asterisk

Commit Graph

Author	SHA1	Message	Date
Jonathan Rose	beae2df26e	Merged revisions 336307 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r336307 \| jrose \| 2011-09-16 16:09:20 -0500 (Fri, 16 Sep 2011) \| 20 lines Merged revisions 336294 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336294 \| jrose \| 2011-09-16 14:53:40 -0500 (Fri, 16 Sep 2011) \| 13 lines Fix bad RTP media bridges in directmedia calls on peers separated by multiple Asterisk nodes. In a situation involving devices on separate Asterisk trunks, the remote RTP bridge would break when starting a call with directmedia. This patch queues a new type of control frame so that our RTP bridge loop can properly detect when these situations occur and check to see if peers need to be updated in order to send their media to the proper location. (Closes issue ASTERISK-18340) Reported by: Thomas Arimont (Closes issue ASTERISK-17725) Reported by: kwk Tested by: twilson, jrose ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@336311 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	b5a641d1fe	Merged revisions 336167 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r336167 \| irroot \| 2011-09-16 12:12:03 +0200 (Fri, 16 Sep 2011) \| 22 lines Merged revisions 336166 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336166 \| irroot \| 2011-09-16 12:09:17 +0200 (Fri, 16 Sep 2011) \| 16 lines The round robin routing routine in chan_misdn.c is broken. it rotates between ports but never checks the channels in the ports. i have extensivly tested it and verified it works on 1 upto 4 ports. before the patch only 1 out of each port was used now all are used as expected. (closes issue ASTERISK-18413) Reported by: irroot Tested by: irroot Reviewed by: irroot Review: https://reviewboard.asterisk.org/r/1410/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@336168 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	40b76b6893	Merged revisions 335991 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r335991 \| irroot \| 2011-09-15 10:29:12 +0200 (Thu, 15 Sep 2011) \| 17 lines Merged revisions 335978 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335978 \| irroot \| 2011-09-15 10:15:22 +0200 (Thu, 15 Sep 2011) \| 11 lines lock the channel before calling ast_bridged_channel() to prevent a seg fault. AMI agents list called on shutdown causes a segfault, introducing proper locking will prevent this. (closes issue ASTERISK-18092) Reported by: agustina Patches: chan_agent.patch (License #5041) patch uploaded by irroot ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335993 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	a27555687b	Merged revisions 335852 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r335852 \| rmudgett \| 2011-09-14 11:00:37 -0500 (Wed, 14 Sep 2011) \| 18 lines Merged revisions 335851 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335851 \| rmudgett \| 2011-09-14 10:53:25 -0500 (Wed, 14 Sep 2011) \| 11 lines Fixed cut-n-paste regression using the wrong variable. Fixes the missing DAHDI channels when using the newer chan_dahdi.conf sections for channel configuration. (closes issue ASTERISK-18496) Reported by: Sean Darcy Patches: jira_asterisk_18496_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: Sean Darcy, rmudgett ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335853 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Olle Johansson	5c6d438231	Documentation updates git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335385 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Olle Johansson	55b060fb35	Small documentation updates git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335349 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Olle Johansson	404151ad65	New sip.conf option for setting default tonezone for channel or individual devices Review: https://reviewboard.asterisk.org/r/1429/ (closes issue ASTERISK-18497) Thanks to russellb for peer review. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335325 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Olle Johansson	e4a11bcb6e	Merged revisions 335323 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r335323 \| oej \| 2011-09-12 15:47:13 +0200 (Mån, 12 Sep 2011) \| 19 lines Merged revisions 335319 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335319 \| oej \| 2011-09-12 15:25:30 +0200 (Mån, 12 Sep 2011) \| 12 lines Lock the peer->mvipvt to avoid crashes with SIP history enabled After the launch of 1.6 event-based MWI we have two threads handling the peer->mwipvt, which cause issues with SIP history additions in combination with the max limit for number of history entries. Review: https://reviewboard.asterisk.org/r/1373/ (closes issue ASTERISK-18288) Thanks to irrot for peer review. Work with this bug funded by IPvision AS ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335324 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	c5c1fed9b6	Merged revisions 335321 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r335321 \| kmoore \| 2011-09-12 08:27:04 -0500 (Mon, 12 Sep 2011) \| 16 lines Merged revisions 335320 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335320 \| kmoore \| 2011-09-12 08:25:42 -0500 (Mon, 12 Sep 2011) \| 9 lines Prevent IAX2 from getting IPv6 addresses via DNS IAX2 does not support IPv6 and getting such addresses from DNS can cause error messages on the remote end involving bad IPv4 address casts in the presence of IPv6/IPv4 tunnels. This patch ensures that IAX2 will not encounter IPv6 addresses via DNS queries. (closes issue ASTERISK-18090) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335322 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Stefan Schmidt	986f2d8836	Merged revisions 335260 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r335260 \| schmidts \| 2011-09-12 11:11:45 +0000 (Mon, 12 Sep 2011) \| 12 lines Merged revisions 335259 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335259 \| schmidts \| 2011-09-12 11:09:19 +0000 (Mon, 12 Sep 2011) \| 6 lines build_peer doesnt unlink a peer object from peers_by_ip container which leads to a wrong refcounter value. adding an ao2_unlink from the peers_by_ip container fix it. Review: https://reviewboard.asterisk.org/r/1428/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335261 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	8b5ba33fe0	Merged revisions 335078 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r335078 \| mjordan \| 2011-09-09 11:27:01 -0500 (Fri, 09 Sep 2011) \| 29 lines Merged revisions 335064 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335064 \| mjordan \| 2011-09-09 11:09:09 -0500 (Fri, 09 Sep 2011) \| 23 lines Updated SIP 484 handling; added Incomplete control frame When a SIP phone uses the dial application and receives a 484 Address Incomplete response, if overlapped dialing is enabled for SIP, then the 484 Address Incomplete is forwarded back to the SIP phone and the HANGUPCAUSE channel variable is set to 28. Previously, the Incomplete application dialplan logic was automatically triggered; now, explicit dialplan usage of the application is required. Additionally, this patch adds a new AST_CONTOL_FRAME type called AST_CONTROL_INCOMPLETE. If a channel driver receives this control frame, it is an indication that the dialplan expects more digits back from the device. If the device supports overlap dialing it should attempt to notify the device that the dialplan is waiting for more digits; otherwise, it can handle the frame in a manner appropriate to the channel driver. (closes issue ASTERISK-17288) Reported by: Mikael Carlsson Tested by: Matthew Jordan Review: https://reviewboard.asterisk.org/r/1416/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335079 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Paul Belanger	272afe432b	Merged revisions 334844 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r334844 \| pabelanger \| 2011-09-07 15:37:24 -0400 (Wed, 07 Sep 2011) \| 11 lines Merged revisions 334843 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334843 \| pabelanger \| 2011-09-07 15:35:52 -0400 (Wed, 07 Sep 2011) \| 4 lines Cleanup chan_iax2.c log messages Review: https://code.asterisk.org/code/cru/CR-AST-11 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@334845 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Paul Belanger	39ac2e639f	Merged revisions 334514 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r334514 \| pabelanger \| 2011-09-06 11:47:59 -0400 (Tue, 06 Sep 2011) \| 6 lines authdebug is now disabled by default To enable this functionaility again set authdebug = yes in iax.conf Review: https://reviewboard.asterisk.org/r/1414/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@334517 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	9dd15059f6	Merged revisions 334157 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r334157 \| mnicholson \| 2011-08-31 13:53:40 -0500 (Wed, 31 Aug 2011) \| 11 lines Merged revisions 334156 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334156 \| mnicholson \| 2011-08-31 13:50:33 -0500 (Wed, 31 Aug 2011) \| 4 lines Disable T.38 when we get a invite with image media port set to 0 ASTERISK-17678 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@334158 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	89e79698fd	Optimize chan_sip.c check_rtp_timeout() function. * Make check_rtp_timeout() remember the values returned by ast_rtp_instance_get_timeout(), ast_rtp_instance_get_hold_timeout(), and ast_rtp_instance_get_keepalive() instead of repeatedly calling them. (closes issue ASTERISK-18319) Reported by: Rob Gagnon Patches: issue-18319-trunk-r333066.diff (License #6159) patch uploaded by Rob Gagnon Review: https://reviewboard.asterisk.org/r/1377/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@334115 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	1961bb6160	Merged revisions 334013 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r334013 \| rmudgett \| 2011-08-31 11:00:49 -0500 (Wed, 31 Aug 2011) \| 30 lines Merged revisions 334012 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334012 \| rmudgett \| 2011-08-31 10:57:12 -0500 (Wed, 31 Aug 2011) \| 23 lines No DAHDI channel available for conference, user introduction disabled. The following error will consistently occur when trying to dial into a MeetMe conference when the server does not have DAHDI hardware installed: app_meetme.c: No DAHDI channel available for conference, user introduction disabled (is chan_dahdi loaded?) While chan_dahdi is loaded correctly during compilation and install of Asterisk/Dahdi, including associated modules, etc., a chan_dahdi.conf configuration file in /etc/asterisk is not created by FreePBX if hardware does not exist, causing MeetMe to be unable to open a DAHDI pseudo channel. * Allow chan_dahdi to create a pseudo channel when there is no chan_dahdi.conf file to load. (closes issue ASTERISK-17398) Reported by: Preston Edwards Patches: jira_asterisk_17398_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: rmudgett ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@334014 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	ab17a27f97	Merged revisions 334010 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r334010 \| rmudgett \| 2011-08-31 10:23:11 -0500 (Wed, 31 Aug 2011) \| 50 lines Merged revisions 334009 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334009 \| rmudgett \| 2011-08-31 10:20:31 -0500 (Wed, 31 Aug 2011) \| 43 lines Call pickup race leaves orphaned channels or crashes. Multiple users attempting to pickup a call that has been forked to multiple extensions either crashes or fails a masquerade with a "bad things may happen" message. This is the scenario that is causing all the grief: 1) Pickup target is selected 2) target is marked as being picked up in ast_do_pickup() 3) target is unlocked by ast_do_pickup() 4) app dial or queue gets a chance to hang up losing calls and calls ast_hangup() on target 5) SINCE A MASQUERADE HAS NOT BEEN SETUP YET BY ast_do_pickup() with ast_channel_masquerade(), ast_hangup() completes successfully and the channel is no longer in the channels container. 6) ast_do_pickup() then calls ast_channel_masquerade() to schedule the masquerade on the dead channel. 7) ast_do_pickup() then calls ast_do_masquerade() on the dead channel 8) bad things happen while doing the masquerade and in the process ast_do_masquerade() puts the dead channel back into the channels container 9) The "orphaned" channel is visible in the channels list if a crash does not happen. This patch does the following: * Made ast_hangup() set AST_FLAG_ZOMBIE on a successfully hung-up channel and not release the channel lock until that has happened. * Made __ast_channel_masquerade() not setup a masquerade if either channel has AST_FLAG_ZOMBIE set. * Fix chan_agent misuse of AST_FLAG_ZOMBIE since it would no longer work. (closes issue ASTERISK-18222) Reported by: Alec Davis Tested by: rmudgett, Alec Davis, irroot, Karsten Wemheuer (closes issue ASTERISK-18273) Reported by: Karsten Wemheuer Tested by: rmudgett, Alec Davis, irroot, Karsten Wemheuer Review: https://reviewboard.asterisk.org/r/1400/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@334011 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	82229cc690	Merged revisions 334007 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r334007 \| kmoore \| 2011-08-31 10:19:30 -0500 (Wed, 31 Aug 2011) \| 14 lines Merged revisions 334006 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334006 \| kmoore \| 2011-08-31 10:18:37 -0500 (Wed, 31 Aug 2011) \| 7 lines Correct an AMI protocol violation with SIPshowpeer The response of SIPshowpeer ends with "\r\n\r\n". Since other commands are ended by using \r\n this confuses any interfacing script. (closes issue ASTERISK-17486) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@334008 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	ba3d34708e	Merged revisions 333837 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r333837 \| twilson \| 2011-08-29 16:41:13 -0500 (Mon, 29 Aug 2011) \| 22 lines Merged revisions 333836 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r333836 \| twilson \| 2011-08-29 16:38:31 -0500 (Mon, 29 Aug 2011) \| 15 lines Refresh peer address if DNS unavailable at peer creation If Asterisk starts and no DNS is available, outbound registrations will fail indefinitely. This patch copies the address from the sip_registry struct, which will be updated, to the peer->addr when necessary. If dnsmgr is enabled, the registration fails without the patch because even though the address on the registry is updated via dnsmgr, the address is just copied on the first try. Since we use ast_sockaddr_copy, dnsmgr can't update the address that is copied to the sip_pvt or peers. Closes issue ASTERISK-18000 Review: https://reviewboard.asterisk.org/r/1335/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@333838 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Tzafrir Cohen	ad02be1f2b	chan_vpb: remove unused variables (gcc4.6) GCC 4.6 detects variables that get assined to, but never used later. Also removes some remmed-out lines that become invalid. (closes issue ASTERISK-18336) Signed-off-by: Tzafrir Cohen (License #5035) <tzafrir.cohen@xorcom.com>, git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@333509 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jason Parker	590391c038	Fix typo from r333070 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@333159 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Olle Johansson	82e1dda364	Formatting changes - Removing some red white space and adding some curly brackets. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@333075 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Olle Johansson	64fb851843	Add manager event for local channel semi-bridge (issue AST-17623) Review: https://reviewboard.asterisk.org/r/1154 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@333070 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Paul Belanger	eb46a4128a	Merged revisions 332877 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r332877 \| pabelanger \| 2011-08-22 15:43:33 -0400 (Mon, 22 Aug 2011) \| 13 lines Merged revisions 332876 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332876 \| pabelanger \| 2011-08-22 15:41:24 -0400 (Mon, 22 Aug 2011) \| 6 lines Revert previous commit It seems google is still making changes to the protocol. (issue ASTERISK-18301) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@332880 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Paul Belanger	3a50145436	Merged revisions 332700 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r332700 \| pabelanger \| 2011-08-21 10:33:23 -0400 (Sun, 21 Aug 2011) \| 12 lines Merged revisions 332699 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332699 \| pabelanger \| 2011-08-21 10:31:31 -0400 (Sun, 21 Aug 2011) \| 5 lines Fix outgoing calls in chan_gtalk (closes issue ASTERISK-18301) Reported by: az1324 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@332701 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	75bb8797f5	Merged revisions 332504 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r332504 \| kmoore \| 2011-08-18 14:29:15 -0500 (Thu, 18 Aug 2011) \| 15 lines Merged revisions 332503 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332503 \| kmoore \| 2011-08-18 14:28:00 -0500 (Thu, 18 Aug 2011) \| 8 lines CRC4 in "dahdi show status" gives wrong impression to T1 users Change CRC4 to CRC in the output of "dahdi show status" so that it can apply in more situations without confusing users, especially since T1 lines use CRC6 instead of CRC4. (closes issue AST-471) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@332505 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	265102faf8	Merged revisions 332265 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r332265 \| rmudgett \| 2011-08-17 11:01:29 -0500 (Wed, 17 Aug 2011) \| 33 lines Merged revisions 332264 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332264 \| rmudgett \| 2011-08-17 10:51:08 -0500 (Wed, 17 Aug 2011) \| 26 lines Outgoing BRI calls fail when using Asterisk 1.8 with HA8, HB8, and B410P cards. France Telecom brings layer 2 and layer 1 down on BRI lines when the line is idle. When layer 1 goes down Asterisk cannot make outgoing calls and the HA8 and HB8 cards also get IRQ misses. The inability to make outgoing calls is because the line is in red alarm and Asterisk will not make calls over a line it considers unavailable. The IRQ misses for the HA8 and HB8 card are because the hardware is switching clock sources from the line which just brought layer 1 down to internal timing. There is a DAHDI option for the B410P card to not tell Asterisk that layer 1 went down so Asterisk will allow outgoing calls: "modprobe wcb4xxp teignored=1". There is a similar DAHDI option for the HA8 and HB8 cards: "modprobe wctdm24xxp bri_teignored=1". Unfortunately that will not clear up the IRQ misses when the telco brings layer 1 down. * Add layer 2 persistence option to customize the layer 2 behavior on BRI PTMP lines. The new option has three settings: 1) Use libpri default layer 2 setting. 2) Keep layer 2 up. Bring layer 2 back up when the peer brings it down. 3) Leave layer 2 down when the peer brings it down. Layer 2 will be brought up as needed for outgoing calls. JIRA AST-598 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@332270 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	269082f035	Merged revisions 332119 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r332119 \| jrose \| 2011-08-16 12:45:38 -0500 (Tue, 16 Aug 2011) \| 23 lines Merged revisions 332118 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332118 \| jrose \| 2011-08-16 12:38:19 -0500 (Tue, 16 Aug 2011) \| 16 lines ASTERISK-18067 ASTERISK-15479 - White Space affects mailbox value, multiple MWI subs Before, having multiple subscriptions to mailboxes on a sip peer set via the mailbox setting in sip.conf would only result in updates being sent on whichever mailbox triggered the mwi event. Now all of them get counted regardless. Also fixes a bug involving parsing of the mailbox option in sip.conf so that trailing and leading spaces before/after commas are trimmed. (closes issue ASTERISK-18067) Reported by: aragon (closes issue ASTERISK-15479) Reported by: Ben Winslow Patches: chan_sip.c-mwi_multi_mailbox_fix-1.6.2.13.diff (License #5288) patch uploaded by Ben Winslow ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@332120 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	ea5262eeb5	Merged revisions 332042 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r332042 \| mnicholson \| 2011-08-16 10:20:48 -0500 (Tue, 16 Aug 2011) \| 2 lines fix a code comment AST-580 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@332044 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	1858e274e3	Merged revisions 332027 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r332027 \| mnicholson \| 2011-08-16 10:08:40 -0500 (Tue, 16 Aug 2011) \| 9 lines Merged revisions 332026 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332026 \| mnicholson \| 2011-08-16 10:06:31 -0500 (Tue, 16 Aug 2011) \| 2 lines use DEFAULT_STORE_SIP_CAUSE to set the default value for the 'storesipcause' option AST-580 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@332028 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Olle Johansson	4403a89f50	Formatting changes while working with DTMF... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@332024 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	8f2e8d4b8a	Merged revisions 332022 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r332022 \| mnicholson \| 2011-08-16 09:40:37 -0500 (Tue, 16 Aug 2011) \| 16 lines In 10 and trunk this option is disabled by default. Merged revisions 332021 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332021 \| mnicholson \| 2011-08-16 09:20:43 -0500 (Tue, 16 Aug 2011) \| 7 lines Added the 'storesipcause' option to sip.conf to allow the user to disable the setting of HASH(SIP_CAUSE,<chan name>) on the channel. Having chan_sip set HASH(SIP_CAUSE,<chan name>) on the channel carries a significant performance penalty because of the usage of the MASTER_CHANNEL() dialplan function. AST-580 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@332023 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	3d42d45f25	Merged revisions 331956 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r331956 \| rmudgett \| 2011-08-15 12:35:03 -0500 (Mon, 15 Aug 2011) \| 20 lines Merged revisions 331955 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331955 \| rmudgett \| 2011-08-15 12:24:08 -0500 (Mon, 15 Aug 2011) \| 13 lines Fix some minor chan_dahdi config load issues. * Address chan_dahdi.conf dahdichan option todo item about needing line number. * Make ignore_failed_channels option also apply to dahdichan option. * Don't attempt to create a default pseudo channel if the chan_dahdi.conf channel/channels option is not allowed. * Add a similar check for dahdichan in normal chan_dahdi.conf sections as is done in users.conf. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331957 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	30b2f36c72	Merged revisions 331868 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r331868 \| dvossel \| 2011-08-15 10:14:13 -0500 (Mon, 15 Aug 2011) \| 12 lines Merged revisions 331867 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331867 \| dvossel \| 2011-08-15 10:12:16 -0500 (Mon, 15 Aug 2011) \| 6 lines Fixes locking inversion issues present in the handling of the sip REFER method. (closes issue ASTERISK-18082) Reported by: James Van Vleet ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331869 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Olle Johansson	6b7e997df2	Formatting guideline fixes git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331830 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	28e2aa76b2	Merged revisions 331772 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r331772 \| rmudgett \| 2011-08-12 13:59:45 -0500 (Fri, 12 Aug 2011) \| 15 lines Merged revisions 331771 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331771 \| rmudgett \| 2011-08-12 13:58:40 -0500 (Fri, 12 Aug 2011) \| 8 lines Suppress warning message when using DAHDITransfer or DAHDIHangup. * The fake event should only be processed by the channel that currently owns the private and not the associated call waiting or 3-way channel. JIRA AST-620 JIRA SWP-3616 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331773 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	452f198609	Merged revisions 331715 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r331715 \| rmudgett \| 2011-08-12 12:54:47 -0500 (Fri, 12 Aug 2011) \| 29 lines Merged revisions 331714 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331714 \| rmudgett \| 2011-08-12 12:47:57 -0500 (Fri, 12 Aug 2011) \| 22 lines AMI actions DAHDIHangup and DAHDITransfer have no effect. The AMI actions DAHDIHangup and DAHDITransfer have no effect on a DAHDI channel. These two AMI actions are highly specialized to analog channels and appear to make the channel behave like a jack port for headsets. * Made the faked DAHDI event get processed before a normal media stream read in dahdi_read() instead of trying to trigger an exception read by setting the AST_FLAG_EXCEPTION flag. Apparently a change was made long ago that changed how AST_FLAG_EXCEPTION is processed in the core. Unfortunately, the faked DAHDI events no longer worked when that happened. * Updated the DAHDI AMI action documentation for the following actions: DAHDITransfer, DAHDIHangup, DAHDIDialOffhook, DAHDIDNDon, DAHDIDNDoff, DAHDIShowChannels, and DAHDIRestart. * Made use sscanf() instead of atoi() for better error checking of the DAHDIChannel header string. JIRA AST-620 JIRA SWP-3616 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331716 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	a6ea606a78	Merged revisions 331518 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r331518 \| kmoore \| 2011-08-10 17:23:49 -0500 (Wed, 10 Aug 2011) \| 17 lines Merged revisions 331517 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331517 \| kmoore \| 2011-08-10 17:23:08 -0500 (Wed, 10 Aug 2011) \| 10 lines SIP Notify via AMI or CLI leaks SIP PVTs Any SIP notify sent via AMI or CLI leaks a SIP PVT with ref count +2. Removing the additional ref just before the invite and adding an unref following it corrects the issue as seen via REF_DEBUG. The unref existed in a distant revision and it appears as though the wrong ref operation was removed. (closes issue ASTERISK-18091) Review: https://reviewboard.asterisk.org/r/1332/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331519 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	dc9513a69d	SIP display-name needed to be empty for Avaya IP500 In order to address a compatability issue with certain features on certain devices which rely on display name content to change behavior, initreqprep in chan_sip.c has been changed to no longer substitute cid_number into the display name when cid_name isn't present. Instead, it will send no display name in that case. (closes issue ASTERISK-16198) Reported by: Walter Doekes Review: https://reviewboard.asterisk.org/r/1341/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331371 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	b99b1116be	Merged revisions 331265 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r331265 \| rmudgett \| 2011-08-09 18:12:49 -0500 (Tue, 09 Aug 2011) \| 22 lines Merged revisions 331248 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331248 \| rmudgett \| 2011-08-09 17:12:59 -0500 (Tue, 09 Aug 2011) \| 15 lines Misc minor items found in code. * Add some reentrancy protection in pbx.c when creating the contexts_table hash table. * Fix inverted test in chan_sip.c conditional code. * Fix uninitialized variable and use of the wrong variable in chan_iax2.c. * Fix test of return value in app_parkandannounce.c. Explicitly testing for -1 is bad if the function does not actually return that value when it fails. * Fixup some comments and add some curly braces in features.c. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331266 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	db3d113414	Merged revisions 330706 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r330706 \| kmoore \| 2011-08-03 08:39:06 -0500 (Wed, 03 Aug 2011) \| 17 lines Merged revisions 330705 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r330705 \| kmoore \| 2011-08-03 08:38:17 -0500 (Wed, 03 Aug 2011) \| 10 lines Call pickup broken for DAHDI channels when beginning with # The call pickup feature did not work on DAHDI devices for anything other than feature codes beginning with * since all feature codes in chan_dahdi were originally hard-coded to begin with *. This patch is also applied to chan_dahdi.c to fix this bug with radio modes. (closes issue AST-621) Review: https://reviewboard.asterisk.org/r/1336/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@330707 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	e128ee2567	Merged revisions 330586 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r330586 \| dvossel \| 2011-08-02 11:17:59 -0500 (Tue, 02 Aug 2011) \| 15 lines Merged revisions 330581 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r330581 \| dvossel \| 2011-08-02 11:15:08 -0500 (Tue, 02 Aug 2011) \| 8 lines Fixes crash in chan_iax2. Fixes crash in chan_iax2 resulting from an edge case in the way control frames are queued during calltoken negotiation is complete. (closes issue ASTERISK-17610) Reported by: mgrobecker ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@330593 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	6f112cce0d	Merged revisions 330579 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r330579 \| dvossel \| 2011-08-02 11:08:57 -0500 (Tue, 02 Aug 2011) \| 9 lines Merged revisions 330578 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r330578 \| dvossel \| 2011-08-02 11:07:02 -0500 (Tue, 02 Aug 2011) \| 2 lines Optimization to buffer initialization fix. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@330580 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
David Vossel	d50e68c827	Merged revisions 330576 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r330576 \| dvossel \| 2011-08-02 10:55:36 -0500 (Tue, 02 Aug 2011) \| 12 lines Merged revisions 330575 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r330575 \| dvossel \| 2011-08-02 10:53:21 -0500 (Tue, 02 Aug 2011) \| 5 lines Fixes uninitialized string buffer in log message. (closes issue ASTERISK-17200) Reported by: lmadsen ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@330577 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	6a15e95a32	astobj2: Avoid using temporary objects + ao2_find() with OBJ_POINTER. There is a fairly common pattern making its way through the code base where we put a temporary object on the stack so we can call ao2_find() with OBJ_POINTER. The purpose is so that it can be passed into the object hash function. However, this really seems like a hack and potentially error prone. This patch is a first stab at approach to avoid having to do that. It adds a new flag, OBJ_KEY, which can be used instead of OBJ_POINTER in these situations. Then, the hash function can know whether it was given an object or some custom data to hash. The patch also changes some uses of ao2_find() for iax2_user and iax2_peer objects to reflect how OBJ_KEY would be used. So long, and thanks for all the fish. Review: https://reviewboard.asterisk.org/r/1184/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@330273 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	a3ba55e7c7	Merged revisions 330051 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r330051 \| rmudgett \| 2011-07-28 12:10:37 -0500 (Thu, 28 Jul 2011) \| 29 lines Merged revisions 330050 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r330050 \| rmudgett \| 2011-07-28 12:04:24 -0500 (Thu, 28 Jul 2011) \| 22 lines Merged revisions 330033 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier .......... r330033 \| rmudgett \| 2011-07-28 11:26:38 -0500 (Thu, 28 Jul 2011) \| 15 lines Datacalls with B410P fail. Incoming and outgoing call legs of a data call are using different formats: a-law, u-law. When the call is bridged, the media stream is run through translation to convert the media formats. The translation is bad for data calls. * Make incoming call that does not explicitly specify u-law or a-law use the DAHDI channel's default law. The outgoing call always uses the default law from the DAHDI channel. (closes issue ABE-2800) Patches: jira_abe_2800_companding.patch (license #5621) patch uploaded by rmudgett .......... ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@330052 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jason Parker	16a32f5030	Merged revisions 329995 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r329995 \| qwell \| 2011-07-28 10:45:49 -0500 (Thu, 28 Jul 2011) \| 13 lines Merged revisions 329994 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329994 \| qwell \| 2011-07-28 10:45:24 -0500 (Thu, 28 Jul 2011) \| 6 lines Fix a SIP transfer deadlock. The locking in this function is very scary. There are like 6 structs involved. (closes issue AST-470) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@329996 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Sean Bright	73462b32dd	Merged revisions 329896 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r329896 \| seanbright \| 2011-07-28 07:35:27 -0400 (Thu, 28 Jul 2011) \| 9 lines Merged revisions 329895 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329895 \| seanbright \| 2011-07-28 07:34:33 -0400 (Thu, 28 Jul 2011) \| 2 lines Make the output of Externhost in 'sip show settings' more consistent. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@329897 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	5c627eba2b	Remove lastmsgssent from sip it has not been working since 1.6	14 years ago
Russell Bryant	f243d129c9	Merged revisions 329257 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r329257 \| russell \| 2011-07-21 15:22:36 -0500 (Thu, 21 Jul 2011) \| 2 lines s/1.10/10.0/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@329258 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago

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7093 Commits (beae2df26eaba06d91b99e1365596081bcd30ca0)