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1.4.5
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1.4.7
1.4.7.1
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1.4.9
1.6.0
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1.6.0.1
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1.6.0.11-rc1
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${ noResults }
101 Commits (19eef2a6dc26e960059c46b7fc8adb08984845c1)
Author | SHA1 | Message | Date |
---|---|---|---|
|
acb7ce4fe7 |
pjsip: Make modify_local_offer2 tolerate previous failed SDP.
If a remote side is broken and sends an SDP that can not be negotiated the call will be torn down but there is a window where a second 183 Session Progress or 200 OK that is forked can be received that also attempts to negotiate SDP. Since the code marked the SDP negotiation as being done and complete prior to this it assumes that there is an active local and remote SDP which it can modify, while in fact there is not as the SDP did not successfully negotiate. Since there is no local or remote SDP a crash occurs. This patch changes the pjmedia_sdp_neg_modify_local_offer2 function to no longer assume that a previous SDP negotiation was successful. ASTERISK-29196 Change-Id: I22de45916d3b05fdc2a67da92b3a38271ee5949e |
5 years ago |
|
7a6cfde4db |
res_pjsip_nat.c: Create deep copies of strings when appropriate
In rewrite_uri asterisk was not making deep copies of strings when changing the uri. This was in some cases causing garbage in the route header and in other cases even crashing asterisk when receiving a message with a record-route header set. Thanks to Ralf Kubis for pointing out why this happens. A similar problem was found in res_pjsip_transport_websocket.c. Pjproject needs as well to be patched to avoid garbage in CANCEL messages. ASTERISK-29024 #close Change-Id: Ic5acd7fa2fbda3080f5f36ef12e46804939b198b |
5 years ago |
|
6482ab5bea |
pjproject: clone sdp to protect against (nat) modifications
PJSIP, UDP transport with external_media_address and session timers enabled. Connected to SIP server that is not in local net. Asterisk initiated the connection and is refreshing the session after 150s (timeout 300s). The 2nd refresh-INVITE triggered by the pjsip timer has a malformed IP address in its SDP (garbage string). This only happens when the SDP is modified by the nat-code to replace the local IP address with the configured external_media_address. Analysis: the code to modify the SDP (in res_pjsip_session.c:session_outgoing_nat_hook() and also (redundantly?) in res_pjsip_sdp_rtp.c:change_outgoing_sdp_stream_media_address()) uses the tdata->pool to allocate the replacement string. But the same pjmedia_sdp_stream that was modified for the 1st refresh-INVITE is also used for the 2nd refresh-INVITE (because it is stored in pjmedia's pjmedia_sdp_neg structure). The problem is, that at that moment, the tdata->pool that holds the stringified external_media_address from the 1. refresh-INVITE has long been reused for something else. Fix by Sauw Ming of pjproject (see https://github.com/pjsip/pjproject/pull/2476): the local, potentially modified pjmedia_sdp_stream is cloned in pjproject/source/pjsip/src/pjmedia/sip_neg.c:process_answer() and the clone is stored, thereby detaching from the tdata->pool (which is only released *after* process_answer()) ASTERISK-28973 Reported-by: Michael Neuhauser Change-Id: I272ac22436076596e06aa51b9fa23fd1c7734a0e |
5 years ago |
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af70bbb13a |
websocket / pjsip: Increase maximum packet size.
When dealing with a lot of video streams on WebRTC the resulting SDPs can grow to be quite large. This effectively doubles the maximum size to allow more streams to exist. The res_http_websocket module has also been changed to use a buffer on the session for reading in packets to ensure that the stack space usage is not excessive. Change-Id: I31d4351d70c8e2c11564807a7528b984f3fbdd01 |
5 years ago |
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8d15f72721 |
pjsip: Include timer patch to prevent cancelling timer 0.
I noticed this while looking at another issue and brought it up with Teluu. It was possible for an uninitialized timer to be cancelled, resulting in the invalid timer id of 0 being placed into the timer heap causing issues. This change is a backport from the pjproject repository preventing this from happening. Change-Id: I1ba318b1f153a6dd7458846396e2867282b428e7 |
5 years ago |
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415b55af5a |
pjproject: Upgrade bundled version to pjproject 2.10
This patch makes the usual necessary changes when upgrading to a new version pjproject. For instance, version number bump, patches removed from third-party, new *.md5 file added, etc.. This patch also includes a change to the Asterisk pjproject Makefile to explicitly create the 'source/pjsip-apps/lib' directory. This directory is no longer there by default so needs to be added so the Asterisk malloc debug can be built. This patch also includes some minor changes to Asterisk that were a result of the upgrade. Specifically, there was a backward incompatibility change made in 2.10 that modified the "expires header" variable field from a signed to an unsigned value. This potentially effects comparison. Namely, those check for a value less than zero. This patch modified a few locations in the Asterisk code that may have been affected. Lastly, this patch adds a new macro PJSIP_MINVERSION that can be used to check a minimum version of pjproject at compile time. ASTERISK-28899 #close Change-Id: Iec8821c6cbbc08c369d0e3cd2f14e691b41d0c81 |
5 years ago |
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801d570f6e |
pjproject: Fix race condition when building with parallel make
Pjproject makefiles miss some dependencies which can cause race conditions when building with parallel make processes. This patch adds such dependencies correctly. ASTERISK-28879 #close Reported-by: Dmitry Wagin <dmitry.wagin@ya.ru> Change-Id: Ie1b0dc365dafe4a84c5248097fe8d73804043c22 |
6 years ago |
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3078a00a6d |
pjsip: Increase maximum ICE candidate count.
In practice it has been seen that some users come close to our maximum ICE candidate count of 32. In case people have gone over this increases the count to 64, giving ample room. ASTERISK-28859 Change-Id: I35cd68948ec0ada86c14eb53092cdaf8b62996cf |
6 years ago |
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168637cc0c |
RTP/ICE: Send on first valid pair.
When handling ICE negotiations, it's possible that there can be a delay between STUN binding requests which in turn will cause a delay in ICE completion, preventing media from flowing. It should be possible to send media when there is at least one valid pair, preventing this scenario from occurring. A change was added to PJPROJECT that adds an optional callback (on_valid_pair) that will be called when the first valid pair is found during ICE negotiation. Asterisk uses this to start the DTLS handshake, allowing media to flow. It will only be called once, either on the first valid pair, or when ICE negotiation is complete. ASTERISK-28716 Change-Id: Ia7b68c34f06d2a1d91c5ed51627b66fd0363d867 |
6 years ago |
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9d9bde76a9 |
pjproject_bundled: Allow brackets in via parameters
ASTERISK-26955 #close Reported by: Peter Sokolov Change-Id: Ib2803640905a77b65d0cee2d0ed2c7b310d470ac |
6 years ago |
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5d9f9f4871 |
pjproject_bundled: Replace earlier reverts with official fixes.
Issues in pjproject 2.9 caused us to revert some of their changes as a work around. This introduced another issue where pjproject wouldn't build with older gcc versions such as that found on CentOS 6. This commit replaces the reverts with the official fixes for the original issues and allows pjproject to be built on CentOS 6 again. ASTERISK-28574 Reported-by: Niklas Larsson Change-Id: I06f8507bea553d1a01b0b8874197d35b9d47ec4c |
6 years ago |
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cc83e76aa5 |
pjproject_bundled: Revert pjproject 2.9 commits causing leaks
We've found a connection re-use regression in pjproject 2.9 introduced by commit "Close #1019: Support for multiple listeners." https://trac.pjsip.org/repos/changeset/6002 https://trac.pjsip.org/repos/ticket/1019 Normally, multiple SSL requests should reuse the same connection if one already exists to the remote server. When a transport error occurs, the next request should establish a new connection and any following requests should use that same one. With this patch, when a transport error occurs, every new request creates a new connection so you can wind up with thousands of open tcp sockets, possibly exhausting file handles, and increasing memory usage. Reverting pjproject commit 6002 (and related 6021) restores the expected behavior. We also found a memory leak in SSL processing that was introduced by commit "Fixed #2204: Add OpenSSL remote certificate chain info" https://trac.pjsip.org/repos/changeset/6014 https://trac.pjsip.org/repos/ticket/2204 Apparently the remote certificate chain is continually recreated causing the leak. Reverting pjproject commit 6014 (and related 6022) restores the expected behavior. Both of these issues have been acknowledged by Teluu. ASTERISK-28521 Change-Id: I8ae7233c3ac4ec29a3b991f738e655dabcaba9f1 |
6 years ago |
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0844d6b127 |
pjproject: Configurable setting for cnonce to include hyphens or not
NEC SIP Station interface with authenticated registration only supports cnonce up to 32 characters. In Linux, PJSIP would generate 36 character cnonce which included hyphens. Teluu developed this patch adding a compile time setting to default to not include the hyphens. They felt it best to still generate the UUID and strip the hyphens. They have indicated it will be part of PJSIP 2.10. ASTERISK-28509 Reported-by: Dan Cropp Change-Id: Ibdfcf845d4f8c0a14df09fd983b11f2d72c5f470 |
6 years ago |
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8b3ee7fe61 |
pjproject_bundled: Add peer information to most SSL/TLS errors
Most SSL/TLS error messages coming from pjproject now have either the peer address:port or peer hostname, depending on what was available at the time and code location where the error was generated. ASTERISK-28444 Reported by: Bernhard Schmidt Change-Id: I41770e8a1ea5e96f6e16b236692c4269ce1ba91e |
6 years ago |
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c70d874f7d |
pjproject: Update to 2.9 release
Relies on https://github.com/asterisk/third-party/pull/4 Change-Id: Iec9cad42cb4ae109a86a3d4dae61e8bce4424ce3 |
6 years ago |
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3853fab3f5 |
pjproject-bundled: Add upstream timer fixes
Fixed #2191: - Stricter double timer entry scheduling prevention. - Integrate group lock in SIP transport, e.g: for add/dec ref, for timer scheduling. ASTERISK-28161 Reported-by: Ross Beer Change-Id: I2e09aa66de0dda9414d8a8259a649c4d2d96a9f5 |
7 years ago |
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7043ed6ac9 |
pjproject: Add timer patch from pjproject r5934
ASTERISK-28161 #close Reported by: Ross Beer Change-Id: I65331d554695753005eaa66c1d5d4807fe9009c8 |
7 years ago |
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deffb8a6e0 |
pjproject_bundled: Add patch for double free issue in timer heap
Fixed #2172: Avoid double reference counter decrements in timer in the scenario of race condition between pj_timer_heap_cancel() and pj_timer_heap_poll(). Change-Id: If000e9438c83ac5084b678eb811e902c035bd2d8 |
7 years ago |
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ecb9ed0958 |
pjproject_bundled: check whether UPDATE is supported on outgoing calls
In ASTERISK-27095 an issue had been fixed because of which chan_pjsip was not trying to send UPDATE messages when connected_line_method was set to invite. However this only solved the issue for incoming INVITES. For outgoing INVITES (important when transferring calls) the options variable needs to be updated at a different place. ASTERISK-28182 #close Reported-by: nappsoft Change-Id: I76cc06da4ca76ddd6dce814a8b97cc66b98aaf29 |
7 years ago |
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37b2e68628 |
res_pjsip: Implement additional SIP RFCs for Google Voice trunk compatability
This change implements a few different generic things which were brought on by Google Voice SIP. 1. The concept of flow transports have been introduced. These are configurable transports in pjsip.conf which can be used to reference a flow of signaling to a target. These have runtime configuration that can be changed by the signaling itself (such as Service-Routes and P-Preferred-Identity). When used these guarantee an individual connection (in the case of TCP or TLS) even if multiple flow transports exist to the same target. 2. Service-Routes (RFC 3608) support has been added to the outbound registration module which when received will be stored on the flow transport and used for requests referencing it. 3. P-Associated-URI / P-Preferred-Identity (RFC 3325) support has been added to the outbound registration module. If a P-Associated-URI header is received it will be used on requests as the P-Preferred-Identity. 4. Configurable outbound extension support has been added to the outbound registration module. When set the extension will be placed in the Supported header. 5. Header parameters can now be configured on an outbound registration which will be placed in the Contact header. 6. Google specific OAuth / Bearer token authentication (draft-ietf-sipcore-sip-authn-02) has been added to the outbound registration module. All functionality changes are controlled by pjsip.conf configuration options and do not affect non-configured pjsip endpoints otherwise. ASTERISK-27971 #close Change-Id: Id214c2d1c550a41fcf564b7df8f3da7be565bd58 |
7 years ago |
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915861b431 |
bundled pjproject: Remove timer cleanup usage patch.
This patch is not in the upstream pjproject and does unsafe things with the timer->_timer_id and timer->_grp_lock values in pj_timer_entry_reset() outside of the timer heap lock. pj_timer_entry_reset() is also called for timers that are not about to be rescheduled in a few places. Change-Id: I4fe0b4bc648f7be5903cf4531b94fc87275713c1 |
7 years ago |
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58035702cb |
pjproject: Update initial 2.8 patches to apply cleanly.
ASTERISK-28059 Change-Id: I027472f2753391646dde594a709a75f14422db93 |
7 years ago |
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ce9a980be6 |
pjproject: Upgrade to 2.8.
This change brings in PJSIP 2.8, removes all the patches that were merged upstream, and makes a minor change to support a breaking change that was done. ASTERISK-28059 Change-Id: I5097772b11b0f95c3c1f52df6400158666f0a189 |
7 years ago |
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b002b85762 |
Merge "pjproject_bundled: Fix for Solaris builds. Do not undef s_addr."
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7 years ago |
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603d1e8d4b |
pjproject_bundled: Fix for Solaris builds. Do not undef s_addr.
The authors of PJProject undef s_addr because of some issue in Microsoft Windows. However in Oracle Solaris, s_addr is not a structure member, but defined to map to the real structure member. Updates the patch from ASTERISK_20366 ASTERISK-27997 Change-Id: I8223026d4d54e2a46521085fcc94bfa6ebe35b11 |
7 years ago |
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1c7c867ce0 |
pjproject_bundled: Find shared libraries in root --with-ssl=PATH.
The script configure from Teluu expects shared libraries (.so) in a subfolder called 'lib', when --with-xyz=PATH is specified. However for OpenSSL, the default location is the root of the source folder = PATH. Furthermore, Asterisk supports both, 'lib' and root. For consistency and because Asterisk is using (only) OpenSSL in PJProject, it is enhanced to support both locations, just like Asterisk. ASTERISK-27995 Change-Id: I8eb916a88b6b8c22e29bb40bee8faaca6c73406f |
7 years ago |
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3424795f3a |
thirdparty/pjproject: fix deadlock in response retransmissions
The tdata containing the response can be shared by both the dialog object and the tsx object. In order to prevent the race condition between the tsx retransmission and the dialog sending a response, clone the tdata before modifying it for the dialog send response. ASTERISK-27966 #close Change-Id: Ic381004a3a212fe1d8eca0e707fe09dba4a6ab4e |
7 years ago |
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0e8976116f |
res_pjsip: Remove spurious error logging when printing silent headers
Asterisk patched the pjproject source to avoid crashing when pjproject sip_msg headers are encountered with NULL vptr's, but the patch also output error messages for some valid headers which simply did not need to be added to the message itself, such as hidden route headers. pjproject has since applied a similar patch to their baseline to avoid crashes, but their version also avoids the spurious error logging. Lets use their patch instead. ASTERISK-27961 #close Change-Id: I2ddbd82c8da10e0dcc9807a48089d1f3c2d6e389 |
7 years ago |
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e19080a184 |
Bundled PJPROJECT: Disable internal connection oriented keep-alive.
Turn off the periodic sending of CRLNCRLN. Default is on (90 seconds), which conflicts with the global section's keep_alive_interval option in pjsip.conf. patches: pjsip_keep_not_alive.patch submitted by Alexander Traud (License 6520) ASTERISK-27347 Change-Id: I6a197f56e1830d3b7e5ec70f17025840a290b057 |
7 years ago |
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880fbff6b7 |
res_pjsip_session: Add ability to accept multiple sdp answers
pjproject by default currently will follow media forked during an INVITE on outbound calls if the To tag is different on a subsequent response as that on an earlier response. We handle this correctly. There have been reported cases where the To tag is the same but we still need to follow the media. The pjproject patch in this commit adds the capability to sip_inv and also adds the capability to control it at runtime. The original "different tag" behavior was always controllable at runtime but we never did anything with it and left it to default to TRUE. So, along with the pjproject patch, this commit adds options to both the system and endpoint objects to control the two behaviors, and a small logic change to session_inv_on_media_update in res_pjsip_session to control the behavior at the endpoint level. The default behavior for "different tags" remains the same at TRUE and the default for "same tag" is FALSE. Change-Id: I64d071942b79adb2f0a4e13137389b19404fe3d6 ASTERISK-27936 Reported-by: Ross Beer |
7 years ago |
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48720e7def |
pjroject_bundled: Add already-destroyed check to tsx_timer_callback
There have been cases that when the transaction timer callback is called the tsx is already destroyed. This causes a crash. We now check the tsx state and return if the tsx is already destroyed. Change-Id: If93acd5e48d9ca5bb553f2405d5afc836842fe1c |
8 years ago |
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7c03b2713e |
pjproject_bundled: timer: Clean up usage of timer heap
Added a new pj_timer_entry_reset function that resets a timer_entry for re-use. Changed direct settings of timer_entry fields to use pj_timer_entry_init and pj_timer_entry_reset. Fixed issues where timers were being rescheduled incorrectly. Change-Id: I5b624bfbc5c1429117484b9b24567293002148e6 |
8 years ago |
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a87141ddfd |
pjproject_bundled: Add patch for pj_atomic crashes
There have been some crashes in the past where something attempts to use a pj_atomic after it's already been destroyed. This patch tries to prevent it by making sure that pj_atomic_destroy sets its mutex to NULL when it's done. The pj_mutex functions already check for a NULL mutex and just return PJ_EINVAL. Teluu also added some checks to the win32 implementation as well. Change-Id: Id25f70b79fdedf44ead6e6e1763a4417d3b3f825 |
8 years ago |
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4d1c9d8711 |
core: Stop using AST_INLINE_API for allocator functions.
This replaces AST_INLINE_API allocators in utils.h with real functions implemented in astmm.c. Associated macro's are also moved from utils.h to astmm.h. Remove menuselect conflicts between MALLOC_DEBUG and DEBUG_CHAOS as they can now be combined. This has multiple benefits: * Simplifies asterisk/utils.h by removing inline functions and use of the logger. * Removal of these inline functions decreases size of Asterisk and module binaries by 1% or more. * Puts memory management functions together with and without MALLOC_DEBUG enabled, simplifying management of the code. * Enables DEBUG_CHAOS for ASTMM_REDIRECT and bundled pjproject. Change-Id: If9df4377f74bdbb627461b27a473123e05525887 |
8 years ago |
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4b7872c9db |
Merge "core: Remove ABI effects of MALLOC_DEBUG."
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8 years ago |
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7f4354c10f |
res_pjproject.c: Upgrade bundled PJPROJECT to 2.7.2
Update patches included in bundled PJPROJECT for the new version. ASTERISK-27730 Change-Id: Id3c8c8ad82126846bcd9768bc3d0a18d89be8944 |
8 years ago |
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c711e4076a |
core: Remove ABI effects of MALLOC_DEBUG.
This allows asterisk to be compiled with MALLOC_DEBUG to load modules built without MALLOC_DEBUG. Now pre-compiled third-party modules will still work regardless of MALLOC_DEBUG being enabled or not. Change-Id: Ic07ad80b2c2df894db984cf27b16a69383ce0e10 |
8 years ago |
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bd549cf936 |
Merge "AST-2018-003: Crash with an invalid SDP fmtp attribute"
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8 years ago |
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880c69f00f |
AST-2018-003: Crash with an invalid SDP fmtp attribute
pjproject's fmtp retrieval function failed to catch invalid fmtp attributes. Because of this Asterisk would crash if given an SDP with an invalid fmtp attribute. When retrieving the format this patch now makes sure the fmtp attribute is available. If not available it now returns an error status. ASTERISK-27583 #close Change-Id: I5cebe000ce2d846cae3af33b6d72c416e51caf2f |
8 years ago |
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d3a398cf90 |
AST-2018-002: Crash with an invalid SDP media format description
pjproject's media format parsing algorithm failed to catch invalid values. Because of this Asterisk would crash if given an SDP with a invalid media format description. When parsing the media format description this patch now properly parses the value and returns an error status if it can't successfully parse/convert the value. ASTERISK-27582 #close Change-Id: I883b3a4ef85b6972397f7b56bf46c5779c55fdd6 |
8 years ago |
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f0a3c977d6 |
pjproject_bundled: Prevent crash on bad outgoing header
We still need to figure out how a bad header is getting into the outgoing message but this patch to pjproject prevents attempting to print that header and causing a crash. For several users, this crash happens when sending 183 progress messages. ASTERISK-26832 Reported by: Ross Beer, Jan Rozhon Change-Id: Ie5c5a921c890c843587763e7f33f987dfe66bd16 |
8 years ago |
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cff3add680 |
BuildSystem: Really do not pass unknown-warning options to the compiler.
When an older GCC version is called with a too new warning option, GCC exited with an error and Asterisk was not built. Therefore, the configure script tests the installed compiler whether it supports that warning option. If not, Asterisk does not pass it to the installed compiler. However, some compilers (like clang) do not exit (error) but give just a warning in such a case. Because the compiler did not exit, Asterisk passed the unknown-warning option. ASTERISK-27560 Change-Id: Ia9d148e689c173df4e91699113605dab2de36038 |
8 years ago |
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fd0ca1c3f9 |
Remove as much trailing whitespace as possible.
Change-Id: I873c1c6d00f447269bd841494459efccdd2c19c0 |
8 years ago |
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0b532367bd |
pjsip: Ignore state changes from old transactions.
When we fail over to a new target we create a new transaction and it becomes the current INVITE transaction. This does not prevent the previous transaction from raising state changes and causing the session to be prematurely disconnected if a transport error occurs immediately. This change backports a fix from PJSIP that eliminates the incorrect state change and reduces when they would be raised in the first place. ASTERISK-27408 Change-Id: Id22d087591782eee31311753d11e7eca4b95ef34 |
8 years ago |
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e2715d2cd4 |
pjproject: Clean up disabling of WebRTC support.
The definition in config_site.h and the argument to the configure script are not necessary to disable WebRTC support. The correct argument, --disable-libwebrtc, is already passed. ASTERISK-26980 Change-Id: I27da2c894f87914956a72710222e17462d8a44bc |
8 years ago |
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db21f7f2e1 |
pjsip: 183 without To tag does not negotiate media
If a 183 with sdp response is receive without a To tag the sdp is not negotiated. According to RFC 3261 section 12.1.2 while a To tag is required, the client needs to still be able to handle the missing tag case for backwards compatibility. This patch, accepted by and applied to pjproject, makes it so if an incoming 180/183 with SDP comes in without a To tag it gets appropriately handled. ASTERISK-27442 #close Change-Id: Ic9d6b01e05e8f4874eebbd7adfe05d932025d203 |
8 years ago |
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aa65be95c0 |
Merge "bundled_pjproject: Update to 2.7.1"
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8 years ago |
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995b8c1a38 |
Merge "bundled_pjproject: sip_parser: Fix return code in pjsip_find_msg"
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8 years ago |
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f6ebd16bb8 |
bundled_pjproject: sip_parser: Fix return code in pjsip_find_msg
The default return code for pjsip_find_msg was PJ_SUCCESS so if a Content-Length header wasn't found at all, pjsip_find_msg was returning PJ_SUCCESS instead of PJSIP_EMISSINGHDR. Also added the volatile keyword to a few variables that are used both inside and outside the PJ_TRY/PJ_CATCH block. Partial fix for ASTERISK_27408 Change-Id: If82ba9de921e3d57df9c68cf96ee45ccc1491f7a |
8 years ago |
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2e7f6cd31b |
bundled_pjproject: Update to 2.7.1
Update from 2.7 to 2.7.1 for bundled pjproject. Changed version and removed patch files included in the update. Change-Id: I55cea8e734b318c2df9daf86aa0802c559ec8357 |
8 years ago |