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asterisk/channels/pjsip/dialplan_functions.c
Matt Jordan 4a58261694 git migration: Refactor the ASTERISK_FILE_VERSION macro 
Git does not support the ability to replace a token with a version
string during check-in. While it does have support for replacing a
token on clone, this is somewhat sub-optimal: the token is replaced
with the object hash, which is not particularly easy for human
consumption. What's more, in practice, the source file version was often
not terribly useful. Generally, when triaging bugs, the overall version
of Asterisk is far more useful than an individual SVN version of a file. As a
result, this patch removes Asterisk's support for showing source file
versions.

Specifically, it does the following:

* Rename ASTERISK_FILE_VERSION macro to ASTERISK_REGISTER_FILE, and
  remove passing the version in with the macro. Other facilities
  than 'core show file version' make use of the file names, such as
  setting a debug level only on a specific file. As such, the act of
  registering source files with the Asterisk core still has use. The
  macro rename now reflects the new macro purpose.

* main/asterisk:
  - Refactor the file_version structure to reflect that it no longer
    tracks a version field.
  - Remove the "core show file version" CLI command. Without the file
    version, it is no longer useful.
  - Remove the ast_file_version_find function. The file version is no
    longer tracked.
  - Rename ast_register_file_version/ast_unregister_file_version to
    ast_register_file/ast_unregister_file, respectively.

* main/manager: Remove value from the Version key of the ModuleCheck
  Action. The actual key itself has not been removed, as doing so would
  absolutely constitute a backwards incompatible change. However, since
  the file version is no longer tracked, there is no need to attempt to
  include it in the Version key.

* UPGRADE: Add notes for:
  - Modification to the ModuleCheck AMI Action
  - Removal of the "core show file version" CLI command

Change-Id: I6cf0ff280e1668bf4957dc21f32a5ff43444a40e
2015-04-13 03:48:57 -04:00

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/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 2013, Digium, Inc.
*
* 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
*
* \author \verbatim Joshua Colp <jcolp@digium.com> \endverbatim
* \author \verbatim Matt Jordan <mjordan@digium.com> \endverbatim
*
* \ingroup functions
*
* \brief PJSIP channel dialplan functions
*/
/*** MODULEINFO
<support_level>core</support_level>
***/
/*** DOCUMENTATION
<function name="PJSIP_DIAL_CONTACTS" language="en_US">
<synopsis>
Return a dial string for dialing all contacts on an AOR.
</synopsis>
<syntax>
<parameter name="endpoint" required="true">
<para>Name of the endpoint</para>
</parameter>
<parameter name="aor" required="false">
<para>Name of an AOR to use, if not specified the configured AORs on the endpoint are used</para>
</parameter>
<parameter name="request_user" required="false">
<para>Optional request user to use in the request URI</para>
</parameter>
</syntax>
<description>
<para>Returns a properly formatted dial string for dialing all contacts on an AOR.</para>
</description>
</function>
<function name="PJSIP_MEDIA_OFFER" language="en_US">
<synopsis>
Media and codec offerings to be set on an outbound SIP channel prior to dialing.
</synopsis>
<syntax>
<parameter name="media" required="true">
<para>types of media offered</para>
</parameter>
</syntax>
<description>
<para>Returns the codecs offered based upon the media choice</para>
</description>
</function>
<info name="PJSIPCHANNEL" language="en_US" tech="PJSIP">
<enumlist>
<enum name="rtp">
<para>R/O Retrieve media related information.</para>
<parameter name="type" required="true">
<para>When <replaceable>rtp</replaceable> is specified, the
<literal>type</literal> parameter must be provided. It specifies
which RTP parameter to read.</para>
<enumlist>
<enum name="src">
<para>Retrieve the local address for RTP.</para>
</enum>
<enum name="dest">
<para>Retrieve the remote address for RTP.</para>
</enum>
<enum name="direct">
<para>If direct media is enabled, this address is the remote address
used for RTP.</para>
</enum>
<enum name="secure">
<para>Whether or not the media stream is encrypted.</para>
<enumlist>
<enum name="0">
<para>The media stream is not encrypted.</para>
</enum>
<enum name="1">
<para>The media stream is encrypted.</para>
</enum>
</enumlist>
</enum>
<enum name="hold">
<para>Whether or not the media stream is currently restricted
due to a call hold.</para>
<enumlist>
<enum name="0">
<para>The media stream is not held.</para>
</enum>
<enum name="1">
<para>The media stream is held.</para>
</enum>
</enumlist>
</enum>
</enumlist>
</parameter>
<parameter name="media_type" required="false">
<para>When <replaceable>rtp</replaceable> is specified, the
<literal>media_type</literal> parameter may be provided. It specifies
which media stream the chosen RTP parameter should be retrieved
from.</para>
<enumlist>
<enum name="audio">
<para>Retrieve information from the audio media stream.</para>
<note><para>If not specified, <literal>audio</literal> is used
by default.</para></note>
</enum>
<enum name="video">
<para>Retrieve information from the video media stream.</para>
</enum>
</enumlist>
</parameter>
</enum>
<enum name="rtcp">
<para>R/O Retrieve RTCP statistics.</para>
<parameter name="statistic" required="true">
<para>When <replaceable>rtcp</replaceable> is specified, the
<literal>statistic</literal> parameter must be provided. It specifies
which RTCP statistic parameter to read.</para>
<enumlist>
<enum name="all">
<para>Retrieve a summary of all RTCP statistics.</para>
<para>The following data items are returned in a semi-colon
delineated list:</para>
<enumlist>
<enum name="ssrc">
<para>Our Synchronization Source identifier</para>
</enum>
<enum name="themssrc">
<para>Their Synchronization Source identifier</para>
</enum>
<enum name="lp">
<para>Our lost packet count</para>
</enum>
<enum name="rxjitter">
<para>Received packet jitter</para>
</enum>
<enum name="rxcount">
<para>Received packet count</para>
</enum>
<enum name="txjitter">
<para>Transmitted packet jitter</para>
</enum>
<enum name="txcount">
<para>Transmitted packet count</para>
</enum>
<enum name="rlp">
<para>Remote lost packet count</para>
</enum>
<enum name="rtt">
<para>Round trip time</para>
</enum>
</enumlist>
</enum>
<enum name="all_jitter">
<para>Retrieve a summary of all RTCP Jitter statistics.</para>
<para>The following data items are returned in a semi-colon
delineated list:</para>
<enumlist>
<enum name="minrxjitter">
<para>Our minimum jitter</para>
</enum>
<enum name="maxrxjitter">
<para>Our max jitter</para>
</enum>
<enum name="avgrxjitter">
<para>Our average jitter</para>
</enum>
<enum name="stdevrxjitter">
<para>Our jitter standard deviation</para>
</enum>
<enum name="reported_minjitter">
<para>Their minimum jitter</para>
</enum>
<enum name="reported_maxjitter">
<para>Their max jitter</para>
</enum>
<enum name="reported_avgjitter">
<para>Their average jitter</para>
</enum>
<enum name="reported_stdevjitter">
<para>Their jitter standard deviation</para>
</enum>
</enumlist>
</enum>
<enum name="all_loss">
<para>Retrieve a summary of all RTCP packet loss statistics.</para>
<para>The following data items are returned in a semi-colon
delineated list:</para>
<enumlist>
<enum name="minrxlost">
<para>Our minimum lost packets</para>
</enum>
<enum name="maxrxlost">
<para>Our max lost packets</para>
</enum>
<enum name="avgrxlost">
<para>Our average lost packets</para>
</enum>
<enum name="stdevrxlost">
<para>Our lost packets standard deviation</para>
</enum>
<enum name="reported_minlost">
<para>Their minimum lost packets</para>
</enum>
<enum name="reported_maxlost">
<para>Their max lost packets</para>
</enum>
<enum name="reported_avglost">
<para>Their average lost packets</para>
</enum>
<enum name="reported_stdevlost">
<para>Their lost packets standard deviation</para>
</enum>
</enumlist>
</enum>
<enum name="all_rtt">
<para>Retrieve a summary of all RTCP round trip time information.</para>
<para>The following data items are returned in a semi-colon
delineated list:</para>
<enumlist>
<enum name="minrtt">
<para>Minimum round trip time</para>
</enum>
<enum name="maxrtt">
<para>Maximum round trip time</para>
</enum>
<enum name="avgrtt">
<para>Average round trip time</para>
</enum>
<enum name="stdevrtt">
<para>Standard deviation round trip time</para>
</enum>
</enumlist>
</enum>
<enum name="txcount"><para>Transmitted packet count</para></enum>
<enum name="rxcount"><para>Received packet count</para></enum>
<enum name="txjitter"><para>Transmitted packet jitter</para></enum>
<enum name="rxjitter"><para>Received packet jitter</para></enum>
<enum name="remote_maxjitter"><para>Their max jitter</para></enum>
<enum name="remote_minjitter"><para>Their minimum jitter</para></enum>
<enum name="remote_normdevjitter"><para>Their average jitter</para></enum>
<enum name="remote_stdevjitter"><para>Their jitter standard deviation</para></enum>
<enum name="local_maxjitter"><para>Our max jitter</para></enum>
<enum name="local_minjitter"><para>Our minimum jitter</para></enum>
<enum name="local_normdevjitter"><para>Our average jitter</para></enum>
<enum name="local_stdevjitter"><para>Our jitter standard deviation</para></enum>
<enum name="txploss"><para>Transmitted packet loss</para></enum>
<enum name="rxploss"><para>Received packet loss</para></enum>
<enum name="remote_maxrxploss"><para>Their max lost packets</para></enum>
<enum name="remote_minrxploss"><para>Their minimum lost packets</para></enum>
<enum name="remote_normdevrxploss"><para>Their average lost packets</para></enum>
<enum name="remote_stdevrxploss"><para>Their lost packets standard deviation</para></enum>
<enum name="local_maxrxploss"><para>Our max lost packets</para></enum>
<enum name="local_minrxploss"><para>Our minimum lost packets</para></enum>
<enum name="local_normdevrxploss"><para>Our average lost packets</para></enum>
<enum name="local_stdevrxploss"><para>Our lost packets standard deviation</para></enum>
<enum name="rtt"><para>Round trip time</para></enum>
<enum name="maxrtt"><para>Maximum round trip time</para></enum>
<enum name="minrtt"><para>Minimum round trip time</para></enum>
<enum name="normdevrtt"><para>Average round trip time</para></enum>
<enum name="stdevrtt"><para>Standard deviation round trip time</para></enum>
<enum name="local_ssrc"><para>Our Synchronization Source identifier</para></enum>
<enum name="remote_ssrc"><para>Their Synchronization Source identifier</para></enum>
</enumlist>
</parameter>
<parameter name="media_type" required="false">
<para>When <replaceable>rtcp</replaceable> is specified, the
<literal>media_type</literal> parameter may be provided. It specifies
which media stream the chosen RTCP parameter should be retrieved
from.</para>
<enumlist>
<enum name="audio">
<para>Retrieve information from the audio media stream.</para>
<note><para>If not specified, <literal>audio</literal> is used
by default.</para></note>
</enum>
<enum name="video">
<para>Retrieve information from the video media stream.</para>
</enum>
</enumlist>
</parameter>
</enum>
<enum name="endpoint">
<para>R/O The name of the endpoint associated with this channel.
Use the <replaceable>PJSIP_ENDPOINT</replaceable> function to obtain
further endpoint related information.</para>
</enum>
<enum name="contact">
<para>R/O The name of the contact associated with this channel.
Use the <replaceable>PJSIP_CONTACT</replaceable> function to obtain
further contact related information. Note this may not be present and if so
is only available on outgoing legs.</para>
</enum>
<enum name="aor">
<para>R/O The name of the AOR associated with this channel.
Use the <replaceable>PJSIP_AOR</replaceable> function to obtain
further AOR related information. Note this may not be present and if so
is only available on outgoing legs.</para>
</enum>
<enum name="pjsip">
<para>R/O Obtain information about the current PJSIP channel and its
session.</para>
<parameter name="type" required="true">
<para>When <replaceable>pjsip</replaceable> is specified, the
<literal>type</literal> parameter must be provided. It specifies
which signalling parameter to read.</para>
<enumlist>
<enum name="secure">
<para>Whether or not the signalling uses a secure transport.</para>
<enumlist>
<enum name="0"><para>The signalling uses a non-secure transport.</para></enum>
<enum name="1"><para>The signalling uses a secure transport.</para></enum>
</enumlist>
</enum>
<enum name="target_uri">
<para>The request URI of the <literal>INVITE</literal> request associated with the creation of this channel.</para>
</enum>
<enum name="local_uri">
<para>The local URI.</para>
</enum>
<enum name="remote_uri">
<para>The remote URI.</para>
</enum>
<enum name="t38state">
<para>The current state of any T.38 fax on this channel.</para>
<enumlist>
<enum name="DISABLED"><para>T.38 faxing is disabled on this channel.</para></enum>
<enum name="LOCAL_REINVITE"><para>Asterisk has sent a <literal>re-INVITE</literal> to the remote end to initiate a T.38 fax.</para></enum>
<enum name="REMOTE_REINVITE"><para>The remote end has sent a <literal>re-INVITE</literal> to Asterisk to initiate a T.38 fax.</para></enum>
<enum name="ENABLED"><para>A T.38 fax session has been enabled.</para></enum>
<enum name="REJECTED"><para>A T.38 fax session was attempted but was rejected.</para></enum>
</enumlist>
</enum>
<enum name="local_addr">
<para>On inbound calls, the full IP address and port number that
the <literal>INVITE</literal> request was received on. On outbound
calls, the full IP address and port number that the <literal>INVITE</literal>
request was transmitted from.</para>
</enum>
<enum name="remote_addr">
<para>On inbound calls, the full IP address and port number that
the <literal>INVITE</literal> request was received from. On outbound
calls, the full IP address and port number that the <literal>INVITE</literal>
request was transmitted to.</para>
</enum>
</enumlist>
</parameter>
</enum>
</enumlist>
</info>
***/
#include "asterisk.h"
#include <pjsip.h>
#include <pjlib.h>
#include <pjsip_ua.h>
ASTERISK_REGISTER_FILE()
#include "asterisk/astobj2.h"
#include "asterisk/module.h"
#include "asterisk/acl.h"
#include "asterisk/app.h"
#include "asterisk/channel.h"
#include "asterisk/format.h"
#include "asterisk/pbx.h"
#include "asterisk/res_pjsip.h"
#include "asterisk/res_pjsip_session.h"
#include "include/chan_pjsip.h"
#include "include/dialplan_functions.h"
/*!
* \brief String representations of the T.38 state enum
*/
static const char *t38state_to_string[T38_MAX_ENUM] = {
[T38_DISABLED] = "DISABLED",
[T38_LOCAL_REINVITE] = "LOCAL_REINVITE",
[T38_PEER_REINVITE] = "REMOTE_REINVITE",
[T38_ENABLED] = "ENABLED",
[T38_REJECTED] = "REJECTED",
};
/*!
* \internal \brief Handle reading RTP information
*/
static int channel_read_rtp(struct ast_channel *chan, const char *type, const char *field, char *buf, size_t buflen)
{
struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
struct chan_pjsip_pvt *pvt;
struct ast_sip_session_media *media = NULL;
struct ast_sockaddr addr;
if (!channel) {
ast_log(AST_LOG_WARNING, "Channel %s has no pvt!\n", ast_channel_name(chan));
return -1;
}
pvt = channel->pvt;
if (!pvt) {
ast_log(AST_LOG_WARNING, "Channel %s has no chan_pjsip pvt!\n", ast_channel_name(chan));
return -1;
}
if (ast_strlen_zero(type)) {
ast_log(AST_LOG_WARNING, "You must supply a type field for 'rtp' information\n");
return -1;
}
if (ast_strlen_zero(field) || !strcmp(field, "audio")) {
media = pvt->media[SIP_MEDIA_AUDIO];
} else if (!strcmp(field, "video")) {
media = pvt->media[SIP_MEDIA_VIDEO];
} else {
ast_log(AST_LOG_WARNING, "Unknown media type field '%s' for 'rtp' information\n", field);
return -1;
}
if (!media || !media->rtp) {
ast_log(AST_LOG_WARNING, "Channel %s has no %s media/RTP session\n",
ast_channel_name(chan), S_OR(field, "audio"));
return -1;
}
if (!strcmp(type, "src")) {
ast_rtp_instance_get_local_address(media->rtp, &addr);
ast_copy_string(buf, ast_sockaddr_stringify(&addr), buflen);
} else if (!strcmp(type, "dest")) {
ast_rtp_instance_get_remote_address(media->rtp, &addr);
ast_copy_string(buf, ast_sockaddr_stringify(&addr), buflen);
} else if (!strcmp(type, "direct")) {
ast_copy_string(buf, ast_sockaddr_stringify(&media->direct_media_addr), buflen);
} else if (!strcmp(type, "secure")) {
snprintf(buf, buflen, "%d", media->srtp ? 1 : 0);
} else if (!strcmp(type, "hold")) {
snprintf(buf, buflen, "%d", media->remotely_held ? 1 : 0);
} else {
ast_log(AST_LOG_WARNING, "Unknown type field '%s' specified for 'rtp' information\n", type);
return -1;
}
return 0;
}
/*!
* \internal \brief Handle reading RTCP information
*/
static int channel_read_rtcp(struct ast_channel *chan, const char *type, const char *field, char *buf, size_t buflen)
{
struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
struct chan_pjsip_pvt *pvt;
struct ast_sip_session_media *media = NULL;
if (!channel) {
ast_log(AST_LOG_WARNING, "Channel %s has no pvt!\n", ast_channel_name(chan));
return -1;
}
pvt = channel->pvt;
if (!pvt) {
ast_log(AST_LOG_WARNING, "Channel %s has no chan_pjsip pvt!\n", ast_channel_name(chan));
return -1;
}
if (ast_strlen_zero(type)) {
ast_log(AST_LOG_WARNING, "You must supply a type field for 'rtcp' information\n");
return -1;
}
if (ast_strlen_zero(field) || !strcmp(field, "audio")) {
media = pvt->media[SIP_MEDIA_AUDIO];
} else if (!strcmp(field, "video")) {
media = pvt->media[SIP_MEDIA_VIDEO];
} else {
ast_log(AST_LOG_WARNING, "Unknown media type field '%s' for 'rtcp' information\n", field);
return -1;
}
if (!media || !media->rtp) {
ast_log(AST_LOG_WARNING, "Channel %s has no %s media/RTP session\n",
ast_channel_name(chan), S_OR(field, "audio"));
return -1;
}
if (!strncasecmp(type, "all", 3)) {
enum ast_rtp_instance_stat_field stat_field = AST_RTP_INSTANCE_STAT_FIELD_QUALITY;
if (!strcasecmp(type, "all_jitter")) {
stat_field = AST_RTP_INSTANCE_STAT_FIELD_QUALITY_JITTER;
} else if (!strcasecmp(type, "all_rtt")) {
stat_field = AST_RTP_INSTANCE_STAT_FIELD_QUALITY_RTT;
} else if (!strcasecmp(type, "all_loss")) {
stat_field = AST_RTP_INSTANCE_STAT_FIELD_QUALITY_LOSS;
}
if (!ast_rtp_instance_get_quality(media->rtp, stat_field, buf, buflen)) {
ast_log(AST_LOG_WARNING, "Unable to retrieve 'rtcp' statistics for %s\n", ast_channel_name(chan));
return -1;
}
} else {
struct ast_rtp_instance_stats stats;
int i;
struct {
const char *name;
enum { INT, DBL } type;
union {
unsigned int *i4;
double *d8;
};
} lookup[] = {
{ "txcount", INT, { .i4 = &stats.txcount, }, },
{ "rxcount", INT, { .i4 = &stats.rxcount, }, },
{ "txjitter", DBL, { .d8 = &stats.txjitter, }, },
{ "rxjitter", DBL, { .d8 = &stats.rxjitter, }, },
{ "remote_maxjitter", DBL, { .d8 = &stats.remote_maxjitter, }, },
{ "remote_minjitter", DBL, { .d8 = &stats.remote_minjitter, }, },
{ "remote_normdevjitter", DBL, { .d8 = &stats.remote_normdevjitter, }, },
{ "remote_stdevjitter", DBL, { .d8 = &stats.remote_stdevjitter, }, },
{ "local_maxjitter", DBL, { .d8 = &stats.local_maxjitter, }, },
{ "local_minjitter", DBL, { .d8 = &stats.local_minjitter, }, },
{ "local_normdevjitter", DBL, { .d8 = &stats.local_normdevjitter, }, },
{ "local_stdevjitter", DBL, { .d8 = &stats.local_stdevjitter, }, },
{ "txploss", INT, { .i4 = &stats.txploss, }, },
{ "rxploss", INT, { .i4 = &stats.rxploss, }, },
{ "remote_maxrxploss", DBL, { .d8 = &stats.remote_maxrxploss, }, },
{ "remote_minrxploss", DBL, { .d8 = &stats.remote_minrxploss, }, },
{ "remote_normdevrxploss", DBL, { .d8 = &stats.remote_normdevrxploss, }, },
{ "remote_stdevrxploss", DBL, { .d8 = &stats.remote_stdevrxploss, }, },
{ "local_maxrxploss", DBL, { .d8 = &stats.local_maxrxploss, }, },
{ "local_minrxploss", DBL, { .d8 = &stats.local_minrxploss, }, },
{ "local_normdevrxploss", DBL, { .d8 = &stats.local_normdevrxploss, }, },
{ "local_stdevrxploss", DBL, { .d8 = &stats.local_stdevrxploss, }, },
{ "rtt", DBL, { .d8 = &stats.rtt, }, },
{ "maxrtt", DBL, { .d8 = &stats.maxrtt, }, },
{ "minrtt", DBL, { .d8 = &stats.minrtt, }, },
{ "normdevrtt", DBL, { .d8 = &stats.normdevrtt, }, },
{ "stdevrtt", DBL, { .d8 = &stats.stdevrtt, }, },
{ "local_ssrc", INT, { .i4 = &stats.local_ssrc, }, },
{ "remote_ssrc", INT, { .i4 = &stats.remote_ssrc, }, },
{ NULL, },
};
if (ast_rtp_instance_get_stats(media->rtp, &stats, AST_RTP_INSTANCE_STAT_ALL)) {
ast_log(AST_LOG_WARNING, "Unable to retrieve 'rtcp' statistics for %s\n", ast_channel_name(chan));
return -1;
}
for (i = 0; !ast_strlen_zero(lookup[i].name); i++) {
if (!strcasecmp(type, lookup[i].name)) {
if (lookup[i].type == INT) {
snprintf(buf, buflen, "%u", *lookup[i].i4);
} else {
snprintf(buf, buflen, "%f", *lookup[i].d8);
}
return 0;
}
}
ast_log(AST_LOG_WARNING, "Unrecognized argument '%s' for 'rtcp' information\n", type);
return -1;
}
return 0;
}
/*!
* \internal \brief Handle reading signalling information
*/
static int channel_read_pjsip(struct ast_channel *chan, const char *type, const char *field, char *buf, size_t buflen)
{
struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
char *buf_copy;
pjsip_dialog *dlg;
if (!channel) {
ast_log(AST_LOG_WARNING, "Channel %s has no pvt!\n", ast_channel_name(chan));
return -1;
}
dlg = channel->session->inv_session->dlg;
if (ast_strlen_zero(type)) {
ast_log(LOG_WARNING, "You must supply a type field for 'pjsip' information\n");
return -1;
} else if (!strcmp(type, "secure")) {
#ifdef HAVE_PJSIP_GET_DEST_INFO
pjsip_host_info dest;
pj_pool_t *pool = pjsip_endpt_create_pool(ast_sip_get_pjsip_endpoint(), "secure-check", 128, 128);
pjsip_get_dest_info(dlg->target, NULL, pool, &dest);
snprintf(buf, buflen, "%d", dest.flag & PJSIP_TRANSPORT_SECURE ? 1 : 0);
pjsip_endpt_release_pool(ast_sip_get_pjsip_endpoint(), pool);
#else
ast_log(LOG_WARNING, "Asterisk has been built against a version of pjproject which does not have the required functionality to support the 'secure' argument. Please upgrade to version 2.3 or later.\n");
return -1;
#endif
} else if (!strcmp(type, "target_uri")) {
pjsip_uri_print(PJSIP_URI_IN_REQ_URI, dlg->target, buf, buflen);
buf_copy = ast_strdupa(buf);
ast_escape_quoted(buf_copy, buf, buflen);
} else if (!strcmp(type, "local_uri")) {
pjsip_uri_print(PJSIP_URI_IN_FROMTO_HDR, dlg->local.info->uri, buf, buflen);
buf_copy = ast_strdupa(buf);
ast_escape_quoted(buf_copy, buf, buflen);
} else if (!strcmp(type, "remote_uri")) {
pjsip_uri_print(PJSIP_URI_IN_FROMTO_HDR, dlg->remote.info->uri, buf, buflen);
buf_copy = ast_strdupa(buf);
ast_escape_quoted(buf_copy, buf, buflen);
} else if (!strcmp(type, "t38state")) {
ast_copy_string(buf, t38state_to_string[channel->session->t38state], buflen);
} else if (!strcmp(type, "local_addr")) {
RAII_VAR(struct ast_datastore *, datastore, NULL, ao2_cleanup);
struct transport_info_data *transport_data;
datastore = ast_sip_session_get_datastore(channel->session, "transport_info");
if (!datastore) {
ast_log(AST_LOG_WARNING, "No transport information for channel %s\n", ast_channel_name(chan));
return -1;
}
transport_data = datastore->data;
if (pj_sockaddr_has_addr(&transport_data->local_addr)) {
pj_sockaddr_print(&transport_data->local_addr, buf, buflen, 3);
}
} else if (!strcmp(type, "remote_addr")) {
RAII_VAR(struct ast_datastore *, datastore, NULL, ao2_cleanup);
struct transport_info_data *transport_data;
datastore = ast_sip_session_get_datastore(channel->session, "transport_info");
if (!datastore) {
ast_log(AST_LOG_WARNING, "No transport information for channel %s\n", ast_channel_name(chan));
return -1;
}
transport_data = datastore->data;
if (pj_sockaddr_has_addr(&transport_data->remote_addr)) {
pj_sockaddr_print(&transport_data->remote_addr, buf, buflen, 3);
}
} else {
ast_log(AST_LOG_WARNING, "Unrecognized argument '%s' for 'pjsip' information\n", type);
return -1;
}
return 0;
}
/*! \brief Struct used to push function arguments to task processor */
struct pjsip_func_args {
struct ast_channel *chan;
const char *param;
const char *type;
const char *field;
char *buf;
size_t len;
int ret;
};
/*! \internal \brief Taskprocessor callback that handles the read on a PJSIP thread */
static int read_pjsip(void *data)
{
struct pjsip_func_args *func_args = data;
if (!strcmp(func_args->param, "rtp")) {
func_args->ret = channel_read_rtp(func_args->chan, func_args->type,
func_args->field, func_args->buf,
func_args->len);
} else if (!strcmp(func_args->param, "rtcp")) {
func_args->ret = channel_read_rtcp(func_args->chan, func_args->type,
func_args->field, func_args->buf,
func_args->len);
} else if (!strcmp(func_args->param, "endpoint")) {
struct ast_sip_channel_pvt *pvt = ast_channel_tech_pvt(func_args->chan);
if (!pvt) {
ast_log(AST_LOG_WARNING, "Channel %s has no pvt!\n", ast_channel_name(func_args->chan));
return -1;
}
if (!pvt->session || !pvt->session->endpoint) {
ast_log(AST_LOG_WARNING, "Channel %s has no endpoint!\n", ast_channel_name(func_args->chan));
return -1;
}
snprintf(func_args->buf, func_args->len, "%s", ast_sorcery_object_get_id(pvt->session->endpoint));
} else if (!strcmp(func_args->param, "contact")) {
struct ast_sip_channel_pvt *pvt = ast_channel_tech_pvt(func_args->chan);
if (!pvt) {
ast_log(AST_LOG_WARNING, "Channel %s has no pvt!\n", ast_channel_name(func_args->chan));
return -1;
}
if (!pvt->session || !pvt->session->contact) {
return 0;
}
snprintf(func_args->buf, func_args->len, "%s", ast_sorcery_object_get_id(pvt->session->contact));
} else if (!strcmp(func_args->param, "aor")) {
struct ast_sip_channel_pvt *pvt = ast_channel_tech_pvt(func_args->chan);
if (!pvt) {
ast_log(AST_LOG_WARNING, "Channel %s has no pvt!\n", ast_channel_name(func_args->chan));
return -1;
}
if (!pvt->session || !pvt->session->aor) {
return 0;
}
snprintf(func_args->buf, func_args->len, "%s", ast_sorcery_object_get_id(pvt->session->aor));
} else if (!strcmp(func_args->param, "pjsip")) {
func_args->ret = channel_read_pjsip(func_args->chan, func_args->type,
func_args->field, func_args->buf,
func_args->len);
} else {
func_args->ret = -1;
}
return 0;
}
int pjsip_acf_channel_read(struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len)
{
struct pjsip_func_args func_args = { 0, };
struct ast_sip_channel_pvt *channel;
char *parse = ast_strdupa(data);
AST_DECLARE_APP_ARGS(args,
AST_APP_ARG(param);
AST_APP_ARG(type);
AST_APP_ARG(field);
);
if (!chan) {
ast_log(LOG_WARNING, "No channel was provided to %s function.\n", cmd);
return -1;
}
channel = ast_channel_tech_pvt(chan);
/* Check for zero arguments */
if (ast_strlen_zero(parse)) {
ast_log(LOG_ERROR, "Cannot call %s without arguments\n", cmd);
return -1;
}
AST_STANDARD_APP_ARGS(args, parse);
/* Sanity check */
if (strcmp(ast_channel_tech(chan)->type, "PJSIP")) {
ast_log(LOG_WARNING, "Cannot call %s on a non-PJSIP channel\n", cmd);
return 0;
}
if (!channel) {
ast_log(AST_LOG_WARNING, "Channel %s has no pvt!\n", ast_channel_name(chan));
return -1;
}
memset(buf, 0, len);
func_args.chan = chan;
func_args.param = args.param;
func_args.type = args.type;
func_args.field = args.field;
func_args.buf = buf;
func_args.len = len;
if (ast_sip_push_task_synchronous(channel->session->serializer, read_pjsip, &func_args)) {
ast_log(LOG_WARNING, "Unable to read properties of channel %s: failed to push task\n", ast_channel_name(chan));
return -1;
}
return func_args.ret;
}
int pjsip_acf_dial_contacts_read(struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len)
{
RAII_VAR(struct ast_sip_endpoint *, endpoint, NULL, ao2_cleanup);
RAII_VAR(struct ast_str *, dial, NULL, ast_free_ptr);
const char *aor_name;
char *rest;
AST_DECLARE_APP_ARGS(args,
AST_APP_ARG(endpoint_name);
AST_APP_ARG(aor_name);
AST_APP_ARG(request_user);
);
AST_STANDARD_APP_ARGS(args, data);
if (ast_strlen_zero(args.endpoint_name)) {
ast_log(LOG_WARNING, "An endpoint name must be specified when using the '%s' dialplan function\n", cmd);
return -1;
} else if (!(endpoint = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "endpoint", args.endpoint_name))) {
ast_log(LOG_WARNING, "Specified endpoint '%s' was not found\n", args.endpoint_name);
return -1;
}
aor_name = S_OR(args.aor_name, endpoint->aors);
if (ast_strlen_zero(aor_name)) {
ast_log(LOG_WARNING, "No AOR has been provided and no AORs are configured on endpoint '%s'\n", args.endpoint_name);
return -1;
} else if (!(dial = ast_str_create(len))) {
ast_log(LOG_WARNING, "Could not get enough buffer space for dialing contacts\n");
return -1;
} else if (!(rest = ast_strdupa(aor_name))) {
