8053 Commits (d04445c24acb316840108fefd92e6b43c0a92115)

Author	SHA1	Message	Date
Matthew Jordan	1106e8fd0f	main/stasis: Allow subscriptions to use a threadpool for message delivery ... Prior to this patch, all Stasis subscriptions would receive a dedicated thread for servicing published messages. In contrast, prior to r400178 (see review https://reviewboard.asterisk.org/r/2881/), the subscriptions shared a thread pool. It was discovered during some initial work on Stasis that, for a low subscription count with high message throughput, the threadpool was not as performant as simply having a dedicated thread per subscriber. For situations where a subscriber receives a substantial number of messages and is always present, the model of having a dedicated thread per subscriber makes sense. While we still have plenty of subscriptions that would follow this model, e.g., AMI, CDRs, CEL, etc., there are plenty that also fall into the following two categories: * Large number of subscriptions, specifically those tied to endpoints/peers. * Low number of messages. Some subscriptions exist specifically to coordinate a single message - the subscription is created, a message is published, the delivery is synchronized, and the subscription is destroyed. In both of the latter two cases, creating a dedicated thread is wasteful (and in the case of a large number of peers/endpoints, harmful). In those cases, having shared delivery threads is far more performant. This patch adds the ability of a subscriber to Stasis to choose whether or not their messages are dispatched on a dedicated thread or on a threadpool. The threadpool is configurable through stasis.conf. Review: https://reviewboard.asterisk.org/r/4193 ASTERISK-24533 #close Reported by: xrobau Tested by: xrobau ........ Merged revisions 428681 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 428687 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428688 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	d25eda5fb2	AST-2014-015: Fix race condition in chan_pjsip when sending responses after a CANCEL has been received. ... Due to the serialized architecture of chan_pjsip there exists a race condition where a CANCEL may be received and processed before responses (such as 180 Ringing, 183 Session Progress, and 200 OK) are sent. Since the session is in an unexpected state PJSIP will assert when this is attempted. This change makes it so that these responses are not sent on disconnected sessions. ASTERISK-24471 #close Reported by: yaron nahum ........ Merged revisions 428301 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 428302 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428303 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	a7c9f4c668	ast_str: Fix improper member access to struct ast_str members. ... Accessing members of struct ast_str outside of the string manipulation API routines is invalid since struct ast_str is supposed to be treated as opaque. Review: https://reviewboard.asterisk.org/r/4194/ ........ Merged revisions 428244 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 428245 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 428246 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428255 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	4cea5fd4ba	chan_sip: Fix theoretical leak of p->refer. ... If transmit_refer is called when p->refer is already allocated, it leaks the previous allocation. Updated code to always free previous allocation during a new allocation. Also instead of checking if we have a previous allocation, always create a clean record. ASTERISK-15242 #close Reported by: David Woolley Review: https://reviewboard.asterisk.org/r/4160/ ........ Merged revisions 428117 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 428118 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 428119 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428120 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	656601d8c4	chan_pjsip: Remove AOR check when dialing and one is specified. ... The AOR value may contain the name of an AOR or a full SIP URI. Checking if the AOR exists can't be done as a result of this. ........ Merged revisions 428051 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 428052 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428053 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	bc02cbabd9	chan_sip: Fix bug where DTLS configuration from general would copy dtlsenable. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428034 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	ece61f5ed1	chan_pjsip: Add additional log message when an AOR is specified when dialing and it does not exist. ... ASTERISK-24499 #close Reported by: Rusty Newton ........ Merged revisions 428007 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 428008 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428009 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	49e63a191d	chan_motif / chan_pjsip: Fix incorrect "No such module" messages when reloading. ... For chan_motif the direct return value of the underlying config options framework was passed back. This can relay various states which the module loader would not interpet as success. It has been changed so only on errors will it report back an error. For chan_pjsip the code implemented a dummy reload function which always returned an error. This has been removed as all configuration is held within res_pjsip instead. ASTERISK-23651 #close Reported by: Rusty Newton ........ Merged revisions 427981 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 427982 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427983 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	d0523b4b3c	chan_sip: Add support for setting DTLS configuration in the general section. ... Configuration of DTLS in the general section will be applied to any users or peers. If configuration exists at their level it overrides the general section values. ASTERISK-24128 #close Reported by: Michael K. patches: dtls_default_settings.patch submitted by Michael K. (license 6621) Review: https://reviewboard.asterisk.org/r/3867/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427950 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	f4392c4b6d	channels/chan_mgcp: Fix regression which causes gateways to be skipped ... In r227276, a while loop was turned into a for loop. Unfortunately, a portion of the while loop was left in the code such that, when a static gateway is encountered in the list of MGCP gateways, the next gateway would be skipped. At best, we would simply flip past a gateway; at worst, this could lead to a crash. ASTERISK-24500 #close Reported by: Xavier Hienne patches: chan_mgcp.patch uploaded by Xavier Hienne (License 6657) ........ Merged revisions 427613 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 427614 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 427615 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427616 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	d4fd0774f4	chan_console: Fix reference leaks to pvt. ... Fix a bunch of calls to get_active_pvt where the reference is never released. ASTERISK-24504 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4152/ ........ Merged revisions 427554 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 427555 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 427557 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427566 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	ac091d4184	chan_pjsip: Add support for passing hold and unhold requests through. ... This change adds an option, moh_passthrough, that when enabled will pass hold and unhold requests through using a SIP re-invite. When placing on hold a re-invite with sendonly will be sent and when taking off hold a re-invite with sendrecv will be sent. This allows remote servers to handle the musiconhold instead of the local Asterisk instance being responsible. Review: https://reviewboard.asterisk.org/r/4103/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427112 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	d88282af40	channels/sip/reqresp_parser: Fix unit tests for r426594 ... When r426594 was made, it did not take into account a unit test that verified that the function properly populated the unsupported buffer. The function would previously memset the buffer if it detected it had any contents; since this function can now be called iteratively on successive headers, the unit tests would now fail. This patch updates the unit tests to reset the buffer themselves between successive calls, and updates the documentation of the function to note that this is now required. ........ Merged revisions 426858 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 426860 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 426863 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 426865 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426868 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Igor Goncharovskiy	c866ced76b	Add additional checks for NULL pointers to fix several crashes reported. ...	11 years ago
Matthew Jordan	0ddc3bde24	channels/chan_sip: Add improved support for 4xx error codes ... This patch adds support for 414, 493, 479, and a stray 400 response in REGISTER response handling. This helps interoperability in a number of scenarios. Review: https://reviewboard.asterisk.org/r/3437 patches: rb3437.patch uploaded by oej (License 5267) ........ Merged revisions 426599 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 426600 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 426601 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 426602 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426603 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	ff83ff564c	channels/chan_sip: Support mutltiple Supported and Required headers ... A SIP request may contain multiple Supported: and Required: headers. Currently, chan_sip only parses the first Supported/Required header it finds. This patch adds support for multiple Supported/Required headers for INVITE requests. Review: https://reviewboard.asterisk.org/r/2478 ASTERISK-21721 #close Reported by: Olle Johansson patches: rb2478.patch uploaded by oej (License 5267) ........ Merged revisions 426594 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 426595 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 426596 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 426597 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426598 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Tzafrir Cohen	8a69aedd17	Fix building chan_phone on big endian systems ... A left over from the formats conversion (Corey Farrell). ASTERISK-24458 #close Review: https://reviewboard.asterisk.org/r/4117/ ........ Merged revisions 426570 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426573 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	86eea19c8f	channels/chan_sip: Respect outboundproxy setting when sending qualify requests ... The outboundproxy setting is currently ignored when sending OPTIONS requests as a result of the qualify setting. This means that if an Asterisk server is unable to send the packet directly to a peer, it is unable to qualify any non-inbound registered peer (e.g. a peer SIP Trunk). This patch grabs the outboundproxy information for a peer when a qualify attempt is being constructed and, if it finds the information, uses it when sending the OPTIONS request. Review: https://reviewboard.asterisk.org/r/3948 ASTERISK-24063 #close Reported by: Damian Ivereigh patches: outboundproxy-dai.patch uploaded by Damian Ivereigh (License 6632) ........ Merged revisions 425818 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 425819 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 425820 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425821 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425822 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	86a4ce4957	PJSIP: Enforce module load dependencies ... This enforces that res_pjsip, res_pjsip_session, and res_pjsip_pubsub have loaded properly before attempting to load any modules that depend on them since the module loader system is not currently capable of resolving module dependencies on its own. ASTERISK-24312 #close Reported by: Dafi Ni Review: https://reviewboard.asterisk.org/r/4062/ ........ Merged revisions 425690 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425691 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425700 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Igor Goncharovskiy	a770ca168d	Fix loss of voice after second call drops (on a second line) in case using multiple lines on unistim phones. There is regression was introduced in r391379. ...	11 years ago
Richard Mudgett	28c11fff78	chan_motif: Cleanup jingle_tech.capabilities only once. ... ........ Merged revisions 425627 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425628 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	9e72c74db5	chan_sip: Fix so asterisk won't send reINVITE after a BYE. ... After a reINVITE glare situation, Asterisk would re-send the reINVITE even though the call had been hung up in the mean time. This patch unschedules the reinvite when handling the BYE. ASTERISK-22791 #close Reported by: Paolo Compagnini Tested by: Paolo Compagnini Review: https://reviewboard.asterisk.org/r/4056/ (testcase is in review r4055) ........ Merged revisions 425296 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 425297 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 425298 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425299 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425300 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	d3f525fd8f	chan_sip: Fix dialog leak resulting from missing ACK to re-INVITE. ... If a device re-INVITEs at the same time as the dialog is hung up, and if then the ACK to the re-INVITE never reaches Asterisk, chan_sip would fail to destroy the dialog after a while. This resulted in (most prominently) file handle leaks. (Patch reindented by me.) ASTERISK-20784 #close ASTERISK-15879 #close Reported by: Torrey Searle, Nitesh Bansal Patches: reinvite_ack_timeout.patch uploaded by Torrey Searle (License #5334) patch_asterisk_20784.txt uploaded by Nitesh Bansal (License #6418) Reviewboard: https://reviewboard.asterisk.org/r/4052/ (testcase can be found at r4051) ........ Merged revisions 425068 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 425069 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 425070 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425071 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425072 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	c013916869	pjsip/dialplan_functions: Handle PJSIP_MEDIA_OFFER called on non-PJSIP channels ... Calling PJSIP_MEDIA_OFFER on a non-PJSIP channel is hazardous to your health. It will treat the channels as a PJSIP channel, eventually hitting an ao2 error, FRACKing on assertion error, and quite likely crashing. This patch adds checks to the read/write callbacks that ensure that the channel technology is of type 'PJSIP' before attempting to operate on the channel. #SIPit31 ASTERISK-24382 #close Reported by: Matt Jordan ........ Merged revisions 424621 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 424622 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424623 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	1b0902caa4	chan_motif: Correct last commit to use ao2_cleanup to free format cap ... This fix applies to 13 and trunk. ASTERISK-24384 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4043/ ........ Merged revisions 424554 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424555 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	0cea12b9e8	chan_motif: Release format capabilities and config on module load error ... ASTERISK-24384 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4043/ ........ Merged revisions 424550 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 424551 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 424552 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424553 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	0165c5f95a	chan_pjsip: Fix deadlock when masquerading PJSIP channels. ... Performing a directed call pickup resulted in a deadlock when PJSIP channels were involved. A masquerade needs to hold onto the channel locks while it swaps channel information between the two channels involved in the masquerade. With PJSIP channels, the fixup routine needed to push a fixup task onto the PJSIP channel's serializer. Unfortunately, if the serializer was also processing a task that needed to lock the channel, you get deadlock. * Added a new control frame that is used to notify the channels that a masquerade is about to start and when it has completed. * Added the ability to query taskprocessors if the current thread is the taskprocessor thread. * Added the ability to suspend/unsuspend the PJSIP serializer thread so a masquerade could fixup the PJSIP channel without using the serializer. ASTERISK-24356 #close Reported by: rmudgett Review: https://reviewboard.asterisk.org/r/4034/ ........ Merged revisions 424471 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 424472 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424473 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	2f570094b7	chan_pjsip: Fix an assertion for channels that lack formats on creation ... ASTERISK-24222 #close Reported by: Mark Michelson Review: https://reviewboard.asterisk.org/r/4017/ ........ Merged revisions 424333 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424358 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	c3a7524457	chan_sip: Simplify some unref code by removing unlink_peer_from_tables. ... ASTERISK-22945 #related Reported by: ibercom Patches: asterisk11-chan_sip-simplifies.patch uploaded by ibercom (License #6599) ........ Merged revisions 424181 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 424182 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 424183 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 424184 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424185 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	841d978a30	chan_sip: Remove excess ref of realtime peer before sip_poke_peer. ... The peer is referenced at the end of sip_poke_peer, it should not get an extra ref before the call to sip_poke_peer. This fixes a memory leak. ASTERISK-22945 #close Reported by: ibercom Tested by: Yuriy Gorlichenko Patches: asterisk11.patch uploaded by ibercom (License #6599) Review: https://reviewboard.asterisk.org/r/4031/ ........ Merged revisions 424176 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 424177 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 424178 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 424179 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424180 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	76744543b4	res_pjsip_session: Add additional checks for delaying session refreshes. ... There are certain situations which no checks existed for which need to prevent session refreshes. This includes sending a session refresh with SDP before SDP negotiation has completed and sending a session refresh before the dialog itself has been established. Checks for these have been added. Additionally COLP related UPDATEs were including SDP when it is not needed. Review: https://reviewboard.asterisk.org/r/4008/ ........ Merged revisions 424056 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 424057 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424058 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	37179a2b1f	core: Don't allow free to mean ast_free (and malloc, etc..). ... This gets rid of most old libc free/malloc/realloc and replaces them with ast_free and friends. When compiling with MALLOC_DEBUG you'll notice it when you're mistakenly using one of the libc variants. For the legacy cases you can define WRAP_LIBC_MALLOC before including asterisk.h. Even better would be if the errors were also enabled when compiling without MALLOC_DEBUG, but that's a slightly more invasive header file change. Those compiling addons/format_mp3 will need to rerun ./contrib/scripts/get_mp3_source.sh. ASTERISK-24348 #related Review: https://reviewboard.asterisk.org/r/4015/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423978 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	39fada4dc9	chan_sip: Unref outbound proxy structure on dialog/pvt destruction. ... Make sure outbound proxy refs are always unreffed on dialog destruction. Review: https://reviewboard.asterisk.org/r/4016/ ........ Merged revisions 423800 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 423801 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 423802 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 423803 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423804 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	593455621b	chan_sip: On INVITE retransmission, don't add an extra 503 response. ... INVITE arrives to asterisk, asterisk responds Busy(). If the INVITE is retransmitted, asterisk would generate a 503 in addition to the 486. Thanks Torrey Searle for providing a working regression test. ASTERISK-24335 #close Review: https://reviewboard.asterisk.org/r/4003/ Patches: retrans_486_invite.patch uploaded by Torrey Searle (License #5334) ........ Merged revisions 423720 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 423721 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 423722 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 423723 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423724 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	ec0313c411	res_pjsip_sdp_rtp.c: Fix native formats containing formats that were not negotiated. ... Outgoing PJSIP calls can result in non-negotiated formats listed in the channel's native formats if video formats are listed in the endpoint's configuration. The resulting call could then use a non-negotiated format resulting in one way audio. * Simplified the update of session->req_caps in set_caps(). Why do something in five steps when only one is needed? AFS-162 #close Review: https://reviewboard.asterisk.org/r/4000/ ........ Merged revisions 423561 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423563 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	7e602175ff	chan_iax2: Fix a crash when using chan_iax2 jitterbuffer settings ... Caused by format changes in Asterisk 13 ASTERISK-24265 #close Reported by: Dafi Ni Review: https://reviewboard.asterisk.org/r/3999/ ........ Merged revisions 423524 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423526 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	02295456ef	chan_rtp: Add unicast RTP support. ... This module supports sending both unicast and multicast RTP to a specified target. Multicast functionality is the same as chan_multicast_rtp was. In the case of unicast a specific IP address and port can be specified, along with optional RTP engine and format in the form of: UnicastRTP/<ip address>:<port>/<engine>/<format> This can be useful for sending a copy of a media stream to another application for processing. Review: https://reviewboard.asterisk.org/r/3981/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423004 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	5a1de68b9a	devicestate.c: Minor tweaks ... * In ast_state_chan2dev() use ARRAY_LEN() instead of a sentinel value in chan2dev[]. * Fix some comments in chan_iax2.c. ........ Merged revisions 422661 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422663 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	1b64f353f1	Resolve race condition where channels enter dialplan application before media has been negotiated. ... Testsuite tests will occasionally fail because on reception of a 200 OK SIP response, an AST_CONTROL_ANSWER frame is queued prior to when media has finished being negotiated. This is because session supplements are called into before PJSIP's inv_session code has told us that media has been updated. Sometimes the queued answer frame is handled by the PBX thread before the ensuing media negotiations occur, causing a test failure. As it turns out, there is another place that session supplements could be called into, which is after media has finished getting negotiated. What this commit introduces is a means for session supplements to indicate when they wish to be called into when handling an incoming SIP response. By default, all session supplements will be run at the same point that they were prior to this commit. However, session supplements may indicate that they wish to be handled earlier than normal on redirects, or they may indicate they wish to be handled after media has been negotiated. In this changeset, two session supplements have been updated to indicate a preference for when they should be run: res_pjsip_diversion executes before handling redirection in order to get information from the Diversion header, and chan_pjsip now handles responses to INVITEs after media negotiation to fix the race condition mentioned previously. ASTERISK-24212 #close Reported by Matt Jordan Review: https://reviewboard.asterisk.org/r/3930 ........ Merged revisions 422536 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 422542 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422543 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Scott Griepentrog	2df2d785b7	The assertion that peer was not found on final event ... message was being triggered on configuration reload. This patch changes that case to just return instead. Review: https://reviewboard.asterisk.org/r/3953/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422358 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Michael L. Young	c5916fb39f	chan_iax2: Fix Dynamic IAX2 Registrations After Temporary DNS Failure ... The reporter on the issue found some issues when upgrading from version 10 to 11 on 55 hosts. Two situations that can occur with dynamic registrations. 1. With dnsmgr disabled, if the host is not resolvable we are not trying to resolve the host again when it is time to attempt to register again. This results in never registering to the host. 2. With dnsmgr enabled, when the host is temporarily not resolvable the address is set to 0.0.0.0:0 and then when the host is resolvable the port is not being restored and stays set to 0. This patch resolves these two issues by: * Storing the hostname so that it can be used for resolving with DNS. * Resolve the hostname on the next scheduled attempt to register. * Storing the port used to reach the host so that when the hostname is resolvable again, we can set the port again if the port is still unset after looking up the host. ASTERISK-23767 #close Reported by: David Herselman Tested by: David Herselman, Michael L. Young Patches: asterisk-23767-dns_reg_retry_and_set_port_11_v3.diff uploaded by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/3856/ ........ Merged revisions 422274 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 422275 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 422276 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422277 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Paul Belanger	ef28cc0d43	chan_sip.c: Add 'rtpbindaddr' setting ... Users now have the ability to bind the rtpengine instance to a specific IP address. For example, you want chan_sip (call control) on eth0 but rtp (media) on eth1. ASTERISK-24280 #close Reported by: Paul Belanger Tested by: Paul Belanger Review: https://reviewboard.asterisk.org/r/3952/ Patches: rtpengine.diff uploaded by Paul Belanger git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422241 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	bf85018107	CallerID: Fix parsing of malformed callerid ... This allows the callerid parsing function to handle malformed input strings and strings containing escaped and unescaped double quotes. This also adds a unittest to cover many of the cases where the parsing algorithm previously failed. Review: https://reviewboard.asterisk.org/r/3923/ Review: https://reviewboard.asterisk.org/r/3933/ ........ Merged revisions 422112 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 422113 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 422114 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 422154 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422158 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	cee660dadf	chan_sip: Use the server reflexive ICE candidate RTCP port as provided. ... This code originally worked around an issue within res_rtp_asterisk itself. The wrong socket was being used for the STUN check for RTCP, causing the port to be the same as RTP. This was subsequently fixed and the RTCP port provided for the ICE candidate is correct and does not need to be incremented. ASTERISK-23997 #close Reported by: Badalian Vyacheslav Patches: plus1.diff submitted by Badalian Vyacheslav (license 5249) ........ Merged revisions 421909 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 421910 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 421911 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421912 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	77ddc5b713	chan_sip: Don't use port derived from fromdomain if it isn't set ... If a user does not provide a port in the fromdomain setting, chan_sip will set the fromdomainport to STANDARD_SIP_PORT (5060). The fromdomainport value will then get used unilaterally in certain places. This causes issues with TLS, where the default port is expected to be 5061. This patch modifies chan_sip such that fromdomainport is only used if it is not the standard SIP port; otherwise, the port from the SIP pvt's recorded self IP address is used. Review: https://reviewboard.asterisk.org/r/3893/ ASTERISK-24178 #close Reported by: Elazar Broad patches: fromdomainport_fix.diff uploaded by Elazar Broad (License 5835) ........ Merged revisions 421717 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 421718 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 421719 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 421720 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421721 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	b7f98c3da4	chan_pjsip: Update media translation paths when new SDP negotiated. ... On a SIP reinvite that changes media strams, the PJSIP channel driver was flooding the log with "Asked to transmit frame type %s, while native formats is %s" warnings. * Fixes PJSIP not setting up translation paths when the formats change on a reinvite. AFS-63 was effectively reintroduced because of the media formats work. res_pjsip_sdp_rtp.c:set_caps() * Improved the unexpected frame format WARNING message to include more information. * Added protective locking while altering formats on a channel. Reworked set_format() to simplify and protect the formats under manipulation. * Restored some code that got lost in the media_formats work. (channel.c:set_format() and res_pjsip_sdp_rtp.c:set_caps()) AFS-137 #close Reported by: Mark Michelson Review: https://reviewboard.asterisk.org/r/3906/ ........ Merged revisions 421645 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421646 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	d0640ad7df	Move evaluation of set_var options in pjsip to the end of channel initialization. ... This allows for set_var to override certain defaults such as caller ID and codec values. This also fixes a test suite regression. The "set_var" test suite test attempted to use set_var to override caller ID, but a recent change caused that to no longer work. ........ Merged revisions 421565 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 421566 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421567 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	83a9b91da9	chan_pjsip: Fix attended transfer connected line name update. ... A calls B B answers B SIP attended transfers to C C answers, B and C can see each other's connected line information B completes the transfer A has number but no name connected line information about C while C has the full information about A I examined the incoming and outgoing party id information handling of chan_pjsip and found several issues: * Fixed ast_sip_session_create_outgoing() not setting up the configured endpoint id as the new channel's caller id. This is why party A got default connected line information. * Made update_initial_connected_line() use the channel's CALLERID(id) information. The core, app_dial, or predial routine may have filled in or changed the endpoint caller id information. * Fixed chan_pjsip_new() not setting the full party id information available on the caller id and ANI party id. This includes the configured callerid_tag string and other party id fields. * Fixed accessing channel party id information without the channel lock held. * Fixed using the effective connected line id without doing a deep copy outside of holding the channel lock. Shallow copy string pointers can become stale if the channel lock is not held. * Made queue_connected_line_update() also update the channel's CALLERID(id) information. Moving the channel to another bridge would need the information there for the new bridge peer. * Fixed off nominal memory leak in update_incoming_connected_line(). * Added pjsip.conf callerid_tag string to party id information from enabled trust_inbound endpoint in caller_id_incoming_request(). AFS-98 #close Reported by: Mark Michelson Review: https://reviewboard.asterisk.org/r/3913/ ........ Merged revisions 421400 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 421403 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421404 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Damien Wedhorn	c4c9d4ad6c	Skinny: Fixup compile warning for non dev-mode. ... ........ Merged revisions 421376 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421380 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	969982b878	chan_sip: Fix type mismatch when the format is changed. ... Symptom is most likely an invalid ao2 object bad magic number message or a less likely crash. ........ Merged revisions 420881 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420882 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago