asterisk

Commit Graph

Author	SHA1	Message	Date
Walter Doekes	7e6f444e38	res_calendar_ews: Relax neon version check to work with 0.30 too. Allow res_calendar_ews to work not only with libneon-0.29 but also with 0.30. ASTERISK-24325 #close Reported by: Tzafrir Cohen Review: https://reviewboard.asterisk.org/r/4068/ ........ Merged revisions 425286 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@425287 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	e6bd251926	CallerID: Fix parsing regression This fixes a regression in callerid parsing introduced when another bug was fixed. This bug occurred when the name was composed entirely of DTMF keys and quoted without a number section (<>). ASTERISK-24406 #close Reported by: Etienne Lessard Tested by: Etienne Lessard Patches: callerid_fix.diff uploaded by Kinsey Moore Review: https://reviewboard.asterisk.org/r/4067/ ........ Merged revisions 425152 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@425153 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	3b4fcfe6de	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@425069 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kevin Harwell	6b2ee3002f	res_rtp_asterisk: Crash if no candidates received for component When starting ice if there is not at least one remote ice candidate with an RTP component asterisk will crash. This is due to an assertion in pjnath as it expects at least one candidate with an RTP component. Added a check to make sure at least one candidate contains an RTP component and at least one candidate has an RTCP component. ASTERISK-24383 #close Review: https://reviewboard.asterisk.org/r/4039/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@425029 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	f49780f7e5	safe_asterisk: Don't automatically exceed MAXFILES value of 2^20. On systems with lots of RAM (e.g. 24GB) /proc/sys/fs/file-max divided by two can exceed the per-process file limit of 2^20. This patch ensures the value is capped. (Patch cleaned up by me.) ASTERISK-24011 #close Reported by: Michael Myles Patches: safe_asterisk-ulimit.diff uploaded by Michael Myles (License #6626) ........ Merged revisions 424875 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@424878 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	bc34e681e6	Blocked revisions 424876 ........ Ouch! Accidental commit of wrong file in 424875. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@424877 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	89247cec8e	res_rtp_asterisk: Allow only UDP ICE candidates. The underlying library, pjnath, that res_rtp_asterisk uses for ICE support does not have support for ICE-TCP. As candidates are passed through directly to it this can cause error messages to occur when it receives something unexpected (such as a TCP candidate). This change merely ignores all non-UDP candidates so they never reach pjnath. ASTERISK-24326 #close Reported by: Joshua Colp git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@424852 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	5454c620e0	astobj2: Correct REF_DEBUG false leak report When ao2_callback is run with OBJ_MULTIPLE and not OBJ_NODATA it allocates a temporary container in a way that does not record REF_DEBUG log entries. This changes that container to correctly record unref's when the container is freed. ASTERISK-24390 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4047/ ........ Merged revisions 424786 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@424787 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	478e64f1e4	message: Don't close an AMI connection on SendMessage action error If SendMessage encounters an error (such as incorrect input provided to the action), it will currently return -1. Actions should only return -1 if the connection to the AMI client should be closed. In this case, SendMessage causing the client to disconnect is inappropriate. This patch causes the action to return 0, which simply causes the action to fail. Review: https://reviewboard.asterisk.org/r/4024 ASTERISK-24354 #close Reported by: Peter Katzmann patches: sendMessage.patch uploaded by Peter Katzmann (License 5968) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@424690 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	7d06282b9e	Release AMI connections on shutdown. ASTERISK-24378 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4037/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@424578 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	8b405b7b34	chan_sip: Clean leak on error path of process_sdp Resolve leak in process_sdp that occurs in 2 error path's where crypto lines are expected but not provided. ASTERISK-24385 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4045/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@424569 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	4d1c38ef37	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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@424550 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	2e62e7b1ef	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@424182 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	a819ae6b8a	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@424177 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	fd4191e016	res_rtp_asterisk: Ensure that the base and mapped address for candidates is present in SDP. This change fixes an issue where ICE candidates put into the SDP did not contain the 'raddr' and 'rport' information for server reflexive and relay candidates. #SIPit31 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@424151 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	59224edc78	autosupport: Fix bashism. '==' is bashism (bashspecific, fails when dash is /bin/sh). Anyway, a 'case' works better there. Originally committed in r375059 and r375060 on 2012-10-16 21:13:08. ASTERISK-20567 #close Reported by: Tzafrir Cohen git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@424117 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	660080c262	res_fax: Fix out of bounds error in update_modem_bits(). ASTERISK-24357 #close Reported by: Jeremy Laine Patches: res_fax_bounds.patch (license #6561) patch uploaded by Jeremy Laine Modified patch to not use magic numbers. ........ Merged revisions 423979 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@423983 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	d05d8eb8a3	docs: Escape unescaped minus sign in asterisk.8 manpage. ASTERISK-23768 #close Reported by: Jeremy Lainé Patches: escape_manpage_hyphen.patch uploaded by Jeremy Lainé (License #6561) ........ Merged revisions 423915 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@423916 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	9bb255bc7c	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@423801 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	559b43bd6d	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@423721 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	a1fab60fcc	cli.c: Fix tab completion "module load" when MALLOC_DEBUG is enabled. r421600 conflicted with r155763. ASTERISK-24348 #close ........ Merged revisions 423657 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@423658 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	7d0f1a26e6	astobj2.c/refcounter.py: Fix to deal with invalid object refs. * Make astob2 REF_DEBUG output an invalid object line when an invalid ao2 object ref/unref is attempted. This is similar to the constructor/destructor lines. * Fixed refcounter.py to handle skewed objects that have constructor/destructor states. * Made refcounter.py highlight the invalid ao2 object refs by putting them in their own section of the processed output file. * Made refcounter.py highlight unreffing an object by more than one that results in a negative ref count and the object being destroyed. The abnormally destroyed object is reported in the invalid and finalized object sections of the output. Review: https://reviewboard.asterisk.org/r/3971/ ........ Merged revisions 423349 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@423400 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	8667ce10d0	res_fax_spandsp: Properly handle cleanup before starting FAXes. If faxing fails at a very early stage, then it is possible for us to pass a NULL t30 state pointer to spandsp, which spandsp is none too pleased with. This patch ensures that we pass the correct pointer to spandsp in the situation where we have not yet set our local t30 state pointer. ASTERISK-24301 #close Reported by Matt Jordan Patches: ASTERISK-24301-fax.diff Uploaded by Mark Michelson (License #5049) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@423360 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	c4cf004f24	config: bug: Fix SEGV in ast_category_insert when matching category isn't found If you call ast_category_insert with a match category that doesn't exist, the list traverse runs out of 'next' categories and you get a SEGV. This patch adds check for the end-of-list condition and changes the signature to return an int for success/failure indication instead of a void. The only consumer of this function is manager and it was also changed to use the return value. Tested by: George Joseph Review: https://reviewboard.asterisk.org/r/3993/ ........ Merged revisions 423276 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@423277 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	0ea0c38600	res_rtp_asterisk: Ensure that the thread terminating pj stuff is registered. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@423253 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	38885ac8bd	res_rtp_asterisk: Fix 100% CPU usage due to timer heap thread spinning. Side note: I need a vacation. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@423210 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	222d6ac4ab	res_rtp_asterisk: Fix building when pjproject is not used. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@423207 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	ff4371e45d	res_rtp_asterisk: Fix a myriad of TURN client issues. 1. The number of file descriptors an ioqueue instance can handle is fixed, so we now spawn the required number to handle the load. 2. Our transport identifiers were exceeding the range supported by pjnath. 3. The TURN client did not set up client binding causing needless bandwidth usage. 4. The code no longer updates address information on each packet. 5. STUN traffic was getting looped back to Asterisk instead of going through the TURN server. 6. Synchronization now ensures things are completely setup or destroyed. 7. Logging now reflects the target the TURN server is sending to/receiving from on our behalf. ASTERISK-23577 #close Reported by: Jay Jideliov ASTERISK-23634 #close Reported by: Roman Skvirsky Review: https://reviewboard.asterisk.org/r/3982/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@423150 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	e994f7fd37	chan_sip: Clarify that sipdebug=yes cannot be undone by the CLI. Document it in sip.conf. ASTERISK-24249 #close Reported by: Avinash Mohod Review: https://reviewboard.asterisk.org/r/3926/ ........ Merged revisions 423066 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@423067 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	6badd734d3	Bridging: Fix bouncing native bridge This fixes a situation in Asterisk 1.8 and 11 where ast_channel_bridge could cause a bouncing native bridge. In the case of the dial_LS_options test, this was a remote RTP bridge which caused the audio path to continually cycle between Asterisk and the remote endpoints generating a large number of SIP messages and delaying the test long enough to cause it to fail (checking timing was part of the test). The root cause was that the code to decide whether to use native bridging was expecting a time-remaining value of 0 to be the default instead of the actual default value of -1. A value of 0 or negative numbers could also be generated by preceding code in some circumstances. Both issues are addressed in this patch. ASTERISK-24211 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3987/ ........ Merged revisions 423006 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@423010 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	2fbdeafb19	config: bug: fix truncation of included config files on permissions error ast_config_text_file_save() currently truncates include files as they are processed. If a subsequent include file or the main config file has a permissions error that prevents writing, earlier include files are left truncated resulting in a frantic search for backups. This patch causes ast_config_text_file_save to check for write access on all files before it truncates any of them. Will be applied 1.8 > trunk. Tested by: George Joseph Review: https://reviewboard.asterisk.org/r/3986/ ........ Merged revisions 422900 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@422903 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Rusty Newton	61b5e7c669	Sounds/BuildSystem: Modifications to include new releases and Japanese language. Modifying Makefile and sounds.xml to include new core 1.4.26 and extra 1.4.15 sound prompt releases, plus the new Japanese core sound prompts contributed by QLOOG. ASTERISK-23324 Reported by: Kevin McCoy Tested by: Rusty Newton ........ Merged revisions 422789 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@422790 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	388a13abb8	Manager: Require read permission for SYSTEM in order to send FullyBooted Review: https://reviewboard.asterisk.org/r/3969/ ........ Merged revisions 422584 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@422625 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	cc3fb5bcb6	manager: Make WaitEvent action respect eventfilters A WaitEvent issued via an http session isn't respecting eventfilters defined for the user. I just added a match_filter to the predicate that controls astman_append. Tested by: George Joseph Review: https://reviewboard.asterisk.org/r/3958/ ........ Merged revisions 422439 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@422440 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	0a4ae50c62	doc: Add a manpage for the smsq utility This patch adds a manpage for the smsq utility. Note that this is one of the patches the Debian distro applies for the Asterisk project, as per ASTERISK-24191. Review: https://reviewboard.asterisk.org/r/3895/ ASTERISK-24171 #close Reported by: Jeremy Laine patches: smsq.8 uploaded by Jeremy Laine (License 6561) ........ Merged revisions 422376 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@422377 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	e0b38b6d07	doc: Add a manpage for the aelparse utility This patch adds a manpage for the aelparse utility. Note that this is one of the patches the Debian distro applies for the Asterisk project, as per ASTERISK-24191. Review: https://reviewboard.asterisk.org/r/3896/ ASTERISK-24171 #close Reported by: Jeremy Laine patches: aelparse.8 uploaded by Jeremy Laine (License 6561) ........ Merged revisions 422371 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@422372 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	6b51f717a7	LICENSE: Clarify language in Asterisk's LICENSE to allow for linking to UniMRCP The UniMRCP project distributes Asterisk modules that integrate Asterisk with UniMRCP, and other Asterisk users use the UniMRCP library as well. Unfortunately, the UniMRCP license is Apache 2.0, which per the Free Software Foundation, is not a compatible license with the GPLv2. "Please note that this license is not compatible with GPL version 2, because it has some requirements that are not in that GPL version. These include certain patent termination and indemnification provisions. The patent termination provision is a good thing, which is why we recommend the Apache 2.0 license for substantial programs over other lax permissive licenses." On the other hand, UniMRCP is a great project and we'd like to let people use it with Asterisk. This patch updates the LICENSE text to allow users to link Asterisk with UniMRCP and distribute the resulting binaries. ........ Merged revisions 422293 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@422294 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Michael L. Young	fef6e06991	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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@422274 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	5cdd1413f1	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/ ........ Merged revisions 422112 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@422113 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	b3572f3828	res_musiconhold: Fix MOH restarting where it left off from the last hold. Restore code removed by https://reviewboard.asterisk.org/r/3536/ that introduced a regression that prevents MOH from restarting were it left off the last time. ASTERISK-24019 #close Reported by: Jason Richards Patches: jira_asterisk_24019_v1.8.patch (license #5621) patch uploaded by rmudgett Review: https://reviewboard.asterisk.org/r/3928/ ........ Merged revisions 421976 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@421977 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	905f1f61e1	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) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@421909 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	6f9257db92	res_musiconhold.c: Remove obsolete REF_DEBUG code. Remove unneeded code that writes to the wrong file location in an obsolete format. ........ Merged revisions 421799 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@421800 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	3045568e47	res_musiconhold: Fix reference leaks caused when reloading with REF_DEBUG set Due to a faulty function for debugging reference decrementing, it was possible to reduce the refcount on the wrong object if two moh classes of the same name were in the moh class container. (closes issue ASTERISK-22252) Reported by: Walter Doekes Patches: 18_moh_debug_ref_patch.diff Uploaded by Jonathan Rose (license 6182) ........ Merged revisions 398937 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@421777 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	6c4c543d48	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@421718 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	05c31108a5	cli.c: Fix tab completion of "module load" when MALLOC_DEBUG is enabled. filename_completion_function() returns memory that was not allocated by the MALLOC_DEBUG allocation tracker so the memory must be freed by ast_std_free(). ........ Merged revisions 421600 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@421602 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	16a3e1cc3f	AMI Docs: Fix Status channel parameter optionality ........ Merged revisions 421442 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@421443 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	9498a686f3	func_config: Change 'Not Found' message from ERROR to DEBUG When you call the CONFIG dialplan function with the name of a variable that doesn't exist in the target context you get an ERROR. This does nothing but clutter up the logs with messages that may be perfectly acceptable. Just because a variable wasn't in the context doesn't mean it's an error. Maybei t's optional or just needs to be defaulted or ignored. This patch changes the log level from ERROR to DEBUG. If a dialplan developer wants to debug their dialplan they still canby setting the console debug level as needed. Tested by: George Joseph Review: https://reviewboard.asterisk.org/r/3919/ ........ Merged revisions 421327 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@421328 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	a640d70ae8	apps/app_dial: Fix Dial 'z' option The 'z' option is supposed to disable the dial timeout in the case of a call forward. Unfortunately, the wrong timeout timer was passed to the do_forward function, resulting in the option not working. ASTERISK-24225 #close Reported by: dimitripietro Tested by: dimitripietro patches: jira_asterisk_24225_v1.8.patch uploaded by rmudgett (License 5621) jira_asterisk_24225_v11.patch uploaded by rmudgett (License 5621) ........ Merged revisions 421232 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@421233 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	1969d911fe	configure: Undefine FORTIFY_SOURCE prior to defining it for patched gcc Some distributions of Linux patch gcc to define FORTIFY_SOURCE when gcc is executed with optimization. This "help" unfortunately results in re-definition warnings when FORTIFY_SOURCE is later defined in Asterisk's build system. This patch undefines FORTIFY_SOURCE prior to defining it to prevent this warning. Review: https://reviewboard.asterisk.org/r/3912/ ASTERISK-24032 #close Reported by: Kilburn Tested by: Kilburn, wdoekes patches: 1.8.diff uploaded by cloos (License 5956) 10.diff uploaded by cloos (License 5956) 11.diff uploaded by cloos (License 5956) 12.diff uploaded by cloos (License 5956) 13.diff uploaded by cloos (License 5956) ........ Merged revisions 421227 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@421228 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	252ead3b13	app_voicemail/app: Remove test events that were duplicated by r421059 Moving the test event raised when a file is played back (which occurred in r421059) broke the ever loving snot out of the voicemail tests. This caused duplicate test events to get raised, as app_voicemail and main/app were raising events prior to call ast_streamfile. The voicemail tests did not enjoy getting multiple events. Since raising the playback event in ast_streamfile is far more useful to the vast majority of tests, this patch keeps the call there and simply removes the extraneous calls that duplicated the event. ........ Merged revisions 421125 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@421164 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago

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