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Author	SHA1	Message	Date
Matthew Jordan	d68012d1a3	channels/sip/sdp_crypto: Handle SRTP keys negotiated with key lifetime/MKI ... Prior to this patch, SDP offers negotiating SDES-SRTP crypto attributes would be rejected if those crypto attributes contained either a key lifetime or a MKI parameter. While from a theoretical point of view this was defensible - Asterisk does not support key lifetimes or multiple crypto keys - from a practical point of view, this is quite a problem. A large number of endpoints offer lifetimes/MKI, which Asterisk can tolerate so long as it doesn't actually have to support anything more than a single key or refresh the key. In reality, this is (so far as we've seen) always the case. This patch is a forward port of Olle's work in the lingon-srtp-key-lifetime-1.8 branch. To quote Olle from ASTERISK-17721, it handles lifetime/MKI parameters in the following fashion: > The Lingon branch now handle lifetime and MKI parameters. > > We only accept lifetimes up to max for the crypto and higher than 10 hours > for packetization of 20 ms (50 pps). > > We only handle MKI with index 1. > > We do not really bother with counting packets and reinviting at end of > lifetime, so the min of 10 hours kind of takes care of most calls. If there > are longer ones, we rely on the other side for re-invites. > > It's still not perfect, but I personally think this is an improvement. A > configuration option for minimum lifetime accepted could be added. When the patch was ported forward, I decided against adding a configuration option as Olle's handling was more than sufficient for every case I've seen come through the issue tracker or through interoperability testing. We can revisit that decision if it proves to be false. A few small other tweaks were made to the surrounding code to reduce indentation and provide better type safety for the 'tag' parameter. Review: https://reviewboard.asterisk.org/r/4419/ Review: https://reviewboard.asterisk.org/r/4418/ ASTERISK-17721 #close Reported by: Terry Wilson ASTERISK-17899 #close Reported by: Dwayne Hubbard patches: lingon-srtp-key-lifetime-1.8.diff uploaded by oej (License 5267) ASTERISK-20233 Reported by: tootai ASTERISK-22748 Reported by: Alejandro Mejia ........ Merged revisions 432239 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 432258 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432259 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
David M. Lee	ff642289f4	Increase WebSocket frame size and improve large read handling ... Some WebSocket applications, like [chan_respoke][], require a larger frame size than the default 8k; this patch bumps the default to 16k. This patch also fixes some problems exacerbated by large frames. The sanity counter was decremented on every fread attempt in ws_safe_read(), regardless of whether data was read from the socket or not. For large frames, this could result in loss of sanity prior to reading the entire frame. (16k frame / 1448 bytes per segment = 12 segments). This patch changes the sanity counter so that it only decrements when fread() doesn't read any bytes. This more closely matches the original intention of ws_safe_read(), given that the error message is "Websocket seems unresponsive". This patch also properly logs EOF conditions, so disconnects are no longer confused with unresponsive connections. [chan_respoke]: https://github.com/respoke/chan_respoke Review: https://reviewboard.asterisk.org/r/4431/ ........ Merged revisions 432236 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 432237 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432238 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	57525c3cf2	config.h: Use real parameter names for ast_variable_new() define. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432220 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	8574c4d197	channels/chan_sip: Fix crash when transmitting packet after thread shutdown ... When the monitor thread is stopped, its pthread ID is set to a specific value (AST_PTHREADT_STOP) so that later portions of the code can determine whether or not it is safe to manipulate the thread. Unfortunately, __sip_reliable_xmit failed to check for that value, checking instead only for AST_PTHREAD_STOP. Passing the invalid yet very specific value to pthread_kill causes a crash. This patch adds a check for AST_PTHREADT_STOP in __sip_reliable_xmit such that it doesn't attempt to poke the thread if the thread has already been stopped. ASTERISK-24800 #close Reported by: JoshE ........ Merged revisions 432198 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 432199 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432200 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	a528dfc9a7	ARI/PJSIP: Apply requesting channel's format cap to created channels ... This patch addresses the following problems: * ari/resource_channels: In ARI, we currently create a format capability structure of SLIN and apply it to the new channel being created. This was originally done when the PBX core was used to create the channel, as there was a condition where a newly created channel could be created without any formats. Unfortunately, now that the Dial API is being used, this has two drawbacks: (a) SLIN, while it will ensure audio will flows, can cause a lot of needless transcodings to occur, particularly when a Local channel is created to the dialplan. When no format capabilities are available, the Dial API handles this better by handing all audio formats to the requsted channels. As such, we defer to that API to provide the format capabilities. (b) If a channel (requester) is causing this channel to be created, we currently don't use its format capabilities as we are passing in our own. However, the Dial API will use the requester channel's formats if none are passed into it, and the requester channel exists and has format capabilities. This is the "best" scenario, as it is the most likely to create a media path that minimizes transcoding. Fixing this simply entails removing the providing of the format capabilities structure to the Dial API. * chan_pjsip: Rather than blindly picking the first format in the format capability structure - which actually *can* be a video or text format - we select an audio format, and only pick the first format if that fails. That minimizes the weird scenario where we attempt to transcode between video/audio. * res_pjsip_sdp_rtp: Applied the joint capapbilites to the format structure. Since ast_request already limits us down to one format capability once the format capabilities are passed along, there's no reason to squelch it here. * channel: Fixed a comment. The reason we have to minimize our requested format capabilities down to a single format is due to Asterisk's inability to convey the format to be used back "up" a channel chain. Consider the following: PJSIP/A => L;1 <=> L;2 => PJSIP/B g,u,a g,u,a g,u,a u That is, we have PJSIP/A dialing a Local channel, where the Local;2 dials PJSIP/B. PJSIP/A has native format capabilities g722,ulaw,alaw; the Local channel has inherited those format capabilities down the line; PJSIP/B supports only ulaw. According to these format capabilities, ulaw is acceptable and should be selected across all the channels, and no transcoding should occur. However, there is no way to convey this: when L;2 and PJSIP/B are put into a bridge, we will select ulaw, but that is not conveyed to PJSIP/A and L;1. Thus, we end up with: PJSIP/A <=> L;1 <=> L;2 <=> PJSIP/B g g X u u Which causes g722 to be written to PJSIP/B. Even if we can convey the 'ulaw' choice back up the chain (which through some severe hacking in Local channels was accomplished), such that the chain looks like: PJSIP/A <=> L;1 <=> L;2 <=> PJSIP/B u u u u We have no way to tell PJSIP/A's *channel driver* to Answer in the SDP back with only 'ulaw'. This results in all the channel structures being set up correctly, but PJSIP/A *still* sending g722 and causing the chain to fall apart. There's a lot of difficulty just in setting this up, as there are numerous race conditions in the act of bridging, and no clean mechanism to pass the selected format backwards down an established channel chain. As such, the best that can be done at this point in time is clarifying the comment. Review: https://reviewboard.asterisk.org/r/4434/ ASTERISK-24812 #close Reported by: Matt Jordan ........ Merged revisions 432195 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432196 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kevin Harwell	91733b5d15	bridge_softmix: G.729 codec license held ... When more than one call using the same codec type enters into a softmix bridge and no audio is present for a channel the bridge optimizes the out frame by using the same one for all channels with the same codec type. Unfortunately, when that number (channels with same codec type) dropped to <= 1 the codec was not dereferenced. At least not until all parties left the bridge. Thus in the case of G.729 the license was not released. This patch ensures that the codec is dereferenced immediately when the optimization no longer applies. ASTERISK-24797 #close Reported by: Luke Hulsey Review: https://reviewboard.asterisk.org/r/4429/ ........ Merged revisions 432174 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 432175 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432176 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	bedf51b2ce	res_ari_channels: Return a 404 response when a requested channel variable does not exist. ... This change makes it so that if a channel variable is requested and it does not exist a 404 response will be returned instead of an allocation failed response. This makes it easier to debug and figure out what is going on for a user. ASTERISK-24677 #close Reported by: Joshua Colp ........ Merged revisions 432154 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432155 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	87b7060f36	res_pjsip_registrar: Add Expires header to 200 OK if present in REGISTER. ... Some implementations don't pay attention to the expires for individual contacts. In this case they may consider the lack of an Expires header in the 200 OK as unregistered. This change makes it so if an Expires header is present in the REGISTER we will add one in the 200 OK. ASTERISK-24785 #close Reported by: Ross Beer ........ Merged revisions 432136 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432137 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	283bb15c16	res_pjsip: Add a log message when creating a UAC dialog to a target URI that is invalid. ... ASTERISK-24499 #close Reported by: Rusty Newton ........ Merged revisions 432118 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432119 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	b3c1ad5d73	apps/app_voicemail: Demote an ERROR message to a WARNING message ... When using IMAP voicemail with FreePBX, you will often get ERROR messages complaining about not being able to find a mailbox. This is due to how FreePBX handles voicemail mailboxes. Unfortunately, app_voicemail has to consider this a configuration error, as in any other system it would be indicative of someone misconfiguring their system. Regardless, a misconfiguration is a WARNING, and not an ERROR. This patch demotes the message so that system administrators can hopefully reduce some of the noise in their log files. Note that in the original patch this was made into a NOTICE, but that's a too forgiving. ASTERISK-24790 #close Reported by: Graham Barnett patches: app_voicemail.c.patch_noise uploaded by Graham Barnett (License 6685) ........ Merged revisions 432098 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 432099 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432100 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	2ea7ccbf70	http: Add missing html tag to 'httpstatus' functionality. ... ASTERISK-24724 #close Reported by: Ashley Sanders ........ Merged revisions 432078 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 432079 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432080 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	e66b874f5d	Allow shutdown to unload modules that register bucket scheme's or codec's. ... * Change __ast_module_shutdown_ref to be NULL safe (11+). * Allow modules that call ast_bucket_scheme_register or ast_codec_register to be unloaded during graceful shutdown only (13+ only). ASTERISK-24796 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4428/ ........ Merged revisions 432058 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 432059 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432060 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	bb71672a47	main/asterisk.c: Reverse #if statement in listener() to fix code folding. ... listener() opens the same code block in two places (#if and #else). This confuses some folding editors causing it to think that an extra code block was opened. Folding in 'geany' causes all code after listener() to be folded as if it were part of that procedure. ASTERISK-24813 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4435/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432057 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	ce50fa314a	asterisk/lock.h: Fix syntax errors for non-gcc OSX with 64-bit integers. ... Add a couple of missing closing brackets / parenthesis. ASTERISK-24814 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4436/ ........ Merged revisions 432054 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 432055 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432056 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	bb06603d5f	chan_dahdi/sig_analog: Put log message strings on one line. ... With the log messages on one line, you can search for the log message seen in the log and expect to find it. ........ Merged revisions 432032 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 432034 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432036 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	340818ad12	ASTERISK-24811: Add ast_sorcery_apply_config() to res_pjsip_publish_asterisk. ... Matt Hoskins reported that res_pjsip_publish_asterisk wouldn't pull config from realtime. Turns out it was just missing a call ast_sorcery_apply_config(). res_pjsip_acl was missing it as well, so I added it. The other pjsip modules looked OK. ASTERISK-24811 #close Reported-by: Matt Hoskins Tested-by: George Joseph Tested-by: Matt Hoskins patches: res_pjsip_publish_asterisk.c.patch submitted by Matt Hoskins (license 6688) Review: https://reviewboard.asterisk.org/r/4433/ ........ Merged revisions 432033 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432035 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	4dab71831f	apps/app_voicemail: Fix IMAP header compatibility issue with Microsoft Exchange ... When interfacing with Microsoft Exchange, custom headers will be returned as all lower case. Currently, the IMAP header code will fail to parse the returned custom headers, as it will be performing a case sensitive comparison. This can cause playback of messages to fail, as needed information - such as origtime - will not be present. This patch updates app_voicemail's header parsing code to perform a case insensitive lookup for the requested custom headers. Since the headers are specific to Asterisk, e.g., 'x-asterisk-vm-orig-time', and headers should be unique in an IMAP message, this should cause no issues with other systems. ASTERISK-24787 #close Reported by: Graham Barnett patches: app_voicemail.c.patch_MSExchange uploaded by Graham Barnett (License 6685) ........ Merged revisions 432012 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 432013 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432014 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	05cc6d6d55	chan_dahdi: Remove some dead code. ... ........ Merged revisions 431992 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 431993 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431994 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	252aee4228	ISDN AOC: Fix crash from an AOC-E message that doesn't have a channel association. ... Processing an AOC-E event that does not or no longer has a channel association causes a crash. The problem with posting AOC events to the channel topic is that AOC-E events don't always have a channel association and posting the event to the all channels topic is just wrong. AOC-E events do however have their own charging association method to refer to the agreement with the charging entity. * Changed the AOC events to post to the AMI manager topic instead of the channel topics. If a channel is associated with the event then channel snapshot information is supplied with the AMI event. * Eliminated RAII_VAR() usage in aoc_to_ami() and ast_aoc_manager_event(). This patch supercedes the patch on Review: https://reviewboard.asterisk.org/r/4427/ ASTERISK-22670 #close Reported by: klaus3000 ASTERISK-24689 #close Reported by: Marcel Manz ASTERISK-24740 #close Reported by: Panos Gkikakis Review: https://reviewboard.asterisk.org/r/4430/ ........ Merged revisions 431974 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431975 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	6992b2e8fa	res_pjsip_refer: Handle INVITE with Replaces failure after answer. ... * Fixed hangup handling of the session->channel after answer if the ast_channel_move() or ast_bridge_impart() fails. We are still the thread controlling the session->channel so we need to call ast_hangup() to kill the channel. * Fixed debug messages in refer_incoming_invite_request() referencing incorrect channnels on success. Code comments now say why the session->channel cannot be used. Review: https://reviewboard.asterisk.org/r/4422/ ........ Merged revisions 431956 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431957 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	e3fd826cdb	tcptls: Handle new OpenSSL compile time option to disable SSLv3 ... Some distributions are going to disable SSLv3 at compile time. This option can be checked using the directive OPENSSL_NO_SSL3_METHOD. This patch updates the TCP/TLS handling in Asterisk to look for that directive before attempting to use the SSLv3 specific methods. ASTERISK-24799 #close Reported by: Alexander Traud patches: no-ssl3-method.patch uploaded by Alexander Traud (License 6520) ........ Merged revisions 431936 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 431937 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431938 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	a4774ceaa5	Create work around for scheduler leaks during shutdown. ... * Added ast_sched_clean_by_callback for cleanup of scheduled events that have not yet fired. * Run all pending peercnt_remove_cb and replace_callno events in chan_iax2. Cleanup of replace_callno events is only run 11, since it no longer releases any references or allocations in 13+. ASTERISK-24451 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4425/ ........ Merged revisions 431916 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 431917 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431918 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	09bfe4b208	res_pjsip_refer: Fix crash from a REFER and BYE collision. ... Analyzing a one-off crash on a busy system showed that processing a REFER request had a NULL session channel pointer. The only way I can think of that could cause this is if an outgoing BYE transaction overlapped the incoming REFER transaction in a collision. Asterisk sends a BYE while the phone sends a REFER to complete an attended transfer. * Made check the session channel pointer before processing an incoming REFER request in res_pjsip_refer. * Fixed similar crash potential for res_pjsip supplement incoming request processing for res_pjsip_sdp_rtp INFO, res_pjsip_caller_id INVITE/UPDATE, res_pjsip_messaging MESSAGE, and res_pjsip_send_to_voicemail REFER messages. * Made res_pjsip_messaging respond to a message body too large with a 413 instead of ignoring it. ASTERISK-24700 #close Reported by: Zane Conkle Review: https://reviewboard.asterisk.org/r/4417/ ........ Merged revisions 431898 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431899 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	d808eace5c	res/res_rtp_asterisk: Fix crash in debug from RTCP reports without report block ... When RTCP debugging was enabled, an RTCP report without a report block would cause a crash. This was due to the verbose output not checking to see if the report_block pointer was NULl before dereferencing it. This patch adds the necessary check to prevent printing any verbose output if the far side hasn't provided us the information they should have. ASTERISK-24791 #close Reported by: JoshE Tested by: JoshE ........ Merged revisions 431879 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431880 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	55eb8fc068	pjsip: Remove "contact" type from pjsip.conf.sample ... The "contact" object is not meant to be configured from the pjsip.conf configuration file. It is meant to be created as a result of a registration and stored elsewhere. ASTERISK-24085 #close Reported by: Rusty Newton ........ Merged revisions 431860 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431861 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	55709bc1f7	install_prereq: Tweak flags when configuring pjproject. ... This change does two things: 1. Disables debugging so assertions which can return an error do, instead of asserting. 2. Enables IPv6 support. ASTERISK-24632 #close Reported by: Rusty Newton ........ Merged revisions 431843 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431844 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	e78dd39885	res_sorcery_config: Improve object lookup times. ... The res_sorcery_config module currently uses a fixed bucket size of 53. This means that depending on the number of objects you either end up with excess buckets or a lot of collisions. Due to the way that res_sorcery_config is implemented it's actually possible to make the bucket size dynamic based on the number of objects. This is due to the fact that each loading of the config file produces a new container and does not modify the existing one. This change uses the number of expected objects and finds a prime number near it. In practice depending on the number of objects this can speed up lookups anywhere from 2X to 15X. This change also removes the lock from the container as it is not needed. Review: https://reviewboard.asterisk.org/r/4423/ ........ Merged revisions 431841 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431842 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	e6fe69b76c	res_pjsip: Add "pjsip show version" CLI command. ... When debugging things it can be useful to know absolutely what version of pjproject res_pjsip is running against. This change adds a "pjsip show version" CLI command which can be used to query for this. ASTERISK-24685 #close Reported by: Joshua Colp Review: https://reviewboard.asterisk.org/r/4424/ ........ Merged revisions 431824 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431825 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	17f9e0cacc	res_timing_pthread: Fix leaky pipes. ... During some refactoring the way private information for timers was stored was changed. As a result of this the action which normally removed the timer upon closure in res_timing_pthread was also removed causing the timer to remain after it should using up resources. This change ensures that the timer is removed upon closure. ASTERISK-24768 #close Reported by: Matthias Urlichs patches: timer.patch submitted by Matthias Urlichs (license 5508) ........ Merged revisions 431807 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431808 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	d1bd8b091b	apps/app_mixmonitor: Move Test Event for MIXMONITOR_END to after it finishes ... The Test Event for MIXMONITOR_END - which signals that a MixMonitor has completed - technically fired before the filestream was closed. If a test used this to trigger a condition to verify that the file was written, it could result in a race condition where the file size would not be what the test expected. Luckily, no tests were using this (although they should have been). Since the test event needed to be moved after the point where the MixMonitor autochan has been destroyed, the test event no longer emits the channel name. Luckily, nothing needs it. ........ Merged revisions 431788 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 431789 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431790 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	455a98a2f8	sorcery: Output an error message if a wizard is specified for an object type and it isn't found. ... ASTERISK-24612 #close Reported by: Joshua Colp ........ Merged revisions 431771 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431772 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	fae6bf8ace	res_pjsip_exten_state: Improve log message when a subscription is attempted to a non-existent extension. ... ASTERISK-24716 #close Reported by: Rusty Newton ........ Merged revisions 431754 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431755 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	cc96e4a7ef	Multiple revisions 431751-431752 ... ........ r431751 | file | 2015-02-14 14:19:07 -0400 (Sat, 14 Feb 2015) | 5 lines chan_pjsip: Fix crash when CHANNEL dialplan function is invoked with pjsip argument and no type. ASTERISK-24771 #close Reported by: Niklas Larsson ........ r431752 | file | 2015-02-14 14:20:27 -0400 (Sat, 14 Feb 2015) | 2 lines 'information' ends with an 'n'. ........ Merged revisions 431751-431752 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431753 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	f00ebf0a2d	res_pjsip_session: Fix double re-INVITE collision crash. ... A multi-asterisk box setup with direct media enabled would occasionally crash when two re-INVITE collisions on a call leg happen in a row. The re-INVITE logic only had one timer struct to defer the re-INVITE. When the second collision happens the timer struct is overwritten and put into the timer heap again. Resources for the first timer are leaked and the heap has two positions occupied by the same timer struct. Now the heap ordering is potentially corrupted, the timer will fire twice, and any resources allocated for the second timer will be released twice. * The solution is to put the collided re-INVITE into the delayed requests queue with all the other delayed requests and cherry pick the next request that can come off the queue when an event happens. * Changed to put delayed BYE requests at the head of the delayed queue. There is no sense in processing delayed UPDATEs and re-INVITEs when a BYE has been requested. * Made the start of a BYE request flush the delayed requests queue to prevent a delayed request from overlapping the BYE transaction. I saw a few cases where a delayed re-INVITE got started after the BYE transaction started. * Changed the delayed_request struct to use an enum instead of a string for the request method. Cherry picking the queue is easier with an enum than string comparisons and the compiler can warn if a switch statement does not cover all defined enum values. * Improved the debug output to give more information. It helps to know which channel is involved with an endpoint. Trunks can have many channels associated with the endpoint at the same time. ASTERISK-24727 #close Reported by: Mark Michelson Review: https://reviewboard.asterisk.org/r/4414/ ........ Merged revisions 431734 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431735 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	29f66b0429	ARI/PJSIP: Add the ability to redirect (transfer) a channel in a Stasis app ... This patch adds a new feature to ARI to redirect a channel to another server, and fixes a few bugs in PJSIP's handling of the Transfer dialplan application/ARI redirect capability. *New Feature* A new operation has been added to the ARI channels resource, redirect. With this, a channel in a Stasis application can be redirected to another endpoint of the same underlying channel technology. *Bug fixes* In the process of writing this new feature, two bugs were fixed in the PJSIP stack: (1) The existing .transfer channel callback had the limitation that it could only transfer channels to a SIP URI, i.e., you had to pass 'PJSIP/sip:foo@my_provider.com' to the dialplan application. While this is still supported, it is somewhat unintuitive - particularly in a world full of endpoints. As such, we now also support specifying the PJSIP endpoint to transfer to. (2) res_pjsip_multihomed was, unfortunately, trying to 'help' a 302 redirect by updating its Contact header. Alas, that resulted in the forwarding destination set by the dialplan application/ARI resource/whatever being rewritten with very incorrect information. Hence, we now don't bother updating an outgoing response if it is a 302. Since this took a looong time to find, some additional debug statements have been added to those modules that update the Contact headers. Review: https://reviewboard.asterisk.org/r/4316/ ASTERISK-24015 #close Reported by: Private Name ASTERISK-24703 #close Reported by: Matt Jordan ........ Merged revisions 431717 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431718 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kevin Harwell	9d081ed06c	res_pjsip: dtls_handler causes Asterisk to crash ... There have been a couple of times where a crash occurred in the dtls_handler section of the code for res_pjsip. Unfortunately, in working this issue the problem was unable to be reproduced. After looking at the backtraces and through the code the current best guess as to why this happened might be due to a reentrance problem and the strtok function. So, the current fix is to convert the strtok function into the reentrant version of the function, strtok_r. ASTERISK-24741 #close Reported by: Zane Conkle Review: https://reviewboard.asterisk.org/r/4409/ ........ Merged revisions 431698 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431699 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kevin Harwell	cc85e55d88	ari_websockets: removed extra check on websocket session read ... When merging the websocket timeout issue (ASTERISK-24701) an extra, almost duplicate, check was left in the code that should not have been. This removes it. ASTERISK-24701 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/4412/ ........ Merged revisions 431693 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431695 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	e2d3215b83	HTTP: Stop accepting requests on final system shutdown. ... There are three CLI commands to stop and restart Asterisk each. 1) core stop/restart now - Hangup all calls and stop or restart Asterisk. New channels are prevented while the shutdown request is pending. 2) core stop/restart gracefully - Stop or restart Asterisk when there are no calls remaining in the system. New channels are prevented while the shutdown request is pending. 3) core stop/restart when convenient - Stop or restart Asterisk when there are no calls in the system. New calls are not prevented while the shutdown request is pending. ARI has made stopping/restarting Asterisk more problematic. While a shutdown request is pending it is desirable to continue to process ARI HTTP requests for current calls. To handle the current calls while a shutdown request is pending, a new committed to shutdown phase is needed so ARI applications can deal with the calls until the system is fully committed to shutdown. * Added a new shutdown committed phase so ARI applications can deal with calls until the final committed to shutdown phase is reached. * Made refuse new HTTP requests when the system has reached the final system shutdown phase. Starting anything while the system is actively releasing resources and unloading modules is not a good thing. * Split the bridging framework shutdown to not cleanup the global bridging containers when shutting down in a hurry. This is similar to how other modules prevent crashes on rapid system shutdown. * Moved ast_begin_shutdown(), ast_cancel_shutdown(), and ast_shutting_down(). You should not have to include channel.h just to access these system functions. ASTERISK-24752 #close Reported by: Matthew Jordan Review: https://reviewboard.asterisk.org/r/4399/ ........ Merged revisions 431692 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431694 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	5a17ed7a38	channels/chan_sip: Fix RealTime error during SIP unregistration with MariaDB ... When a SIP device that has its registration stored in RealTime unregisters, the entry for that device is updated with blank values, i.e., "", indicating that it is no longer registered. Unfortunately, one of those values that is 'blanked' is the device's port. If the column type for the port is not a string datatype (the recommended type is integer), an ODBC or database error will be thrown. MariaDB does not coerce empty strings to a valid integer value. This patch updates the query run from chan_sip such that it replaces the port value with a value of '0', as opposed to a blank value. This is the value that other database backends coerce the empty string ("") to already, and the handling of reading a RealTime registration value from a backend already anticipates receiving a port of '0' from the backends. ASTERISK-24772 #close Reported by: Richard Miller patches: chan_sip.diff uploaded by Richard Miller (License 5685) ........ Merged revisions 431673 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 431674 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431675 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	8cc50b1ebc	Enable REF_DEBUG for ast_module_ref / ast_module_unref. ... Add ast_module_shutdown_ref for use by modules that can only be unloaded during graceful shutdown. When REF_DEBUG is enabled: * Add an empty ao2 object to struct ast_module. * Allocate ao2 object when the module is loaded. * Perform an ao2_ref in each place where mod->usecount is manipulated. * ao2_cleanup on module unload. ASTERISK-24479 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4141/ ........ Merged revisions 431662 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 431663 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431672 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kevin Harwell	137c4b0778	res_http_websocket: websocket write timeout fails to fully disconnect ... When writing to a websocket if a timeout occurred the underlying socket did not get closed/disconnected. This patch makes sure the websocket gets disconnected on a write timeout. Also a notice is logged stating that the websocket was disconnected. ASTERISK-24701 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/4412/ ........ Merged revisions 431669 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 431670 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431671 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	49161d8df8	res_pjsip_config_wizard: Add ability to auto-create hints. ... Looking at the Super Awesome Company sample reminded me that creating hints is just plain gruntwork. So you can now have the pjsip conifg wizard auto-create them for you. Specifying 'hint_exten' in the wizard will create 'exten => <hint_exten>,hint/PJSIP/<wizard_id>' in whatever is specified for 'hint_context'. Specifying 'hint_application' in the wizard will create 'exten => <hint_exten>,1,<hint_application>' in whatever is specified for 'hint_context'. The default for 'hint_context' is the endpoint's context. There's no default for 'hint_application'. If not specified, no app is added. There's no default for 'hint_exten'. If not specified, neither the hint itself nor the application will be created. Some may think this is the slippery slope to users.conf but hints are a basic necessity for phones unlike voicemail, manager, etc that users.conf creates. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4383/ ........ Merged revisions 431643 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431644 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	858e825568	res/ari/resource_channels: Add missing 'no_answer' reason to DELETE /channels ... One of the canonical reasons for hanging up a channel is because the far end failed to answer - or because someone else answered, and we want to get rid of this channel. This patch adds the missing value to the 'reason' query parameter for the DELETE /channels operation. Review: https://reviewboard.asterisk.org/r/4400 ASTERISK-24745 #close Reported by: Ben Merrills patches: add_no_answer_ari_hangup_cause.diff uploaded by Ben Merrills (License 6678) ........ Merged revisions 431622 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431623 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	17247daae6	res/res_odbc: Remove unneeded queries when determining if a table exists ... This patch modifies the ast_odbc_find_table function such that it only performs a lookup of the requested table if the table is not already known. Prior to this patch, a queries would be executed against the database even if the table was already known and cached. Review: https://reviewboard.asterisk.org/r/4405/ ASTERISK-24742 #close Reported by: ibercom patches: patch.diff uploaded by ibercom (License 6599) ........ Merged revisions 431617 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 431618 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431619 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	2ebe811d80	res/res_pjsip_sdp_rtp: Fix leak of local ICE candidates when applying to SDP ... When an SDP is created for an outgoing request/response, the ICE candidates obtained from the RTP instance are currently leaked. This causes the ao2 container that holds the candidates to never properly be reclaimed when the RTP instance is destroyed. This patch properly decrements the ICE candidates' container if it is successfully obtained. ASTERISK-24769 #close Reported by: Matt Jordan ........ Merged revisions 431600 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431601 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Scott Griepentrog	7ca1a0da04	various: cleanup issues found during leak hunt ... In this collection of small patches to prevent Valgrind errors are: fixes for reference leaks in config hooks, evaluating a parameter beyond bounds, and accessing a structure after a lock where it could have been already free'd. Review: https://reviewboard.asterisk.org/r/4407/ ........ Merged revisions 431583 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431584 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	a79c920aa1	res_pjsip_keepalive: Don't crash if PJSIP module is not loaded. ... ........ Merged revisions 431555 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431556 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	03ce56d6c5	sorcery: Don't try to load object types which haven't been defined. ... The act of defining wizards for an object type in sorcery.conf will create a minimal object type. This can cause a problem when a module has multiple sorcery instances (which all get the wizards from sorcery.conf applied) but the sorcery instances do not all contain full information about the object types. Upon loading errors will occur stating that the objects can not be created. This is confusing and is actually perfectly fine. This change makes it so that only object types which have been fully defined will be loaded. ASTERISK-24748 #close Reported by: Joshua Colp ........ Merged revisions 431538 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431539 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	14a57782a6	res_format_attr_h264: Fix crash when determining joint capability. ... The res_format_attr_h264 module currently incorrectly attempts to copy SPS and PPS information from the wrong attribute. This change fixes that. ASTERISK-24616 #close Reported by: Yura Kocyuba Review: https://reviewboard.asterisk.org/r/4392/ ........ Merged revisions 431521 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431522 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	23bb5f6a73	app_agent_pool: Fix initial module load agent device state reporting. ... When the app_agent_pool module initially loads there is a race condition between the thread loading agents.conf and the device state internal processing thread. If the device state internal processing thread handles the agent creation state updates before the thread that loaded agents.conf registers the device state provider callback then the cached agent state is "Invalid". When a consumer module like app_queue asks for the agent state it gets the cached "Invalid" state instead of the real state from the provider. * Moved loading the agents.conf configuration to the last thing setup by app_agent_pool in load_module(). Now the device state provider callback is registered before the config is loaded so the agent creation state updates are guaranteed to get the initial device state. * Removed some now redundant config cleanup on error in load_config(). * Added lock protection when accessing the device state in agent_pvt_devstate_get() and eliminated the RAII_VAR() usage. ASTERISK-24737 #close Reported by: Steve Pitts Review: https://reviewboard.asterisk.org/r/4390/ ........ Merged revisions 431492 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431493 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago