chan_vpb was trying to use sizeof(*p->play_dtmf), where
p->play_dtmf is defined as char[16], to get the length of the array
but since p->play_dtmf is an actual array, sizeof(*p->play_dtmf)
returns the size of the first array element, which is 1. gcc7
validly complains because the context in which it's used could
cause an out-of-bounds condition.
Change-Id: If9c4bfdb6b02fa72d39e0c09bf88900663c000ba
Currently privacy requests are only granted if the Privacy header
value is exactly "id" (defined in RFC 3325). It ignores any other
possible value (or a combination there of). This patch reverses the
logic from testing for "id" to grant privacy, to testing for "none" and
granting privacy for any other value. "none" must not be used in
combination with any other value (RFC 3323 section 4.2).
ASTERISK-27284 #close
Change-Id: If438a21f31a962da32d7a33ff33bdeb1e776fe56
Provide a way to get the contents of the the Request URI from the initial SIP
INVITE in dial plan function call. (In this case "${CHANNEL(ruri)}")
ASTERISK-27278
Reported by: David J. Pryke
Tested by: David J. Pryke
Change-Id: I1dd4d6988eed1b6c98a9701e0e833a15ef0dac3e
Multicast/Unicast RTP do not use SDP so we need to use a format that
cleanly maps to one of the static RTP payload types. Without this
change, an Originate to a Multicast or Unicast channel without a format
specified would produce no audio on the receiving device.
ASTERISK-21399 #close
Reported by: Tzafrir Cohen
Change-Id: I97e332b566e85da04b0004b9b0daae746cfca0e3
In handle_request_invite, when processing a pickup, a call
is made to get_sip_pvt_from_replaces to locate the pvt for
the subscription. The pvt is assumed to be valid when zero
is returned indicating no error, and is dereferenced which
can cause a crash if it was not found.
This change checks the not found case and returns -1 which
allows the calling code to fail appropriately.
ASTERISK-27217 #close
Reported-by: Bryan Walters
Change-Id: I6bee92b8b8b85fcac3fd66f8c00ab18bc1765612
If directmedia=yes is configured, when call is answered, Asterisk sends reINVITE
to both parties to set up media path directly between the endpoints.
In this reINVITE msg SDP origin line (o=) contains IP address of endpoint
instead of IP of asterisk. This behavior violates RFC3264, sec 8:
"When issuing an offer that modifies the session,
the "o=" line of the new SDP MUST be identical to that in the
previous SDP, except that the version in the origin field MUST
increment by one from the previous SDP."
This patch assures IP address of Asterisk is always sent in
SDP origin line.
ASTERISK-17540
Reported by: saghul
Change-Id: I533a047490c43dcff32eeca8378b2ba02345b64e
When rtp_keepalive is on for a PJSIP endpoint dialing to another
Asterisk instance also using PJSIP, Asterisk will continue to print
warning messages about not being able to send frames of a certain
type. This suppresses that warning message.
Change-Id: I0332a05519d7bda9cacfa26d433909ff1909be67
Create local_tag and remote_tag in CHANNEL info to get tag from From and
To headers of a SIP dialog.
ASTERISK-27220
Change-Id: I59b16c4b928896fcbde02ad88f0e98922b15d524
If an SDP answer hasn't been sent yet, it's legal to change it.
This is required for PJSIP_DTMF_MODE to work correctly, and can
also have use in the future for updating codecs too.
ASTERISK-27209 #close
Change-Id: Idbbfb7cb3f72fbd96c94d10d93540f69bd51e7a1
Introduce a new property to rtp-engine to make it aware of
the desire for assymetric codecs or not. If asymmetric codecs
is not allowed, the bridge will compare read/write formats
and shut down the p2p bridge if needed
ASTERISK-26745 #close
Change-Id: I0d9c83e5356df81661e58d40a8db565833501a6f
* chan_sip: channel in test_sip_rtpqos_1.
* test_config: config hook, config info and global config holder.
* test_core_format: format in format_attribute_set_without_interface.
* test_stream: unneeded frame duplication.
* test_taskprocessor: task_data.
Change-Id: I94d364d195cf3b3b5de2bf3ad565343275c7ad31
GCC 7 has added capability to produce warnings, this fixes most of those
warnings. The specific warnings are disabled in a few places:
* app_voicemail.c: truncation of paths more than 4096 chars in many places.
* chan_mgcp.c: callid truncated to 80 chars.
* cdr.c: two userfields are combined to cdr copy, fix would break ABI.
* tcptls.c: ignore use of deprecated method SSLv3_client_method().
ASTERISK-27156 #close
Change-Id: I65f280e7d3cfad279d16f41823a4d6fddcbc4c88
This function is a replica of SIPDtmfMode, allowing the DTMF mode of a
PJSIP call to be modified on a per-call basis
ASTERISK-27085 #close
Change-Id: I20eef5da3e5d1d3e58b304416bc79683f87e7612
On every reload of chan_iax2 module, MWI subscription was added, which
results in additional taskprocessors being accumulated over time.
This commit fixes it by making sure we check for existing subscription
first.
This was verified with 'core show taskprocessors' CLI command.
ASTERISK-27122 #close
Change-Id: Ie2ef528fd5ca01b933eeb88188cc10967899cfb9
When sip.conf contained tcpenable=yes and autodomain=yes, the TCP domain was
added in any case, because of a local Boolean-negation error of the return value
of ast_sockaddr_cmp. After fixing this error for TCP and TLS, the TLS domain was
still always added with tlsenable=yes, because the domains were not compared
just on the address but also on the port – and TLS is always on a different port
than UDP/TCP.
ASTERISK-27106
Change-Id: I14fe9e319e238320b094016980445ef3a5b3337c
Because of a copy-and-paste error when the struct ast_sockaddr changed,
tlsbindaddr was not added, when sip.conf contained autodomain=yes; see
"show sip domains" on the command-line interface (CLI) of Asterisk.
ASTERISK-27106
Change-Id: I3d0957150017c223136968ef1266f275d0d6695e
When connected_line_method is "invite", we're supposed to determine
if the client can support UPDATE and if it can, send UPDATE instead
of INVITE to avoid the SDP renegotiation. Not only was pjproject
not setting the PJSIP_INV_SUPPORT_UPDATE flag, we were testing
that invite_tsx wasn't NULL which isn't always the case.
* Updated chan_pjsip/update_connected_line_information to drop the
requirement that invite_tsx isn't NULL.
* Submitted patch to pjproject sip_inv.c that sets the
PJSIP_INV_SUPPORT_UPDATE flag correctly.
* Updated pjsip.conf.sample to clarify what happens when "invite"
is specified.
ASTERISK-27095
Change-Id: Ic2381b3567b8052c616d96fbe79564c530e81560
The existing auto dtmf mode reverts to inband if 4733 fails to be
negotiated. This patch adds a new mode auto_info which will
switch to INFO instead of inband if 4733 is not available.
ASTERISK-27066 #close
Change-Id: Id185b11e84afd9191a2f269e8443019047765e91
The construction of the returned string assumed incorrectly that the
supplied buffer would always be initialized as an empty string. If it is
not an empty string we could overrun the supplied buffer by the length of
the non-empty buffer string plus one. It is also theoreticaly possible
for the supplied buffer to be overrun by a string terminator during a read
operation even if the supplied buffer is an empty string.
* Fix the assumption that the supplied buffer would already be an empty
string. The buffer is not guaranteed to contain an empty string by all
possible callers.
* Fix string terminator buffer overrun potential.
Change-Id: If6a0806806527678c8554b1dcb34fd7808aa95c9
The chan_pjsip module uses a calculation approach for
determining device state. This means that in situations
where we would expect device state to change we need to
tell the core to query. A scenario that was missed is
when early media was signaled.
This change adds the notification for the core to
query device state when we are told that early media
is being provided.
ASTERISK-27039
Change-Id: Iafebfd152894966344ff2e950a3cee9f59a3eb6f
PJSIP support in Asterisk differs from chan_sip in that it
allows media to be sent as-is without transcoding provided
the codecs were negotiated in the SDP. This is allowed
according to the RFC. Support for this differs quite a lot
though and some endpoints do not handle it well.
This change extends the 'asymmetric_rtp_codec' option to
also cover this case. When set to no (the default) the code
behaves as chan_sip does - the best codec is selected and
we will only ever send that, unless we change what we are
sending if the remote side changes. When set to yes we
will send media as-is without transcoding if the codec
has been negotiated in the SDP.
ASTERISK-26996
Change-Id: Ib1647f6902a0843e8c435946f831c2159e8d1d51
This introduces the ability for PJSIP code to specify filtering flags
when retrieving PJSIP contacts. The first flag for use causes the
query code to only retrieve contacts that are not unreachable. This
change has been leveraged by both the Dial() process and the
PJSIP_DIAL_CONTACTS dialplan function so they will now only attempt
calls to contacts which are not unreachable.
ASTERISK-26281
Change-Id: I8233b4faa21ba3db114f5a42e946e4b191446f6c
If we are offered or are offering RTCP-MUX, don't consider RTCP ICE
candidates. This confuses certain browsers (current Firefox for
example) and causes intial audio setup delays.
ASTERISK-26982 #close
Change-Id: Ifeaf47e83972fe8dbe58b7fb3d6d1823400cfb91
The while(1) loop in skinny_session wasn't checking for EOF so
a packet that was longer than a header but still truncated
would spin the while loop infinitely. Not only does this
permanently tie up a thread and drive a core to 100% utilization,
the call of ast_log() in such a tight loop eats all available
process memory.
Added poll with timeout to top of read loop
ASTERISK-26940 #close
Reported-by: Sandro Gauci
Change-Id: I2ce65f3c5cb24b4943a9f75b64d545a1e2cd2898
For outgoing TCP connections, Asterisk uses the first IP address of the
interface instead of the IP address we asked him to bind to.
ASTERISK-26922 #close
Reported-by: Ksenia
Change-Id: I43c71ca89211dbf1838e5bcdb9be8d06d98e54eb
If you use ast_request to create a PJSIP channel but then hang it
up without causing a transaction to be sent, the session will
never be destroyed. This is due ot the fact that it's pjproject
that triggers the session cleanup when the transaction ends.
app_chanisavail was doing this to get more granular channel state
and it's also possible for this to happen via ARI.
* ast_sip_session_terminate was modified to explicitly call the
cleanup tasks and unreference session if the invite state is NULL
AND invite_tsx is NULL (meaning we never sent a transaction).
* chan_pjsip/hangup was modified to bump session before it calls
ast_sip_session_terminate to insure that session stays valid
while it does its own cleanup.
* Added test events to session_destructor for a future testsuite
test.
ASTERISK-26908 #close
Reported-by: Richard Mudgett
Change-Id: I52daf6f757184e5544c261f64f6fe9602c4680a9
Some equipments may send a re-INVITE containing an SDP in the final ACK
request. If this happens in the context of direct media, the remote end
should be updated with a re-INVITE.
This patch queues an "update RTP peer" frame to trigger the re-INVITE,
instead of the "source change" frame wich was used previously.
ASTERISK-26951
Change-Id: I3644d2025f20e086ea9f8f62b486172c52b5b2e6
In all non-pbx modules, AST_MODULE_LOAD_FAILURE has been changed
to AST_MODULE_LOAD_DECLINE. This prevents asterisk from exiting
if a module can't be loaded. If the user wishes to retain the
FAILURE behavior for a specific module, they can use the "require"
or "preload-require" keyword in modules.conf.
A new API was added to logger: ast_is_logger_initialized(). This
allows asterisk.c/check_init() to print to the error log once the
logger subsystem is ready instead of just to stdout. If something
does fail before the logger is initialized, we now print to stderr
instead of stdout.
Change-Id: I5f4b50623d9b5a6cb7c5624a8c5c1274c13b2b25
SIP user-agents indicate which protocol extensions are allowed in headers
like Supported and Required. Such protocol extensions are Session Timers
(RFC 4028) for example. Session Timers are supported since Mantis-10665.
Since ASTERISK-21721, not only the first but multiple Supported/Required
headers in a message are parsed. In that change, an existing variable was
re-used within a newly added do-loop. Currently, at the end of that loop,
that variable is an empty string always. Previously, that variable was used
within log output. However, the log output was not changed.
ASTERISK-26915 #close
Change-Id: I09315f31b4d78fb214bb2a9fb6c0f5e143eae990
POSIX does not require getprotobyname() to be thread safe and some
implementations use static memory which causes issues when multiple
threads are used.
Further, our usage of it today is just to ultimately get IPPROTO_TCP
for calls to setsockopt(). So instead we just use IPPROTO_TCP directly.
Change-Id: I2e14e58674808f7ce99b2f5e900d0f90d0d8da48
When using a non-dynamic peer address, build_peer() invalidates the
peer address structure by setting the address family to unspecified.
However, if dnsmgr is enabled, the subsequent call to ast_dnsmgr_lookup()
will not amend the peer address if the cache is still valid, resulting
in peer connectivity failures.
To fix this, we call ast_dnsmgr_refresh() instead.
ASTERISK-26865
Change-Id: Id8a89a2f771ebbaf32255a35fe596a6dcb97a082
When querying for PJSIP specific information using the dialplan
function CHANNEL() it is possible that the underlying session
will no longer have a channel associated with it. This is
most likely to occur when the RTCP HEP module attempts to get
the channel name. If this happens then a crash will occur.
This change just adds a check that the channel exists on the
session before querying it.
ASTERISK-26857
Change-Id: I113479cffff6ae64cf8ed089e9e1565223426f01
When receiving a 422 response, the invitestate variable must be reset to
INV_CALLING.
ASTERISK-26841
Change-Id: Ia0502d6b02192664cefa4e75bafdd2645ce56099