ast_log(LOG_WARNING, "Could not duplicate provided AORs\n");
return -1;
}
while ((aor_name = strsep(&rest, ","))) {
RAII_VAR(struct ast_sip_aor *, aor, ast_sip_location_retrieve_aor(aor_name), ao2_cleanup);
RAII_VAR(struct ao2_container *, contacts, NULL, ao2_cleanup);
struct ao2_iterator it_contacts;
struct ast_sip_contact *contact;
if (!aor) {
/* If the AOR provided is not found skip it, there may be more */
continue;
} else if (!(contacts = ast_sip_location_retrieve_aor_contacts(aor))) {
/* No contacts are available, skip it as well */
continue;
} else if (!ao2_container_count(contacts)) {
/* We were given a container but no contacts are in it... */
continue;
}
it_contacts = ao2_iterator_init(contacts, 0);
for (; (contact = ao2_iterator_next(&it_contacts)); ao2_ref(contact, -1)) {
ast_str_append(&dial, -1, "PJSIP/");
if (!ast_strlen_zero(args.request_user)) {
ast_str_append(&dial, -1, "%s@", args.request_user);
}
ast_str_append(&dial, -1, "%s/%s&", args.endpoint_name, contact->uri);
}
ao2_iterator_destroy(&it_contacts);
}
/* Trim the '&' at the end off */
ast_str_truncate(dial, ast_str_strlen(dial) - 1);
ast_copy_string(buf, ast_str_buffer(dial), len);
return 0;
}
static int media_offer_read_av(struct ast_sip_session *session, char *buf,
size_t len, enum ast_media_type media_type)
{
int i, size = 0;
for (i = 0; i < ast_format_cap_count(session->req_caps); i++) {
struct ast_format *fmt = ast_format_cap_get_format(session->req_caps, i);
if (ast_format_get_type(fmt) != media_type) {
ao2_ref(fmt, -1);
continue;
}
/* add one since we'll include a comma */
size = strlen(ast_format_get_name(fmt)) + 1;
if (len < size) {
ao2_ref(fmt, -1);
break;
}
len -= size;
/* no reason to use strncat here since we have already ensured buf has
enough space, so strcat can be safely used */
strcat(buf, ast_format_get_name(fmt));
strcat(buf, ",");
ao2_ref(fmt, -1);
}
if (size) {
/* remove the extra comma */
buf[strlen(buf) - 1] = '\0';
}
return 0;
}
struct media_offer_data {
struct ast_sip_session *session;
enum ast_media_type media_type;
const char *value;
};
static int media_offer_write_av(void *obj)
{
struct media_offer_data *data = obj;
ast_format_cap_remove_by_type(data->session->req_caps, data->media_type);
ast_format_cap_update_by_allow_disallow(data->session->req_caps, data->value, 1);
return 0;
}
int pjsip_acf_media_offer_read(struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len)
{
struct ast_sip_channel_pvt *channel;
if (!chan) {
ast_log(LOG_WARNING, "No channel was provided to %s function.\n", cmd);
return -1;
}
if (strcmp(ast_channel_tech(chan)->type, "PJSIP")) {
ast_log(LOG_WARNING, "Cannot call %s on a non-PJSIP channel\n", cmd);
return -1;
}
channel = ast_channel_tech_pvt(chan);
if (!strcmp(data, "audio")) {
return media_offer_read_av(channel->session, buf, len, AST_MEDIA_TYPE_AUDIO);
} else if (!strcmp(data, "video")) {
return media_offer_read_av(channel->session, buf, len, AST_MEDIA_TYPE_VIDEO);
}
return 0;
}
int pjsip_acf_media_offer_write(struct ast_channel *chan, const char *cmd, char *data, const char *value)
{
struct ast_sip_channel_pvt *channel;
struct media_offer_data mdata = {
.value = value
};
if (!chan) {
ast_log(LOG_WARNING, "No channel was provided to %s function.\n", cmd);
return -1;
}
if (strcmp(ast_channel_tech(chan)->type, "PJSIP")) {
ast_log(LOG_WARNING, "Cannot call %s on a non-PJSIP channel\n", cmd);
return -1;
}
channel = ast_channel_tech_pvt(chan);
mdata.session = channel->session;
if (!strcmp(data, "audio")) {
mdata.media_type = AST_MEDIA_TYPE_AUDIO;
} else if (!strcmp(data, "video")) {
mdata.media_type = AST_MEDIA_TYPE_VIDEO;
}
return ast_sip_push_task_synchronous(channel->session->serializer, media_offer_write_av, &mdata);
}
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