8336 Commits (b5eca43b29f0eb33d1e8305a8851a4295e4211b0)

Author	SHA1	Message	Date
Joshua Colp	b5eca43b29	Merge "res_rtp_asterisk: Make P2P bridge Asymmetric codec aware" into 13	8 years ago
Jenkins2	6dd12aaacd	Merge "Fix compile error for old versions of GCC." into 13	8 years ago
Torrey Searle	c4f201cd73	res_rtp_asterisk: Make P2P bridge Asymmetric codec aware ... 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	8 years ago
Corey Farrell	84b6a5efd7	Correct some leaks in unit tests. ... * 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	8 years ago
Corey Farrell	d066758a4c	Fix compile error for old versions of GCC. ... Use -Wno-format-truncation only if supported by compiler. ASTERISK-27171 #close Change-Id: Iac0aed7a5bcaa16c21b7d62c4e4678d244c4ccb6	8 years ago
Corey Farrell	0f49e6ee2e	Fix compiler warnings on Fedora 26 / GCC 7. ... 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	8 years ago
Torrey Searle	423d01cf16	chan_pjsip: add a new function PJSIP_DTMF_MODE ... 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	8 years ago
Joshua Colp	24bb5a8908	core: Add VP9 passthrough support. ... This change adds VP9 as a known codec and creates a cached "vp9" media format for use. Change-Id: I025a93ed05cf96153d66f36db1839109cc24c5cc	8 years ago
Sergej Kasumovic	cdd6ca488a	chan_iax2: On reload make sure to check for existing MWI subscription ... 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	8 years ago
Jenkins2	cf6e0b8f8b	Merge "chan_sip: Only when different, add TCP|TLS in autodomain (SIP Domain Support)." into 13	8 years ago
George Joseph	40490768cc	Merge "chan_pjsip: Fix ability to send UPDATE on COLP" into 13	8 years ago
Alexander Traud	39d2ebbf56	chan_sip: Only when different, add TCP|TLS in autodomain (SIP Domain Support). ... 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	8 years ago
Alexander Traud	9f4b3b966e	chan_sip: Fix a typo for tlsbindaddr in autodomain (SIP Domain Support). ... 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	8 years ago
George Joseph	6bd7c0f37c	chan_pjsip: Fix ability to send UPDATE on COLP ... 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	8 years ago
Torrey Searle	9fbc34d2bd	res_pjsip: Add DTMF INFO Failback mode ... 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	8 years ago
Richard Mudgett	b9a4ab8c8c	chan_pjsip: Fix PJSIP_MEDIA_OFFER dialplan function read. ... 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	8 years ago
Joshua Colp	bc51d4324a	Merge "pjsip: Extend 'asymmetric_rtp_codec' option to include us changing." into 13	8 years ago
Joshua Colp	1f10c6b3b0	chan_pjsip: Update device state when in early media. ... 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	8 years ago
Joshua Colp	996a4791ff	pjsip: Extend 'asymmetric_rtp_codec' option to include us changing. ... 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	8 years ago
Joshua Colp	746c2c5745	res_pjsip: Add support for returning only reachable contacts and use it. ... 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	8 years ago
Sean Bright	4479038073	chan_sip: Better ICE handling for RTCP-MUX ... 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	8 years ago
George Joseph	1cc18d4025	AST-2017-004: chan_skinny: Add EOF check in skinny_session ... 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	8 years ago
Vitezslav Novy	1bcce442d0	chan_sip: Change sip_get_codec() to return correct codec list ... Return cahnnel nativeformats to fix bridge technology selection process. Same approach as in pjsip module. ASTERISK-26143 Reported-by: Henning Holtschneider Change-Id: I64e863753954d6ad67a9e722df2ebc328705ad48	8 years ago
Thierry Magnien	23db04ed93	channels/chan_sip.c: use binding IP address for outgoing TCP SIP connections ... 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	8 years ago
Jenkins2	9bb683242c	Merge "res_pjsip_session: Add cleanup to ast_sip_session_terminate" into 13	8 years ago
George Joseph	c5b9ed20fd	res_pjsip_session: Add cleanup to ast_sip_session_terminate ... 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	8 years ago
Jean Aunis	566ad7c35d	chan_sip: Trigger reinvite if the SDP answer is included in the SIP ACK ... 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	8 years ago
George Joseph	f882ca2572	modules: change module LOAD_FAILUREs to LOAD_DECLINES ... 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	8 years ago
Alexander Traud	94bd529f9e	chan_sip: Session Timers required but refused wrongly. ... 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	8 years ago
Sean Bright	79a2c26c03	core: Remove embedded module support ... This has not worked for some time and is no longer actively maintained. Change-Id: I5110b0db69c152761b58fa025cb0a53b0e544d99	8 years ago
Sean Bright	38cebc73a3	thread safety: Don't use getprotobyname() ... 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	8 years ago
Sean Bright	8721d0bf1b	chan_sip: Add rtcp-mux support ... ASTERISK-26846 #close Change-Id: I541a1602ff55ab73684e9f8002edb9e0e745d639	8 years ago
Joshua Colp	701b753a0b	Merge "chan_iax2: Reload of iax peer results in loss of host address/port" into 13	8 years ago
Joshua Colp	947f1ebf86	Merge "chan_pjsip: Don't assume a session will have a channel." into 13	8 years ago
Richard Begg	5389666d6f	chan_iax2: Reload of iax peer results in loss of host address/port ... 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	8 years ago
Joshua Colp	c8d1b915d7	chan_pjsip: Don't assume a session will have a channel. ... 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	8 years ago
Jean Aunis	d3ef833b3b	chan_sip: Call not cancelled after receiving a 422 response ... When receiving a 422 response, the invitestate variable must be reset to INV_CALLING. ASTERISK-26841 Change-Id: Ia0502d6b02192664cefa4e75bafdd2645ce56099	8 years ago
Joshua Colp	8386a38e06	Merge "pjsip/cli_commands: pjsip show channelstats shows wrong codec" into 13	8 years ago
Daniel Journo	67c989ce78	pjsip/cli_commands: pjsip show channelstats shows wrong codec ... * cli_commands.c Fixed CLI output ASTERISK-26822 #close Change-Id: I3889ef6a8f6738fc312fab42db5efacd6e452b01	8 years ago
Richard Mudgett	4271c700f7	core: Cleanup ast_get_hint() usage. ... * manager.c:manager_state_cb() Fix potential use of uninitialized hint[] if a hint does not exist for the requested extension. Ran into this when developing a testsuite test. The AMI event ExtensionStatus came out with the hint header value containing garbage. The AMI event PresenceStatus also had the same issue. * manager.c:action_extensionstate() no need to completely initialize the hint[]. Only initialize the first element. * pbx.c:ast_add_hint() Remove unnecessary assignment. * chan_sip.c: Eliminate an unneeded hint[] local variable. We only care about the return value of ast_get_hint() there. Change-Id: Ia9a8786f01f93f1f917200f0a50bead0319af97b	8 years ago
Vitezslav Novy	d91f61f0b5	chan_sip: Allow DTLS to be disabled when reloading. ... This change fixes a problem where removing the DTLS configuration options and reloading would not disable DTLS. This occurred because the DTLS configuration was not reset to an unconfigured state on reload. ASTERISK-26313 Change-Id: I10952709cc4a7727fb50534b042bce9d64894b39	8 years ago
Igor Goncharovsky	7aa731c1c7	chan_unistim: fix char type to have consistent behavior on ARM ... There is difference exists in behaviour of char type on x86 and ARM. On x86 by default char variable type means signed char, but in ARM unsigned char used. This make binary calculations and negative values works wrong on ARM. This patch change type of char variables used for store negative values and binary calculations to signed char. ASTERISK-26714 Change-Id: Id78716dee9568a58419d4ef63c038affc3dfc7ab	8 years ago
zuul	0d6c99e715	Merge "cli: Fix various CLI documentation and completion issues" into 13	8 years ago
zuul	6958241b3f	Merge "core: Cleanup some channel snapshot staging anomalies." into 13	8 years ago
Sean Bright	ea8a610776	cli: Fix various CLI documentation and completion issues ... * app_minivm: Use built-in completion facilities to complete optional arguments. * app_voicemail: Use built-in completion facilities to complete optional arguments. * app_confbridge: Add missing colons after 'Usage' text. * chan_alsa: Use built-in completion facilities to complete optional arguments. * chan_sip: Use built-in completion facilities to complete optional arguments. Add completions for 'load' for 'sip show user', 'sip show peer', and 'sip qualify peer.' * chan_skinny: Correct and extend completions for 'skinny reset' and 'skinny show line.' * func_odbc: Correct completions for 'odbc read' and 'odbc write' * main/asterisk: Correct and extend completions for 'core show file version.' * main/astmm: Use built-in completion facilities to complete arguments for 'memory' commands. * main/bridge: Correct completions for 'bridge kick.' * main/ccss: Use built-in completion facilities to complete arguments for 'cc cancel' command. * main/cli: Add 'all' completion for 'channel request hangup.' Correct completions for 'core set debug channel.' Correct completions for 'core show calls.' * main/pbx_app: Remove redundant completions for 'core show applications.' * main/pbx_hangup_handler: Remove unused completions for 'core show hanguphandlers all.' * res_sorcery_memory_cache: Add completion for 'reload' argument of 'sorcery memory cache stale' and properly implement. Change-Id: Iee58c7392f6fec34ad9d596109117af87697bbca	8 years ago
zuul	c38cd504ad	Merge "chan_pjsip: Multidomain endpoint finding on call" into 13	8 years ago
Norbert Varga	17030100ca	chan_pjsip: Multidomain endpoint finding on call ... When PJSIP tries to call an endpoint with a domain (e.g. 1000@test.com), the user part is stripped down as it would be a trunk with a specified user, and only the host part is called as a PJSIP endpoint and can't be found. This is not correct in the case of a multidomain SIP account, so the stripping after the @ sign is done only if the whole endpoint (in multidomain case 1000@test.com) can't be found. ASTERISK-26248 Change-Id: I3a2dd6f57f3bd042df46b961eccd81d31ab202e6	8 years ago
Richard Mudgett	2817f87d27	core: Cleanup some channel snapshot staging anomalies. ... We shouldn't unlock the channel after starting a snapshot staging because another thread may interfere and do its own snapshot staging. * app_dial.c:dial_exec_full() made hold the channel lock while setting up the outgoing channel staging. Made hold the channel lock after the called party answers while updating the caller channel staging. * chan_sip.c:sip_new() completed the channel staging on off-nominal exit. Also we need to use ast_hangup() instead of ast_channel_unref() at that location. * channel.c:__ast_channel_alloc_ap() added a comment about not needing to complete the channel snapshot staging on off-nominal exit paths. * rtp_engine.c:ast_rtp_instance_set_stats_vars() made hold the channel locks while staging the channels for the stats channel variables. Change-Id: Iefb6336893163f6447bad65568722ad5d5d8212a	8 years ago
zuul	484e8ed5e3	Merge "debug_utilities: Add ast_logescalator" into 13	8 years ago
George Joseph	456bc3c704	debug_utilities: Add ast_logescalator ... The escalator works by creating a set of startup commands in cli.conf that set up logger channels and issue the debug commands for the subsystems specified. If asterisk is running when it is executed, the same commands will be issued to the running instance. The original cli.conf is saved before any changes are made and can be restored by executing '$prog --reset'. The log output will be stored in... $astlogdir/message.$uniqueid $astlogdir/debug.$uniqueid $astlogdir/dtmf.$uniqueid $astlogdir/fax.$uniqueid $astlogdir/security.$uniqueid $astlogdir/pjsip_history.$uniqueid $astlogdir/sip_history.$uniqueid Some minor tweaks were made to chan_sip, and res_pjsip_history so their history output could be send to a log channel as packets are captured. A minor tweak was also made to manager so events are output to verbose when "manager set debug on" is issued. Change-Id: I799f8e5013b86dc5282961b27383d134bf09e543	8 years ago