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Author	SHA1	Message	Date
Michael L. Young	a317b5fc93	Adding a note to UPGRADE.txt about a change made to res_agi in order to ... indicate when streaming an audio file fails like it is done in other parts of the code to indicate an error. Note was requested by Paul Belanger: http://lists.digium.com/pipermail/asterisk-dev/2013-July/061420.html (related to issue ASTERISK-21903) ........ Merged revisions 396196 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@396197 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	3754c3a7dd	Update copyright year to 2013 in asterisk.c; some whitespace fixes ... (closes issue ASTERISK-22179) Reported by: Malcolm Davenport ........ Merged revisions 395032 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@395033 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	06f0da01f5	Add an upgrade note for libuuid dependency; remove note in CHANGES ... This patch notes that libuuid is now a dependency for res_rtp_asterisk; this was introduced in between 11.4.0 and 11.5.0 to resolve a dependency for pjproject, which res_rtp_asterisk uses for ICE/STUN/TURN support. It also removes a conflicting note from CHANGES. While support for playing prompts to the first participant was added for app_queue, it was disabled by default and an option added to enable it. That was properly noted in the UPGRADE.txt file. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@395020 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	5b3d9a9b20	Clean up documentation ... This patch cleans up documentation in func_channel for the following items: * rtpsource * secure_signaling * secure_media * various OOH323 parameters (closes issue ASTERISK-20969) Reported by: snuffy patches: func_chan-update.diff uploaded by snuffy (License 5024) ........ Merged revisions 394980 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@394981 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	07edcbb23a	Provide proper ring tone in indications.conf for Malaysia ... The ring tone provided in the sample indications.conf was incorrect. This patch modifies the sample ring tone to be what it should: ring = 425/400,0/200,425/400,0/2000 This brings it in line with the tone definition in DAHDI 2.7.0. (zonedata.c) (closes issue ASTERISK-21997) Reported by: Filip Jenicek patches: malaysia_ring.patch uploaded by phill (License 6277) ........ Merged revisions 394940 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@394941 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	03a1e40d8e	Tolerate presence of RFC2965 Cookie2 header by ignoring it ... This patch modifies parsing of cookies in Asterisk's http server by doing an explicit comparison of the "Cookie" header instead of looking at the first 6 characters to determine if the header is a cookie header. This avoids parsing "Cookie2" headers and overwriting the previously parsed "Cookie" header. Note that we probably should be appending the cookies in each "Cookie" header to the parsed results; however, while clients can send multiple cookie headers they never really do. While this patch doesn't improve Asterisk's behavior in that regard, it shouldn't make it any worse either. Note that the solution in this patch was pointed out on the issue by the issue reporter, Stuart Henderson. (closes issue ASTERISK-21789) Reported by: Stuart Henderson Tested by: mjordan, Stuart Henderson ........ Merged revisions 394899 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@394900 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	e3661c7fe4	Update PostgreSQL realtime scripts with schema for queue_log table ... This patch updates the realtime SQL scripts with an entry that will create the queue_log table. This brings the PostgreSQL scripts inline with the MySQL scripts, with respect to what tables they will create. (closes issue ASTERISK-21021) Reported by: Eugene patches: queue_log.sql uploaded by varnav (license 6360) ........ Merged revisions 394896 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@394897 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	35caf6f92b	Document connectedline parameter for chan_iax2 ... The connectedline parameter for a chan_iax2 peer was undocumented. This patch documents the options in the sample configuration file. (closes issue ASTERISK-21953) Reported by: Birger "WIMPy" Harzenetter ........ Merged revisions 394886 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@394890 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Michael L. Young	3b731ff2d0	Properly indicate failure to open an audio stream in res_agi ... If there is an error streaming an audio file, the current return status makes it difficult for an AGI script to determine that there was an error with the audio file. This patches changes the result to return -1 and the function returns RESULT_FAILURE instead of RESULT_SUCCESS. From looking at other parts of res_agi, this would appear to be the proper way to handle an error. (closes issue ASTERISK-21903) Reported by: Ariel Wainer Tested by: Ariel Wainer Patches: asterisk-21903-return-stream-res_1.8.diff by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2625/ ........ Merged revisions 394640 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@394641 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	7bb2991218	Provide error message for QUEUE_MEMBER when member is not in queue ... When QUEUE_MEMBER is used and the member specified is not in the queue, Asterisk provides an ERROR message that indicates that the option specified is not valid. This patch now properly displays an ERROR message that the member is not in the queue if an interface is specified. (closes issue ASTERISK-21980) Reported by: Avraam David git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@394345 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	4c8e75bcf0	Clarify documentation for function PASSTHRU ... It is not apparent to the average user that the PASSTHRU function should not be passed as ${PASSTHRU(string)} but just as PASSTHRU(string) to functions which take a variable name and not its contents. This patch clarifies the behavior in the documentation and provides an example. (closes issue ASTERISK-21717) Reported by: Richard Miller patches: func_strings.diff uploaded by Richard Miller (license 5685) ........ Merged revisions 394302 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@394303 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Moises Silva	bc78bfee41	Fix a longstanding issue with MFC-R2 configuration that prevented users ... from mixing different variants or general MFC-R2 settings within the same E1 line. Most users do not have a problem with this since MFC-R2 lines are usually fractional E1s, or the whole E1 has the same country variant and R2 settings. In Venezuela however is common to have inbound MFC-R2 and outbound DTMF-R2 within the same E1. This fix now properly parses the chan_dahdi.conf file to generate a new openr2 context every time a new channel => section is found and the configuration was changed. (closes issue ASTERISK-21117) Reported by: Rafael Angulo Related Elastix issue: http://bugs.elastix.org/view.php?id=1612 ........ Merged revisions 394106 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@394173 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Russell Bryant	9959caa047	astobj2-ify the SLA code ... The SLA code within app_meetme was written before asotbj2 had been merged into Asterisk. Worse, support for reloads did not exist at first and was added later as a bolt-on feature. I knew at the time that reloading was not safe at all while SLA was in use, so the reload would be queued up to execute when the system was idle. Unfortunately, this approach was still prone to errors beyond the fact that this was the only place in Asterisk where configuration was not reloaded instantly when requested. This patch converts various SLA objects to be reference counted objects using astobj2. This allows reloads to be processed while the system is in use. The code ensures that the objects will not disappear while one of the other threads is using them. However, they will be immediately removed from the global trunk and station containers so no new calls will use them if removed from configuration. Review: https://reviewboard.asterisk.org/r/2581/ ........ Merged revisions 393928 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@393929 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	802c27b394	MixMonitor: Fix refleak in manager_stop_mixmonitor() if could not stop monitoring. ... ........ Merged revisions 393490 from http://svn.asterisk.org/svn/asterisk/trunk git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@393630 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	c03b11466d	chan_dahdi: Fix segfault reloading chan_dahdi when round robin is used. ... * Clear round_robin[] in dahdi_restart(). (closes issue ASTERISK-21847) Reported by: Ivo Andonov Patches: jira_asterisk_21847_v1.8.patch (license #5621) patch uploaded by rmudgett ........ Merged revisions 393627 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@393628 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Igor Goncharovskiy	9ce8896d15	Fix issue with inability to cancell call transfer made by on-sceen menus. ...	12 years ago
Matthew Jordan	2ffb648a20	Fix memory/ref counting leaks in a variety of locations ... This patch fixes the following memory leaks: * http.c: The structure containing the addresses to bind to was not being deallocated when no longer used * named_acl.c: The global configuration information was not disposed of * config_options.c: An invalid read was occurring for certain option types. * res_calendar.c: The loaded calendars on module unload were not being properly disposed of. * chan_motif.c: The format capabilities needed to be disposed of on module unload. In addition, this now specifies the default options for the maxpayloads and maxicecandidates in such a way that it doesn't cause the invalid read in config_options.c to occur. (issue ASTERISK-21906) Reported by: John Hardin patches: http.patch uploaded by jhardin (license 6512) named_acl.patch uploaded by jhardin (license 6512) config_options.patch uploaded by jhardin (license 6512) res_calendar.patch uploaded by jhardin (license 6512) chan_motif.patch uploaded by jhardin (license 6512) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@392810 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	be6c8d267c	app_mixmonitor: Fix crashes caused by unloading app_mixmonitor ... Unloading app_mixmonitor while active mixmonitors were running would cause a segfault. This patch fixes that by making it impossible to unload app_mixmonitor while mixmonitors are active. Review: https://reviewboard.asterisk.org/r/2624/ ........ Merged revisions 391778 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@391794 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	365e81053b	app_confbridge: Fix memory leak on reload. ... The config framework options should not be registered multiple times. Instead the configuration just needs to be reprocessed by the config framework. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@391700 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	7e0ebaa2e0	Fix segfault for certain invalid WebSocket input. ... The WebSocket code would allocate, on the stack, a string large enough to hold a key provided by the client, and the WEBSOCKET_GUID. If the key is NULL, this causes a segfault. If the key is too large, it could overflow the stack. This patch checks the key for NULL and checks the length of the key to avoid stack smashing nastiness. (closes issue ASTERISK-21825) Reported by: Alfred Farrugia Tested by: Alfred Farrugia, David M. Lee Patches: issueA21825_check_if_key_is_sent.patch uploaded by Walter Doekes (license 5674) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@391560 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	56a4d03ee2	Fix memory leak while loading priority modules and adding formats ... This patch fixes two memory leaks: * When we load a module with the LOAD_PRIORITY flag, we remove its entry from the load order list. Unfortunately, we don't free the memory associated with entry in the list. This patch corrects that and properly frees the memory for the module in the list. * When adding a custom format (such as SILK or CELT), the routine for adding the format was leaking a reference. RAII_VAR cleans this up properly. ........ Merged revisions 391489 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@391507 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Igor Goncharovskiy	13b2c25687	Fix issue with no sound in both way in case of previous call to chan_unistim phone was canceled. ...	12 years ago
Alec L Davis	a90ad16e55	IAX2: Transfer Reject: Lock bridgecallno before touching it, refactor ... 1). When touching the bridgecallno, we need to lock it. 2). Remove magic number '0' and replace with TRANSFER_NONE. 3). Exit early if no bridgecallno. 4). Reduce indentation. Reported by: alecdavis Tested by: alecdavis alecdavis (license 585) Review https://reviewboard.asterisk.org/r/2613/ ........ Merged revisions 391333 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@391334 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	4a99d74105	Add announce-to-first-user option for app_queue ... In r386792, the ability to play prompts to the first caller in a call queue was added. While this is arguably a bug fix for those who expect the first caller to continue receiving prompts while the agent is dialed, it has the side effect of preventing the first caller from hearing the agent immediately upon bridging. This may not be a problem for those who really want this option, but for those who didn't care whether or not the first caller in queue heard their position, it was an issue. This patch disables the ability for the first caller in the queue to hear prompts and adds a new option, announce-to-first-user, to queues.conf. Those who the behavior can enable it by setting this value to True. Note that if we ever implement the ability to have the prompts be stopped upon bridging, this option can be removed. (closes issue ASTERISK-21782) Reported by: Remi Quezada ........ Merged revisions 391215 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@391241 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Alec L Davis	f09521a0d5	chan_iax2: nativebridge refactor, missed unlock bridgecallno ... ........ Merged revisions 391143 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@391148 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Alec L Davis	30cfce07f7	fix bad edit after conflict resolution ... ........ Merged revisions 391107 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@391111 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Alec L Davis	20b9dac9fc	IAX2: refactor nativebridge transfer ... remove triple checking of iaxs[fr->callno]->transferring reduce indentation. Reported by: alecdavis Tested by: alecdavis alecdavis (license 585) Review https://reviewboard.asterisk.org/r/2602/ ........ Merged revisions 391065 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@391084 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Alec L Davis	9fca44e6d4	IAX2: fix race condition with nativebridge transfers. ... 1). When touching the bridgecallno, we need to lock it. 2). stop_stuff() which calls iax2_destroy_helper() Assumes the lock on the pvt is already held, when iax2_destroy_helper() is called. Thus we need to lock the bridgecallno pvt before we call stop_stuff(iaxs[fr->callno]->bridgecallno); 3). When evaluating the state of 'callno->transferring' of the current leg, we can't change it to READY unless the bridgecallno is locked. Why, if we are interrupted by the other call leg before 'transferring = TRANSFER_RELEASED', the interrupt will find that it is READY and that the bridgecallno is also READY so Releases the legs. (closes issue ASTERISK-21409) Reported by: alecdavis Tested by: alecdavis alecdavis (license 585) Review https://reviewboard.asterisk.org/r/2594/ ........ Merged revisions 391062 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@391063 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Alexandr Anikin	55b2eebe39	remove unnecessary declarations ... (issue ASTERISK-21800) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@390229 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Alexandr Anikin	5a9206c43b	reject call attempts when gatekeeper is configured but not registered ... (closes issue ASTERISK-21800) Reported by: Dmitry Melekhov Patches: ASTERISK-21800-1.patch Tested by: Dmitry Melekhov ........ Merged revisions 390181 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 390223 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@390228 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	829edeffe9	Fix segfault when dealing with chan_agent channels. ... Check the returned bridged pointer for NULL to avoid a crash. It looks like chan_agent is returning a NULL pointer when it probably should be returning a pointer to the channel the Agent channel is pretending to be. (closes issue ASTERISK-21793) Reported by: Rodrigo P. Telles Patches: jira_asterisk_21793_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: Rodrigo P. Telles ........ Merged revisions 390044 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@390047 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	249d7c2790	Fix a memory copying bug in slinfactory which was causing mixmonitor issues. ... Reported by: Michael Walton Tested by: Jonathan Rose Patches: slinfactory.c.ASTERISK-21799.patch uploaded by Michael Walton (license 6502) (closes issue ASTERISK-21799) ........ Merged revisions 389895 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@389896 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	1aed26bc0a	Print all logger messages on shutdown ... When Asterisk shuts down and shuts down the loggin gsubsystem, any messages currently in flight will not get logged. This patch prevents the loop writing messages from breaking out prematurely, such that all of the messages are logged. (closes issue ASTERISK-21716) Reported by: Corey Farrell patches: logger-process-all-messages.patch uploaded by Corey Farrell (license 5909) ........ Merged revisions 389676 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@389677 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Igor Goncharovskiy	97ca159774	Fix several problems caused by multiple line usage with i2004 phones. ...	12 years ago
Jason Parker	54bd3a2a9f	Add doxygen.log to svn:ignore property. ... ........ Merged revisions 389244 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@389245 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	faeea4c0f2	Fix for segfault in __ast_rwlock_destroy with DEBUG_THREADS ... If DEBUG_THREADS is enabled __ast_rwlock_destroy causes a segfault while trying to access a possible NULL t->track object. A NULL check has been added before trying to access the memory. (closes issue ASTERISK-21724) Reported by: Corey Farrell Fixed by: Corey Farrell Patches: ast_rwlock_destroy-segv.patch uploaded by Corey Farrell (license 5909) ........ Merged revisions 388838 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@388839 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jason Parker	67729f6249	Fix VM snapshot handling for combined INBOX. ... The snapshot API contains an option that allow for combining of new and old messages within a single snapshot. New messages, however, include options beyond just 'INBOX' - it also includes the Urgent folder. A previous patch that combined INBOX and Urgent accidentally impacted snapshots that attempted to gain messages from just the Old folder. This patch fixes the snapshot gathering such that the API returns the appropriate messages for the folder selected, with and without the combine option. This should make it more clear about what's happening. Review: https://reviewboard.asterisk.org/r/2539/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@388816 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	457d5c39dc	Use srtp_shutdown when available ... This allows the SRTP library to be shut down properly when the functionality is offered by libsrtp. Review: https://reviewboard.asterisk.org/r/2538/ (closes issue ASTERISK-21719) ........ Merged revisions 388768 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@388769 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	a58b7639dd	Make ao2 global objects not always use the debug version of the ao2_ref() calls. ... The debug versions of ao2_ref() should only be used if REF_DEBUG is enabled so nothing is written to /tmp/refs unexpectedly. (closes issue ASTERISK-21785) Reported by: abelbeck Patches: jira_asterisk_21785_v11.patch (license #5621) patch uploaded by rmudgett Tested by: abelbeck git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@388700 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Michael L. Young	672040ab2e	Fix Missing CALL-ID When Logging Through Syslog ... The CALL-ID (ie [C-00000074]) is missing when logging to syslog. This was just an oversight when this feature was added. * Add CALL-IDs when using syslog (closes issue ASTERISK-21430) Reported by: Nikola Ciprich Tested by: Nikola Ciprich, Michael L. Young Patches: asterisk-21430-syslog-callid_trunk.diff by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2526/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@388605 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Michael L. Young	eec46f56f4	Fix Crash Caused By One-way Audio With auto_* NAT Settings Fix ... The prior code committed, r385473, failed to take into consideration that not all outgoing calls will be to a peer. My fault. This patch does the following: * Check if there is a related peer involved. If there is, check and set NAT settings according to the peer's settings. * Fix a problem with realtime peers. If the global setting has auto_force_rport set and we issued a "sip reload" while a peer is still registered, the peer's flags for NAT are reset to off. When this happens, we were always setting the contact address of the peer to that of the full contact info that we had. (closes issue ASTERISK-21374) Reported by: jmls Tested by: Michael L. Young Patches: asterisk-21374-fix-crash-and-rt-peers.diff by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2524/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@388601 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	6b5b35b756	Revert r388529 for now ... Adding the cleanup function needs some deeper thought since it apparently doesn't exist for all variants of libsrtp. ........ Merged revisions 388596 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@388597 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	5287afb80c	pbx: Fix lack of cleanup on macrolock and context_table ... (closes issue ASTERISK-21723) Reported by: Corey Farrell Patches: core-pbx-cleanup.patch uploaded by Correy Farrell (license 5909) ........ Merged revisions 388532 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@388578 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	c3a0ce8338	Close libsrtp properly ... Ensure that libsrtp is shutdown properly when res_srtp is unloaded. (closes issue ASTERISK-21719) Reported by: Corey Farrell Patches: res_srtp-library-shutdown.patch uploaded by Corey Farrell ........ Merged revisions 388529 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@388530 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	e712c4320f	Fix SendText AMI action to never return non-zero. ... AMI actions must never return non-zero unless they intend to close the AMI connection. (Which is almost never.) (closes issue ASTERISK-21779) Reported by: Paul Goldbaum ........ Merged revisions 388477 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@388478 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	f296671ec5	Allow mISDN to send PROGRESS messsage. ... * Made isdn_msg_parser.c build a progress message with the mandatory progress indicator IE. (The mISDNuser NT state machine rejected sending the incomplete message.) Note: The associated mISDN and mISDNuser patches respectively are viewable here: http://svnview.digium.com/svn/thirdparty?view=rev&rev=200 http://svnview.digium.com/svn/thirdparty?view=rev&rev=201 (closes issue AST-1153) Reported by: Guenther Kelleter Patches: progress-chan_misdn.diff (license #6372) patch uploaded by Guenther Kelleter progress-misdn.diff (license #6372) mISDN patch uploaded by Guenther Kelleter progress-misdnuser.diff (license #6372) mISDNuser patch uploaded by Guenther Kelleter ........ Merged revisions 388425 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@388426 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	2e52e2aa20	Add version.c to list of ignored files in the utils directory. ... ........ Merged revisions 388423 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@388424 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	3c28267b7e	Fix memory leak in pbx_dundi ... pbx_dundi added an io context without removing it. This caused a memory leak when the module was unloaded. (closes ASTERISK-21718) Reported by Corey Farrell Patches: pbx_dundi-ast_io_remove.patch uploaded by Corey Farrell (License #5909) ........ Merged revisions 388376 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@388378 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Sean Bright	771ce9e1e7	Fix copy/paste error in one-touch-recording implementation. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@388253 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Michael L. Young	08c2a533f2	Fix The Payload Being Set On CN Packets And Do Not Set Marker Bit ... When we send out a CN packet (for instance, in the case of using rtpkeepalives), we are not setting the payload code properly. Also, we are setting the marker bit when we shouldn't be according to RFC 3389, section 4. AST_RTP_CN is not defined by AST_FORMAT codes. Therefore, we should be using ast_rtp_codecs_payload_code() rather than ast_rtp_codecs_payload_lookup(). 11 and trunk already use the appropriate function. * In 1.8, use ast_rtp_codecs_payload_code() * Remove the setting of the marker bit * Fix the debug message by incrementing the seqno after the debug message is set in order to display the correct seqno that was sent out (closes issue ASTERISK-21246) Reported by: Peter Katzmann Tested by: Peter Katzmann, Michael L. Young Patches: asterisk-21246-rtp-cng-payload-error_1.8_v2.diff uploaded by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2500/ ........ Merged revisions 388111 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@388112 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago