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2011-09-28Merged revisions 338228 via svnmerge from Jason Parker
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r338228 | qwell | 2011-09-28 15:54:35 -0500 (Wed, 28 Sep 2011) | 9 lines Merged revisions 338227 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r338227 | qwell | 2011-09-28 15:52:47 -0500 (Wed, 28 Sep 2011) | 1 line Add support levels to non-module sections of menuselect (cflags, utils, etc). ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@338229 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-28Merged revisions 338225 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r338225 | rmudgett | 2011-09-28 15:26:39 -0500 (Wed, 28 Sep 2011) | 12 lines Merged revisions 338224 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r338224 | rmudgett | 2011-09-28 15:24:41 -0500 (Wed, 28 Sep 2011) | 5 lines Fix chan_dahd compiling with gcc 4.6 when PRI and SS7 not present. (closes issue ASTERISK-18357) Reported by: Matthew Nicholson ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@338226 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-27Whitespace (red blobs) fixesOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@338042 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-22Merged revisions 337721 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r337721 | rmudgett | 2011-09-22 16:37:41 -0500 (Thu, 22 Sep 2011) | 25 lines Merged revisions 337720 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337720 | rmudgett | 2011-09-22 16:29:46 -0500 (Thu, 22 Sep 2011) | 18 lines Made ISDN not add numbering plan prefix strings to empty numbers. When the Caller-ID is restricted, the expected behavior is for the Caller-ID to be blank. In chan_dahdi, the national prefix is placed onto the Caller-ID number even if it is restricted (empty) causing the Caller-ID to be the national prefix rather than blank. This behavior was lost when sig_pri was extracted from chan_dahdi. * Made not add prefix strings to empty connected line, calling, and ANI number strings. (closes issue ASTERISK-18577) Reported by: Kris Shaw Patches: jira_asterisk_18577_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: Kris Shaw ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337722 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-22Merged revisions 337595,337597 via svnmerge from Jonathan Rose
https://origsvn.digium.com/svn/asterisk/branches/10 ........ r337595 | jrose | 2011-09-22 10:35:50 -0500 (Thu, 22 Sep 2011) | 12 lines Generate Security events in chan_sip using new Security Events Framework Security Events Framework was added in 1.8 and support was added for AMI to generate events at that time. This patch adds support for chan_sip to generate security events. (closes issue ASTERISK-18264) Reported by: Michael L. Young Patches: security_events_chan_sip_v4.patch (license #5026) by Michael L. Young Review: https://reviewboard.asterisk.org/r/1362/ ........ r337597 | jrose | 2011-09-22 10:47:05 -0500 (Thu, 22 Sep 2011) | 10 lines Forgot to svn add new files to r337595 Part of Generating security events for chan_sip (issue ASTERISK-18264) Reported by: Michael L. Young Patches: security_events_chan_sip_v4.patch (License #5026) by Michael L. Young Reviewboard: https://reviewboard.asterisk.org/r/1362/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337600 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-22Merged revisions 337487 via svnmerge from Gregory Nietsky
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r337487 | irroot | 2011-09-22 11:26:26 +0200 (Thu, 22 Sep 2011) | 16 lines Merged revisions 337486 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337486 | irroot | 2011-09-22 11:22:26 +0200 (Thu, 22 Sep 2011) | 10 lines If IP address is used in chan_h323 host parameter of peer configuration. module tries to resolve IP address to IP address and fails. Simple fix to set family of socket this is a hangover from ipv6 changes. (closes issue ASTERISK-18237) (issue ASTERISK-17278) (issue ASTERISK-17500) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337488 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-20Merged revisions 337008 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r337008 | rmudgett | 2011-09-20 14:12:24 -0500 (Tue, 20 Sep 2011) | 22 lines Merged revisions 337007 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337007 | rmudgett | 2011-09-20 14:10:30 -0500 (Tue, 20 Sep 2011) | 15 lines Check if a channel was created before using the pointer in sig_ss7_new_ast_channel(). Fixes the crash in ASTERISK-17955 gdb-11918.txt backtrace. * Added some missing libss7 access lock protection. * Prevent cancelling the ss7_linkset() thread at inoportune times just like the pri_dchannel() thread. (issue ASTERISK-17955) Reported by: Ian M Sherman Patches: jira_asterisk_17955_v1.8.patch (license #5621) patch uploaded by rmudgett (attached to related ASTERISK-17966) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337009 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-20Merged revisions 336978 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r336978 | rmudgett | 2011-09-20 13:14:40 -0500 (Tue, 20 Sep 2011) | 28 lines Merged revisions 336977 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336977 | rmudgett | 2011-09-20 13:12:17 -0500 (Tue, 20 Sep 2011) | 21 lines Fix deadlock from not releasing SS7 linkset lock. sig_ss7_hangup() failed to release the SS7 linkset lock if the call had the alreadyhungup flag set. * Made unlock the SS7 linkset lock in sig_ss7_hangup() if the alreadyhungup flag is set. * Made ss7_start_call() not hold any locks while creating the channel for an incoming call to prevent deadlock. * Made ss7_grab() a void function, since it could never fail, to simplify calling code. * Made obtain the channel lock to do softhangup in some places. Patches: jira_ast_668_v1.8.patch (license #5621) patch uploaded by rmudgett JIRA AST-668 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@336988 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-20Merged revisions 336936 via svnmerge from Gregory Nietsky
https://origsvn.digium.com/svn/asterisk/branches/10 ........ r336936 | irroot | 2011-09-20 18:51:59 +0200 (Tue, 20 Sep 2011) | 14 lines Allow Setting Auth Tag Bit length Based on invite or config option Update the SIP SRTP API to allow use of 32 or 80 bit taglen. Curently only 80 bit is supported. The outgoing invite will use the taglen of the incoming invite preventing one-way audio. (Closes issue ASTERISK-17895) Review: https://reviewboard.asterisk.org/r/1173/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@336937 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-19Merged revisions 336792 via svnmerge from Terry Wilson
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r336792 | twilson | 2011-09-19 17:13:34 -0500 (Mon, 19 Sep 2011) | 9 lines Merged revisions 336791 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336791 | twilson | 2011-09-19 17:07:58 -0500 (Mon, 19 Sep 2011) | 2 lines Don't interfere with T.38 reinvites This is an update to the fix for ASTERISK-18340 and ASTERISK-17725 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@336837 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-19Merged revisions 336570 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r336570 | rmudgett | 2011-09-19 10:32:00 -0500 (Mon, 19 Sep 2011) | 11 lines Merged revisions 336569 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336569 | rmudgett | 2011-09-19 10:25:34 -0500 (Mon, 19 Sep 2011) | 4 lines Rework sig_pri_hangup() to be simpler and clearer. JIRA AST-675 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@336571 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-19Merged revisions 336502 via svnmerge from Olle Johansson
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r336502 | oej | 2011-09-19 15:38:53 +0200 (Mån, 19 Sep 2011) | 12 lines Merged revisions 336501 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336501 | oej | 2011-09-19 15:33:50 +0200 (Mån, 19 Sep 2011) | 5 lines Add diversion header to a 302 redirect response if we have diversion data (closes issue ASTERISK-18143) patch by oej ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@336505 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-19Merged revisions 336500 via svnmerge from Gregory Nietsky
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r336500 | irroot | 2011-09-19 15:31:50 +0200 (Mon, 19 Sep 2011) | 19 lines Merged revisions 336499 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336499 | irroot | 2011-09-19 15:27:52 +0200 (Mon, 19 Sep 2011) | 13 lines A long time ago in a galaxy far far away a IPv6 update was made, chan_h323 was not updated causeing all to flee to chan_ooh323. the brave Jedi [asterisk developers] pondered this miscarrige of justice and restored order to the force for the sake of closing out 2 old issues. (closes issue ASTERISK-17278) (closes issue ASTERISK-17500) Reported by: dread, sybasesql Tested by: irroot Reviewed by: IRC (russellb, kpfleming) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@336503 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-19Merged revisions 336381 via svnmerge from Olle Johansson
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r336381 | oej | 2011-09-19 12:05:00 +0200 (Mån, 19 Sep 2011) | 16 lines Merged revisions 336378 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336378 | oej | 2011-09-19 11:40:44 +0200 (Mån, 19 Sep 2011) | 9 lines Add missing unlock at MWI message sending time (closes issue ASTERISK-18573) Patches: sip_mwi_lock.patch (license #5041) by Gregory Hinton Nietsky Thanks to irrot for the reminder, to Gregory for the patch! ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@336382 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-16Merged revisions 336307 via svnmerge from Jonathan Rose
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
2011-09-16Merged revisions 336167 via svnmerge from Gregory Nietsky
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
2011-09-15Merged revisions 335991 via svnmerge from Gregory Nietsky
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
2011-09-14Merged revisions 335852 via svnmerge from Richard Mudgett
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
2011-09-12Documentation updatesOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-12Small documentation updatesOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335349 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-12New sip.conf option for setting default tonezone for channel or individual ↵Olle Johansson
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
2011-09-12Merged revisions 335323 via svnmerge from Olle Johansson
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
2011-09-12Merged revisions 335321 via svnmerge from Kinsey Moore
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
2011-09-12Merged revisions 335260 via svnmerge from Stefan Schmidt
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
2011-09-09Merged revisions 335078 via svnmerge from Matthew Jordan
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
2011-09-07Merged revisions 334844 via svnmerge from Paul Belanger
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
2011-09-06Merged revisions 334514 via svnmerge from Paul Belanger
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
2011-08-31Merged revisions 334157 via svnmerge from Matthew Nicholson
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
2011-08-31Optimize chan_sip.c check_rtp_timeout() function.Richard Mudgett
* 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
2011-08-31Merged revisions 334013 via svnmerge from Richard Mudgett
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
2011-08-31Merged revisions 334010 via svnmerge from Richard Mudgett
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
2011-08-31Merged revisions 334007 via svnmerge from Kinsey Moore
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
2011-08-29Merged revisions 333837 via svnmerge from Terry Wilson
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
2011-08-28chan_vpb: remove unused variables (gcc4.6)Tzafrir Cohen
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
2011-08-24Fix typo from r333070Jason Parker
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@333159 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-08-24Formatting changes - Removing some red white space and adding some curly ↵Olle Johansson
brackets. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@333075 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-08-24Add manager event for local channel semi-bridgeOlle Johansson
(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
2011-08-22Merged revisions 332877 via svnmerge from Paul Belanger
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
2011-08-21Merged revisions 332700 via svnmerge from Paul Belanger
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
2011-08-18Merged revisions 332504 via svnmerge from Kinsey Moore
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
2011-08-17Merged revisions 332265 via svnmerge from Richard Mudgett
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
2011-08-16Merged revisions 332119 via svnmerge from Jonathan Rose
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
2011-08-16Merged revisions 332042 via svnmerge from Matthew Nicholson
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
2011-08-16Merged revisions 332027 via svnmerge from Matthew Nicholson
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
2011-08-16Formatting changes while working with DTMF...Olle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@332024 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-08-16Merged revisions 332022 via svnmerge from Matthew Nicholson
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
2011-08-15Merged revisions 331956 via svnmerge from Richard Mudgett
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
2011-08-15Merged revisions 331868 via svnmerge from David Vossel
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
2011-08-15Formatting guideline fixesOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331830 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-08-12Merged revisions 331772 via svnmerge from Richard Mudgett
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
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