop_session_timer(p); /* Stop Session-Timer */
if (!ast_strlen_zero(get_header(req, "Also"))) {
ast_log(LOG_NOTICE, "Client '%s' using deprecated BYE/Also transfer method. Ask vendor to support REFER instead\n",
ast_inet_ntoa(p->recv.sin_addr));
if (ast_strlen_zero(p->context))
ast_string_field_set(p, context, sip_cfg.default_context);
res = get_also_info(p, req);
if (!res) {
c = p->owner;
if (c) {
bridged_to = ast_bridged_channel(c);
if (bridged_to) {
/* Don't actually hangup here... */
ast_queue_control(c, AST_CONTROL_UNHOLD);
ast_async_goto(bridged_to, p->context, p->refer->refer_to, 1);
} else
ast_queue_hangup(p->owner);
}
} else {
ast_log(LOG_WARNING, "Invalid transfer information from '%s'\n", ast_inet_ntoa(p->recv.sin_addr));
if (p->owner)
ast_queue_hangup_with_cause(p->owner, AST_CAUSE_PROTOCOL_ERROR);
}
} else if (p->owner) {
ast_queue_hangup(p->owner);
ast_debug(3, "Received bye, issuing owner hangup\n");
} else {
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
ast_debug(3, "Received bye, no owner, selfdestruct soon.\n");
}
ast_clear_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
transmit_response(p, "200 OK", req);
return 1;
}
/*! \brief Handle incoming MESSAGE request */
static int handle_request_message(struct sip_pvt *p, struct sip_request *req)
{
if (!req->ignore) {
if (req->debug)
ast_verbose("Receiving message!\n");
receive_message(p, req);
} else
transmit_response(p, "202 Accepted", req);
return 1;
}
static void add_peer_mwi_subs(struct sip_peer *peer)
{
struct sip_mailbox *mailbox;
AST_LIST_TRAVERSE(&peer->mailboxes, mailbox, entry) {
mailbox->event_sub = ast_event_subscribe(AST_EVENT_MWI, mwi_event_cb, peer,
AST_EVENT_IE_MAILBOX, AST_EVENT_IE_PLTYPE_STR, mailbox->mailbox,
AST_EVENT_IE_CONTEXT, AST_EVENT_IE_PLTYPE_STR, S_OR(mailbox->context, "default"),
AST_EVENT_IE_END);
}
}
/*! \brief Handle incoming SUBSCRIBE request */
static int handle_request_subscribe(struct sip_pvt *p, struct sip_request *req, struct sockaddr_in *sin, int seqno, char *e)
{
int gotdest = 0;
int res = 0;
int firststate = AST_EXTENSION_REMOVED;
struct sip_peer *authpeer = NULL;
const char *eventheader = get_header(req, "Event"); /* Get Event package name */
const char *acceptheader = get_header(req, "Accept");
int resubscribe = (p->subscribed != NONE);
char *temp, *event;
struct ao2_iterator i;
if (p->initreq.headers) {
/* We already have a dialog */
if (p->initreq.method != SIP_SUBSCRIBE) {
/* This is a SUBSCRIBE within another SIP dialog, which we do not support */
/* For transfers, this could happen, but since we haven't seen it happening, let us just refuse this */
transmit_response(p, "403 Forbidden (within dialog)", req);
/* Do not destroy session, since we will break the call if we do */
ast_debug(1, "Got a subscription within the context of another call, can't handle that - %s (Method %s)\n", p->callid, sip_methods[p->initreq.method].text);
return 0;
} else if (req->debug) {
if (resubscribe)
ast_debug(1, "Got a re-subscribe on existing subscription %s\n", p->callid);
else
ast_debug(1, "Got a new subscription %s (possibly with auth)\n", p->callid);
}
}
/* Check if we have a global disallow setting on subscriptions.
if so, we don't have to check peer settings after auth, which saves a lot of processing
*/
if (!sip_cfg.allowsubscribe) {
transmit_response(p, "403 Forbidden (policy)", req);
pvt_set_needdestroy(p, "forbidden");
return 0;
}
if (!req->ignore && !resubscribe) { /* Set up dialog, new subscription */
const char *to = get_header(req, "To");
char totag[128];
/* Check to see if a tag was provided, if so this is actually a resubscription of a dialog we no longer know about */
if (!ast_strlen_zero(to) && gettag(req, "To", totag, sizeof(totag))) {
if (req->debug)
ast_verbose("Received resubscription for a dialog we no longer know about. Telling remote side to subscribe again.\n");
transmit_response(p, "481 Subscription does not exist", req);
pvt_set_needdestroy(p, "subscription does not exist");
return 0;
}
/* Use this as the basis */
if (req->debug)
ast_verbose("Creating new subscription\n");
copy_request(&p->initreq, req);
if (sipdebug)
ast_debug(4, "Initializing initreq for method %s - callid %s\n", sip_methods[req->method].text, p->callid);
check_via(p, req);
} else if (req->debug && req->ignore)
ast_verbose("Ignoring this SUBSCRIBE request\n");
/* Find parameters to Event: header value and remove them for now */
if (ast_strlen_zero(eventheader)) {
transmit_response(p, "489 Bad Event", req);
ast_debug(2, "Received SIP subscribe for unknown event package: <none>\n");
pvt_set_needdestroy(p, "unknown event package in subscribe");
return 0;
}
if ( (strchr(eventheader, ';'))) {
event = ast_strdupa(eventheader); /* Since eventheader is a const, we can't change it */
temp = strchr(event, ';');
*temp = '\0'; /* Remove any options for now */
/* We might need to use them later :-) */
} else
event = (char *) eventheader; /* XXX is this legal ? */
/* Handle authentication */
res = check_user_full(p, req, SIP_SUBSCRIBE, e, 0, sin, &authpeer);
/* if an authentication response was sent, we are done here */
if (res == AUTH_CHALLENGE_SENT) /* authpeer = NULL here */
return 0;
if (res < 0) {
if (res == AUTH_FAKE_AUTH) {
ast_log(LOG_NOTICE, "Sending fake auth rejection for device %s\n", get_header(req, "From"));
transmit_fake_auth_response(p, SIP_SUBSCRIBE, req, XMIT_UNRELIABLE);
} else {
ast_log(LOG_NOTICE, "Failed to authenticate device %s for SUBSCRIBE\n", get_header(req, "From"));
transmit_response_reliable(p, "403 Forbidden", req);
}
pvt_set_needdestroy(p, "authentication failed");
return 0;
}
/* At this point, authpeer cannot be NULL. Remember we hold a reference,
* so we must release it when done.
* XXX must remove all the checks for authpeer == NULL.
*/
/* Check if this device is allowed to subscribe at all */
if (!ast_test_flag(&p->flags[1], SIP_PAGE2_ALLOWSUBSCRIBE)) {
transmit_response(p, "403 Forbidden (policy)", req);
pvt_set_needdestroy(p, "subscription not allowed");
if (authpeer)
unref_peer(authpeer, "unref_peer, from handle_request_subscribe (authpeer 1)");
return 0;
}
if (strcmp(event, "message-summary")) {
/* Get destination right away */
gotdest = get_destination(p, NULL);
}
/* Get full contact header - this needs to be used as a request URI in NOTIFY's */
parse_ok_contact(p, req);
build_contact(p);
if (gotdest) {
transmit_response(p, "404 Not Found", req);
pvt_set_needdestroy(p, "subscription target not found");
if (authpeer)
unref_peer(authpeer, "unref_peer, from handle_request_subscribe (authpeer 2)");
return 0;
}
/* Initialize tag for new subscriptions */
if (ast_strlen_zero(p->tag))
make_our_tag(p->tag, sizeof(p->tag));
if (!strcmp(event, "presence") || !strcmp(event, "dialog")) { /* Presence, RFC 3842 */
if (authpeer) /* We do not need the authpeer any more */
unref_peer(authpeer, "unref_peer, from handle_request_subscribe (authpeer 2)");
/* Header from Xten Eye-beam Accept: multipart/related, application/rlmi+xml, application/pidf+xml, application/xpidf+xml */
/* Polycom phones only handle xpidf+xml, even if they say they can
handle pidf+xml as well
*/
if (strstr(p->useragent, "Polycom")) {
p->subscribed = XPIDF_XML;
} else if (strstr(acceptheader, "application/pidf+xml")) {
p->subscribed = PIDF_XML; /* RFC 3863 format */
} else if (strstr(acceptheader, "application/dialog-info+xml")) {
p->subscribed = DIALOG_INFO_XML;
/* IETF draft: draft-ietf-sipping-dialog-package-05.txt */
} else if (strstr(acceptheader, "application/cpim-pidf+xml")) {
p->subscribed = CPIM_PIDF_XML; /* RFC 3863 format */
} else if (strstr(acceptheader, "application/xpidf+xml")) {
p->subscribed = XPIDF_XML; /* Early pre-RFC 3863 format with MSN additions (Microsoft Messenger) */
} else if (ast_strlen_zero(acceptheader)) {
if (p->subscribed == NONE) { /* if the subscribed field is not already set, and there is no accept header... */
transmit_response(p, "489 Bad Event", req);
ast_log(LOG_WARNING, "SUBSCRIBE failure: no Accept header: pvt: stateid: %d, laststate: %d, dialogver: %d, subscribecont: '%s', subscribeuri: '%s'\n",
p->stateid, p->laststate, p->dialogver, p->subscribecontext, p->subscribeuri);
pvt_set_needdestroy(p, "no Accept header");
return 0;
}
/* if p->subscribed is non-zero, then accept is not obligatory; according to rfc 3265 section 3.1.3, at least.
so, we'll just let it ride, keeping the value from a previous subscription, and not abort the subscription */
} else {
/* Can't find a format for events that we know about */
char mybuf[200];
snprintf(mybuf, sizeof(mybuf), "489 Bad Event (format %s)", acceptheader);
transmit_response(p, mybuf, req);
ast_log(LOG_WARNING, "SUBSCRIBE failure: unrecognized format: '%s' pvt: subscribed: %d, stateid: %d, laststate: %d, dialogver: %d, subscribecont: '%s', subscribeuri: '%s'\n",
acceptheader, (int)p->subscribed, p->stateid, p->laststate, p->dialogver, p->subscribecontext, p->subscribeuri);
pvt_set_needdestroy(p, "unrecognized format");
return 0;
}
} else if (!strcmp(event, "message-summary")) {
if (!ast_strlen_zero(acceptheader) && strcmp(acceptheader, "application/simple-message-summary")) {
/* Format requested that we do not support */
transmit_response(p, "406 Not Acceptable", req);
ast_debug(2, "Received SIP mailbox subscription for unknown format: %s\n", acceptheader);
pvt_set_needdestroy(p, "unknown format");
if (authpeer)
unref_peer(authpeer, "unref_peer, from handle_request_subscribe (authpeer 3)");
return 0;
}
/* Looks like they actually want a mailbox status
This version of Asterisk supports mailbox subscriptions
The subscribed URI needs to exist in the dial plan
In most devices, this is configurable to the voicemailmain extension you use
*/
if (!authpeer || AST_LIST_EMPTY(&authpeer->mailboxes)) {
transmit_response(p, "404 Not found (no mailbox)", req);
pvt_set_needdestroy(p, "received 404 response");
ast_log(LOG_NOTICE, "Received SIP subscribe for peer without mailbox: %s\n", authpeer->name);
if (authpeer)
unref_peer(authpeer, "unref_peer, from handle_request_subscribe (authpeer 4)");
return 0;
}
p->subscribed = MWI_NOTIFICATION;
if (ast_test_flag(&authpeer->flags[1], SIP_PAGE2_SUBSCRIBEMWIONLY)) {
add_peer_mwi_subs(authpeer);
}
if (authpeer->mwipvt && authpeer->mwipvt != p) { /* Destroy old PVT if this is a new one */
/* We only allow one subscription per peer */
dialog_unlink_all(authpeer->mwipvt, TRUE, TRUE);
authpeer->mwipvt = dialog_unref(authpeer->mwipvt, "unref dialog authpeer->mwipvt");
/* sip_destroy(authpeer->mwipvt); */
}
if (authpeer->mwipvt)
dialog_unref(authpeer->mwipvt, "Unref previously stored mwipvt dialog pointer");
authpeer->mwipvt = dialog_ref(p, "setting peers' mwipvt to p"); /* Link from peer to pvt UH- should this be dialog_ref()? */
if (p->relatedpeer)
unref_peer(p->relatedpeer, "Unref previously stored relatedpeer ptr");
p->relatedpeer = ref_peer(authpeer, "setting dialog's relatedpeer pointer"); /* already refcounted...Link from pvt to peer UH- should this be dialog_ref()? */
/* Do not release authpeer here */
} else { /* At this point, Asterisk does not understand the specified event */
transmit_response(p, "489 Bad Event", req);
ast_debug(2, "Received SIP subscribe for unknown event package: %s\n", event);
pvt_set_needdestroy(p, "unknown event package");
if (authpeer)
unref_peer(authpeer, "unref_peer, from handle_request_subscribe (authpeer 5)");
return 0;
}
/* Add subscription for extension state from the PBX core */
if (p->subscribed != MWI_NOTIFICATION && !resubscribe) {
if (p->stateid > -1) {
ast_extension_state_del(p->stateid, cb_extensionstate);
/* we need to dec the refcount, now that the extensionstate is removed */
dialog_unref(p, "the extensionstate containing this dialog ptr was deleted");
}
p->stateid = ast_extension_state_add(p->context, p->exten, cb_extensionstate, dialog_ref(p,"copying dialog ptr into extension state struct"));
}
if (!req->ignore && p)
p->lastinvite = seqno;
if (p && !p->needdestroy) {
p->expiry = atoi(get_header(req, "Expires"));
/* check if the requested expiry-time is within the approved limits from sip.conf */
if (p->expiry > max_expiry)
p->expiry = max_expiry;
if (p->expiry < min_expiry && p->expiry > 0)
p->expiry = min_expiry;
if (sipdebug) {
if (p->subscribed == MWI_NOTIFICATION && p->relatedpeer)
ast_debug(2, "Adding subscription for mailbox notification - peer %s\n", p->relatedpeer->name);
else
ast_debug(2, "Adding subscription for extension %s context %s for peer %s\n", p->exten, p->context, p->username);
}
if (p->autokillid > -1 && sip_cancel_destroy(p)) /* Remove subscription expiry for renewals */
ast_log(LOG_WARNING, "Unable to cancel SIP destruction. Expect bad things.\n");
if (p->expiry > 0)
sip_scheddestroy(p, (p->expiry + 10) * 1000); /* Set timer for destruction of call at expiration */
if (p->subscribed == MWI_NOTIFICATION) {
ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
transmit_response(p, "200 OK", req);
if (p->relatedpeer) { /* Send first notification */
ao2_lock(p->relatedpeer); /* was WRLOCK */
sip_send_mwi_to_peer(p->relatedpeer, NULL, 0);
ao2_unlock(p->relatedpeer);
}
} else {
struct sip_pvt *p_old;
if ((firststate = ast_extension_state(NULL, p->context, p->exten)) < 0) {
ast_log(LOG_NOTICE, "Got SUBSCRIBE for extension %s@%s from %s, but there is no hint for that extension.\n", p->exten, p->context, ast_inet_ntoa(p->sa.sin_addr));
transmit_response(p, "404 Not found", req);
pvt_set_needdestroy(p, "no extension for SUBSCRIBE");
return 0;
}
ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
transmit_response(p, "200 OK", req);
transmit_state_notify(p, firststate, 1, FALSE); /* Send first notification */
append_history(p, "Subscribestatus", "%s", ast_extension_state2str(firststate));
/* hide the 'complete' exten/context in the refer_to field for later display */
ast_string_field_build(p, subscribeuri, "%s@%s", p->exten, p->context);
/* remove any old subscription from this peer for the same exten/context,
as the peer has obviously forgotten about it and it's wasteful to wait
for it to expire and send NOTIFY messages to the peer only to have them
ignored (or generate errors)
*/
i = ao2_iterator_init(dialogs, 0);
while ((p_old = ao2_t_iterator_next(&i, "iterate thru dialogs"))) {
if (p_old == p) {
ao2_t_ref(p_old, -1, "toss dialog ptr from iterator_next before continue");
continue;
}
if (p_old->initreq.method != SIP_SUBSCRIBE) {
ao2_t_ref(p_old, -1, "toss dialog ptr from iterator_next before continue");
continue;
}
if (p_old->subscribed == NONE) {
ao2_t_ref(p_old, -1, "toss dialog ptr from iterator_next before continue");
continue;
}
sip_pvt_lock(p_old);
if (!strcmp(p_old->username, p->username)) {
if (!strcmp(p_old->exten, p->exten) &&
!strcmp(p_old->context, p->context)) {
pvt_set_needdestroy(p_old, "replacing subscription");
sip_pvt_unlock(p_old);
ao2_t_ref(p_old, -1, "toss dialog ptr from iterator_next before break");
break;
}
}
sip_pvt_unlock(p_old);
ao2_t_ref(p_old, -1, "toss dialog ptr from iterator_next");
}
}
if (!p->expiry) {
pvt_set_needdestroy(p, "forcing expiration");
}
}
return 1;
}
/*! \brief Handle incoming REGISTER request */
static int handle_request_register(struct sip_pvt *p, struct sip_request *req, struct sockaddr_in *sin, char *e)
{
enum check_auth_result res;
/* Use this as the basis */
copy_request(&p->initreq, req);
if (sipdebug)
ast_debug(4, "Initializing initreq for method %s - callid %s\n", sip_methods[req->method].text, p->callid);
check_via(p, req);
if ((res = register_verify(p, sin, req, e)) < 0) {
const char *reason;
switch (res) {
case AUTH_SECRET_FAILED:
reason = "Wrong password";
break;
case AUTH_USERNAME_MISMATCH:
reason = "Username/auth name mismatch";
break;
case AUTH_NOT_FOUND:
reason = "No matching peer found";
break;
case AUTH_UNKNOWN_DOMAIN:
reason = "Not a local domain";
break;
case AUTH_PEER_NOT_DYNAMIC:
reason = "Peer is not supposed to register";
break;
case AUTH_ACL_FAILED:
reason = "Device does not match ACL";
break;
case AUTH_BAD_TRANSPORT:
reason = "Device not configured to use this transport type";
break;
default:
reason = "Unknown failure";
break;
}
ast_log(LOG_NOTICE, "Registration from '%s' failed for '%s' - %s\n",
get_header(req, "To"), ast_inet_ntoa(sin->sin_addr),
reason);
append_history(p, "RegRequest", "Failed : Account %s : %s", get_header(req, "To"), reason);
} else
append_history(p, "RegRequest", "Succeeded : Account %s", get_header(req, "To"));
if (res < 1) {
/* Destroy the session, but keep us around for just a bit in case they don't
get our 200 OK */
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
}
return res;
}
/*! \brief Handle incoming SIP requests (methods)
\note This is where all incoming requests go first */
/* called with p and p->owner locked */
static int handle_incoming(struct sip_pvt *p, struct sip_request *req, struct sockaddr_in *sin, int *recount, int *nounlock)
{
/* Called with p->lock held, as well as p->owner->lock if appropriate, keeping things
relatively static */
const char *cmd;
const char *cseq;
const char *useragent;
int seqno;
int len;
int respid;
int res = 0;
int debug = sip_debug_test_pvt(p);
char *e;
int error = 0;
/* Get Method and Cseq */
cseq = get_header(req, "Cseq");
cmd = REQ_OFFSET_TO_STR(req, header[0]);
/* Must have Cseq */
if (ast_strlen_zero(cmd) || ast_strlen_zero(cseq)) {
ast_log(LOG_ERROR, "Missing Cseq. Dropping this SIP message, it's incomplete.\n");
error = 1;
}
if (!error && sscanf(cseq, "%d%n", &seqno, &len) != 1) {
ast_log(LOG_ERROR, "No seqno in '%s'. Dropping incomplete message.\n", cmd);
error = 1;
}
if (error) {
if (!p->initreq.headers) { /* New call */
pvt_set_needdestroy(p, "no headers");
}
return -1;
}
/* Get the command XXX */
cmd = REQ_OFFSET_TO_STR(req, rlPart1);
e = REQ_OFFSET_TO_STR(req, rlPart2);
/* Save useragent of the client */
useragent = get_header(req, "User-Agent");
if (!ast_strlen_zero(useragent))
ast_string_field_set(p, useragent, useragent);
/* Find out SIP method for incoming request */
if (req->method == SIP_RESPONSE) { /* Response to our request */
/* When we get here, we know this is a SIP dialog where we've sent
* a request and have a response, or at least get a response
* within an existing dialog. Do some sanity checks, then
* possibly process the request. In all cases, there function
* terminates at the end of this block
*/
int ret = 0;
if (p->ocseq < seqno && p->lastinvite != seqno && p->lastnoninvite != seqno) {
ast_debug(1, "Ignoring out of order response %d (expecting %d)\n", seqno, p->ocseq);
ret = -1;
} else if (p->ocseq != seqno && p->lastinvite != seqno && p->lastnoninvite != seqno) {
/* ignore means "don't do anything with it" but still have to
* respond appropriately.
* But in this case this is a response already, so we really
* have nothing to do with this message, and even setting the
* ignore flag is pointless.
*/
req->ignore = 1;
append_history(p, "Ignore", "Ignoring this retransmit\n");
} else if (e) {
e = ast_skip_blanks(e);
if (sscanf(e, "%d %n", &respid, &len) != 1) {
ast_log(LOG_WARNING, "Invalid response: '%s'\n", e);
/* XXX maybe should do ret = -1; */
} else if (respid <= 0) {
ast_log(LOG_WARNING, "Invalid SIP response code: '%d'\n", respid);
/* XXX maybe should do ret = -1; */
} else { /* finally, something worth processing */
/* More SIP ridiculousness, we have to ignore bogus contacts in 100 etc responses */
if ((respid == 200) || ((respid >= 300) && (respid <= 399)))
extract_uri(p, req);
handle_response(p, respid, e + len, req, seqno);
}
}
return 0;
}
/* New SIP request coming in
(could be new request in existing SIP dialog as well...)
*/
p->method = req->method; /* Find out which SIP method they are using */
ast_debug(4, "**** Received %s (%d) - Command in SIP %s\n", sip_methods[p->method].text, sip_methods[p->method].id, cmd);
if (p->icseq && (p->icseq > seqno) ) {
if (p->pendinginvite && seqno == p->pendinginvite && (req->method == SIP_ACK || req->method == SIP_CANCEL)) {
ast_debug(2, "Got CANCEL or ACK on INVITE with transactions in between.\n");
} else {
ast_debug(1, "Ignoring too old SIP packet packet %d (expecting >= %d)\n", seqno, p->icseq);
if (req->method != SIP_ACK)
transmit_response(p, "503 Server error", req); /* We must respond according to RFC 3261 sec 12.2 */
return -1;
}
} else if (p->icseq &&
p->icseq == seqno &&
req->method != SIP_ACK &&
(p->method != SIP_CANCEL || p->alreadygone)) {
/* ignore means "don't do anything with it" but still have to
respond appropriately. We do this if we receive a repeat of
the last sequence number */
req->ignore = 1;
ast_debug(3, "Ignoring SIP message because of retransmit (%s Seqno %d, ours %d)\n", sip_methods[p->method].text, p->icseq, seqno);
}
if (seqno >= p->icseq)
/* Next should follow monotonically (but not necessarily
incrementally -- thanks again to the genius authors of SIP --
increasing */
p->icseq = seqno;
/* Find their tag if we haven't got it */
if (ast_strlen_zero(p->theirtag)) {
char tag[128];
gettag(req, "From", tag, sizeof(tag));
ast_string_field_set(p, theirtag, tag);
}
snprintf(p->lastmsg, sizeof(p->lastmsg), "Rx: %s", cmd);
if (sip_cfg.pedanticsipchecking) {
/* If this is a request packet without a from tag, it's not
correct according to RFC 3261 */
/* Check if this a new request in a new dialog with a totag already attached to it,
RFC 3261 - section 12.2 - and we don't want to mess with recovery */
if (!p->initreq.headers && req->has_to_tag) {
/* If this is a first request and it got a to-tag, it is not for us */
if (!req->ignore && req->method == SIP_INVITE) {
transmit_response_reliable(p, "481 Call/Transaction Does Not Exist", req);
/* Will cease to exist after ACK */
} else if (req->method != SIP_ACK) {
transmit_response(p, "481 Call/Transaction Does Not Exist", req);
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
} else {
ast_debug(1, "Got ACK for unknown dialog... strange.\n");
}
return res;
}
}
if (!e && (p->method == SIP_INVITE || p->method == SIP_SUBSCRIBE || p->method == SIP_REGISTER || p->method == SIP_NOTIFY)) {
transmit_response(p, "400 Bad request", req);
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
return -1;
}
/* Handle various incoming SIP methods in requests */
switch (p->method) {
case SIP_OPTIONS:
res = handle_request_options(p, req);
break;
case SIP_INVITE:
res = handle_request_invite(p, req, debug, seqno, sin, recount, e, nounlock);
break;
case SIP_REFER:
res = handle_request_refer(p, req, debug, seqno, nounlock);
break;
case SIP_CANCEL:
res = handle_request_cancel(p, req);
break;
case SIP_BYE:
res = handle_request_bye(p, req);
break;
case SIP_MESSAGE:
res = handle_request_message(p, req);
break;
case SIP_SUBSCRIBE:
res = handle_request_subscribe(p, req, sin, seqno, e);
break;
case SIP_REGISTER:
res = handle_request_register(p, req, sin, e);
break;
case SIP_INFO:
if (req->debug)
ast_verbose("Receiving INFO!\n");
if (!req->ignore)
handle_request_info(p, req);
else /* if ignoring, transmit response */
transmit_response(p, "200 OK", req);
break;
case SIP_NOTIFY:
res = handle_request_notify(p, req, sin, seqno, e);
break;
case SIP_ACK:
/* Make sure we don't ignore this */
if (seqno == p->pendinginvite) {
p->invitestate = INV_TERMINATED;
p->pendinginvite = 0;
__sip_ack(p, seqno, 1 /* response */, 0);
if (find_sdp(req)) {
if (process_sdp(p, req, SDP_T38_NONE))
return -1;
}
check_pendings(p);
} else if (p->glareinvite == seqno) {
/* handle ack for the 491 pending sent for glareinvite */
p->glareinvite = 0;
__sip_ack(p, seqno, 1, 0);
}
/* Got an ACK that we did not match. Ignore silently */
if (!p->lastinvite && ast_strlen_zero(p->randdata)) {
pvt_set_needdestroy(p, "unmatched ACK");
}
break;
default:
transmit_response_with_allow(p, "501 Method Not Implemented", req, 0);
ast_log(LOG_NOTICE, "Unknown SIP command '%s' from '%s'\n",
cmd, ast_inet_ntoa(p->sa.sin_addr));
/* If this is some new method, and we don't have a call, destroy it now */
if (!p->initreq.headers) {
pvt_set_needdestroy(p, "unimplemented method");
}
break;
}
return res;
}
static void process_request_queue(struct sip_pvt *p, int *recount, int *nounlock)
{
struct sip_request *req;
while ((req = AST_LIST_REMOVE_HEAD(&p->request_queue, next))) {
if (handle_incoming(p, req, &p->recv, recount, nounlock) == -1) {
/* Request failed */
if (option_debug) {
ast_log(LOG_DEBUG, "SIP message could not be handled, bad request: %-70.70s\n", p->callid[0] ? p->callid : "<no callid>");
}
}
ast_free(req);
}
}
static int scheduler_process_request_queue(const void *data)
{
struct sip_pvt *p = (struct sip_pvt *) data;
int recount = 0;
int nounlock = 0;
int lockretry;
for (lockretry = 10; lockretry > 0; lockretry--) {
sip_pvt_lock(p);
/* lock the owner if it has one -- we may need it */
/* because this is deadlock-prone, we need to try and unlock if failed */
if (!p->owner || !ast_channel_trylock(p->owner)) {
break; /* locking succeeded */
}
if (lockretry != 1) {
sip_pvt_unlock(p);
/* Sleep for a very short amount of time */
usleep(1);
}
}
if (!lockretry) {
int retry = !AST_LIST_EMPTY(&p->request_queue);
/* we couldn't get the owner lock, which is needed to process
the queued requests, so return a non-zero value, which will
cause the scheduler to run this request again later if there
still requests to be processed
*/
sip_pvt_unlock(p);
if (!retry) {
dialog_unref(p, "The ref to a dialog passed to this sched callback is going out of scope; unref it.");
}
return retry;
};
process_request_queue(p, &recount, &nounlock);
p->request_queue_sched_id = -1;
if (p->owner && !nounlock) {
ast_channel_unlock(p->owner);
}
sip_pvt_unlock(p);
if (recount) {
ast_update_use_count();
}
dialog_unref(p, "The ref to a dialog passed to this sched callback is going out of scope; unref it.");
return 0;
}
static int queue_request(struct sip_pvt *p, const struct sip_request *req)
{
struct sip_request *newreq;
if (!(newreq = ast_calloc(1, sizeof(*newreq)))) {
return -1;
}
copy_request(newreq, req);
AST_LIST_INSERT_TAIL(&p->request_queue, newreq, next);
if (p->request_queue_sched_id == -1) {
if ((p->request_queue_sched_id = ast_sched_add(sched, 10, scheduler_process_request_queue, dialog_ref(p, "Increment refcount to pass dialog pointer to sched callback"))) == -1) {
dialog_unref(p, "Decrement refcount due to sched_add failure");
}
}
return 0;
}
/*! \brief Read data from SIP UDP socket
\note sipsock_read locks the owner channel while we are processing the SIP message
\return 1 on error, 0 on success
\note Successful messages is connected to SIP call and forwarded to handle_incoming()
*/
static int sipsock_read(int *id, int fd, short events, void *ignore)
{
struct sip_request req;
struct sockaddr_in sin = { 0, };
int res;
socklen_t len = sizeof(sin);
static char readbuf[65535];
memset(&req, 0, sizeof(req));
res = recvfrom(fd, readbuf, sizeof(readbuf) - 1, 0, (struct sockaddr *)&sin, &len);
if (res < 0) {
#if !defined(__FreeBSD__)
if (errno == EAGAIN)
ast_log(LOG_NOTICE, "SIP: Received packet with bad UDP checksum\n");
else
#endif
if (errno != ECONNREFUSED)
ast_log(LOG_WARNING, "Recv error: %s\n", strerror(errno));
return 1;
}
readbuf[res] = '\0';
if (!(req.data = ast_str_create(SIP_MIN_PACKET))) {
return 1;
}
if (ast_str_set(&req.data, 0, "%s", readbuf) == AST_DYNSTR_BUILD_FAILED) {
return -1;
}
req.len = res;
req.socket.fd = sipsock;
req.socket.type = SIP_TRANSPORT_UDP;
req.socket.tcptls_session = NULL;
req.socket.port = bindaddr.sin_port;
handle_request_do(&req, &sin);
if (req.data) {
ast_free(req.data);
req.data = NULL;
}
return 1;
}
/*! \brief Handle incoming SIP message - request or response
This is used for all transports (udp, tcp and tcp/tls)
*/
static int handle_request_do(struct sip_request *req, struct sockaddr_in *sin)
{
struct sip_pvt *p;
int recount = 0;
int nounlock = 0;
int lockretry;
if (sip_debug_test_addr(sin)) /* Set the debug flag early on packet level */
req->debug = 1;
if (sip_cfg.pedanticsipchecking)
req->len = lws2sws(req->data->str, req->len); /* Fix multiline headers */
if (req->debug) {
ast_verbose("\n<--- SIP read from %s:%s:%d --->\n%s\n<------------->\n",
get_transport(req->socket.type), ast_inet_ntoa(sin->sin_addr),
ntohs(sin->sin_port), req->data->str);
}
if (parse_request(req) == -1) { /* Bad packet, can't parse */
ast_str_reset(req->data); /* nulling this out is NOT a good idea here. */
return 1;
}
req->method = find_sip_method(REQ_OFFSET_TO_STR(req, rlPart1));
if (req->debug)
ast_verbose("--- (%d headers %d lines)%s ---\n", req->headers, req->lines, (req->headers + req->lines == 0) ? " Nat keepalive" : "");
if (req->headers < 2) { /* Must have at least two headers */
ast_str_reset(req->data); /* nulling this out is NOT a good idea here. */
return 1;
}
/* Process request, with netlock held, and with usual deadlock avoidance */
for (lockretry = 10; lockretry > 0; lockretry--) {
ast_mutex_lock(&netlock);
/* Find the active SIP dialog or create a new one */
p = find_call(req, sin, req->method); /* returns p locked */
if (p == NULL) {
ast_debug(1, "Invalid SIP message - rejected , no callid, len %d\n", req->len);
ast_mutex_unlock(&netlock);
return 1;
}
copy_socket_data(&p->socket, &req->socket);
/* Go ahead and lock the owner if it has one -- we may need it */
/* becaues this is deadlock-prone, we need to try and unlock if failed */
if (!p->owner || !ast_channel_trylock(p->owner))
break; /* locking succeeded */
if (lockretry != 1) {
sip_pvt_unlock(p);
ao2_t_ref(p, -1, "release p (from find_call) inside lockretry loop"); /* we'll look for it again, but p is dead now */
ast_mutex_unlock(&netlock);
/* Sleep for a very short amount of time */
usleep(1);
}
}
p->recv = *sin;
if (p->do_history) /* This is a request or response, note what it was for */
append_history(p, "Rx", "%s / %s / %s", req->data->str, get_header(req, "CSeq"), REQ_OFFSET_TO_STR(req, rlPart2));
if (!lockretry) {
if (!queue_request(p, req)) {
/* the request has been queued for later handling */
sip_pvt_unlock(p);
ao2_t_ref(p, -1, "release p (from find_call) after queueing request");
ast_mutex_unlock(&netlock);
return 1;
}
if (p->owner)
ast_log(LOG_ERROR, "Channel lock for %s could not be obtained, and request was unable to be queued.\n", S_OR(p->owner->name, "- no channel name ??? - "));
ast_log(LOG_ERROR, "SIP transaction failed: %s \n", p->callid);
if (req->method != SIP_ACK)
transmit_response(p, "503 Server error", req); /* We must respond according to RFC 3261 sec 12.2 */
/* XXX We could add retry-after to make sure they come back */
append_history(p, "LockFail", "Owner lock failed, transaction failed.");
sip_pvt_unlock(p);
ao2_t_ref(p, -1, "release p (from find_call) at end of lockretry"); /* p is gone after the return */
ast_mutex_unlock(&netlock);
return 1;
}
/* if there are queued requests on this sip_pvt, process them first, so that everything is
handled in order
*/
if (!AST_LIST_EMPTY(&p->request_queue)) {
AST_SCHED_DEL_UNREF(sched, p->request_queue_sched_id, dialog_unref(p, "when you delete the request_queue_sched_id sched, you should dec the refcount for the stored dialog ptr"));
process_request_queue(p, &recount, &nounlock);
}
if (handle_incoming(p, req, sin, &recount, &nounlock) == -1) {
/* Request failed */
ast_debug(1, "SIP message could not be handled, bad request: %-70.70s\n", p->callid[0] ? p->callid : "<no callid>");
}
if (recount)
ast_update_use_count();
if (p->owner && !nounlock)
ast_channel_unlock(p->owner);
sip_pvt_unlock(p);
ast_mutex_unlock(&netlock);
ao2_t_ref(p, -1, "throw away dialog ptr from find_call at end of routine"); /* p is gone after the return */
return 1;
}
/*! \brief Returns the port to use for this socket
*
* \param type The type of transport used
* \param port Port we are checking to see if it's the standard port.
* \note port is expected in host byte order
*/
static int sip_standard_port(enum sip_transport type, int port)
{
if (type & SIP_TRANSPORT_TLS)
return port == STANDARD_TLS_PORT;
else
return port == STANDARD_SIP_PORT;
}
/*!
* \brief Find thread for TCP/TLS session (based on IP/Port
*
* \note This function returns an astobj2 reference
*/
static struct ast_tcptls_session_instance *sip_tcp_locate(struct sockaddr_in *s)
{
struct sip_threadinfo *th;
struct ast_tcptls_session_instance *tcptls_instance = NULL;
AST_LIST_LOCK(&threadl);
AST_LIST_TRAVERSE(&threadl, th, list) {
if ((s->sin_family == th->tcptls_session->remote_address.sin_family) &&
(s->sin_addr.s_addr == th->tcptls_session->remote_address.sin_addr.s_addr) &&
(s->sin_port == th->tcptls_session->remote_address.sin_port)) {
tcptls_instance = (ao2_ref(th->tcptls_session, +1), th->tcptls_session);
break;
}
}
AST_LIST_UNLOCK(&threadl);
return tcptls_instance;
}
/*! \todo Get socket for dialog, prepare if needed, and return file handle */
static int sip_prepare_socket(struct sip_pvt *p)
{
struct sip_socket *s = &p->socket;
static const char name[] = "SIP socket";
struct ast_tcptls_session_instance *tcptls_session;
struct ast_tcptls_session_args ca = {
.name = name,
.accept_fd = -1,
};
if (s->fd != -1)
return s->fd; /* This socket is already active */
/*! \todo Check this... This might be wrong, depending on the proxy configuration
If proxy is in "force" mode its correct.
*/
if (p->outboundproxy && p->outboundproxy->transport) {
s->type = p->outboundproxy->transport;
}
if (s->type & SIP_TRANSPORT_UDP) {
s->fd = sipsock;
return s->fd;
}
ca.remote_address = *(sip_real_dst(p));
if ((tcptls_session = sip_tcp_locate(&ca.remote_address))) { /* Check if we have a thread handling a socket connected to this IP/port */
s->fd = tcptls_session->fd;
if (s->tcptls_session) {
ao2_ref(s->tcptls_session, -1);
s->tcptls_session = NULL;
}
s->tcptls_session = tcptls_session;
return s->fd;
}
if (s->tcptls_session && s->tcptls_session->parent->tls_cfg) {
ca.tls_cfg = s->tcptls_session->parent->tls_cfg;
} else {
if (s->type & SIP_TRANSPORT_TLS) {
ca.tls_cfg = ast_calloc(1, sizeof(*ca.tls_cfg));
if (!ca.tls_cfg)
return -1;
memcpy(ca.tls_cfg, &default_tls_cfg, sizeof(*ca.tls_cfg));
if (!ast_strlen_zero(p->tohost))
ast_copy_string(ca.hostname, p->tohost, sizeof(ca.hostname));
}
}
if (s->tcptls_session) {
/* the pvt socket already has a server instance ... */
} else {
s->tcptls_session = ast_tcptls_client_start(&ca); /* Start a client connection to this address */
}
if (!s->tcptls_session) {
if (ca.tls_cfg)
ast_free(ca.tls_cfg);
return -1;
}
s->fd = ca.accept_fd;
/* Give the new thread a reference */
ao2_ref(s->tcptls_session, +1);
if (ast_pthread_create_background(&ca.master, NULL, sip_tcp_worker_fn, s->tcptls_session)) {
ast_debug(1, "Unable to launch '%s'.", ca.name);
ao2_ref(s->tcptls_session, -1);
close(ca.accept_fd);
s->fd = ca.accept_fd = -1;
}
return s->fd;
}
/*!
* \brief Small function to parse a config line for a host with a transport
* i.e. tls://www.google.com:8056
*/
static int sip_parse_host(char *line, int lineno, char **hostname, int *portnum, enum sip_transport *transport)
{
char *port;
if ((*hostname = strstr(line, "://"))) {
*hostname += 3;
if (!strncasecmp(line, "tcp", 3))
*transport = SIP_TRANSPORT_TCP;
else if (!strncasecmp(line, "tls", 3))
*transport = SIP_TRANSPORT_TLS;
else if (!strncasecmp(line, "udp", 3))
*transport = SIP_TRANSPORT_UDP;
else
ast_log(LOG_NOTICE, "'%.3s' is not a valid transport type on line %d of sip.conf. defaulting to udp.\n", line, lineno);
} else {
*hostname = line;
*transport = SIP_TRANSPORT_UDP;
}
if ((line = strrchr(*hostname, '@')))
line++;
else
line = *hostname;
if ((port = strrchr(line, ':'))) {
*port++ = '\0';
if (!sscanf(port, "%u", portnum)) {
ast_log(LOG_NOTICE, "'%s' is not a valid port number on line %d of sip.conf. using default.\n", port, lineno);
port = NULL;
}
}
if (!port) {
if (*transport & SIP_TRANSPORT_TLS) {
*portnum = STANDARD_TLS_PORT;
} else {
*portnum = STANDARD_SIP_PORT;
}
}
return 0;
}
/*!
* \brief Get cached MWI info
* \retval 0 At least one message is waiting
* \retval 1 no messages waiting
*/
static int get_cached_mwi(struct sip_peer *peer, int *new, int *old)
{
struct sip_mailbox *mailbox;
AST_LIST_TRAVERSE(&peer->mailboxes, mailbox, entry) {
struct ast_event *event;
event = ast_event_get_cached(AST_EVENT_MWI,
AST_EVENT_IE_MAILBOX, AST_EVENT_IE_PLTYPE_STR, mailbox->mailbox,
AST_EVENT_IE_CONTEXT, AST_EVENT_IE_PLTYPE_STR, S_OR(mailbox->context, "default"),
AST_EVENT_IE_END);
if (!event)
continue;
*new += ast_event_get_ie_uint(event, AST_EVENT_IE_NEWMSGS);
*old += ast_event_get_ie_uint(event, AST_EVENT_IE_OLDMSGS);
ast_event_destroy(event);
}
return (*new || *old) ? 0 : 1;
}
/*! \brief Send message waiting indication to alert peer that they've got voicemail */
static int sip_send_mwi_to_peer(struct sip_peer *peer, const struct ast_event *event, int cache_only)
{
/* Called with peerl lock, but releases it */
struct sip_pvt *p;
int newmsgs = 0, oldmsgs = 0;
if (ast_test_flag((&peer->flags[1]), SIP_PAGE2_SUBSCRIBEMWIONLY) && !peer->mwipvt)
return 0;
/* Do we have an IP address? If not, skip this peer */
if (!peer->addr.sin_addr.s_addr && !peer->defaddr.sin_addr.s_addr)
return 0;
if (event) {
newmsgs = ast_event_get_ie_uint(event, AST_EVENT_IE_NEWMSGS);
oldmsgs = ast_event_get_ie_uint(event, AST_EVENT_IE_OLDMSGS);
} else if (!get_cached_mwi(peer, &newmsgs, &oldmsgs)) {
/* got it! Don't keep looking. */
} else if (cache_only) {
return 0;
} else { /* Fall back to manually checking the mailbox */
struct ast_str *mailbox_str = ast_str_alloca(512);
peer_mailboxes_to_str(&mailbox_str, peer);
ast_app_inboxcount(mailbox_str->str, &newmsgs, &oldmsgs);
}
if (peer->mwipvt) {
/* Base message on subscription */
p = dialog_ref(peer->mwipvt, "sip_send_mwi_to_peer: Setting dialog ptr p from peer->mwipvt-- should this be done?");
} else {
/* Build temporary dialog for this message */
if (!(p = sip_alloc(NULL, NULL, 0, SIP_NOTIFY)))
return -1;
/* If we don't set the socket type to 0, then create_addr_from_peer will fail immediately if the peer
* uses any transport other than UDP. We set the type to 0 here and then let create_addr_from_peer copy
* the peer's socket information to the sip_pvt we just allocated
*/
p->socket.type = 0;
if (create_addr_from_peer(p, peer)) {
/* Maybe they're not registered, etc. */
dialog_unlink_all(p, TRUE, TRUE);
dialog_unref(p, "unref dialog p just created via sip_alloc");
/* sip_destroy(p); */
return 0;
}
/* Recalculate our side, and recalculate Call ID */
ast_sip_ouraddrfor(&p->sa.sin_addr, &p->ourip);
build_via(p);
ao2_t_unlink(dialogs, p, "About to change the callid -- remove the old name");
build_callid_pvt(p);
if (!ast_strlen_zero(peer->mwi_from)) {
ast_string_field_set(p, mwi_from, peer->mwi_from);
} else if (!ast_strlen_zero(default_mwi_from)) {
ast_string_field_set(p, mwi_from, default_mwi_from);
}
ao2_t_link(dialogs, p, "Linking in under new name");
/* Destroy this session after 32 secs */
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
}
/* Send MWI */
ast_set_flag(&p->flags[0], SIP_OUTGOING);
/* the following will decrement the refcount on p as it finishes */
transmit_notify_with_mwi(p, newmsgs, oldmsgs, peer->vmexten);
dialog_unref(p, "unref dialog ptr p just before it goes out of scope at the end of sip_send_mwi_to_peer.");
return 0;
}
/*! \brief helper function for the monitoring thread -- seems to be called with the assumption that the dialog is locked */
static void check_rtp_timeout(struct sip_pvt *dialog, time_t t)
{
/* If we have no RTP or no active owner, no need to check timers */
if (!dialog->rtp || !dialog->owner)
return;
/* If the call is not in UP state or redirected outside Asterisk, no need to check timers */
if (dialog->owner->_state != AST_STATE_UP || dialog->redirip.sin_addr.s_addr)
return;
/* If the call is involved in a T38 fax session do not check RTP timeout */
if (dialog->t38.state == T38_ENABLED)
return;
/* If we have no timers set, return now */
if (!ast_rtp_instance_get_timeout(dialog->rtp) && !ast_rtp_instance_get_hold_timeout(dialog->rtp)) {
return;
}
/*! \todo Check video RTP keepalives
Do we need to move the lastrtptx to the RTP structure to have one for audio and one
for video? It really does belong to the RTP structure.
*/
/* Check AUDIO RTP timers */
if (dialog->lastrtprx && (ast_rtp_instance_get_timeout(dialog->rtp) || ast_rtp_instance_get_hold_timeout(dialog->rtp)) && (t > dialog->lastrtprx + ast_rtp_instance_get_timeout(dialog->rtp))) {
if (!ast_test_flag(&dialog->flags[1], SIP_PAGE2_CALL_ONHOLD) || (ast_rtp_instance_get_hold_timeout(dialog->rtp) && (t > dialog->lastrtprx + ast_rtp_instance_get_hold_timeout(dialog->rtp)))) {
/* Needs a hangup */
if (ast_rtp_instance_get_timeout(dialog->rtp)) {
while (dialog->owner && ast_channel_trylock(dialog->owner)) {
sip_pvt_unlock(dialog);
usleep(1);
sip_pvt_lock(dialog);
}
ast_log(LOG_NOTICE, "Disconnecting call '%s' for lack of RTP activity in %ld seconds\n",
dialog->owner->name, (long) (t - dialog->lastrtprx));
/* Issue a softhangup */
ast_softhangup_nolock(dialog->owner, AST_SOFTHANGUP_DEV);
ast_channel_unlock(dialog->owner);
/* forget the timeouts for this call, since a hangup
has already been requested and we don't want to
repeatedly request hangups
*/
ast_rtp_instance_set_timeout(dialog->rtp, 0);
ast_rtp_instance_set_hold_timeout(dialog->rtp, 0);
if (dialog->vrtp) {
ast_rtp_instance_set_timeout(dialog->vrtp, 0);
ast_rtp_instance_set_hold_timeout(dialog->vrtp, 0);
}
}
}
}
}
/*! \brief The SIP monitoring thread
\note This thread monitors all the SIP sessions and peers that needs notification of mwi
(and thus do not have a separate thread) indefinitely
*/
static void *do_monitor(void *data)
{
int res;
time_t t;
int reloading;
/* Add an I/O event to our SIP UDP socket */
if (sipsock > -1)
sipsock_read_id = ast_io_add(io, sipsock, sipsock_read, AST_IO_IN, NULL);
/* From here on out, we die whenever asked */
for(;;) {
/* Check for a reload request */
ast_mutex_lock(&sip_reload_lock);
reloading = sip_reloading;
sip_reloading = FALSE;
ast_mutex_unlock(&sip_reload_lock);
if (reloading) {
ast_verb(1, "Reloading SIP\n");
sip_do_reload(sip_reloadreason);
/* Change the I/O fd of our UDP socket */
if (sipsock > -1) {
if (sipsock_read_id)
sipsock_read_id = ast_io_change(io, sipsock_read_id, sipsock, NULL, 0, NULL);
else
sipsock_read_id = ast_io_add(io, sipsock, sipsock_read, AST_IO_IN, NULL);
} else if (sipsock_read_id) {
ast_io_remove(io, sipsock_read_id);
sipsock_read_id = NULL;
}
}
/* Check for dialogs needing to be killed */
t = time(NULL);
/* don't scan the dialogs list if it hasn't been a reasonable period
of time since the last time we did it (when MWI is being sent, we can
get back to this point every millisecond or less)
*/
ao2_t_callback(dialogs, OBJ_UNLINK | OBJ_NODATA | OBJ_MULTIPLE, dialog_needdestroy, &t,
"callback to remove dialogs w/needdestroy");
/* the old methodology would be to restart the search for dialogs to delete with every
dialog that was found and destroyed, probably because the list contents would change,
so we'd need to restart. This isn't the best thing to do with callbacks. */
/* XXX TODO The scheduler usage in this module does not have sufficient
* synchronization being done between running the scheduler and places
* scheduling tasks. As it is written, any scheduled item may not run
* any sooner than about 1 second, regardless of whether a sooner time
* was asked for. */
pthread_testcancel();
/* Wait for sched or io */
res = ast_sched_wait(sched);
if ((res < 0) || (res > 1000))
res = 1000;
res = ast_io_wait(io, res);
if (res > 20)
ast_debug(1, "chan_sip: ast_io_wait ran %d all at once\n", res);
ast_mutex_lock(&monlock);
res = ast_sched_runq(sched);
if (res >= 20)
ast_debug(1, "chan_sip: ast_sched_runq ran %d all at once\n", res);
ast_mutex_unlock(&monlock);
}
/* Never reached */
return NULL;
}
/*! \brief Start the channel monitor thread */
static int restart_monitor(void)
{
/* If we're supposed to be stopped -- stay stopped */
if (monitor_thread == AST_PTHREADT_STOP)
return 0;
ast_mutex_lock(&monlock);
if (monitor_thread == pthread_self()) {
ast_mutex_unlock(&monlock);
ast_log(LOG_WARNING, "Cannot kill myself\n");
return -1;
}
if (monitor_thread != AST_PTHREADT_NULL) {
/* Wake up the thread */
pthread_kill(monitor_thread, SIGURG);
} else {
/* Start a new monitor */
if (ast_pthread_create_background(&monitor_thread, NULL, do_monitor, NULL) < 0) {
ast_mutex_unlock(&monlock);
ast_log(LOG_ERROR, "Unable to start monitor thread.\n");
return -1;
}
}
ast_mutex_unlock(&monlock);
return 0;
}
/*! \brief Session-Timers: Restart session timer */
static void restart_session_timer(struct sip_pvt *p)
{
if (!p->stimer) {
ast_log(LOG_WARNING, "Null stimer in restart_session_timer - %s\n", p->callid);
return;
}
if (p->stimer->st_active == TRUE) {
AST_SCHED_DEL_UNREF(sched, p->stimer->st_schedid,
dialog_unref(p, "Removing session timer ref"));
ast_debug(2, "Session timer stopped: %d - %s\n", p->stimer->st_schedid, p->callid);
start_session_timer(p);
}
}
/*! \brief Session-Timers: Stop session timer */
static void stop_session_timer(struct sip_pvt *p)
{
if (!p->stimer) {
ast_log(LOG_WARNING, "Null stimer in stop_session_timer - %s\n", p->callid);
return;
}
if (p->stimer->st_active == TRUE) {
p->stimer->st_active = FALSE;
AST_SCHED_DEL_UNREF(sched, p->stimer->st_schedid,
dialog_unref(p, "removing session timer ref"));
ast_debug(2, "Session timer stopped: %d - %s\n", p->stimer->st_schedid, p->callid);
}
}
/*! \brief Session-Timers: Start session timer */
static void start_session_timer(struct sip_pvt *p)
{
if (!p->stimer) {
ast_log(LOG_WARNING, "Null stimer in start_session_timer - %s\n", p->callid);
return;
}
p->stimer->st_schedid = ast_sched_add(sched, p->stimer->st_interval * 1000 / 2, proc_session_timer,
dialog_ref(p, "adding session timer ref"));
if (p->stimer->st_schedid < 0) {
dialog_unref(p, "removing session timer ref");
ast_log(LOG_ERROR, "ast_sched_add failed.\n");
}
ast_debug(2, "Session timer started: %d - %s\n", p->stimer->st_schedid, p->callid);
}
/*! \brief Session-Timers: Process session refresh timeout event */
static int proc_session_timer(const void *vp)
{
struct sip_pvt *p = (struct sip_pvt *) vp;
int sendreinv = FALSE;
int res = 0;
if (!p->stimer) {
ast_log(LOG_WARNING, "Null stimer in proc_session_timer - %s\n", p->callid);
goto return_unref;
}
ast_debug(2, "Session timer expired: %d - %s\n", p->stimer->st_schedid, p->callid);
if (!p->owner) {
goto return_unref;
}
if ((p->stimer->st_active != TRUE) || (p->owner->_state != AST_STATE_UP)) {
goto return_unref;
}
switch (p->stimer->st_ref) {
case SESSION_TIMER_REFRESHER_UAC:
if (p->outgoing_call == TRUE) {
sendreinv = TRUE;
}
break;
case SESSION_TIMER_REFRESHER_UAS:
if (p->outgoing_call != TRUE) {
sendreinv = TRUE;
}
break;
default:
ast_log(LOG_ERROR, "Unknown session refresher %d\n", p->stimer->st_ref);
goto return_unref;
}
if (sendreinv == TRUE) {
res = 1;
transmit_reinvite_with_sdp(p, FALSE, TRUE);
} else {
p->stimer->st_expirys++;
if (p->stimer->st_expirys >= 2) {
ast_log(LOG_WARNING, "Session-Timer expired - %s\n", p->callid);
while (p->owner && ast_channel_trylock(p->owner)) {
sip_pvt_unlock(p);
usleep(1);
sip_pvt_lock(p);
}
ast_softhangup_nolock(p->owner, AST_SOFTHANGUP_DEV);
ast_channel_unlock(p->owner);
}
}
return_unref:
if (!res) {
/* An error occurred. Stop session timer processing */
p->stimer->st_schedid = -1;
stop_session_timer(p);
/* If we are not asking to be rescheduled, then we need to release our
* reference to the dialog. */
dialog_unref(p, "removing session timer ref");
}
return res;
}
/*! \brief Session-Timers: Function for parsing Min-SE header */
int parse_minse (const char *p_hdrval, int *const p_interval)
{
if (ast_strlen_zero(p_hdrval)) {
ast_log(LOG_WARNING, "Null Min-SE header\n");
return -1;
}
*p_interval = 0;
p_hdrval = ast_skip_blanks(p_hdrval);
if (!sscanf(p_hdrval, "%d", p_interval)) {
ast_log(LOG_WARNING, "Parsing of Min-SE header failed %s\n", p_hdrval);
return -1;
}
ast_debug(2, "Received Min-SE: %d\n", *p_interval);
return 0;
}
/*! \brief Session-Timers: Function for parsing Session-Expires header */
int parse_session_expires(const char *p_hdrval, int *const p_interval, enum st_refresher *const p_ref)
{
char *p_token;
int ref_idx;
char *p_se_hdr;
if (ast_strlen_zero(p_hdrval)) {
ast_log(LOG_WARNING, "Null Session-Expires header\n");
return -1;
}
*p_ref = SESSION_TIMER_REFRESHER_AUTO;
*p_interval = 0;
p_se_hdr = ast_strdupa(p_hdrval);
p_se_hdr = ast_skip_blanks(p_se_hdr);
while ((p_token = strsep(&p_se_hdr, ";"))) {
p_token = ast_skip_blanks(p_token);
if (!sscanf(p_token, "%d", p_interval)) {
ast_log(LOG_WARNING, "Parsing of Session-Expires failed\n");
return -1;
}
ast_debug(2, "Session-Expires: %d\n", *p_interval);
if (!p_se_hdr)
continue;
ref_idx = strlen("refresher=");
if (!strncasecmp(p_se_hdr, "refresher=", ref_idx)) {
p_se_hdr += ref_idx;
p_se_hdr = ast_skip_blanks(p_se_hdr);
if (!strncasecmp(p_se_hdr, "uac", strlen("uac"))) {
*p_ref = SESSION_TIMER_REFRESHER_UAC;
ast_debug(2, "Refresher: UAC\n");
} else if (!strncasecmp(p_se_hdr, "uas", strlen("uas"))) {
*p_ref = SESSION_TIMER_REFRESHER_UAS;
ast_debug(2, "Refresher: UAS\n");
} else {
ast_log(LOG_WARNING, "Invalid refresher value %s\n", p_se_hdr);
return -1;
}
break;
}
}
return 0;
}
/*! \brief Handle 422 response to INVITE with session-timer requested
Session-Timers: An INVITE originated by Asterisk that asks for session-timers support
from the UAS can result into a 422 response. This is how a UAS or an intermediary proxy
server tells Asterisk that the session refresh interval offered by Asterisk is too low
for them. The proc_422_rsp() function handles a 422 response. It extracts the Min-SE
header that comes back in 422 and sends a new INVITE accordingly. */
static void proc_422_rsp(struct sip_pvt *p, struct sip_request *rsp)
{
int rtn;
const char *p_hdrval;
int minse;
p_hdrval = get_header(rsp, "Min-SE");
if (ast_strlen_zero(p_hdrval)) {
ast_log(LOG_WARNING, "422 response without a Min-SE header %s\n", p_hdrval);
return;
}
rtn = parse_minse(p_hdrval, &minse);
if (rtn != 0) {
ast_log(LOG_WARNING, "Parsing of Min-SE header failed %s\n", p_hdrval);
return;
}
p->stimer->st_interval = minse;
transmit_invite(p, SIP_INVITE, 1, 2);
}
/*! \brief Get Max or Min SE (session timer expiry)
* \param p pointer to the SIP dialog
* \param max if true, get max se, otherwise min se
*/
int st_get_se(struct sip_pvt *p, int max)
{
if (max == TRUE) {
if (p->stimer->st_cached_max_se) {
return p->stimer->st_cached_max_se;
} else if (p->peername) {
struct sip_peer *pp = find_peer(p->peername, NULL, TRUE, FINDPEERS, FALSE);
if (pp) {
p->stimer->st_cached_max_se = pp->stimer.st_max_se;
unref_peer(pp, "unref peer pointer from find_peer call in st_get_se");
return (p->stimer->st_cached_max_se);
}
}
p->stimer->st_cached_max_se = global_max_se;
return (p->stimer->st_cached_max_se);
} else {
if (p->stimer->st_cached_min_se) {
return p->stimer->st_cached_min_se;
} else if (p->peername) {
struct sip_peer *pp = find_peer(p->peername, NULL, TRUE, FINDPEERS, FALSE);
if (pp) {
p->stimer->st_cached_min_se = pp->stimer.st_min_se;
unref_peer(pp, "unref peer pointer from find_peer call in st_get_se (2)");
return (p->stimer->st_cached_min_se);
}
}
p->stimer->st_cached_min_se = global_min_se;
return (p->stimer->st_cached_min_se);
}
}
/*! \brief Get the entity (UAC or UAS) that's acting as the session-timer refresher
* \param p pointer to the SIP dialog
*/
enum st_refresher st_get_refresher(struct sip_pvt *p)
{
if (p->stimer->st_cached_ref != SESSION_TIMER_REFRESHER_AUTO)
return p->stimer->st_cached_ref;
if (p->peername) {
struct sip_peer *pp = find_peer(p->peername, NULL, TRUE, FINDPEERS, FALSE);
if (pp) {
p->stimer->st_cached_ref = pp->stimer.st_ref;
unref_peer(pp, "unref peer pointer from find_peer call in st_get_refresher");
return pp->stimer.st_ref;
}
}
p->stimer->st_cached_ref = global_st_refresher;
return global_st_refresher;
}
/*! \brief Get the session-timer mode
* \param p pointer to the SIP dialog
*/
enum st_mode st_get_mode(struct sip_pvt *p)
{
if (!p->stimer)
sip_st_alloc(p);
if (p->stimer->st_cached_mode != SESSION_TIMER_MODE_INVALID)
return p->stimer->st_cached_mode;
if (p->peername) {
struct sip_peer *pp = find_peer(p->peername, NULL, TRUE, FINDPEERS, FALSE);
if (pp) {
p->stimer->st_cached_mode = pp->stimer.st_mode_oper;
unref_peer(pp, "unref peer pointer from find_peer call in st_get_mode");
return pp->stimer.st_mode_oper;
}
}
p->stimer->st_cached_mode = global_st_mode;
return global_st_mode;
}
/*! \brief React to lack of answer to Qualify poke */
static int sip_poke_noanswer(const void *data)
{
struct sip_peer *peer = (struct sip_peer *)data;
peer->pokeexpire = -1;
if (peer->lastms > -1) {
ast_log(LOG_NOTICE, "Peer '%s' is now UNREACHABLE! Last qualify: %d\n", peer->name, peer->lastms);
if (sip_cfg.peer_rtupdate) {
ast_update_realtime(ast_check_realtime("sipregs") ? "sipregs" : "sippeers", "name", peer->name, "lastms", "-1", SENTINEL);
}
manager_event(EVENT_FLAG_SYSTEM, "PeerStatus", "ChannelType: SIP\r\nPeer: SIP/%s\r\nPeerStatus: Unreachable\r\nTime: %d\r\n", peer->name, -1);
if (sip_cfg.regextenonqualify) {
register_peer_exten(peer, FALSE);
}
}
if (peer->call) {
dialog_unlink_all(peer->call, TRUE, TRUE);
peer->call = dialog_unref(peer->call, "unref dialog peer->call");
/* peer->call = sip_destroy(peer->call);*/
}
peer->lastms = -1;
ast_devstate_changed(AST_DEVICE_UNKNOWN, "SIP/%s", peer->name);
/* Try again quickly */
AST_SCHED_REPLACE_UNREF(peer->pokeexpire, sched,
DEFAULT_FREQ_NOTOK, sip_poke_peer_s, peer,
unref_peer(_data, "removing poke peer ref"),
unref_peer(peer, "removing poke peer ref"),
ref_peer(peer, "adding poke peer ref"));
/* Release the ref held by the running scheduler entry */
unref_peer(peer, "release peer poke noanswer ref");
return 0;
}
/*! \brief Check availability of peer, also keep NAT open
\note This is done with 60 seconds between each ping,
unless forced by cli or manager. If peer is unreachable,
we check every 10th second by default.
*/
static int sip_poke_peer(struct sip_peer *peer, int force)
{
struct sip_pvt *p;
int xmitres = 0;
if ((!peer->maxms && !force) || !peer->addr.sin_addr.s_addr) {
/* IF we have no IP, or this isn't to be monitored, return
immediately after clearing things out */
AST_SCHED_DEL_UNREF(sched, peer->pokeexpire,
unref_peer(peer, "removing poke peer ref"));
peer->lastms = 0;
if (peer->call) {
peer->call = dialog_unref(peer->call, "unref dialog peer->call");
}
return 0;
}
if (peer->call) {
if (sipdebug) {
ast_log(LOG_NOTICE, "Still have a QUALIFY dialog active, deleting\n");
}
dialog_unlink_all(peer->call, TRUE, TRUE);
peer->call = dialog_unref(peer->call, "unref dialog peer->call");
/* peer->call = sip_destroy(peer->call); */
}
if (!(p = sip_alloc(NULL, NULL, 0, SIP_OPTIONS))) {
return -1;
}
peer->call = dialog_ref(p, "copy sip alloc from p to peer->call");
p->sa = peer->addr;
p->recv = peer->addr;
copy_socket_data(&p->socket, &peer->socket);
ast_copy_flags(&p->flags[0], &peer->flags[0], SIP_FLAGS_TO_COPY);
ast_copy_flags(&p->flags[1], &peer->flags[1], SIP_PAGE2_FLAGS_TO_COPY);
/* Send OPTIONs to peer's fullcontact */
if (!ast_strlen_zero(peer->fullcontact))
ast_string_field_set(p, fullcontact, peer->fullcontact);
if (!ast_strlen_zero(peer->tohost))
ast_string_field_set(p, tohost, peer->tohost);
else
ast_string_field_set(p, tohost, ast_inet_ntoa(peer->addr.sin_addr));
/* Recalculate our side, and recalculate Call ID */
ast_sip_ouraddrfor(&p->sa.sin_addr, &p->ourip);
build_via(p);
ao2_t_unlink(dialogs, p, "About to change the callid -- remove the old name");
build_callid_pvt(p);
ao2_t_link(dialogs, p, "Linking in under new name");
AST_SCHED_DEL_UNREF(sched, peer->pokeexpire,
unref_peer(peer, "removing poke peer ref"));
if (p->relatedpeer)
p->relatedpeer = unref_peer(p->relatedpeer,"unsetting the relatedpeer field in the dialog, before it is set to something else.");
p->relatedpeer = ref_peer(peer, "setting the relatedpeer field in the dialog");
ast_set_flag(&p->flags[0], SIP_OUTGOING);
#ifdef VOCAL_DATA_HACK
ast_copy_string(p->username, "__VOCAL_DATA_SHOULD_READ_THE_SIP_SPEC__", sizeof(p->username));
xmitres = transmit_invite(p, SIP_INVITE, 0, 2); /* sinks the p refcount */
#else
xmitres = transmit_invite(p, SIP_OPTIONS, 0, 2); /* sinks the p refcount */
#endif
peer->ps = ast_tvnow();
if (xmitres == XMIT_ERROR) {
sip_poke_noanswer(peer); /* Immediately unreachable, network problems */
} else if (!force) {
AST_SCHED_REPLACE_UNREF(peer->pokeexpire, sched, peer->maxms * 2, sip_poke_noanswer, peer,
unref_peer(_data, "removing poke peer ref"),
unref_peer(peer, "removing poke peer ref"),
ref_peer(peer, "adding poke peer ref"));
}
dialog_unref(p, "unref dialog at end of sip_poke_peer, obtained from sip_alloc, just before it goes out of scope");
return 0;
}
/*! \brief Part of PBX channel interface
\note
\par Return values:---
If we have qualify on and the device is not reachable, regardless of registration
state we return AST_DEVICE_UNAVAILABLE
For peers with call limit:
- not registered AST_DEVICE_UNAVAILABLE
- registered, no call AST_DEVICE_NOT_INUSE
- registered, active calls AST_DEVICE_INUSE
- registered, call limit reached AST_DEVICE_BUSY
- registered, onhold AST_DEVICE_ONHOLD
- registered, ringing AST_DEVICE_RINGING
For peers without call limit:
- not registered AST_DEVICE_UNAVAILABLE
- registered AST_DEVICE_NOT_INUSE
- fixed IP (!dynamic) AST_DEVICE_NOT_INUSE
Peers that does not have a known call and can't be reached by OPTIONS
- unreachable AST_DEVICE_UNAVAILABLE
If we return AST_DEVICE_UNKNOWN, the device state engine will try to find
out a state by walking the channel list.
The queue system (\ref app_queue.c) treats a member as "active"
if devicestate is != AST_DEVICE_UNAVAILBALE && != AST_DEVICE_INVALID
When placing a call to the queue member, queue system sets a member to busy if
!= AST_DEVICE_NOT_INUSE and != AST_DEVICE_UNKNOWN
*/
static int sip_devicestate(void *data)
{
char *host;
char *tmp;
struct sip_peer *p;
int res = AST_DEVICE_INVALID;
/* make sure data is not null. Maybe unnecessary, but better be safe */
host = ast_strdupa(data ? data : "");
if ((tmp = strchr(host, '@')))
host = tmp + 1;
ast_debug(3, "Checking device state for peer %s\n", host);
/* If find_peer asks for a realtime peer, then this breaks rtautoclear. This
* is because when a peer tries to autoexpire, the last thing it does is to
* queue up an event telling the system that the devicestate has changed
* (presumably to unavailable). If we ask for a realtime peer here, this would
* load it BACK into memory, thus defeating the point of trying to clear dead
* hosts out of memory.
*/
if ((p = find_peer(host, NULL, FALSE, FINDALLDEVICES, TRUE))) {
if (p->addr.sin_addr.s_addr || p->defaddr.sin_addr.s_addr) {
/* we have an address for the peer */
/* Check status in this order
- Hold
- Ringing
- Busy (enforced only by call limit)
- Inuse (we have a call)
- Unreachable (qualify)
If we don't find any of these state, report AST_DEVICE_NOT_INUSE
for registered devices */
if (p->onHold)
/* First check for hold or ring states */
res = AST_DEVICE_ONHOLD;
else if (p->inRinging) {
if (p->inRinging == p->inUse)
res = AST_DEVICE_RINGING;
else
res = AST_DEVICE_RINGINUSE;
} else if (p->call_limit && (p->inUse == p->call_limit))
/* check call limit */
res = AST_DEVICE_BUSY;
else if (p->call_limit && p->busy_level && p->inUse >= p->busy_level)
/* We're forcing busy before we've reached the call limit */
res = AST_DEVICE_BUSY;
else if (p->call_limit && p->inUse)
/* Not busy, but we do have a call */
res = AST_DEVICE_INUSE;
else if (p->maxms && ((p->lastms > p->maxms) || (p->lastms < 0)))
/* We don't have a call. Are we reachable at all? Requires qualify= */
res = AST_DEVICE_UNAVAILABLE;
else /* Default reply if we're registered and have no other data */
res = AST_DEVICE_NOT_INUSE;
} else {
/* there is no address, it's unavailable */
res = AST_DEVICE_UNAVAILABLE;
}
unref_peer(p, "unref_peer, from sip_devicestate, release ref from find_peer");
} else {
res = AST_DEVICE_UNKNOWN;
}
return res;
}
/*! \brief PBX interface function -build SIP pvt structure
* SIP calls initiated by the PBX arrive here.
*
* \verbatim
* SIP Dial string syntax
* SIP/exten@host!dnid
* or SIP/host/exten!dnid
* or SIP/host!dnid
* \endverbatim
*/
static struct ast_channel *sip_request_call(const char *type, int format, void *data, int *cause)
{
struct sip_pvt *p;
struct ast_channel *tmpc = NULL;
char *ext = NULL, *host;
char tmp[256];
char *dest = data;
char *dnid;
char *secret = NULL;
char *md5secret = NULL;
char *authname = NULL;
char *trans = NULL;
enum sip_transport transport = 0;
int oldformat = format;
/* mask request with some set of allowed formats.
* XXX this needs to be fixed.
* The original code uses AST_FORMAT_AUDIO_MASK, but it is
* unclear what to use here. We have global_capabilities, which is
* configured from sip.conf, and sip_tech.capabilities, which is
* hardwired to all audio formats.
*/
format &= AST_FORMAT_AUDIO_MASK;
if (!format) {
ast_log(LOG_NOTICE, "Asked to get a channel of unsupported format %s while capability is %s\n", ast_getformatname(oldformat), ast_getformatname(global_capability));
*cause = AST_CAUSE_BEARERCAPABILITY_NOTAVAIL; /* Can't find codec to connect to host */
return NULL;
}
ast_debug(1, "Asked to create a SIP channel with formats: %s\n", ast_getformatname_multiple(tmp, sizeof(tmp), oldformat));
if (!(p = sip_alloc(NULL, NULL, 0, SIP_INVITE))) {
ast_log(LOG_ERROR, "Unable to build sip pvt data for '%s' (Out of memory or socket error)\n", dest);
*cause = AST_CAUSE_SWITCH_CONGESTION;
return NULL;
}
p->outgoing_call = TRUE;
if (!(p->options = ast_calloc(1, sizeof(*p->options)))) {
dialog_unlink_all(p, TRUE, TRUE);
dialog_unref(p, "unref dialog p from mem fail");
/* sip_destroy(p); */
ast_log(LOG_ERROR, "Unable to build option SIP data structure - Out of memory\n");
*cause = AST_CAUSE_SWITCH_CONGESTION;
return NULL;
}
/* Save the destination, the SIP dial string */
ast_copy_string(tmp, dest, sizeof(tmp));
/* Find DNID and take it away */
dnid = strchr(tmp, '!');
if (dnid != NULL) {
*dnid++ = '\0';
ast_string_field_set(p, todnid, dnid);
}
/* Find at sign - @ */
host = strchr(tmp, '@');
if (host) {
*host++ = '\0';
ext = tmp;
secret = strchr(ext, ':');
}
if (secret) {
*secret++ = '\0';
md5secret = strchr(secret, ':');
}
if (md5secret) {
*md5secret++ = '\0';
authname = strchr(md5secret, ':');
}
if (authname) {
*authname++ = '\0';
trans = strchr(authname, ':');
}
if (trans) {
*trans++ = '\0';
if (!strcasecmp(trans, "tcp"))
transport = SIP_TRANSPORT_TCP;
else if (!strcasecmp(trans, "tls"))
transport = SIP_TRANSPORT_TLS;
else {
if (strcasecmp(trans, "udp"))
ast_log(LOG_WARNING, "'%s' is not a valid transport option to Dial() for SIP calls, using udp by default.\n", trans);
transport = SIP_TRANSPORT_UDP;
}
} else { /* use default */
transport = SIP_TRANSPORT_UDP;
}
if (!host) {
ext = strchr(tmp, '/');
if (ext)
*ext++ = '\0';
host = tmp;
}
p->socket.fd = -1;
p->socket.type = transport;
/* We now have
host = peer name, DNS host name or DNS domain (for SRV)
ext = extension (user part of URI)
dnid = destination of the call (applies to the To: header)
*/
if (create_addr(p, host, NULL, 1)) {
*cause = AST_CAUSE_UNREGISTERED;
ast_debug(3, "Cant create SIP call - target device not registered\n");
dialog_unlink_all(p, TRUE, TRUE);
dialog_unref(p, "unref dialog p UNREGISTERED");
/* sip_destroy(p); */
return NULL;
}
if (ast_strlen_zero(p->peername) && ext)
ast_string_field_set(p, peername, ext);
/* Recalculate our side, and recalculate Call ID */
ast_sip_ouraddrfor(&p->sa.sin_addr, &p->ourip);
build_via(p);
ao2_t_unlink(dialogs, p, "About to change the callid -- remove the old name");
build_callid_pvt(p);
ao2_t_link(dialogs, p, "Linking in under new name");
/* We have an extension to call, don't use the full contact here */
/* This to enable dialing registered peers with extension dialling,
like SIP/peername/extension
SIP/peername will still use the full contact
*/
if (ext) {
ast_string_field_set(p, username, ext);
ast_string_field_set(p, fullcontact, NULL);
}
if (secret && !ast_strlen_zero(secret))
ast_string_field_set(p, peersecret, secret);
if (md5secret && !ast_strlen_zero(md5secret))
ast_string_field_set(p, peermd5secret, md5secret);
if (authname && !ast_strlen_zero(authname))
ast_string_field_set(p, authname, authname);
#if 0
printf("Setting up to call extension '%s' at '%s'\n", ext ? ext : "<none>", host);
#endif
p->prefcodec = oldformat; /* Format for this call */
p->jointcapability = oldformat;
sip_pvt_lock(p);
tmpc = sip_new(p, AST_STATE_DOWN, host); /* Place the call */
if (sip_cfg.callevents)
manager_event(EVENT_FLAG_SYSTEM, "ChannelUpdate",
"Channel: %s\r\nChanneltype: %s\r\nSIPcallid: %s\r\nSIPfullcontact: %s\r\nPeername: %s\r\n",
p->owner? p->owner->name : "", "SIP", p->callid, p->fullcontact, p->peername);
sip_pvt_unlock(p);
if (!tmpc) {
dialog_unlink_all(p, TRUE, TRUE);
/* sip_destroy(p); */
}
dialog_unref(p, "toss pvt ptr at end of sip_request_call");
ast_update_use_count();
restart_monitor();
return tmpc;
}
/*! \brief Parse insecure= setting in sip.conf and set flags according to setting */
static void set_insecure_flags (struct ast_flags *flags, const char *value, int lineno)
{
if (ast_strlen_zero(value))
return;
if (!ast_false(value)) {
char buf[64];
char *word, *next;
ast_copy_string(buf, value, sizeof(buf));
next = buf;
while ((word = strsep(&next, ","))) {
if (!strcasecmp(word, "port"))
ast_set_flag(&flags[0], SIP_INSECURE_PORT);
else if (!strcasecmp(word, "invite"))
ast_set_flag(&flags[0], SIP_INSECURE_INVITE);
else
ast_log(LOG_WARNING, "Unknown insecure mode '%s' on line %d\n", value, lineno);
}
}
}
/*!
\brief Handle flag-type options common to configuration of devices - peers
\param flags array of two struct ast_flags
\param mask array of two struct ast_flags
\param v linked list of config variables to process
\returns non-zero if any config options were handled, zero otherwise
*/
static int handle_common_options(struct ast_flags *flags, struct ast_flags *mask, struct ast_variable *v)
{
int res = 1;
if (!strcasecmp(v->name, "trustrpid")) {
ast_set_flag(&mask[0], SIP_TRUSTRPID);
ast_set2_flag(&flags[0], ast_true(v->value), SIP_TRUSTRPID);
} else if (!strcasecmp(v->name, "sendrpid")) {
ast_set_flag(&mask[0], SIP_SENDRPID);
if (!strcasecmp(v->value, "pai")) {
ast_set_flag(&flags[0], SIP_SENDRPID_PAI);
} else if (!strcasecmp(v->value, "rpid")) {
ast_set_flag(&flags[0], SIP_SENDRPID_RPID);
} else if (ast_true(v->value)) {
ast_set_flag(&flags[0], SIP_SENDRPID_RPID);
}
} else if (!strcasecmp(v->name, "rpid_immediate")) {
ast_set_flag(&mask[1], SIP_PAGE2_RPID_IMMEDIATE);
ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_RPID_IMMEDIATE);
} else if (!strcasecmp(v->name, "g726nonstandard")) {
ast_set_flag(&mask[0], SIP_G726_NONSTANDARD);
ast_set2_flag(&flags[0], ast_true(v->value), SIP_G726_NONSTANDARD);
} else if (!strcasecmp(v->name, "useclientcode")) {
ast_set_flag(&mask[0], SIP_USECLIENTCODE);
ast_set2_flag(&flags[0], ast_true(v->value), SIP_USECLIENTCODE);
} else if (!strcasecmp(v->name, "dtmfmode")) {
ast_set_flag(&mask[0], SIP_DTMF);
ast_clear_flag(&flags[0], SIP_DTMF);
if (!strcasecmp(v->value, "inband"))
ast_set_flag(&flags[0], SIP_DTMF_INBAND);
else if (!strcasecmp(v->value, "rfc2833"))
ast_set_flag(&flags[0], SIP_DTMF_RFC2833);
else if (!strcasecmp(v->value, "info"))
ast_set_flag(&flags[0], SIP_DTMF_INFO);
else if (!strcasecmp(v->value, "shortinfo"))
ast_set_flag(&flags[0], SIP_DTMF_SHORTINFO);
else if (!strcasecmp(v->value, "auto"))
ast_set_flag(&flags[0], SIP_DTMF_AUTO);
else {
ast_log(LOG_WARNING, "Unknown dtmf mode '%s' on line %d, using rfc2833\n", v->value, v->lineno);
ast_set_flag(&flags[0], SIP_DTMF_RFC2833);
}
} else if (!strcasecmp(v->name, "nat")) {
ast_set_flag(&mask[0], SIP_NAT);
ast_clear_flag(&flags[0], SIP_NAT);
if (!strcasecmp(v->value, "never"))
ast_set_flag(&flags[0], SIP_NAT_NEVER);
else if (!strcasecmp(v->value, "route"))
ast_set_flag(&flags[0], SIP_NAT_ROUTE);
else if (ast_true(v->value))
ast_set_flag(&flags[0], SIP_NAT_ALWAYS);
else
ast_set_flag(&flags[0], SIP_NAT_RFC3581);
} else if (!strcasecmp(v->name, "canreinvite")) {
ast_set_flag(&mask[0], SIP_REINVITE);
ast_clear_flag(&flags[0], SIP_REINVITE);
if (ast_true(v->value)) {
ast_set_flag(&flags[0], SIP_CAN_REINVITE | SIP_CAN_REINVITE_NAT);
} else if (!ast_false(v->value)) {
char buf[64];
char *word, *next = buf;
ast_copy_string(buf, v->value, sizeof(buf));
while ((word = strsep(&next, ","))) {
if (!strcasecmp(word, "update")) {
ast_set_flag(&flags[0], SIP_REINVITE_UPDATE | SIP_CAN_REINVITE);
} else if (!strcasecmp(word, "nonat")) {
ast_set_flag(&flags[0], SIP_CAN_REINVITE);
ast_clear_flag(&flags[0], SIP_CAN_REINVITE_NAT);
} else {
ast_log(LOG_WARNING, "Unknown canreinvite mode '%s' on line %d\n", v->value, v->lineno);
}
}
}
} else if (!strcasecmp(v->name, "insecure")) {
ast_set_flag(&mask[0], SIP_INSECURE);
ast_clear_flag(&flags[0], SIP_INSECURE);
set_insecure_flags(&flags[0], v->value, v->lineno);
} else if (!strcasecmp(v->name, "progressinband")) {
ast_set_flag(&mask[0], SIP_PROG_INBAND);
ast_clear_flag(&flags[0], SIP_PROG_INBAND);
if (ast_true(v->value))
ast_set_flag(&flags[0], SIP_PROG_INBAND_YES);
else if (strcasecmp(v->value, "never"))
ast_set_flag(&flags[0], SIP_PROG_INBAND_NO);
} else if (!strcasecmp(v->name, "promiscredir")) {
ast_set_flag(&mask[0], SIP_PROMISCREDIR);
ast_set2_flag(&flags[0], ast_true(v->value), SIP_PROMISCREDIR);
} else if (!strcasecmp(v->name, "videosupport")) {
if (!strcasecmp(v->value, "always")) {
ast_set_flag(&mask[1], SIP_PAGE2_VIDEOSUPPORT_ALWAYS);
ast_set_flag(&flags[1], SIP_PAGE2_VIDEOSUPPORT_ALWAYS);
} else {
ast_set_flag(&mask[1], SIP_PAGE2_VIDEOSUPPORT);
ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_VIDEOSUPPORT);
}
} else if (!strcasecmp(v->name, "textsupport")) {
ast_set_flag(&mask[1], SIP_PAGE2_TEXTSUPPORT);
ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_TEXTSUPPORT);
res = 1;
} else if (!strcasecmp(v->name, "allowoverlap")) {
ast_set_flag(&mask[1], SIP_PAGE2_ALLOWOVERLAP);
ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_ALLOWOVERLAP);
} else if (!strcasecmp(v->name, "allowsubscribe")) {
ast_set_flag(&mask[1], SIP_PAGE2_ALLOWSUBSCRIBE);
ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_ALLOWSUBSCRIBE);
} else if (!strcasecmp(v->name, "ignoresdpversion")) {
ast_set_flag(&mask[1], SIP_PAGE2_IGNORESDPVERSION);
ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_IGNORESDPVERSION);
} else if (!strcasecmp(v->name, "faxdetect")) {
ast_set_flag(&mask[1], SIP_PAGE2_FAX_DETECT);
ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_FAX_DETECT);
} else if (!strcasecmp(v->name, "t38pt_udptl")) {
ast_set_flag(&mask[1], SIP_PAGE2_T38SUPPORT_UDPTL);
ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_T38SUPPORT_UDPTL);
#ifdef WHEN_WE_HAVE_T38_FOR_OTHER_TRANSPORTS
} else if (!strcasecmp(v->name, "t38pt_rtp")) {
ast_set_flag(&mask[1], SIP_PAGE2_T38SUPPORT_RTP);
ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_T38SUPPORT_RTP);
} else if (!strcasecmp(v->name, "t38pt_tcp")) {
ast_set_flag(&mask[1], SIP_PAGE2_T38SUPPORT_TCP);
ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_T38SUPPORT_TCP);
#endif
} else if (!strcasecmp(v->name, "rfc2833compensate")) {
ast_set_flag(&mask[1], SIP_PAGE2_RFC2833_COMPENSATE);
ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_RFC2833_COMPENSATE);
} else if (!strcasecmp(v->name, "buggymwi")) {
ast_set_flag(&mask[1], SIP_PAGE2_BUGGY_MWI);
ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_BUGGY_MWI);
} else if (!strcasecmp(v->name, "t38pt_usertpsource")) {
ast_set_flag(&mask[1], SIP_PAGE2_UDPTL_DESTINATION);
ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_UDPTL_DESTINATION);
} else
res = 0;
return res;
}
/*! \brief Add SIP domain to list of domains we are responsible for */
static int add_sip_domain(const char *domain, const enum domain_mode mode, const char *context)
{
struct domain *d;
if (ast_strlen_zero(domain)) {
ast_log(LOG_WARNING, "Zero length domain.\n");
return 1;
}
if (!(d = ast_calloc(1, sizeof(*d))))
return 0;
ast_copy_string(d->domain, domain, sizeof(d->domain));
if (!ast_strlen_zero(context))
ast_copy_string(d->context, context, sizeof(d->context));
d->mode = mode;
AST_LIST_LOCK(&domain_list);
AST_LIST_INSERT_TAIL(&domain_list, d, list);
AST_LIST_UNLOCK(&domain_list);
if (sipdebug)
ast_debug(1, "Added local SIP domain '%s'\n", domain);
return 1;
}
/*! \brief check_sip_domain: Check if domain part of uri is local to our server */
static int check_sip_domain(const char *domain, char *context, size_t len)
{
struct domain *d;
int result = 0;
AST_LIST_LOCK(&domain_list);
AST_LIST_TRAVERSE(&domain_list, d, list) {
if (strcasecmp(d->domain, domain))
continue;
if (len && !ast_strlen_zero(d->context))
ast_copy_string(context, d->context, len);
result = 1;
break;
}
AST_LIST_UNLOCK(&domain_list);
return result;
}
/*! \brief Clear our domain list (at reload) */
static void clear_sip_domains(void)
{
struct domain *d;
AST_LIST_LOCK(&domain_list);
while ((d = AST_LIST_REMOVE_HEAD(&domain_list, list)))
ast_free(d);
AST_LIST_UNLOCK(&domain_list);
}
/*! \brief Add realm authentication in list */
static struct sip_auth *add_realm_authentication(struct sip_auth *authlist, const char *configuration, int lineno)
{
char authcopy[256];
char *username=NULL, *realm=NULL, *secret=NULL, *md5secret=NULL;
char *stringp;
struct sip_auth *a, *b, *auth;
if (ast_strlen_zero(configuration))
return authlist;
ast_debug(1, "Auth config :: %s\n", configuration);
ast_copy_string(authcopy, configuration, sizeof(authcopy));
stringp = authcopy;
username = stringp;
realm = strrchr(stringp, '@');
if (realm)
*realm++ = '\0';
if (ast_strlen_zero(username) || ast_strlen_zero(realm)) {
ast_log(LOG_WARNING, "Format for authentication entry is user[:secret]@realm at line %d\n", lineno);
return authlist;
}
stringp = username;
username = strsep(&stringp, ":");
if (username) {
secret = strsep(&stringp, ":");
if (!secret) {
stringp = username;
md5secret = strsep(&stringp, "#");
}
}
if (!(auth = ast_calloc(1, sizeof(*auth))))
return authlist;
ast_copy_string(auth->realm, realm, sizeof(auth->realm));
ast_copy_string(auth->username, username, sizeof(auth->username));
if (secret)
ast_copy_string(auth->secret, secret, sizeof(auth->secret));
if (md5secret)
ast_copy_string(auth->md5secret, md5secret, sizeof(auth->md5secret));
/* find the end of the list */
for (b = NULL, a = authlist; a ; b = a, a = a->next)
;
if (b)
b->next = auth; /* Add structure add end of list */
else
authlist = auth;
ast_verb(3, "Added authentication for realm %s\n", realm);
return authlist;
}
/*! \brief Clear realm authentication list (at reload) */
static int clear_realm_authentication(struct sip_auth *authlist)
{
struct sip_auth *a = authlist;
struct sip_auth *b;
while (a) {
b = a;
a = a->next;
ast_free(b);
}
return 1;
}
/*! \brief Find authentication for a specific realm */
static struct sip_auth *find_realm_authentication(struct sip_auth *authlist, const char *realm)
{
struct sip_auth *a;
for (a = authlist; a; a = a->next) {
if (!strcasecmp(a->realm, realm))
break;
}
return a;
}
/*! \brief
* implement the setvar config line
*/
static struct ast_variable *add_var(const char *buf, struct ast_variable *list)
{
struct ast_variable *tmpvar = NULL;
char *varname = ast_strdupa(buf), *varval = NULL;
if ((varval = strchr(varname, '='))) {
*varval++ = '\0';
if ((tmpvar = ast_variable_new(varname, varval, ""))) {
tmpvar->next = list;
list = tmpvar;
}
}
return list;
}
/*! \brief Set peer defaults before configuring specific configurations */
static void set_peer_defaults(struct sip_peer *peer)
{
if (peer->expire == 0) {
/* Don't reset expire or port time during reload
if we have an active registration
*/
peer->expire = -1;
peer->pokeexpire = -1;
peer->addr.sin_port = htons(STANDARD_SIP_PORT);
peer->socket.type = SIP_TRANSPORT_UDP;
peer->socket.fd = -1;
}
peer->type = SIP_TYPE_PEER;
ast_copy_flags(&peer->flags[0], &global_flags[0], SIP_FLAGS_TO_COPY);
ast_copy_flags(&peer->flags[1], &global_flags[1], SIP_PAGE2_FLAGS_TO_COPY);
ast_string_field_set(peer, context, sip_cfg.default_context);
ast_string_field_set(peer, subscribecontext, sip_cfg.default_subscribecontext);
ast_string_field_set(peer, language, default_language);
ast_string_field_set(peer, mohinterpret, default_mohinterpret);
ast_string_field_set(peer, mohsuggest, default_mohsuggest);
ast_string_field_set(peer, engine, default_engine);
peer->addr.sin_family = AF_INET;
peer->defaddr.sin_family = AF_INET;
peer->capability = global_capability;
peer->maxcallbitrate = default_maxcallbitrate;
peer->rtptimeout = global_rtptimeout;
peer->rtpholdtimeout = global_rtpholdtimeout;
peer->rtpkeepalive = global_rtpkeepalive;
peer->allowtransfer = sip_cfg.allowtransfer;
peer->autoframing = global_autoframing;
peer->qualifyfreq = global_qualifyfreq;
if (global_callcounter)
peer->call_limit=999;
ast_string_field_set(peer, vmexten, default_vmexten);
ast_string_field_set(peer, secret, "");
ast_string_field_set(peer, remotesecret, "");
ast_string_field_set(peer, md5secret, "");
ast_string_field_set(peer, cid_num, "");
ast_string_field_set(peer, cid_name, "");
ast_string_field_set(peer, fromdomain, "");
ast_string_field_set(peer, fromuser, "");
ast_string_field_set(peer, regexten, "");
peer->callgroup = 0;
peer->pickupgroup = 0;
peer->maxms = default_qualify;
peer->prefs = default_prefs;
peer->stimer.st_mode_oper = global_st_mode; /* Session-Timers */
peer->stimer.st_ref = global_st_refresher;
peer->stimer.st_min_se = global_min_se;
peer->stimer.st_max_se = global_max_se;
peer->timer_t1 = global_t1;
peer->timer_b = global_timer_b;
clear_peer_mailboxes(peer);
}
/*! \brief Create temporary peer (used in autocreatepeer mode) */
static struct sip_peer *temp_peer(const char *name)
{
struct sip_peer *peer;
if (!(peer = ao2_t_alloc(sizeof(*peer), sip_destroy_peer_fn, "allocate a peer struct")))
return NULL;
if (ast_string_field_init(peer, 512)) {
ao2_t_ref(peer, -1, "failed to string_field_init, drop peer");
return NULL;
}
ast_atomic_fetchadd_int(&apeerobjs, 1);
set_peer_defaults(peer);
ast_copy_string(peer->name, name, sizeof(peer->name));
peer->selfdestruct = TRUE;
peer->host_dynamic = TRUE;
peer->prefs = default_prefs;
reg_source_db(peer);
return peer;
}
/*! \todo document this function */
static void add_peer_mailboxes(struct sip_peer *peer, const char *value)
{
char *next, *mbox, *context;
next = ast_strdupa(value);
while ((mbox = context = strsep(&next, ","))) {
struct sip_mailbox *mailbox;
if (!(mailbox = ast_calloc(1, sizeof(*mailbox))))
continue;
strsep(&context, "@");
if (ast_strlen_zero(mbox)) {
ast_free(mailbox);
continue;
}
mailbox->mailbox = ast_strdup(mbox);
mailbox->context = ast_strdup(context);
AST_LIST_INSERT_TAIL(&peer->mailboxes, mailbox, entry);
}
}
/*! \brief Build peer from configuration (file or realtime static/dynamic) */
static struct sip_peer *build_peer(const char *name, struct ast_variable *v, struct ast_variable *alt, int realtime)
{
struct sip_peer *peer = NULL;
struct ast_ha *oldha = NULL;
int found=0;
int firstpass=1;
int format=0; /* Ama flags */
time_t regseconds = 0;
struct ast_flags peerflags[2] = {{(0)}};
struct ast_flags mask[2] = {{(0)}};
char callback[256] = "";
struct sip_peer tmp_peer;
const char *srvlookup = NULL;
static int deprecation_warning = 1;
struct ast_str *fullcontact = ast_str_alloca(512);
if (!realtime || ast_test_flag(&global_flags[1], SIP_PAGE2_RTCACHEFRIENDS)) {
/* Note we do NOT use find_peer here, to avoid realtime recursion */
/* We also use a case-sensitive comparison (unlike find_peer) so
that case changes made to the peer name will be properly handled
during reload
*/
ast_copy_string(tmp_peer.name, name, sizeof(tmp_peer.name));
peer = ao2_t_find(peers, &tmp_peer, OBJ_POINTER | OBJ_UNLINK, "find and unlink peer from peers table");
}
if (peer) {
/* Already in the list, remove it and it will be added back (or FREE'd) */
found++;
if (!(peer->the_mark))
firstpass = 0;
} else {
if (!(peer = ao2_t_alloc(sizeof(*peer), sip_destroy_peer_fn, "allocate a peer struct")))
return NULL;
if (ast_string_field_init(peer, 512)) {
ao2_t_ref(peer, -1, "failed to string_field_init, drop peer");
return NULL;
}
if (realtime && !ast_test_flag(&global_flags[1], SIP_PAGE2_RTCACHEFRIENDS)) {
ast_atomic_fetchadd_int(&rpeerobjs, 1);
ast_debug(3, "-REALTIME- peer built. Name: %s. Peer objects: %d\n", name, rpeerobjs);
} else
ast_atomic_fetchadd_int(&speerobjs, 1);
}
/* Note that our peer HAS had its reference count increased */
if (firstpass) {
peer->lastmsgssent = -1;
oldha = peer->ha;
peer->ha = NULL;
set_peer_defaults(peer); /* Set peer defaults */
peer->type = 0;
}
if (!found && name)
ast_copy_string(peer->name, name, sizeof(peer->name));
/* If we have channel variables, remove them (reload) */
if (peer->chanvars) {
ast_variables_destroy(peer->chanvars);
peer->chanvars = NULL;
/* XXX should unregister ? */
}
/* If we have realm authentication information, remove them (reload) */
clear_realm_authentication(peer->auth);
peer->auth = NULL;
peer->transports = 0;
peer->socket.type = 0;
for (; v || ((v = alt) && !(alt=NULL)); v = v->next) {
if (handle_common_options(&peerflags[0], &mask[0], v))
continue;
if (!strcasecmp(v->name, "transport") && !ast_strlen_zero(v->value)) {
char *val = ast_strdupa(v->value);
char *trans;
while ((trans = strsep(&val, ","))) {
trans = ast_skip_blanks(trans);
if (!strncasecmp(trans, "udp", 3))
peer->transports |= SIP_TRANSPORT_UDP;
else if (!strncasecmp(trans, "tcp", 3))
peer->transports |= SIP_TRANSPORT_TCP;
else if (!strncasecmp(trans, "tls", 3))
peer->transports |= SIP_TRANSPORT_TLS;
else
ast_log(LOG_NOTICE, "'%s' is not a valid transport type. if no other is specified, udp will be used.\n", trans);
if (!peer->socket.type) { /*!< The first transport listed should be used for outgoing */
peer->socket.type = peer->transports;
peer->socket.fd = -1;
}
}
} else if (realtime && !strcasecmp(v->name, "regseconds")) {
ast_get_time_t(v->value, &regseconds, 0, NULL);
} else if (realtime && !strcasecmp(v->name, "lastms")) {
sscanf(v->value, "%d", &peer->lastms);
} else if (realtime && !strcasecmp(v->name, "ipaddr") && !ast_strlen_zero(v->value) ) {
inet_aton(v->value, &(peer->addr.sin_addr));
} else if (realtime && !strcasecmp(v->name, "name"))
ast_copy_string(peer->name, v->value, sizeof(peer->name));
else if (realtime && !strcasecmp(v->name, "fullcontact")) {
/* Reconstruct field, because realtime separates our value at the ';' */
if (fullcontact->used > 0) {
ast_str_append(&fullcontact, 0, ";%s", v->value);
} else {
ast_str_set(&fullcontact, 0, "%s", v->value);
}
} else if (!strcasecmp(v->name, "type")) {
if (!strcasecmp(v->value, "peer")) {
peer->type |= SIP_TYPE_PEER;
} else if (!strcasecmp(v->value, "user")) {
peer->type |= SIP_TYPE_USER;
} else if (!strcasecmp(v->value, "friend")) {
peer->type = SIP_TYPE_USER | SIP_TYPE_PEER;
}
} else if (!strcasecmp(v->name, "remotesecret")) {
ast_string_field_set(peer, remotesecret, v->value);
} else if (!strcasecmp(v->name, "secret")) {
ast_string_field_set(peer, secret, v->value);
} else if (!strcasecmp(v->name, "md5secret"))
ast_string_field_set(peer, md5secret, v->value);
else if (!strcasecmp(v->name, "auth"))
peer->auth = add_realm_authentication(peer->auth, v->value, v->lineno);
else if (!strcasecmp(v->name, "callerid")) {
char cid_name[80] = { '\0' }, cid_num[80] = { '\0' };
ast_callerid_split(v->value, cid_name, sizeof(cid_name), cid_num, sizeof(cid_num));
ast_string_field_set(peer, cid_name, cid_name);
ast_string_field_set(peer, cid_num, cid_num);
} else if (!strcasecmp(v->name, "mwi_from")) {
ast_string_field_set(peer, mwi_from, v->value);
} else if (!strcasecmp(v->name, "fullname")) {
ast_string_field_set(peer, cid_name, v->value);
} else if (!strcasecmp(v->name, "cid_number")) {
ast_string_field_set(peer, cid_num, v->value);
} else if (!strcasecmp(v->name, "context")) {
ast_string_field_set(peer, context, v->value);
} else if (!strcasecmp(v->name, "subscribecontext")) {
ast_string_field_set(peer, subscribecontext, v->value);
} else if (!strcasecmp(v->name, "fromdomain")) {
ast_string_field_set(peer, fromdomain, v->value);
} else if (!strcasecmp(v->name, "usereqphone")) {
ast_set2_flag(&peer->flags[0], ast_true(v->value), SIP_USEREQPHONE);
} else if (!strcasecmp(v->name, "fromuser")) {
ast_string_field_set(peer, fromuser, v->value);
} else if (!strcasecmp(v->name, "outboundproxy")) {
char *port, *next, *force, *proxyname;
int forceopt = FALSE;
/* Set peer channel variable */
next = proxyname = ast_strdupa(v->value);
if ((port = strchr(proxyname, ':'))) {
*port++ = '\0';
next = port;
}
if ((force = strchr(next, ','))) {
*force++ = '\0';
forceopt = strcmp(force, "force");
}
/* Allocate proxy object */
peer->outboundproxy = proxy_allocate(proxyname, port, forceopt);
} else if (!strcasecmp(v->name, "host")) {
if (!strcasecmp(v->value, "dynamic")) {
/* They'll register with us */
if (!found || !peer->host_dynamic) {
/* Initialize stuff if this is a new peer, or if it used to
* not be dynamic before the reload. */
memset(&peer->addr.sin_addr, 0, 4);
if (peer->addr.sin_port) {
/* If we've already got a port, make it the default rather than absolute */
peer->defaddr.sin_port = peer->addr.sin_port;
peer->addr.sin_port = 0;
}
}
peer->host_dynamic = TRUE;
} else {
/* Non-dynamic. Make sure we become that way if we're not */
AST_SCHED_DEL_UNREF(sched, peer->expire,
unref_peer(peer, "removing register expire ref"));
peer->host_dynamic = FALSE;
srvlookup = v->value;
if (global_dynamic_exclude_static) {
int err = 0;
global_contact_ha = ast_append_ha("deny", (char *)ast_inet_ntoa(peer->addr.sin_addr), global_contact_ha, &err);
if (err) {
ast_log(LOG_ERROR, "Bad ACL entry in configuration line %d : %s\n", v->lineno, v->value);
}
}
}
} else if (!strcasecmp(v->name, "defaultip")) {
if (ast_get_ip(&peer->defaddr, v->value)) {
unref_peer(peer, "unref_peer: from build_peer defaultip");
return NULL;
}
} else if (!strcasecmp(v->name, "permit") || !strcasecmp(v->name, "deny")) {
int ha_error = 0;
peer->ha = ast_append_ha(v->name, v->value, peer->ha, &ha_error);
if (ha_error)
ast_log(LOG_ERROR, "Bad ACL entry in configuration line %d : %s\n", v->lineno, v->value);
} else if (!strcasecmp(v->name, "contactpermit") || !strcasecmp(v->name, "contactdeny")) {
int ha_error = 0;
peer->contactha = ast_append_ha(v->name + 7, v->value, peer->contactha, &ha_error);
if (ha_error) {
ast_log(LOG_ERROR, "Bad ACL entry in configuration line %d : %s\n", v->lineno, v->value);
}
} else if (!strcasecmp(v->name, "port")) {
if (!realtime && peer->host_dynamic)
peer->defaddr.sin_port = htons(atoi(v->value));
else
peer->addr.sin_port = htons(atoi(v->value));
} else if (!strcasecmp(v->name, "callingpres")) {
peer->callingpres = ast_parse_caller_presentation(v->value);
if (peer->callingpres == -1)
peer->callingpres = atoi(v->value);
} else if (!strcasecmp(v->name, "username") || !strcmp(v->name, "defaultuser")) { /* "username" is deprecated */
ast_string_field_set(peer, username, v->value);
if (!strcasecmp(v->name, "username")) {
if (deprecation_warning) {
ast_log(LOG_NOTICE, "The 'username' field for sip peers has been deprecated in favor of the term 'defaultuser'\n");
deprecation_warning = 0;
}
peer->deprecated_username = 1;
}
} else if (!strcasecmp(v->name, "language")) {
ast_string_field_set(peer, language, v->value);
} else if (!strcasecmp(v->name, "regexten")) {
ast_string_field_set(peer, regexten, v->value);
} else if (!strcasecmp(v->name, "callbackextension")) {
ast_copy_string(callback, v->value, sizeof(callback));
} else if (!strcasecmp(v->name, "callcounter")) {
peer->call_limit = ast_true(v->value) ? 999 : 0;
} else if (!strcasecmp(v->name, "call-limit")) {
peer->call_limit = atoi(v->value);
if (peer->call_limit < 0)
peer->call_limit = 0;
} else if (!strcasecmp(v->name, "busylevel")) {
peer->busy_level = atoi(v->value);
if (peer->busy_level < 0)
peer->busy_level = 0;
} else if (!strcasecmp(v->name, "amaflags")) {
format = ast_cdr_amaflags2int(v->value);
if (format < 0) {
ast_log(LOG_WARNING, "Invalid AMA Flags for peer: %s at line %d\n", v->value, v->lineno);
} else {
peer->amaflags = format;
}
} else if (!strcasecmp(v->name, "accountcode")) {
ast_string_field_set(peer, accountcode, v->value);
} else if (!strcasecmp(v->name, "mohinterpret")) {
ast_string_field_set(peer, mohinterpret, v->value);
} else if (!strcasecmp(v->name, "mohsuggest")) {
ast_string_field_set(peer, mohsuggest, v->value);
} else if (!strcasecmp(v->name, "parkinglot")) {
ast_string_field_set(peer, parkinglot, v->value);
} else if (!strcasecmp(v->name, "rtp_engine")) {
ast_string_field_set(peer, engine, v->value);
} else if (!strcasecmp(v->name, "mailbox")) {
add_peer_mailboxes(peer, v->value);
} else if (!strcasecmp(v->name, "hasvoicemail")) {
/* People expect that if 'hasvoicemail' is set, that the mailbox will
* be also set, even if not explicitly specified. */
if (ast_true(v->value) && AST_LIST_EMPTY(&peer->mailboxes)) {
add_peer_mailboxes(peer, name);
}
} else if (!strcasecmp(v->name, "subscribemwi")) {
ast_set2_flag(&peer->flags[1], ast_true(v->value), SIP_PAGE2_SUBSCRIBEMWIONLY);
} else if (!strcasecmp(v->name, "vmexten")) {
ast_string_field_set(peer, vmexten, v->value);
} else if (!strcasecmp(v->name, "callgroup")) {
peer->callgroup = ast_get_group(v->value);
} else if (!strcasecmp(v->name, "allowtransfer")) {
peer->allowtransfer = ast_true(v->value) ? TRANSFER_OPENFORALL : TRANSFER_CLOSED;
} else if (!strcasecmp(v->name, "pickupgroup")) {
peer->pickupgroup = ast_get_group(v->value);
} else if (!strcasecmp(v->name, "allow")) {
int error = ast_parse_allow_disallow(&peer->prefs, &peer->capability, v->value, TRUE);
if (error)
ast_log(LOG_WARNING, "Codec configuration errors found in line %d : %s = %s\n", v->lineno, v->name, v->value);
} else if (!strcasecmp(v->name, "disallow")) {
int error = ast_parse_allow_disallow(&peer->prefs, &peer->capability, v->value, FALSE);
if (error)
ast_log(LOG_WARNING, "Codec configuration errors found in line %d : %s = %s\n", v->lineno, v->name, v->value);
} else if (!strcasecmp(v->name, "preferred_codec_only")) {
ast_set2_flag(&peer->flags[1], ast_true(v->value), SIP_PAGE2_PREFERRED_CODEC);
} else if (!strcasecmp(v->name, "registertrying")) {
ast_set2_flag(&peer->flags[1], ast_true(v->value), SIP_PAGE2_REGISTERTRYING);
} else if (!strcasecmp(v->name, "autoframing")) {
peer->autoframing = ast_true(v->value);
} else if (!strcasecmp(v->name, "rtptimeout")) {
if ((sscanf(v->value, "%d", &peer->rtptimeout) != 1) || (peer->rtptimeout < 0)) {
ast_log(LOG_WARNING, "'%s' is not a valid RTP hold time at line %d. Using default.\n", v->value, v->lineno);
peer->rtptimeout = global_rtptimeout;
}
} else if (!strcasecmp(v->name, "rtpholdtimeout")) {
if ((sscanf(v->value, "%d", &peer->rtpholdtimeout) != 1) || (peer->rtpholdtimeout < 0)) {
ast_log(LOG_WARNING, "'%s' is not a valid RTP hold time at line %d. Using default.\n", v->value, v->lineno);
peer->rtpholdtimeout = global_rtpholdtimeout;
}
} else if (!strcasecmp(v->name, "rtpkeepalive")) {
if ((sscanf(v->value, "%d", &peer->rtpkeepalive) != 1) || (peer->rtpkeepalive < 0)) {
ast_log(LOG_WARNING, "'%s' is not a valid RTP keepalive time at line %d. Using default.\n", v->value, v->lineno);
peer->rtpkeepalive = global_rtpkeepalive;
}
} else if (!strcasecmp(v->name, "timert1")) {
if ((sscanf(v->value, "%d", &peer->timer_t1) != 1) || (peer->timer_t1 < 0)) {
ast_log(LOG_WARNING, "'%s' is not a valid T1 time at line %d. Using default.\n", v->value, v->lineno);
peer->timer_t1 = global_t1;
}
/* Note that Timer B is dependent upon T1 and MUST NOT be lower
* than T1 * 64, according to RFC 3261, Section 17.1.1.2 */
if (peer->timer_b < peer->timer_t1 * 64) {
peer->timer_b = peer->timer_t1 * 64;
}
} else if (!strcasecmp(v->name, "timerb")) {
if ((sscanf(v->value, "%d", &peer->timer_b) != 1) || (peer->timer_b < 0)) {
ast_log(LOG_WARNING, "'%s' is not a valid Timer B time at line %d. Using default.\n", v->value, v->lineno);
peer->timer_b = global_timer_b;
}
if (peer->timer_b < peer->timer_t1 * 64) {
static int warning = 0;
if (warning++ % 20 == 0) {
ast_log(LOG_WARNING, "Timer B has been set lower than recommended. (RFC 3261, 17.1.1.2)\n");
}
}
} else if (!strcasecmp(v->name, "setvar")) {
peer->chanvars = add_var(v->value, peer->chanvars);
} else if (!strcasecmp(v->name, "qualify")) {
if (!strcasecmp(v->value, "no")) {
peer->maxms = 0;
} else if (!strcasecmp(v->value, "yes")) {
peer->maxms = default_qualify ? default_qualify : DEFAULT_MAXMS;
} else if (sscanf(v->value, "%d", &peer->maxms) != 1) {
ast_log(LOG_WARNING, "Qualification of peer '%s' should be 'yes', 'no', or a number of milliseconds at line %d of sip.conf\n", peer->name, v->lineno);
peer->maxms = 0;
}
if (realtime && !ast_test_flag(&global_flags[1], SIP_PAGE2_RTCACHEFRIENDS) && peer->maxms > 0) {
/* This would otherwise cause a network storm, where the
* qualify response refreshes the peer from the database,
* which in turn causes another qualify to be sent, ad
* infinitum. */
ast_log(LOG_WARNING, "Qualify is incompatible with dynamic uncached realtime. Please either turn rtcachefriends on or turn qualify off on peer '%s'\n", peer->name);
peer->maxms = 0;
}
} else if (!strcasecmp(v->name, "qualifyfreq")) {
int i;
if (sscanf(v->value, "%d", &i) == 1)
peer->qualifyfreq = i * 1000;
else {
ast_log(LOG_WARNING, "Invalid qualifyfreq number '%s' at line %d of %s\n", v->value, v->lineno, config);
peer->qualifyfreq = global_qualifyfreq;
}
} else if (!strcasecmp(v->name, "maxcallbitrate")) {
peer->maxcallbitrate = atoi(v->value);
if (peer->maxcallbitrate < 0)
peer->maxcallbitrate = default_maxcallbitrate;
} else if (!strcasecmp(v->name, "session-timers")) {
int i = (int) str2stmode(v->value);
if (i < 0) {
ast_log(LOG_WARNING, "Invalid session-timers '%s' at line %d of %s\n", v->value, v->lineno, config);
peer->stimer.st_mode_oper = global_st_mode;
} else {
peer->stimer.st_mode_oper = i;
}
} else if (!strcasecmp(v->name, "session-expires")) {
if (sscanf(v->value, "%d", &peer->stimer.st_max_se) != 1) {
ast_log(LOG_WARNING, "Invalid session-expires '%s' at line %d of %s\n", v->value, v->lineno, config);
peer->stimer.st_max_se = global_max_se;
}
} else if (!strcasecmp(v->name, "session-minse")) {
if (sscanf(v->value, "%d", &peer->stimer.st_min_se) != 1) {
ast_log(LOG_WARNING, "Invalid session-minse '%s' at line %d of %s\n", v->value, v->lineno, config);
peer->stimer.st_min_se = global_min_se;
}
if (peer->stimer.st_min_se < 90) {
ast_log(LOG_WARNING, "session-minse '%s' at line %d of %s is not allowed to be < 90 secs\n", v->value, v->lineno, config);
peer->stimer.st_min_se = global_min_se;
}
} else if (!strcasecmp(v->name, "session-refresher")) {
int i = (int) str2strefresher(v->value);
if (i < 0) {
ast_log(LOG_WARNING, "Invalid session-refresher '%s' at line %d of %s\n", v->value, v->lineno, config);
peer->stimer.st_ref = global_st_refresher;
} else {
peer->stimer.st_ref = i;
}
}
}
if (!peer->socket.type) {
/* Set default set of transports */
peer->transports = default_transports;
/* Set default primary transport */
peer->socket.type = default_primary_transport;
peer->socket.fd = -1;
}
if (fullcontact->used > 0) {
ast_string_field_set(peer, fullcontact, fullcontact->str);
peer->rt_fromcontact = TRUE;
/* We have a hostname in the fullcontact, but if we don't have an
* address listed on the entry (or if it's 'dynamic'), then we need to
* parse the entry to obtain the IP address, so a dynamic host can be
* contacted immediately after reload (as opposed to waiting for it to
* register once again). */
/* XXX May need to revisit the final argument; does the realtime DB store whether
* the original contact was over TLS or not? XXX */
__set_address_from_contact(fullcontact->str, &peer->addr, 0);
}
if (srvlookup && peer->dnsmgr == NULL) {
char transport[MAXHOSTNAMELEN];
char _srvlookup[MAXHOSTNAMELEN];
char *params;
ast_copy_string(_srvlookup, srvlookup, sizeof(_srvlookup));
if ((params = strchr(_srvlookup, ';'))) {
*params++ = '\0';
}
snprintf(transport, sizeof(transport), "_sip._%s", get_transport(peer->socket.type));
if (ast_dnsmgr_lookup(_srvlookup, &peer->addr, &peer->dnsmgr, sip_cfg.srvlookup ? transport : NULL)) {
unref_peer(peer, "getting rid of a peer pointer");
return NULL;
}
ast_string_field_set(peer, tohost, srvlookup);
}
if (!peer->addr.sin_port)
peer->addr.sin_port = htons(((peer->socket.type & SIP_TRANSPORT_TLS) ? STANDARD_TLS_PORT : STANDARD_SIP_PORT));
if (!peer->socket.port)
peer->socket.port = htons(((peer->socket.type & SIP_TRANSPORT_TLS) ? STANDARD_TLS_PORT : STANDARD_SIP_PORT));
if (!sip_cfg.ignore_regexpire && peer->host_dynamic && realtime) {
time_t nowtime = time(NULL);
if ((nowtime - regseconds) > 0) {
destroy_association(peer);
memset(&peer->addr, 0, sizeof(peer->addr));
peer->lastms = -1;
ast_debug(1, "Bah, we're expired (%d/%d/%d)!\n", (int)(nowtime - regseconds), (int)regseconds, (int)nowtime);
}
}
/* Startup regular pokes */
if (realtime && peer->lastms > 0) {
ref_peer(peer, "schedule qualify");
sip_poke_peer(peer, 0);
}
ast_copy_flags(&peer->flags[0], &peerflags[0], mask[0].flags);
ast_copy_flags(&peer->flags[1], &peerflags[1], mask[1].flags);
if (ast_test_flag(&peer->flags[1], SIP_PAGE2_ALLOWSUBSCRIBE))
sip_cfg.allowsubscribe = TRUE; /* No global ban any more */
if (!found && peer->host_dynamic && !peer->is_realtime)
reg_source_db(peer);
/* If they didn't request that MWI is sent *only* on subscribe, go ahead and
* subscribe to it now. */
if (!ast_test_flag(&peer->flags[1], SIP_PAGE2_SUBSCRIBEMWIONLY) &&
!AST_LIST_EMPTY(&peer->mailboxes)) {
add_peer_mwi_subs(peer);
/* Send MWI from the event cache only. This is so we can send initial
* MWI if app_voicemail got loaded before chan_sip. If it is the other
* way, then we will get events when app_voicemail gets loaded. */
sip_send_mwi_to_peer(peer, NULL, 1);
}
peer->the_mark = 0;
ast_free_ha(oldha);
if (!ast_strlen_zero(callback)) { /* build string from peer info */
char *reg_string;
if (asprintf(&reg_string, "%s:%s@%s/%s", peer->username, peer->remotesecret ? peer->remotesecret : peer->secret, peer->tohost, callback) < 0) {
ast_log(LOG_WARNING, "asprintf() failed: %s\n", strerror(errno));
} else if (reg_string) {
sip_register(reg_string, 0); /* XXX TODO: count in registry_count */
ast_free(reg_string);
}
}
return peer;
}
static int peer_markall_func(void *device, void *arg, int flags)
{
struct sip_peer *peer = device;
peer->the_mark = 1;
return 0;
}
/*! \brief Re-read SIP.conf config file
\note This function reloads all config data, except for
active peers (with registrations). They will only
change configuration data at restart, not at reload.
SIP debug and recordhistory state will not change
*/
static int reload_config(enum channelreloadreason reason)
{
struct ast_config *cfg, *ucfg;
struct ast_variable *v;
struct sip_peer *peer;
char *cat, *stringp, *context, *oldregcontext;
char newcontexts[AST_MAX_CONTEXT], oldcontexts[AST_MAX_CONTEXT];
struct ast_flags dummy[2];
struct ast_flags config_flags = { reason == CHANNEL_MODULE_LOAD ? 0 : CONFIG_FLAG_FILEUNCHANGED };
int auto_sip_domains = FALSE;
struct sockaddr_in old_bindaddr = bindaddr;
int registry_count = 0, peer_count = 0;
time_t run_start, run_end;
run_start = time(0);
ast_unload_realtime("sipregs");
ast_unload_realtime("sippeers");
cfg = ast_config_load(config, config_flags);
/* We *must* have a config file otherwise stop immediately */
if (!cfg) {
ast_log(LOG_NOTICE, "Unable to load config %s\n", config);
return -1;
} else if (cfg == CONFIG_STATUS_FILEUNCHANGED) {
ucfg = ast_config_load("users.conf", config_flags);
if (ucfg == CONFIG_STATUS_FILEUNCHANGED) {
return 1;
} else if (ucfg == CONFIG_STATUS_FILEINVALID) {
ast_log(LOG_ERROR, "Contents of users.conf are invalid and cannot be parsed\n");
return 1;
}
/* Must reread both files, because one changed */
ast_clear_flag(&config_flags, CONFIG_FLAG_FILEUNCHANGED);
if ((cfg = ast_config_load(config, config_flags)) == CONFIG_STATUS_FILEINVALID) {
ast_log(LOG_ERROR, "Contents of %s are invalid and cannot be parsed\n", config);
ast_config_destroy(ucfg);
return 1;
}
} else if (cfg == CONFIG_STATUS_FILEINVALID) {
ast_log(LOG_ERROR, "Contents of %s are invalid and cannot be parsed\n", config);
return 1;
} else {
ast_clear_flag(&config_flags, CONFIG_FLAG_FILEUNCHANGED);
if ((ucfg = ast_config_load("users.conf", config_flags)) == CONFIG_STATUS_FILEINVALID) {
ast_log(LOG_ERROR, "Contents of users.conf are invalid and cannot be parsed\n");
ast_config_destroy(cfg);
return 1;
}
}
/* Initialize tcp sockets */
memset(&sip_tcp_desc.local_address, 0, sizeof(sip_tcp_desc.local_address));
memset(&sip_tls_desc.local_address, 0, sizeof(sip_tls_desc.local_address));
ast_free_ha(global_contact_ha);
global_contact_ha = NULL;
default_tls_cfg.enabled = FALSE; /* Default: Disable TLS */
sip_tcp_desc.local_address.sin_port = htons(STANDARD_SIP_PORT);
sip_tls_desc.local_address.sin_port = htons(STANDARD_TLS_PORT);
if (reason != CHANNEL_MODULE_LOAD) {
ast_debug(4, "--------------- SIP reload started\n");
clear_realm_authentication(authl);
clear_sip_domains();
authl = NULL;
/* First, destroy all outstanding registry calls */
/* This is needed, since otherwise active registry entries will not be destroyed */
ASTOBJ_CONTAINER_TRAVERSE(&regl, 1, do { /* regl is locked */
/* avoid a deadlock in the unlink_all call, if iterator->call's (a dialog) registry entry
is this registry entry. In other words, if the dialog we are pointing to points back to
us, then if we get a lock on this object, and try to UNREF it, we will deadlock, because
we already ... NO. This is not the problem. */
ASTOBJ_RDLOCK(iterator); /* now regl is locked, and the object is also locked */
if (iterator->call) {
ast_debug(3, "Destroying active SIP dialog for registry %s@%s\n", iterator->username, iterator->hostname);
/* This will also remove references to the registry */
dialog_unlink_all(iterator->call, TRUE, TRUE);
iterator->call = dialog_unref(iterator->call, "remove iterator->call from registry traversal");
/* iterator->call = sip_destroy(iterator->call); */
}
ASTOBJ_UNLOCK(iterator);
} while(0));
/* Then, actually destroy users and registry */
ASTOBJ_CONTAINER_DESTROYALL(&regl, sip_registry_destroy);
ast_debug(4, "--------------- Done destroying registry list\n");
ao2_t_callback(peers, OBJ_NODATA, peer_markall_func, NULL, "callback to mark all peers");
}
/* Reset certificate handling for TLS sessions */
if (reason != CHANNEL_MODULE_LOAD) {
ast_free(default_tls_cfg.certfile);
ast_free(default_tls_cfg.pvtfile);
ast_free(default_tls_cfg.cipher);
ast_free(default_tls_cfg.cafile);
ast_free(default_tls_cfg.capath);
}
default_tls_cfg.certfile = ast_strdup(AST_CERTFILE); /*XXX Not sure if this is useful */
default_tls_cfg.pvtfile = ast_strdup("");
default_tls_cfg.cipher = ast_strdup("");
default_tls_cfg.cafile = ast_strdup("");
default_tls_cfg.capath = ast_strdup("");
/* Initialize copy of current global_regcontext for later use in removing stale contexts */
ast_copy_string(oldcontexts, global_regcontext, sizeof(oldcontexts));
oldregcontext = oldcontexts;
/* Clear all flags before setting default values */
/* Preserve debugging settings for console */
sipdebug &= sip_debug_console;
ast_clear_flag(&global_flags[0], AST_FLAGS_ALL);
ast_clear_flag(&global_flags[1], AST_FLAGS_ALL);
/* Reset IP addresses */
memset(&bindaddr, 0, sizeof(bindaddr));
memset(&stunaddr, 0, sizeof(stunaddr));
memset(&internip, 0, sizeof(internip));
/* Free memory for local network address mask */
ast_free_ha(localaddr);
memset(&localaddr, 0, sizeof(localaddr));
memset(&externip, 0, sizeof(externip));
memset(&default_prefs, 0 , sizeof(default_prefs));
memset(&sip_cfg.outboundproxy, 0, sizeof(struct sip_proxy));
sip_cfg.outboundproxy.ip.sin_port = htons(STANDARD_SIP_PORT);
sip_cfg.outboundproxy.ip.sin_family = AF_INET; /*!< Type of address: IPv4 */
sip_cfg.outboundproxy.force = FALSE; /*!< Don't force proxy usage, use route: headers */
default_transports = 0; /*!< Reset default transport to zero here, default value later on */
default_primary_transport = 0; /*!< Reset default primary transport to zero here, default value later on */
ourport_tcp = STANDARD_SIP_PORT;
ourport_tls = STANDARD_TLS_PORT;
bindaddr.sin_port = htons(STANDARD_SIP_PORT);
sip_cfg.srvlookup = DEFAULT_SRVLOOKUP;
global_tos_sip = DEFAULT_TOS_SIP;
global_tos_audio = DEFAULT_TOS_AUDIO;
global_tos_video = DEFAULT_TOS_VIDEO;
global_tos_text = DEFAULT_TOS_TEXT;
global_cos_sip = DEFAULT_COS_SIP;
global_cos_audio = DEFAULT_COS_AUDIO;
global_cos_video = DEFAULT_COS_VIDEO;
global_cos_text = DEFAULT_COS_TEXT;
externhost[0] = '\0'; /* External host name (for behind NAT DynDNS support) */
externexpire = 0; /* Expiration for DNS re-issuing */
externrefresh = 10;
/* Reset channel settings to default before re-configuring */
sip_cfg.allow_external_domains = DEFAULT_ALLOW_EXT_DOM; /* Allow external invites */
global_regcontext[0] = '\0';
sip_cfg.regextenonqualify = DEFAULT_REGEXTENONQUALIFY;
sip_cfg.notifyringing = DEFAULT_NOTIFYRINGING;
sip_cfg.notifycid = DEFAULT_NOTIFYCID;
sip_cfg.notifyhold = FALSE; /*!< Keep track of hold status for a peer */
sip_cfg.directrtpsetup = FALSE; /* Experimental feature, disabled by default */
sip_cfg.alwaysauthreject = DEFAULT_ALWAYSAUTHREJECT;
sip_cfg.allowsubscribe = FALSE;
snprintf(global_useragent, sizeof(global_useragent), "%s %s", DEFAULT_USERAGENT, ast_get_version());
snprintf(global_sdpsession, sizeof(global_sdpsession), "%s %s", DEFAULT_SDPSESSION, ast_get_version());
snprintf(global_sdpowner, sizeof(global_sdpowner), "%s", DEFAULT_SDPOWNER);
ast_copy_string(default_notifymime, DEFAULT_NOTIFYMIME, sizeof(default_notifymime));
ast_copy_string(sip_cfg.realm, S_OR(ast_config_AST_SYSTEM_NAME, DEFAULT_REALM), sizeof(sip_cfg.realm));
ast_copy_string(default_callerid, DEFAULT_CALLERID, sizeof(default_callerid));
ast_copy_string(default_mwi_from, DEFAULT_MWI_FROM, sizeof(default_mwi_from));
sip_cfg.compactheaders = DEFAULT_COMPACTHEADERS;
global_reg_timeout = DEFAULT_REGISTRATION_TIMEOUT;
global_regattempts_max = 0;
sip_cfg.pedanticsipchecking = DEFAULT_PEDANTIC;
sip_cfg.autocreatepeer = DEFAULT_AUTOCREATEPEER;
global_autoframing = 0;
sip_cfg.allowguest = DEFAULT_ALLOWGUEST;
global_callcounter = DEFAULT_CALLCOUNTER;
global_match_auth_username = FALSE; /*!< Match auth username if available instead of From: Default off. */
global_rtptimeout = 0;
global_rtpholdtimeout = 0;
global_rtpkeepalive = DEFAULT_RTPKEEPALIVE;
sip_cfg.allowtransfer = TRANSFER_OPENFORALL; /* Merrily accept all transfers by default */
sip_cfg.rtautoclear = 120;
ast_set_flag(&global_flags[1], SIP_PAGE2_ALLOWSUBSCRIBE); /* Default for all devices: TRUE */
ast_set_flag(&global_flags[1], SIP_PAGE2_ALLOWOVERLAP); /* Default for all devices: TRUE */
sip_cfg.peer_rtupdate = TRUE;
/* Session-Timers */
global_st_mode = SESSION_TIMER_MODE_ACCEPT;
global_st_refresher = SESSION_TIMER_REFRESHER_UAS;
global_min_se = DEFAULT_MIN_SE;
global_max_se = DEFAULT_MAX_SE;
/* Peer poking settings */
global_qualify_gap = DEFAULT_QUALIFY_GAP;
global_qualify_peers = DEFAULT_QUALIFY_PEERS;
/* Initialize some reasonable defaults at SIP reload (used both for channel and as default for devices */
ast_copy_string(sip_cfg.default_context, DEFAULT_CONTEXT, sizeof(sip_cfg.default_context));
sip_cfg.default_subscribecontext[0] = '\0';
default_language[0] = '\0';
default_fromdomain[0] = '\0';
default_qualify = DEFAULT_QUALIFY;
default_maxcallbitrate = DEFAULT_MAX_CALL_BITRATE;
ast_copy_string(default_mohinterpret, DEFAULT_MOHINTERPRET, sizeof(default_mohinterpret));
ast_copy_string(default_mohsuggest, DEFAULT_MOHSUGGEST, sizeof(default_mohsuggest));
ast_copy_string(default_vmexten, DEFAULT_VMEXTEN, sizeof(default_vmexten));
ast_set_flag(&global_flags[0], SIP_DTMF_RFC2833); /*!< Default DTMF setting: RFC2833 */
ast_set_flag(&global_flags[0], SIP_NAT_RFC3581); /*!< NAT support if requested by device with rport */
ast_set_flag(&global_flags[0], SIP_CAN_REINVITE); /*!< Allow re-invites */
ast_copy_string(default_engine, DEFAULT_ENGINE, sizeof(default_engine));
/* Debugging settings, always default to off */
dumphistory = FALSE;
recordhistory = FALSE;
sipdebug &= ~sip_debug_config;
/* Misc settings for the channel */
global_relaxdtmf = FALSE;
sip_cfg.callevents = DEFAULT_CALLEVENTS;
global_authfailureevents = FALSE;
global_t1 = DEFAULT_TIMER_T1;
global_timer_b = 64 * DEFAULT_TIMER_T1;
global_t1min = DEFAULT_T1MIN;
global_qualifyfreq = DEFAULT_QUALIFYFREQ;
sip_cfg.matchexterniplocally = DEFAULT_MATCHEXTERNIPLOCALLY;
/* Copy the default jb config over global_jbconf */
memcpy(&global_jbconf, &default_jbconf, sizeof(struct ast_jb_conf));
ast_clear_flag(&global_flags[1], SIP_PAGE2_FAX_DETECT);
ast_clear_flag(&global_flags[1], SIP_PAGE2_VIDEOSUPPORT | SIP_PAGE2_VIDEOSUPPORT_ALWAYS);
ast_clear_flag(&global_flags[1], SIP_PAGE2_TEXTSUPPORT);
ast_clear_flag(&global_flags[1], SIP_PAGE2_IGNORESDPVERSION);
/* Read the [general] config section of sip.conf (or from realtime config) */
for (v = ast_variable_browse(cfg, "general"); v; v = v->next) {
if (handle_common_options(&global_flags[0], &dummy[0], v))
continue;
/* handle jb conf */
if (!ast_jb_read_conf(&global_jbconf, v->name, v->value))
continue;
/* handle tls conf */
if (!ast_tls_read_conf(&default_tls_cfg, &sip_tls_desc, v->name, v->value)) {
continue;
}
if (!strcasecmp(v->name, "context")) {
ast_copy_string(sip_cfg.default_context, v->value, sizeof(sip_cfg.default_context));
} else if (!strcasecmp(v->name, "subscribecontext")) {
ast_copy_string(sip_cfg.default_subscribecontext, v->value, sizeof(sip_cfg.default_subscribecontext));
} else if (!strcasecmp(v->name, "callcounter")) {
global_callcounter = ast_true(v->value) ? 1 : 0;
} else if (!strcasecmp(v->name, "allowguest")) {
sip_cfg.allowguest = ast_true(v->value) ? 1 : 0;
} else if (!strcasecmp(v->name, "realm")) {
ast_copy_string(sip_cfg.realm, v->value, sizeof(sip_cfg.realm));
} else if (!strcasecmp(v->name, "useragent")) {
ast_copy_string(global_useragent, v->value, sizeof(global_useragent));
ast_debug(1, "Setting SIP channel User-Agent Name to %s\n", global_useragent);
} else if (!strcasecmp(v->name, "sdpsession")) {
ast_copy_string(global_sdpsession, v->value, sizeof(global_sdpsession));
} else if (!strcasecmp(v->name, "sdpowner")) {
/* Field cannot contain spaces */
if (!strstr(v->value, " "))
ast_copy_string(global_sdpowner, v->value, sizeof(global_sdpowner));
else
ast_log(LOG_WARNING, "'%s' must not contain spaces at line %d. Using default.\n", v->value, v->lineno);
} else if (!strcasecmp(v->name, "allowtransfer")) {
sip_cfg.allowtransfer = ast_true(v->value) ? TRANSFER_OPENFORALL : TRANSFER_CLOSED;
} else if (!strcasecmp(v->name, "rtcachefriends")) {
ast_set2_flag(&global_flags[1], ast_true(v->value), SIP_PAGE2_RTCACHEFRIENDS);
} else if (!strcasecmp(v->name, "rtsavesysname")) {
sip_cfg.rtsave_sysname = ast_true(v->value);
} else if (!strcasecmp(v->name, "rtupdate")) {
sip_cfg.peer_rtupdate = ast_true(v->value);
} else if (!strcasecmp(v->name, "ignoreregexpire")) {
sip_cfg.ignore_regexpire = ast_true(v->value);
} else if (!strcasecmp(v->name, "timert1")) {
/* Defaults to 500ms, but RFC 3261 states that it is recommended
* for the value to be set higher, though a lower value is only
* allowed on private networks unconnected to the Internet. */
global_t1 = atoi(v->value);
/* Note that timer B is dependent on the value of T1 */
global_timer_b = global_t1 * 64;
} else if (!strcasecmp(v->name, "t1min")) {
global_t1min = atoi(v->value);
} else if (!strcasecmp(v->name, "transport") && !ast_strlen_zero(v->value)) {
char *val = ast_strdupa(v->value);
char *trans;
while ((trans = strsep(&val, ","))) {
trans = ast_skip_blanks(trans);
if (!strncasecmp(trans, "udp", 3))
default_transports |= SIP_TRANSPORT_UDP;
else if (!strncasecmp(trans, "tcp", 3))
default_transports |= SIP_TRANSPORT_TCP;
else if (!strncasecmp(trans, "tls", 3))
default_transports |= SIP_TRANSPORT_TLS;
else
ast_log(LOG_NOTICE, "'%s' is not a valid transport type. if no other is specified, udp will be used.\n", trans);
if (default_primary_transport == 0) {
default_primary_transport = default_transports;
}
}
} else if (!strcasecmp(v->name, "tcpenable")) {
sip_tcp_desc.local_address.sin_family = ast_false(v->value) ? 0 : AF_INET;
ast_debug(2, "Enabling TCP socket for listening\n");
} else if (!strcasecmp(v->name, "tcpbindaddr")) {
int family = sip_tcp_desc.local_address.sin_family;
if (ast_parse_arg(v->value, PARSE_INADDR, &sip_tcp_desc.local_address))
ast_log(LOG_WARNING, "Invalid %s '%s' at line %d of %s\n", v->name, v->value, v->lineno, config);
sip_tcp_desc.local_address.sin_family = family;
ast_debug(2, "Setting TCP socket address to %s\n", v->value);
} else if (!strcasecmp(v->name, "dynamic_exclude_static") || !strcasecmp(v->name, "dynamic_excludes_static")) {
global_dynamic_exclude_static = ast_true(v->value);
} else if (!strcasecmp(v->name, "contactpermit") || !strcasecmp(v->name, "contactdeny")) {
int ha_error = 0;
global_contact_ha = ast_append_ha(v->name + 7, v->value, global_contact_ha, &ha_error);
if (ha_error) {
ast_log(LOG_ERROR, "Bad ACL entry in configuration line %d : %s\n", v->lineno, v->value);
}
} else if (!strcasecmp(v->name, "rtautoclear")) {
int i = atoi(v->value);
if (i > 0)
sip_cfg.rtautoclear = i;
else
i = 0;
ast_set2_flag(&global_flags[1], i || ast_true(v->value), SIP_PAGE2_RTAUTOCLEAR);
} else if (!strcasecmp(v->name, "usereqphone")) {
ast_set2_flag(&global_flags[0], ast_true(v->value), SIP_USEREQPHONE);
} else if (!strcasecmp(v->name, "relaxdtmf")) {
global_relaxdtmf = ast_true(v->value);
} else if (!strcasecmp(v->name, "vmexten")) {
ast_copy_string(default_vmexten, v->value, sizeof(default_vmexten));
} else if (!strcasecmp(v->name, "rtptimeout")) {
if ((sscanf(v->value, "%d", &global_rtptimeout) != 1) || (global_rtptimeout < 0)) {
ast_log(LOG_WARNING, "'%s' is not a valid RTP hold time at line %d. Using default.\n", v->value, v->lineno);
global_rtptimeout = 0;
}
} else if (!strcasecmp(v->name, "rtpholdtimeout")) {
if ((sscanf(v->value, "%d", &global_rtpholdtimeout) != 1) || (global_rtpholdtimeout < 0)) {
ast_log(LOG_WARNING, "'%s' is not a valid RTP hold time at line %d. Using default.\n", v->value, v->lineno);
global_rtpholdtimeout = 0;
}
} else if (!strcasecmp(v->name, "rtpkeepalive")) {
if ((sscanf(v->value, "%d", &global_rtpkeepalive) != 1) || (global_rtpkeepalive < 0)) {
ast_log(LOG_WARNING, "'%s' is not a valid RTP keepalive time at line %d. Using default.\n", v->value, v->lineno);
global_rtpkeepalive = DEFAULT_RTPKEEPALIVE;
}
} else if (!strcasecmp(v->name, "compactheaders")) {
sip_cfg.compactheaders = ast_true(v->value);
} else if (!strcasecmp(v->name, "notifymimetype")) {
ast_copy_string(default_notifymime, v->value, sizeof(default_notifymime));
} else if (!strcasecmp(v->name, "directrtpsetup")) {
sip_cfg.directrtpsetup = ast_true(v->value);
} else if (!strcasecmp(v->name, "notifyringing")) {
sip_cfg.notifyringing = ast_true(v->value);
} else if (!strcasecmp(v->name, "notifyhold")) {
sip_cfg.notifyhold = ast_true(v->value);
} else if (!strcasecmp(v->name, "notifycid")) {
if (!strcasecmp(v->value, "ignore-context")) {
sip_cfg.notifycid = IGNORE_CONTEXT;
} else {
sip_cfg.notifycid = ast_true(v->value);
}
} else if (!strcasecmp(v->name, "alwaysauthreject")) {
sip_cfg.alwaysauthreject = ast_true(v->value);
} else if (!strcasecmp(v->name, "mohinterpret")) {
ast_copy_string(default_mohinterpret, v->value, sizeof(default_mohinterpret));
} else if (!strcasecmp(v->name, "mohsuggest")) {
ast_copy_string(default_mohsuggest, v->value, sizeof(default_mohsuggest));
} else if (!strcasecmp(v->name, "language")) {
ast_copy_string(default_language, v->value, sizeof(default_language));
} else if (!strcasecmp(v->name, "regcontext")) {
ast_copy_string(newcontexts, v->value, sizeof(newcontexts));
stringp = newcontexts;
/* Let's remove any contexts that are no longer defined in regcontext */
cleanup_stale_contexts(stringp, oldregcontext);
/* Create contexts if they don't exist already */
while ((context = strsep(&stringp, "&"))) {
ast_copy_string(used_context, context, sizeof(used_context));
ast_context_find_or_create(NULL, NULL, context, "SIP");
}
ast_copy_string(global_regcontext, v->value, sizeof(global_regcontext));
} else if (!strcasecmp(v->name, "regextenonqualify")) {
sip_cfg.regextenonqualify = ast_true(v->value);
} else if (!strcasecmp(v->name, "callerid")) {
ast_copy_string(default_callerid, v->value, sizeof(default_callerid));
} else if (!strcasecmp(v->name, "mwi_from")) {
ast_copy_string(default_mwi_from, v->value, sizeof(default_mwi_from));
} else if (!strcasecmp(v->name, "fromdomain")) {
ast_copy_string(default_fromdomain, v->value, sizeof(default_fromdomain));
} else if (!strcasecmp(v->name, "outboundproxy")) {
int portnum;
char *tok, *proxyname;
if (ast_strlen_zero(v->value)) {
ast_log(LOG_WARNING, "no value given for outbound proxy on line %d of sip.conf.", v->lineno);
continue;
}
tok = ast_skip_blanks(strtok(ast_strdupa(v->value), ","));
sip_parse_host(tok, v->lineno, &proxyname, &portnum, &sip_cfg.outboundproxy.transport);
sip_cfg.outboundproxy.ip.sin_port = htons(portnum);
if ((tok = strtok(NULL, ","))) {
sip_cfg.outboundproxy.force = !strncasecmp(ast_skip_blanks(tok), "force", 5);
} else {
sip_cfg.outboundproxy.force = FALSE;
}
if (ast_strlen_zero(proxyname)) {
ast_log(LOG_WARNING, "you must specify a name for the outboundproxy on line %d of sip.conf.", v->lineno);
sip_cfg.outboundproxy.name[0] = '\0';
continue;
}
ast_copy_string(sip_cfg.outboundproxy.name, proxyname, sizeof(sip_cfg.outboundproxy.name));
proxy_update(&sip_cfg.outboundproxy);
} else if (!strcasecmp(v->name, "autocreatepeer")) {
sip_cfg.autocreatepeer = ast_true(v->value);
} else if (!strcasecmp(v->name, "match_auth_username")) {
global_match_auth_username = ast_true(v->value);
} else if (!strcasecmp(v->name, "srvlookup")) {
sip_cfg.srvlookup = ast_true(v->value);
} else if (!strcasecmp(v->name, "pedantic")) {
sip_cfg.pedanticsipchecking = ast_true(v->value);
} else if (!strcasecmp(v->name, "maxexpirey") || !strcasecmp(v->name, "maxexpiry")) {
max_expiry = atoi(v->value);
if (max_expiry < 1)
max_expiry = DEFAULT_MAX_EXPIRY;
} else if (!strcasecmp(v->name, "minexpirey") || !strcasecmp(v->name, "minexpiry")) {
min_expiry = atoi(v->value);
if (min_expiry < 1)
min_expiry = DEFAULT_MIN_EXPIRY;
} else if (!strcasecmp(v->name, "defaultexpiry") || !strcasecmp(v->name, "defaultexpirey")) {
default_expiry = atoi(v->value);
if (default_expiry < 1)
default_expiry = DEFAULT_DEFAULT_EXPIRY;
} else if (!strcasecmp(v->name, "mwiexpiry") || !strcasecmp(v->name, "mwiexpirey")) {
mwi_expiry = atoi(v->value);
if (mwi_expiry < 1)
mwi_expiry = DEFAULT_MWI_EXPIRY;
} else if (!strcasecmp(v->name, "sipdebug")) {
if (ast_true(v->value))
sipdebug |= sip_debug_config;
} else if (!strcasecmp(v->name, "dumphistory")) {
dumphistory = ast_true(v->value);
} else if (!strcasecmp(v->name, "recordhistory")) {
recordhistory = ast_true(v->value);
} else if (!strcasecmp(v->name, "registertimeout")) {
global_reg_timeout = atoi(v->value);
if (global_reg_timeout < 1)
global_reg_timeout = DEFAULT_REGISTRATION_TIMEOUT;
} else if (!strcasecmp(v->name, "registerattempts")) {
global_regattempts_max = atoi(v->value);
} else if (!strcasecmp(v->name, "stunaddr")) {
stunaddr.sin_port = htons(3478);
if (ast_parse_arg(v->value, PARSE_INADDR, &stunaddr))
ast_log(LOG_WARNING, "Invalid STUN server address: %s\n", v->value);
externexpire = time(NULL);
} else if (!strcasecmp(v->name, "bindaddr") || !strcasecmp(v->name, "udpbindaddr")) {
if (ast_parse_arg(v->value, PARSE_INADDR, &bindaddr))
ast_log(LOG_WARNING, "Invalid address: %s\n", v->value);
} else if (!strcasecmp(v->name, "localnet")) {
struct ast_ha *na;
int ha_error = 0;
if (!(na = ast_append_ha("d", v->value, localaddr, &ha_error)))
ast_log(LOG_WARNING, "Invalid localnet value: %s\n", v->value);
else
localaddr = na;
if (ha_error)
ast_log(LOG_ERROR, "Bad localnet configuration value line %d : %s\n", v->lineno, v->value);
} else if (!strcasecmp(v->name, "externip")) {
if (ast_parse_arg(v->value, PARSE_INADDR, &externip))
ast_log(LOG_WARNING, "Invalid address for externip keyword: %s\n", v->value);
externexpire = 0;
/* If no port was specified use the value of bindport */
if (!externip.sin_port)
externip.sin_port = bindaddr.sin_port;
} else if (!strcasecmp(v->name, "externhost")) {
ast_copy_string(externhost, v->value, sizeof(externhost));
if (ast_parse_arg(externhost, PARSE_INADDR, &externip))
ast_log(LOG_WARNING, "Invalid address for externhost keyword: %s\n", externhost);
externexpire = time(NULL);
/* If no port was specified use the value of bindport */
if (!externip.sin_port)
externip.sin_port = bindaddr.sin_port;
} else if (!strcasecmp(v->name, "externrefresh")) {
if (sscanf(v->value, "%d", &externrefresh) != 1) {
ast_log(LOG_WARNING, "Invalid externrefresh value '%s', must be an integer >0 at line %d\n", v->value, v->lineno);
externrefresh = 10;
}
} else if (!strcasecmp(v->name, "allow")) {
int error = ast_parse_allow_disallow(&default_prefs, &global_capability, v->value, TRUE);
if (error)
ast_log(LOG_WARNING, "Codec configuration errors found in line %d : %s = %s\n", v->lineno, v->name, v->value);
} else if (!strcasecmp(v->name, "disallow")) {
int error = ast_parse_allow_disallow(&default_prefs, &global_capability, v->value, FALSE);
if (error)
ast_log(LOG_WARNING, "Codec configuration errors found in line %d : %s = %s\n", v->lineno, v->name, v->value);
} else if (!strcasecmp(v->name, "preferred_codec_only")) {
ast_set2_flag(&global_flags[1], ast_true(v->value), SIP_PAGE2_PREFERRED_CODEC);
} else if (!strcasecmp(v->name, "autoframing")) {
global_autoframing = ast_true(v->value);
} else if (!strcasecmp(v->name, "allowexternaldomains")) {
sip_cfg.allow_external_domains = ast_true(v->value);
} else if (!strcasecmp(v->name, "autodomain")) {
auto_sip_domains = ast_true(v->value);
} else if (!strcasecmp(v->name, "domain")) {
char *domain = ast_strdupa(v->value);
char *cntx = strchr(domain, ',');
if (cntx)
*cntx++ = '\0';
if (ast_strlen_zero(cntx))
ast_debug(1, "No context specified at line %d for domain '%s'\n", v->lineno, domain);
if (ast_strlen_zero(domain))
ast_log(LOG_WARNING, "Empty domain specified at line %d\n", v->lineno);
else
add_sip_domain(ast_strip(domain), SIP_DOMAIN_CONFIG, cntx ? ast_strip(cntx) : "");
} else if (!strcasecmp(v->name, "register")) {
if (sip_register(v->value, v->lineno) == 0)
registry_count++;
} else if (!strcasecmp(v->name, "mwi")) {
sip_subscribe_mwi(v->value, v->lineno);
} else if (!strcasecmp(v->name, "tos_sip")) {
if (ast_str2tos(v->value, &global_tos_sip))
ast_log(LOG_WARNING, "Invalid tos_sip value at line %d, refer to QoS documentation\n", v->lineno);
} else if (!strcasecmp(v->name, "tos_audio")) {
if (ast_str2tos(v->value, &global_tos_audio))
ast_log(LOG_WARNING, "Invalid tos_audio value at line %d, refer to QoS documentation\n", v->lineno);
} else if (!strcasecmp(v->name, "tos_video")) {
if (ast_str2tos(v->value, &global_tos_video))
ast_log(LOG_WARNING, "Invalid tos_video value at line %d, refer to QoS documentation\n", v->lineno);
} else if (!strcasecmp(v->name, "tos_text")) {
if (ast_str2tos(v->value, &global_tos_text))
ast_log(LOG_WARNING, "Invalid tos_text value at line %d, refer to QoS documentation\n", v->lineno);
} else if (!strcasecmp(v->name, "cos_sip")) {
if (ast_str2cos(v->value, &global_cos_sip))
ast_log(LOG_WARNING, "Invalid cos_sip value at line %d, refer to QoS documentation\n", v->lineno);
} else if (!strcasecmp(v->name, "cos_audio")) {
if (ast_str2cos(v->value, &global_cos_audio))
ast_log(LOG_WARNING, "Invalid cos_audio value at line %d, refer to QoS documentation\n", v->lineno);
} else if (!strcasecmp(v->name, "cos_video")) {
if (ast_str2cos(v->value, &global_cos_video))
ast_log(LOG_WARNING, "Invalid cos_video value at line %d, refer to QoS documentation\n", v->lineno);
} else if (!strcasecmp(v->name, "cos_text")) {
if (ast_str2cos(v->value, &global_cos_text))
ast_log(LOG_WARNING, "Invalid cos_text value at line %d, refer to QoS documentation\n", v->lineno);
} else if (!strcasecmp(v->name, "bindport")) {
int i;
if (sscanf(v->value, "%d", &i) == 1) {
bindaddr.sin_port = htons(i);
} else {
ast_log(LOG_WARNING, "Invalid port number '%s' at line %d of %s\n", v->value, v->lineno, config);
}
} else if (!strcasecmp(v->name, "hash_user")) {
int i;
if (sscanf(v->value, "%d", &i) == 1 && i > 2) {
hash_user_size = i;
} else {
ast_log(LOG_WARNING, "Invalid hash_user size '%s' at line %d of %s -- should be much larger than 2\n", v->value, v->lineno, config);
}
} else if (!strcasecmp(v->name, "hash_peer")) {
int i;
if (sscanf(v->value, "%d", &i) == 1 && i > 2) {
hash_peer_size = i;
} else {
ast_log(LOG_WARNING, "Invalid hash_peer size '%s' at line %d of %s -- should be much larger than 2\n", v->value, v->lineno, config);
}
} else if (!strcasecmp(v->name, "hash_dialog")) {
int i;
if (sscanf(v->value, "%d", &i) == 1 && i > 2) {
hash_dialog_size = i;
} else {
ast_log(LOG_WARNING, "Invalid hash_dialog size '%s' at line %d of %s -- should be much larger than 2\n", v->value, v->lineno, config);
}
} else if (!strcasecmp(v->name, "qualify")) {
if (!strcasecmp(v->value, "no")) {
default_qualify = 0;
} else if (!strcasecmp(v->value, "yes")) {
default_qualify = DEFAULT_MAXMS;
} else if (sscanf(v->value, "%d", &default_qualify) != 1) {
ast_log(LOG_WARNING, "Qualification default should be 'yes', 'no', or a number of milliseconds at line %d of sip.conf\n", v->lineno);
default_qualify = 0;
}
} else if (!strcasecmp(v->name, "qualifyfreq")) {
int i;
if (sscanf(v->value, "%d", &i) == 1)
global_qualifyfreq = i * 1000;
else {
ast_log(LOG_WARNING, "Invalid qualifyfreq number '%s' at line %d of %s\n", v->value, v->lineno, config);
global_qualifyfreq = DEFAULT_QUALIFYFREQ;
}
} else if (!strcasecmp(v->name, "callevents")) {
sip_cfg.callevents = ast_true(v->value);
} else if (!strcasecmp(v->name, "authfailureevents")) {
global_authfailureevents = ast_true(v->value);
} else if (!strcasecmp(v->name, "maxcallbitrate")) {
default_maxcallbitrate = atoi(v->value);
if (default_maxcallbitrate < 0)
default_maxcallbitrate = DEFAULT_MAX_CALL_BITRATE;
} else if (!strcasecmp(v->name, "matchexterniplocally")) {
sip_cfg.matchexterniplocally = ast_true(v->value);
} else if (!strcasecmp(v->name, "session-timers")) {
int i = (int) str2stmode(v->value);
if (i < 0) {
ast_log(LOG_WARNING, "Invalid session-timers '%s' at line %d of %s\n", v->value, v->lineno, config);
global_st_mode = SESSION_TIMER_MODE_ACCEPT;
} else {
global_st_mode = i;
}
} else if (!strcasecmp(v->name, "session-expires")) {
if (sscanf(v->value, "%d", &global_max_se) != 1) {
ast_log(LOG_WARNING, "Invalid session-expires '%s' at line %d of %s\n", v->value, v->lineno, config);
global_max_se = DEFAULT_MAX_SE;
}
} else if (!strcasecmp(v->name, "session-minse")) {
if (sscanf(v->value, "%d", &global_min_se) != 1) {
ast_log(LOG_WARNING, "Invalid session-minse '%s' at line %d of %s\n", v->value, v->lineno, config);
global_min_se = DEFAULT_MIN_SE;
}
if (global_min_se < 90) {
ast_log(LOG_WARNING, "session-minse '%s' at line %d of %s is not allowed to be < 90 secs\n", v->value, v->lineno, config);
global_min_se = DEFAULT_MIN_SE;
}
} else if (!strcasecmp(v->name, "session-refresher")) {
int i = (int) str2strefresher(v->value);
if (i < 0) {
ast_log(LOG_WARNING, "Invalid session-refresher '%s' at line %d of %s\n", v->value, v->lineno, config);
global_st_refresher = SESSION_TIMER_REFRESHER_UAS;
} else {
global_st_refresher = i;
}
} else if (!strcasecmp(v->name, "qualifygap")) {
if (sscanf(v->value, "%d", &global_qualify_gap) != 1) {
ast_log(LOG_WARNING, "Invalid qualifygap '%s' at line %d of %s\n", v->value, v->lineno, config);
global_qualify_gap = DEFAULT_QUALIFY_GAP;
}
} else if (!strcasecmp(v->name, "qualifypeers")) {
if (sscanf(v->value, "%d", &global_qualify_peers) != 1) {
ast_log(LOG_WARNING, "Invalid pokepeers '%s' at line %d of %s\n", v->value, v->lineno, config);
global_qualify_peers = DEFAULT_QUALIFY_PEERS;
}
}
}
if (!sip_cfg.allow_external_domains && AST_LIST_EMPTY(&domain_list)) {
ast_log(LOG_WARNING, "To disallow external domains, you need to configure local SIP domains.\n");
sip_cfg.allow_external_domains = 1;
}
/* If not configured, set default transports */
if (default_transports == 0) {
default_transports = default_primary_transport = SIP_TRANSPORT_UDP;
}
/* Build list of authentication to various SIP realms, i.e. service providers */
for (v = ast_variable_browse(cfg, "authentication"); v ; v = v->next) {
/* Format for authentication is auth = username:password@realm */
if (!strcasecmp(v->name, "auth"))
authl = add_realm_authentication(authl, v->value, v->lineno);
}
if (ucfg) {
struct ast_variable *gen;
int genhassip, genregistersip;
const char *hassip, *registersip;
genhassip = ast_true(ast_variable_retrieve(ucfg, "general", "hassip"));
genregistersip = ast_true(ast_variable_retrieve(ucfg, "general", "registersip"));
gen = ast_variable_browse(ucfg, "general");
cat = ast_category_browse(ucfg, NULL);
while (cat) {
if (strcasecmp(cat, "general")) {
hassip = ast_variable_retrieve(ucfg, cat, "hassip");
registersip = ast_variable_retrieve(ucfg, cat, "registersip");
if (ast_true(hassip) || (!hassip && genhassip)) {
peer = build_peer(cat, gen, ast_variable_browse(ucfg, cat), 0);
if (peer) {
ao2_t_link(peers, peer, "link peer into peer table");
if ((peer->type & SIP_TYPE_PEER) && peer->addr.sin_addr.s_addr) {
ao2_t_link(peers_by_ip, peer, "link peer into peers_by_ip table");
}
unref_peer(peer, "unref_peer: from reload_config");
peer_count++;
}
}
if (ast_true(registersip) || (!registersip && genregistersip)) {
char tmp[256];
const char *host = ast_variable_retrieve(ucfg, cat, "host");
const char *username = ast_variable_retrieve(ucfg, cat, "username");
const char *secret = ast_variable_retrieve(ucfg, cat, "secret");
const char *contact = ast_variable_retrieve(ucfg, cat, "contact");
if (!host)
host = ast_variable_retrieve(ucfg, "general", "host");
if (!username)
username = ast_variable_retrieve(ucfg, "general", "username");
if (!secret)
secret = ast_variable_retrieve(ucfg, "general", "secret");
if (!contact)
contact = "s";
if (!ast_strlen_zero(username) && !ast_strlen_zero(host)) {
if (!ast_strlen_zero(secret))
snprintf(tmp, sizeof(tmp), "%s:%s@%s/%s", username, secret, host, contact);
else
snprintf(tmp, sizeof(tmp), "%s@%s/%s", username, host, contact);
if (sip_register(tmp, 0) == 0)
registry_count++;
}
}
}
cat = ast_category_browse(ucfg, cat);
}
ast_config_destroy(ucfg);
}
/* Load peers, users and friends */
cat = NULL;
while ( (cat = ast_category_browse(cfg, cat)) ) {
const char *utype;
if (!strcasecmp(cat, "general") || !strcasecmp(cat, "authentication"))
continue;
utype = ast_variable_retrieve(cfg, cat, "type");
if (!utype) {
ast_log(LOG_WARNING, "Section '%s' lacks type\n", cat);
continue;
} else {
if (!strcasecmp(utype, "user")) {
;
} else if (!strcasecmp(utype, "friend")) {
;
} else if (!strcasecmp(utype, "peer")) {
;
} else {
ast_log(LOG_WARNING, "Unknown type '%s' for '%s' in %s\n", utype, cat, "sip.conf");
continue;
}
peer = build_peer(cat, ast_variable_browse(cfg, cat), NULL, 0);
if (peer) {
ao2_t_link(peers, peer, "link peer into peers table");
if ((peer->type & SIP_TYPE_PEER) && peer->addr.sin_addr.s_addr) {
ao2_t_link(peers_by_ip, peer, "link peer into peers_by_ip table");
}
unref_peer(peer, "unref the result of the build_peer call. Now, the links from the tables are the only ones left.");
peer_count++;
}
}
}
/* Set UDP address and open socket */
bindaddr.sin_family = AF_INET;
internip = bindaddr;
if (ast_find_ourip(&internip.sin_addr, bindaddr)) {
ast_log(LOG_WARNING, "Unable to get own IP address, SIP disabled\n");
ast_config_destroy(cfg);
return 0;
}
ast_mutex_lock(&netlock);
if ((sipsock > -1) && (memcmp(&old_bindaddr, &bindaddr, sizeof(struct sockaddr_in)))) {
close(sipsock);
sipsock = -1;
}
if (sipsock < 0) {
sipsock = socket(AF_INET, SOCK_DGRAM, 0);
if (sipsock < 0) {
ast_log(LOG_WARNING, "Unable to create SIP socket: %s\n", strerror(errno));
ast_config_destroy(cfg);
return -1;
} else {
/* Allow SIP clients on the same host to access us: */
const int reuseFlag = 1;
setsockopt(sipsock, SOL_SOCKET, SO_REUSEADDR,
(const char*)&reuseFlag,
sizeof reuseFlag);
ast_enable_packet_fragmentation(sipsock);
if (bind(sipsock, (struct sockaddr *)&bindaddr, sizeof(bindaddr)) < 0) {
ast_log(LOG_WARNING, "Failed to bind to %s:%d: %s\n",
ast_inet_ntoa(bindaddr.sin_addr), ntohs(bindaddr.sin_port),
strerror(errno));
close(sipsock);
sipsock = -1;
} else {
ast_verb(2, "SIP Listening on %s:%d\n",
ast_inet_ntoa(bindaddr.sin_addr), ntohs(bindaddr.sin_port));
ast_netsock_set_qos(sipsock, global_tos_sip, global_cos_sip, "SIP");
}
}
}
if (stunaddr.sin_addr.s_addr != 0) {
ast_debug(1, "stun to %s:%d\n",
ast_inet_ntoa(stunaddr.sin_addr) , ntohs(stunaddr.sin_port));
ast_stun_request(sipsock, &stunaddr,
NULL, &externip);
ast_debug(1, "STUN sees us at %s:%d\n",
ast_inet_ntoa(externip.sin_addr) , ntohs(externip.sin_port));
}
ast_mutex_unlock(&netlock);
/* Start TCP server */
ast_tcptls_server_start(&sip_tcp_desc);
if (sip_tcp_desc.accept_fd == -1 && sip_tcp_desc.local_address.sin_family == AF_INET) {
/* TCP server start failed. Tell the admin */
ast_log(LOG_ERROR, "SIP TCP Server start failed. Not listening on TCP socket.\n");
sip_tcp_desc.local_address.sin_family = 0;
} else {
ast_debug(2, "SIP TCP server started\n");
}
/* Start TLS server if needed */
memcpy(sip_tls_desc.tls_cfg, &default_tls_cfg, sizeof(default_tls_cfg));
if (ast_ssl_setup(sip_tls_desc.tls_cfg)) {
ast_tcptls_server_start(&sip_tls_desc);
if (default_tls_cfg.enabled && sip_tls_desc.accept_fd == -1) {
ast_log(LOG_ERROR, "TLS Server start failed. Not listening on TLS socket.\n");
sip_tls_desc.tls_cfg = NULL;
}
} else if (sip_tls_desc.tls_cfg->enabled) {
sip_tls_desc.tls_cfg = NULL;
ast_log(LOG_WARNING, "SIP TLS server did not load because of errors.\n");
}
/* Add default domains - host name, IP address and IP:port
* Only do this if user added any sip domain with "localdomains"
* In order to *not* break backwards compatibility
* Some phones address us at IP only, some with additional port number
*/
if (auto_sip_domains) {
char temp[MAXHOSTNAMELEN];
/* First our default IP address */
if (bindaddr.sin_addr.s_addr)
add_sip_domain(ast_inet_ntoa(bindaddr.sin_addr), SIP_DOMAIN_AUTO, NULL);
else
ast_log(LOG_NOTICE, "Can't add wildcard IP address to domain list, please add IP address to domain manually.\n");
/* If TCP is running on a different IP than UDP, then add it too */
if (sip_tcp_desc.local_address.sin_addr.s_addr && !inaddrcmp(&bindaddr, &sip_tcp_desc.local_address))
add_sip_domain(ast_inet_ntoa(sip_tcp_desc.local_address.sin_addr), SIP_DOMAIN_AUTO, NULL);
/* If TLS is running on a differen IP than UDP and TCP, then add that too */
if (sip_tls_desc.local_address.sin_addr.s_addr && !inaddrcmp(&bindaddr, &sip_tls_desc.local_address) && inaddrcmp(&sip_tcp_desc.local_address, &sip_tls_desc.local_address))
add_sip_domain(ast_inet_ntoa(sip_tls_desc.local_address.sin_addr), SIP_DOMAIN_AUTO, NULL);
/* Our extern IP address, if configured */
if (externip.sin_addr.s_addr)
add_sip_domain(ast_inet_ntoa(externip.sin_addr), SIP_DOMAIN_AUTO, NULL);
/* Extern host name (NAT traversal support) */
if (!ast_strlen_zero(externhost))
add_sip_domain(externhost, SIP_DOMAIN_AUTO, NULL);
/* Our host name */
if (!gethostname(temp, sizeof(temp)))
add_sip_domain(temp, SIP_DOMAIN_AUTO, NULL);
}
/* Release configuration from memory */
ast_config_destroy(cfg);
/* Load the list of manual NOTIFY types to support */
if (notify_types)
ast_config_destroy(notify_types);
if ((notify_types = ast_config_load(notify_config, config_flags)) == CONFIG_STATUS_FILEINVALID) {
ast_log(LOG_ERROR, "Contents of %s are invalid and cannot be parsed.\n", notify_config);
notify_types = NULL;
}
/* Done, tell the manager */
manager_event(EVENT_FLAG_SYSTEM, "ChannelReload", "ChannelType: SIP\r\nReloadReason: %s\r\nRegistry_Count: %d\r\nPeer_Count: %d\r\n", channelreloadreason2txt(reason), registry_count, peer_count);
run_end = time(0);
ast_debug(4, "SIP reload_config done...Runtime= %d sec\n", (int)(run_end-run_start));
return 0;
}
static struct ast_udptl *sip_get_udptl_peer(struct ast_channel *chan)
{
struct sip_pvt *p;
struct ast_udptl *udptl = NULL;
p = chan->tech_pvt;
if (!p)
return NULL;
sip_pvt_lock(p);
if (p->udptl && ast_test_flag(&p->flags[0], SIP_CAN_REINVITE))
udptl = p->udptl;
sip_pvt_unlock(p);
return udptl;
}
static int sip_set_udptl_peer(struct ast_channel *chan, struct ast_udptl *udptl)
{
struct sip_pvt *p;
p = chan->tech_pvt;
if (!p)
return -1;
sip_pvt_lock(p);
if (udptl)
ast_udptl_get_peer(udptl, &p->udptlredirip);
else
memset(&p->udptlredirip, 0, sizeof(p->udptlredirip));
if (!ast_test_flag(&p->flags[0], SIP_GOTREFER)) {
if (!p->pendinginvite) {
ast_debug(3, "Sending reinvite on SIP '%s' - It's UDPTL soon redirected to IP %s:%d\n", p->callid, ast_inet_ntoa(udptl ? p->udptlredirip.sin_addr : p->ourip.sin_addr), udptl ? ntohs(p->udptlredirip.sin_port) : 0);
transmit_reinvite_with_sdp(p, TRUE, FALSE);
} else if (!ast_test_flag(&p->flags[0], SIP_PENDINGBYE)) {
ast_debug(3, "Deferring reinvite on SIP '%s' - It's UDPTL will be redirected to IP %s:%d\n", p->callid, ast_inet_ntoa(udptl ? p->udptlredirip.sin_addr : p->ourip.sin_addr), udptl ? ntohs(p->udptlredirip.sin_port) : 0);
ast_set_flag(&p->flags[0], SIP_NEEDREINVITE);
}
}
/* Reset lastrtprx timer */
p->lastrtprx = p->lastrtptx = time(NULL);
sip_pvt_unlock(p);
return 0;
}
static enum ast_rtp_glue_result sip_get_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance **instance)
{
struct sip_pvt *p = NULL;
enum ast_rtp_glue_result res = AST_RTP_GLUE_RESULT_LOCAL;
if (!(p = chan->tech_pvt)) {
return AST_RTP_GLUE_RESULT_FORBID;
}
sip_pvt_lock(p);
if (!(p->rtp)) {
sip_pvt_unlock(p);
return AST_RTP_GLUE_RESULT_FORBID;
}
ao2_ref(p->rtp, +1);
*instance = p->rtp;
if (!ast_test_flag(&p->flags[0], SIP_CAN_REINVITE_NAT)) {
res = AST_RTP_GLUE_RESULT_LOCAL;
} else if (ast_test_flag(&p->flags[0], SIP_CAN_REINVITE)) {
res = AST_RTP_GLUE_RESULT_REMOTE;
} else if (ast_test_flag(&global_jbconf, AST_JB_FORCED)) {
res = AST_RTP_GLUE_RESULT_FORBID;
}
sip_pvt_unlock(p);
return res;
}
static enum ast_rtp_glue_result sip_get_vrtp_peer(struct ast_channel *chan, struct ast_rtp_instance **instance)
{
struct sip_pvt *p = NULL;
enum ast_rtp_glue_result res = AST_RTP_GLUE_RESULT_FORBID;
if (!(p = chan->tech_pvt)) {
return AST_RTP_GLUE_RESULT_FORBID;
}
sip_pvt_lock(p);
if (!(p->vrtp)) {
sip_pvt_unlock(p);
return AST_RTP_GLUE_RESULT_FORBID;
}
ao2_ref(p->vrtp, +1);
*instance = p->vrtp;
if (ast_test_flag(&p->flags[0], SIP_CAN_REINVITE)) {
res = AST_RTP_GLUE_RESULT_REMOTE;
}
sip_pvt_unlock(p);
return res;
}
static enum ast_rtp_glue_result sip_get_trtp_peer(struct ast_channel *chan, struct ast_rtp_instance **instance)
{
struct sip_pvt *p = NULL;
enum ast_rtp_glue_result res = AST_RTP_GLUE_RESULT_FORBID;
if (!(p = chan->tech_pvt)) {
return AST_RTP_GLUE_RESULT_FORBID;
}
sip_pvt_lock(p);
if (!(p->trtp)) {
sip_pvt_unlock(p);
return AST_RTP_GLUE_RESULT_FORBID;
}
ao2_ref(p->trtp, +1);
*instance = p->trtp;
if (ast_test_flag(&p->flags[0], SIP_CAN_REINVITE)) {
res = AST_RTP_GLUE_RESULT_REMOTE;
}
sip_pvt_unlock(p);
return res;
}
static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance *instance, struct ast_rtp_instance *vinstance, struct ast_rtp_instance *tinstance, int codecs, int nat_active)
{
struct sip_pvt *p;
int changed = 0;
p = chan->tech_pvt;
if (!p)
return -1;
/* Disable early RTP bridge */
if (chan->_state != AST_STATE_UP && !sip_cfg.directrtpsetup) /* We are in early state */
return 0;
sip_pvt_lock(p);
if (p->alreadygone) {
/* If we're destroyed, don't bother */
sip_pvt_unlock(p);
return 0;
}
/* if this peer cannot handle reinvites of the media stream to devices
that are known to be behind a NAT, then stop the process now
*/
if (nat_active && !ast_test_flag(&p->flags[0], SIP_CAN_REINVITE_NAT)) {
sip_pvt_unlock(p);
return 0;
}
if (instance) {
changed |= ast_rtp_instance_get_remote_address(instance, &p->redirip);
} else if (p->redirip.sin_addr.s_addr || ntohs(p->redirip.sin_port) != 0) {
memset(&p->redirip, 0, sizeof(p->redirip));
changed = 1;
}
if (vinstance) {
changed |= ast_rtp_instance_get_remote_address(vinstance, &p->vredirip);
} else if (p->vredirip.sin_addr.s_addr || ntohs(p->vredirip.sin_port) != 0) {
memset(&p->vredirip, 0, sizeof(p->vredirip));
changed = 1;
}
if (tinstance) {
changed |= ast_rtp_instance_get_remote_address(tinstance, &p->tredirip);
} else if (p->tredirip.sin_addr.s_addr || ntohs(p->tredirip.sin_port) != 0) {
memset(&p->tredirip, 0, sizeof(p->tredirip));
changed = 1;
}
if (codecs && (p->redircodecs != codecs)) {
p->redircodecs = codecs;
changed = 1;
}
if (changed && !ast_test_flag(&p->flags[0], SIP_GOTREFER) && !ast_test_flag(&p->flags[0], SIP_DEFER_BYE_ON_TRANSFER)) {
if (chan->_state != AST_STATE_UP) { /* We are in early state */
if (p->do_history)
append_history(p, "ExtInv", "Initial invite sent with remote bridge proposal.");
ast_debug(1, "Early remote bridge setting SIP '%s' - Sending media to %s\n", p->callid, ast_inet_ntoa(instance ? p->redirip.sin_addr : p->ourip.sin_addr));
} else if (!p->pendinginvite) { /* We are up, and have no outstanding invite */
ast_debug(3, "Sending reinvite on SIP '%s' - It's audio soon redirected to IP %s\n", p->callid, ast_inet_ntoa(instance ? p->redirip.sin_addr : p->ourip.sin_addr));
transmit_reinvite_with_sdp(p, FALSE, FALSE);
} else if (!ast_test_flag(&p->flags[0], SIP_PENDINGBYE)) {
ast_debug(3, "Deferring reinvite on SIP '%s' - It's audio will be redirected to IP %s\n", p->callid, ast_inet_ntoa(instance ? p->redirip.sin_addr : p->ourip.sin_addr));
/* We have a pending Invite. Send re-invite when we're done with the invite */
ast_set_flag(&p->flags[0], SIP_NEEDREINVITE);
}
}
/* Reset lastrtprx timer */
p->lastrtprx = p->lastrtptx = time(NULL);
sip_pvt_unlock(p);
return 0;
}
static int sip_get_codec(struct ast_channel *chan)
{
struct sip_pvt *p = chan->tech_pvt;
return p->peercapability ? p->peercapability : p->capability;
}
static struct ast_rtp_glue sip_rtp_glue = {
.type = "SIP",
.get_rtp_info = sip_get_rtp_peer,
.get_vrtp_info = sip_get_vrtp_peer,
.get_trtp_info = sip_get_trtp_peer,
.update_peer = sip_set_rtp_peer,
.get_codec = sip_get_codec,
};
static char *app_dtmfmode = "SIPDtmfMode";
static char *app_sipaddheader = "SIPAddHeader";
static char *app_sipremoveheader = "SIPRemoveHeader";
/*! \brief Set the DTMFmode for an outbound SIP call (application) */
static int sip_dtmfmode(struct ast_channel *chan, void *data)
{
struct sip_pvt *p;
char *mode = data;
if (!data) {
ast_log(LOG_WARNING, "This application requires the argument: info, inband, rfc2833\n");
return 0;
}
ast_channel_lock(chan);
if (!IS_SIP_TECH(chan->tech)) {
ast_log(LOG_WARNING, "Call this application only on SIP incoming calls\n");
ast_channel_unlock(chan);
return 0;
}
p = chan->tech_pvt;
if (!p) {
ast_channel_unlock(chan);
return 0;
}
sip_pvt_lock(p);
if (!strcasecmp(mode, "info")) {
ast_clear_flag(&p->flags[0], SIP_DTMF);
ast_set_flag(&p->flags[0], SIP_DTMF_INFO);
p->jointnoncodeccapability &= ~AST_RTP_DTMF;
} else if (!strcasecmp(mode, "shortinfo")) {
ast_clear_flag(&p->flags[0], SIP_DTMF);
ast_set_flag(&p->flags[0], SIP_DTMF_SHORTINFO);
p->jointnoncodeccapability &= ~AST_RTP_DTMF;
} else if (!strcasecmp(mode, "rfc2833")) {
ast_clear_flag(&p->flags[0], SIP_DTMF);
ast_set_flag(&p->flags[0], SIP_DTMF_RFC2833);
p->jointnoncodeccapability |= AST_RTP_DTMF;
} else if (!strcasecmp(mode, "inband")) {
ast_clear_flag(&p->flags[0], SIP_DTMF);
ast_set_flag(&p->flags[0], SIP_DTMF_INBAND);
p->jointnoncodeccapability &= ~AST_RTP_DTMF;
} else
ast_log(LOG_WARNING, "I don't know about this dtmf mode: %s\n", mode);
if (p->rtp)
ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_DTMF, ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833);
if (ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_INBAND) {
if (!p->vad) {
p->vad = ast_dsp_new();
ast_dsp_set_features(p->vad, DSP_FEATURE_DIGIT_DETECT);
}
} else {
if (p->vad) {
ast_dsp_free(p->vad);
p->vad = NULL;
}
}
sip_pvt_unlock(p);
ast_channel_unlock(chan);
return 0;
}
/*! \brief Add a SIP header to an outbound INVITE */
static int sip_addheader(struct ast_channel *chan, void *data)
{
int no = 0;
int ok = FALSE;
char varbuf[30];
char *inbuf = data, *subbuf;
if (ast_strlen_zero(inbuf)) {
ast_log(LOG_WARNING, "This application requires the argument: Header\n");
return 0;
}
ast_channel_lock(chan);
/* Check for headers */
while (!ok && no <= 50) {
no++;
snprintf(varbuf, sizeof(varbuf), "__SIPADDHEADER%.2d", no);
/* Compare without the leading underscores */
if ((pbx_builtin_getvar_helper(chan, (const char *) varbuf + 2) == (const char *) NULL)) {
ok = TRUE;
}
}
if (ok) {
size_t len = strlen(inbuf);
subbuf = alloca(len + 1);
ast_get_encoded_str(inbuf, subbuf, len + 1);
pbx_builtin_setvar_helper(chan, varbuf, subbuf);
if (sipdebug) {
ast_debug(1, "SIP Header added \"%s\" as %s\n", inbuf, varbuf);
}
} else {
ast_log(LOG_WARNING, "Too many SIP headers added, max 50\n");
}
ast_channel_unlock(chan);
return 0;
}
/*! \brief Remove SIP headers added previously with SipAddHeader application */
static int sip_removeheader(struct ast_channel *chan, void *data)
{
struct ast_var_t *newvariable;
struct varshead *headp;
int removeall = 0;
char *inbuf = (char *) data;
if (ast_strlen_zero(inbuf)) {
removeall = 1;
}
ast_channel_lock(chan);
headp=&chan->varshead;
AST_LIST_TRAVERSE_SAFE_BEGIN (headp, newvariable, entries) {
if (strncasecmp(ast_var_name(newvariable), "SIPADDHEADER", strlen("SIPADDHEADER")) == 0) {
if (removeall || (!strncasecmp(ast_var_value(newvariable),inbuf,strlen(inbuf)))) {
if (sipdebug)
ast_log(LOG_DEBUG,"removing SIP Header \"%s\" as %s\n",
ast_var_value(newvariable),
ast_var_name(newvariable));
AST_LIST_REMOVE_CURRENT(entries);
ast_var_delete(newvariable);
}
}
}
AST_LIST_TRAVERSE_SAFE_END;
ast_channel_unlock(chan);
return 0;
}
/*! \brief Transfer call before connect with a 302 redirect
\note Called by the transfer() dialplan application through the sip_transfer()
pbx interface function if the call is in ringing state
\todo Fix this function so that we wait for reply to the REFER and
react to errors, denials or other issues the other end might have.
*/
static int sip_sipredirect(struct sip_pvt *p, const char *dest)
{
char *cdest;
char *extension, *host, *port;
char tmp[80];
cdest = ast_strdupa(dest);
extension = strsep(&cdest, "@");
host = strsep(&cdest, ":");
port = strsep(&cdest, ":");
if (ast_strlen_zero(extension)) {
ast_log(LOG_ERROR, "Missing mandatory argument: extension\n");
return 0;
}
/* we'll issue the redirect message here */
if (!host) {
char *localtmp;
ast_copy_string(tmp, get_header(&p->initreq, "To"), sizeof(tmp));
if (ast_strlen_zero(tmp)) {
ast_log(LOG_ERROR, "Cannot retrieve the 'To' header from the original SIP request!\n");
return 0;
}
if ( ( (localtmp = strcasestr(tmp, "sip:")) || (localtmp = strcasestr(tmp, "sips:")) )
&& (localtmp = strchr(localtmp, '@'))) {
char lhost[80], lport[80];
memset(lhost, 0, sizeof(lhost));
memset(lport, 0, sizeof(lport));
localtmp++;
/* This is okey because lhost and lport are as big as tmp */
sscanf(localtmp, "%[^<>:; ]:%[^<>:; ]", lhost, lport);
if (ast_strlen_zero(lhost)) {
ast_log(LOG_ERROR, "Can't find the host address\n");
return 0;
}
host = ast_strdupa(lhost);
if (!ast_strlen_zero(lport)) {
port = ast_strdupa(lport);
}
}
}
ast_string_field_build(p, our_contact, "Transfer <sip:%s@%s%s%s>", extension, host, port ? ":" : "", port ? port : "");
transmit_response_reliable(p, "302 Moved Temporarily", &p->initreq);
sip_scheddestroy(p, SIP_TRANS_TIMEOUT); /* Make sure we stop send this reply. */
sip_alreadygone(p);
if (p->owner) {
enum ast_control_transfer message = AST_TRANSFER_SUCCESS;
ast_queue_control_data(p->owner, AST_CONTROL_TRANSFER, &message, sizeof(message));
}
/* hangup here */
return 0;
}
/*! \brief Send a poke to all known peers */
static void sip_poke_all_peers(void)
{
int ms = 0, num = 0;
struct ao2_iterator i;
struct sip_peer *peer;
i = ao2_iterator_init(peers, 0);
if (!speerobjs) /* No peers, just give up */
return;
while ((peer = ao2_t_iterator_next(&i, "iterate thru peers table"))) {
ao2_lock(peer);
if (num == global_qualify_peers) {
ms += global_qualify_gap;
num = 0;
} else {
num++;
}
AST_SCHED_REPLACE_UNREF(peer->pokeexpire, sched, ms, sip_poke_peer_s, peer,
unref_peer(_data, "removing poke peer ref"),
unref_peer(peer, "removing poke peer ref"),
ref_peer(peer, "adding poke peer ref"));
ao2_unlock(peer);
unref_peer(peer, "toss iterator peer ptr");
}
}
/*! \brief Send all known registrations */
static void sip_send_all_registers(void)
{
int ms;
int regspacing;
if (!regobjs)
return;
regspacing = default_expiry * 1000/regobjs;
if (regspacing > 100)
regspacing = 100;
ms = regspacing;
ASTOBJ_CONTAINER_TRAVERSE(&regl, 1, do {
ASTOBJ_WRLOCK(iterator);
ms += regspacing;
AST_SCHED_REPLACE_UNREF(iterator->expire, sched, ms, sip_reregister, iterator,
registry_unref(_data, "REPLACE sched del decs the refcount"),
registry_unref(iterator, "REPLACE sched add failure decs the refcount"),
registry_addref(iterator, "REPLACE sched add incs the refcount"));
ASTOBJ_UNLOCK(iterator);
} while (0)
);
}
/*! \brief Send all MWI subscriptions */
static void sip_send_all_mwi_subscriptions(void)
{
ASTOBJ_CONTAINER_TRAVERSE(&submwil, 1, do {
ASTOBJ_WRLOCK(iterator);
AST_SCHED_DEL(sched, iterator->resub);
if ((iterator->resub = ast_sched_add(sched, 1, sip_subscribe_mwi_do, ASTOBJ_REF(iterator))) < 0) {
ASTOBJ_UNREF(iterator, sip_subscribe_mwi_destroy);
}
ASTOBJ_UNLOCK(iterator);
} while (0));
}
/*! \brief Reload module */
static int sip_do_reload(enum channelreloadreason reason)
{
time_t start_poke, end_poke;
reload_config(reason);
ast_sched_dump(sched);
start_poke = time(0);
/* Prune peers who still are supposed to be deleted */
ao2_t_callback(peers, OBJ_NODATA | OBJ_UNLINK | OBJ_MULTIPLE, peer_is_marked, NULL,
"callback to remove marked peers");
ast_debug(4, "--------------- Done destroying pruned peers\n");
/* Send qualify (OPTIONS) to all peers */
sip_poke_all_peers();
/* Register with all services */
sip_send_all_registers();
sip_send_all_mwi_subscriptions();
end_poke = time(0);
ast_debug(4, "do_reload finished. peer poke/prune reg contact time = %d sec.\n", (int)(end_poke-start_poke));
ast_debug(4, "--------------- SIP reload done\n");
return 0;
}
/*! \brief Force reload of module from cli */
static char *sip_reload(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
switch (cmd) {
case CLI_INIT:
e->command = "sip reload";
e->usage =
"Usage: sip reload\n"
" Reloads SIP configuration from sip.conf\n";
return NULL;
case CLI_GENERATE:
return NULL;
}
ast_mutex_lock(&sip_reload_lock);
if (sip_reloading)
ast_verbose("Previous SIP reload not yet done\n");
else {
sip_reloading = TRUE;
sip_reloadreason = (a && a->fd) ? CHANNEL_CLI_RELOAD : CHANNEL_MODULE_RELOAD;
}
ast_mutex_unlock(&sip_reload_lock);
restart_monitor();
return CLI_SUCCESS;
}
/*! \brief Part of Asterisk module interface */
static int reload(void)
{
if (sip_reload(0, 0, NULL))
return 0;
return 1;
}
/*! \brief SIP Cli commands definition */
static struct ast_cli_entry cli_sip[] = {
AST_CLI_DEFINE(sip_show_channels, "List active SIP channels or subscriptions"),
AST_CLI_DEFINE(sip_show_channelstats, "List statistics for active SIP channels"),
AST_CLI_DEFINE(sip_show_domains, "List our local SIP domains"),
AST_CLI_DEFINE(sip_show_inuse, "List all inuse/limits"),
AST_CLI_DEFINE(sip_show_objects, "List all SIP object allocations"),
AST_CLI_DEFINE(sip_show_peers, "List defined SIP peers"),
AST_CLI_DEFINE(sip_show_registry, "List SIP registration status"),
AST_CLI_DEFINE(sip_unregister, "Unregister (force expiration) a SIP peer from the registry"),
AST_CLI_DEFINE(sip_show_settings, "Show SIP global settings"),
AST_CLI_DEFINE(sip_show_mwi, "Show MWI subscriptions"),
AST_CLI_DEFINE(sip_cli_notify, "Send a notify packet to a SIP peer"),
AST_CLI_DEFINE(sip_show_channel, "Show detailed SIP channel info"),
AST_CLI_DEFINE(sip_show_history, "Show SIP dialog history"),
AST_CLI_DEFINE(sip_show_peer, "Show details on specific SIP peer"),
AST_CLI_DEFINE(sip_show_users, "List defined SIP users"),
AST_CLI_DEFINE(sip_show_user, "Show details on specific SIP user"),
AST_CLI_DEFINE(sip_qualify_peer, "Send an OPTIONS packet to a peer"),
AST_CLI_DEFINE(sip_show_sched, "Present a report on the status of the sched queue"),
AST_CLI_DEFINE(sip_prune_realtime, "Prune cached Realtime users/peers"),
AST_CLI_DEFINE(sip_do_debug, "Enable/Disable SIP debugging"),
AST_CLI_DEFINE(sip_set_history, "Enable/Disable SIP history"),
AST_CLI_DEFINE(sip_reload, "Reload SIP configuration"),
AST_CLI_DEFINE(sip_show_tcp, "List TCP Connections")
};
/*! \brief PBX load module - initialization */
static int load_module(void)
{
ast_verbose("SIP channel loading...\n");
/* the fact that ao2_containers can't resize automatically is a major worry! */
/* if the number of objects gets above MAX_XXX_BUCKETS, things will slow down */
peers = ao2_t_container_alloc(hash_peer_size, peer_hash_cb, peer_cmp_cb, "allocate peers");
peers_by_ip = ao2_t_container_alloc(hash_peer_size, peer_iphash_cb, peer_ipcmp_cb, "allocate peers_by_ip");
dialogs = ao2_t_container_alloc(hash_dialog_size, dialog_hash_cb, dialog_cmp_cb, "allocate dialogs");
ASTOBJ_CONTAINER_INIT(&regl); /* Registry object list -- not searched for anything */
ASTOBJ_CONTAINER_INIT(&submwil); /* MWI subscription object list */
if (!(sched = sched_context_create())) {
ast_log(LOG_ERROR, "Unable to create scheduler context\n");
return AST_MODULE_LOAD_FAILURE;
}
if (!(io = io_context_create())) {
ast_log(LOG_ERROR, "Unable to create I/O context\n");
sched_context_destroy(sched);
return AST_MODULE_LOAD_FAILURE;
}
sip_reloadreason = CHANNEL_MODULE_LOAD;
if(reload_config(sip_reloadreason)) /* Load the configuration from sip.conf */
return AST_MODULE_LOAD_DECLINE;
/* Prepare the version that does not require DTMF BEGIN frames.
* We need to use tricks such as memcpy and casts because the variable
* has const fields.
*/
memcpy(&sip_tech_info, &sip_tech, sizeof(sip_tech));
memset((void *) &sip_tech_info.send_digit_begin, 0, sizeof(sip_tech_info.send_digit_begin));
/* Make sure we can register our sip channel type */
if (ast_channel_register(&sip_tech)) {
ast_log(LOG_ERROR, "Unable to register channel type 'SIP'\n");
io_context_destroy(io);
sched_context_destroy(sched);
return AST_MODULE_LOAD_FAILURE;
}
/* Register all CLI functions for SIP */
ast_cli_register_multiple(cli_sip, ARRAY_LEN(cli_sip));
/* Tell the UDPTL subdriver that we're here */
ast_udptl_proto_register(&sip_udptl);
/* Tell the RTP engine about our RTP glue */
ast_rtp_glue_register(&sip_rtp_glue);
/* Register dialplan applications */
ast_register_application_xml(app_dtmfmode, sip_dtmfmode);
ast_register_application_xml(app_sipaddheader, sip_addheader);
ast_register_application_xml(app_sipremoveheader, sip_removeheader);
/* Register dialplan functions */
ast_custom_function_register(&sip_header_function);
ast_custom_function_register(&sippeer_function);
ast_custom_function_register(&sipchaninfo_function);
ast_custom_function_register(&checksipdomain_function);
/* Register manager commands */
ast_manager_register2("SIPpeers", EVENT_FLAG_SYSTEM | EVENT_FLAG_REPORTING, manager_sip_show_peers,
"List SIP peers (text format)", mandescr_show_peers);
ast_manager_register2("SIPshowpeer", EVENT_FLAG_SYSTEM | EVENT_FLAG_REPORTING, manager_sip_show_peer,
"Show SIP peer (text format)", mandescr_show_peer);
ast_manager_register2("SIPqualifypeer", EVENT_FLAG_SYSTEM | EVENT_FLAG_REPORTING, manager_sip_qualify_peer,
"Show SIP peer (text format)", mandescr_show_peer); /*! \todo Fix this XXX This must be all wrong XXXX */
ast_manager_register2("SIPshowregistry", EVENT_FLAG_SYSTEM | EVENT_FLAG_REPORTING, manager_show_registry,
"Show SIP registrations (text format)", mandescr_show_registry);
ast_manager_register2("SIPnotify", EVENT_FLAG_SYSTEM, manager_sipnotify,
"Send a SIP notify", mandescr_sipnotify);
sip_poke_all_peers();
sip_send_all_registers();
sip_send_all_mwi_subscriptions();
/* And start the monitor for the first time */
restart_monitor();
ast_realtime_require_field(ast_check_realtime("sipregs") ? "sipregs" : "sippeers",
"name", RQ_CHAR, 10,
"ipaddr", RQ_CHAR, 15,
"port", RQ_UINTEGER2, 5,
"regseconds", RQ_INTEGER4, 11,
"defaultuser", RQ_CHAR, 10,
"fullcontact", RQ_CHAR, 35,
"regserver", RQ_CHAR, 20,
"useragent", RQ_CHAR, 20,
"lastms", RQ_INTEGER4, 11,
SENTINEL);
return AST_MODULE_LOAD_SUCCESS;
}
/*! \brief PBX unload module API */
static int unload_module(void)
{
struct sip_pvt *p;
struct sip_threadinfo *th;
struct ast_context *con;
struct ao2_iterator i;
ast_sched_dump(sched);
/* First, take us out of the channel type list */
ast_channel_unregister(&sip_tech);
/* Unregister dial plan functions */
ast_custom_function_unregister(&sipchaninfo_function);
ast_custom_function_unregister(&sippeer_function);
ast_custom_function_unregister(&sip_header_function);
ast_custom_function_unregister(&checksipdomain_function);
/* Unregister dial plan applications */
ast_unregister_application(app_dtmfmode);
ast_unregister_application(app_sipaddheader);
ast_unregister_application(app_sipremoveheader);
/* Unregister CLI commands */
ast_cli_unregister_multiple(cli_sip, ARRAY_LEN(cli_sip));
/* Disconnect from UDPTL */
ast_udptl_proto_unregister(&sip_udptl);
/* Disconnect from RTP engine */
ast_rtp_glue_unregister(&sip_rtp_glue);
/* Unregister AMI actions */
ast_manager_unregister("SIPpeers");
ast_manager_unregister("SIPshowpeer");
ast_manager_unregister("SIPqualifypeer");
ast_manager_unregister("SIPshowregistry");
ast_manager_unregister("SIPnotify");
/* Kill TCP/TLS server threads */
if (sip_tcp_desc.master)
ast_tcptls_server_stop(&sip_tcp_desc);
if (sip_tls_desc.master)
ast_tcptls_server_stop(&sip_tls_desc);
/* Kill all existing TCP/TLS threads */
AST_LIST_LOCK(&threadl);
AST_LIST_TRAVERSE_SAFE_BEGIN(&threadl, th, list) {
pthread_t thread = th->threadid;
th->stop = 1;
AST_LIST_UNLOCK(&threadl);
pthread_kill(thread, SIGURG);
pthread_join(thread, NULL);
AST_LIST_LOCK(&threadl);
}
AST_LIST_TRAVERSE_SAFE_END;
AST_LIST_UNLOCK(&threadl);
/* Hangup all dialogs if they have an owner */
i = ao2_iterator_init(dialogs, 0);
while ((p = ao2_t_iterator_next(&i, "iterate thru dialogs"))) {
if (p->owner)
ast_softhangup(p->owner, AST_SOFTHANGUP_APPUNLOAD);
ao2_t_ref(p, -1, "toss dialog ptr from iterator_next");
}
ast_mutex_lock(&monlock);
if (monitor_thread && (monitor_thread != AST_PTHREADT_STOP) && (monitor_thread != AST_PTHREADT_NULL)) {
pthread_cancel(monitor_thread);
pthread_kill(monitor_thread, SIGURG);
pthread_join(monitor_thread, NULL);
}
monitor_thread = AST_PTHREADT_STOP;
ast_mutex_unlock(&monlock);
/* Destroy all the dialogs and free their memory */
i = ao2_iterator_init(dialogs, 0);
while ((p = ao2_t_iterator_next(&i, "iterate thru dialogs"))) {
dialog_unlink_all(p, TRUE, TRUE);
ao2_t_ref(p, -1, "throw away iterator result");
}
/* Free memory for local network address mask */
ast_free_ha(localaddr);
clear_realm_authentication(authl);
if (default_tls_cfg.certfile)
ast_free(default_tls_cfg.certfile);
if (default_tls_cfg.pvtfile)
ast_free(default_tls_cfg.pvtfile);
if (default_tls_cfg.cipher)
ast_free(default_tls_cfg.cipher);
if (default_tls_cfg.cafile)
ast_free(default_tls_cfg.cafile);
if (default_tls_cfg.capath)
ast_free(default_tls_cfg.capath);
ASTOBJ_CONTAINER_DESTROYALL(&regl, sip_registry_destroy);
ASTOBJ_CONTAINER_DESTROY(&regl);
ASTOBJ_CONTAINER_DESTROYALL(&submwil, sip_subscribe_mwi_destroy);
ASTOBJ_CONTAINER_DESTROY(&submwil);
ao2_t_ref(peers, -1, "unref the peers table");
ao2_t_ref(peers_by_ip, -1, "unref the peers_by_ip table");
ao2_t_ref(dialogs, -1, "unref the dialogs table");
clear_sip_domains();
close(sipsock);
sched_context_destroy(sched);
con = ast_context_find(used_context);
if (con)
ast_context_destroy(con, "SIP");
ast_unload_realtime("sipregs");
ast_unload_realtime("sippeers");
return 0;
}
AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_DEFAULT, "Session Initiation Protocol (SIP)",
.load = load_module,
.unload = unload_module,
.reload = reload,
);
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