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Author	SHA1	Message	Date
Richard Mudgett	3423bb637d	Merged revisions 286113 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r286113 | rmudgett | 2010-09-10 15:33:16 -0500 (Fri, 10 Sep 2010) | 11 lines An outgoing call may not get hung up if a pre-connect incoming ISDN call is disconnected. If the ISDN link a pre-connect incoming call is using fails or is reset, the outgoing leg may not hang up or be delayed in hanging up. (Causes: PRI_CAUSE_NETWORK_OUT_OF_ORDER, PRI_CAUSE_DESTINATION_OUT_OF_ORDER, and PRI_CAUSE_NORMAL_TEMPORARY_FAILURE.) Just hang up the call if the incoming call leg hangs up before connecting for any reason. It makes no sense to send a BUSY or CONGESTION control frame to the outgoing call leg under these circumstances. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@286116 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	a6ecd54058	Merged revisions 286059 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r286059 | twilson | 2010-09-10 14:25:08 -0500 (Fri, 10 Sep 2010) | 16 lines Inherit CHANNEL() writes to both sides of a Local channel Having Local (/n) channels as queue members and setting the language in the extension with Set(CHANNEL(language)=fr) sets the language on the Local/...,2 channel. Hold time report playbacks happen on the Local/...,1 channel and therefor do not play in the specified language. This patch modifies func_channel_write to call the setoption callback and pass the CHANNEL() write info to the callback. chan_local uses this information to look up the other side of the channel and apply the same changes to it. (closes issue #17673) Reported by: Guggemand Review: https://reviewboard.asterisk.org/r/903/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@286115 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	a08ad772b9	Blocked revisions 286070 via svnmerge ... ........ r286070 | dvossel | 2010-09-10 15:03:50 -0500 (Fri, 10 Sep 2010) | 32 lines Fixes sip extension state update DEADLOCK PROBLEM: In chan_sip, and all the other channel drivers, it is common for us to hold the tech_pvt lock while we ask the Asterisk core about an extension and context. Every time we do this the locking order becomes, (1. tech_pvt lock ---> 2. global context lock). In chan_sip when a dialog subscribes to a hint, that locking order is reversed in the extensionstate callback which will occur outside of the channel_driver's monitor loop. So, on an extension state update we have (1. global context lock ----> 2. tech_pvt lock). Typically when we have to do a reversed locking order like this we'd just do some sort of deadlock avoidance to fix the problem... That will not work here. There are more locks involved here than just the context and tech_pvt. Those are the two that are colliding, but it is impossible to give up the context lock because the global hints list lock MUST be held as well and we can not give that lock up during the extensionstate callback traversal... The locking order for the context and hints are (1. global context lock ----> 2. hints list lock). Deadlock avoidance is not an option here. SOLUTION: The solution this patch implements is to queue the extension state updates into a list and send the NOTIFY messages out during the do_monitor pvt traversal. This clears out the problem of having to hold the context lock before the tech_pvt lock entirely. (closes issue #17888) Reported by: zerohalo ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@286071 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	ec3f34e375	Merged revisions 286023 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r286023 | tilghman | 2010-09-10 13:22:04 -0500 (Fri, 10 Sep 2010) | 2 lines Missing newline ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@286024 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	6e26576166	Another fix for Mac OS X. ... While trying to fix this the "right" way, I wandered into dependency hell. Two hours later, I backed out, and just removed the offending code. ast_inline_api only goes one level deep and then it breaks. Ouch. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@285961 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	8d4cda01cb	Merged revisions 285889 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r285889 | tilghman | 2010-09-09 19:13:45 -0500 (Thu, 09 Sep 2010) | 7 lines Fix Mac OS X build. This also fixes a rather grievous calculation error for the offset of ast_fdset, which was masked on Linux and FreeBSD, because these platforms check the first 256 FDs regardless of the bitmask setting (due to backwards compatibility). ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@285930 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Paul Belanger	08c90ed82d	Merged revisions 285817 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r285817 | pabelanger | 2010-09-09 18:34:35 -0400 (Thu, 09 Sep 2010) | 8 lines GCC 4.2.x optimizations result in improper behavior of GSM codec (closes issue #17688) Reported by: pprindeville Patches: asterisk-trunk-bugid11243.patch uploaded by pprindeville (license 347) Tested by: mkeuter, pprindeville ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@285818 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jason Parker	63ff776f65	Merged revisions 285742 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r285742 | qwell | 2010-09-09 15:06:31 -0500 (Thu, 09 Sep 2010) | 9 lines Transmit silence when reading DTMF in ast_readstring. Otherwise, you could get issues with DTMF timeouts causing hangups. (closes issue #17370) Reported by: makoto Patches: channel-readstring-silence-generator.patch uploaded by makoto (license 38) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@285744 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Brett Bryant	e05f4b7e20	Fixes an issue with dialplan pattern matching where the specificity for pattern ranges and pattern special characters was inconsistent. ... (closes issue #16903) Reported by: Nick_Lewis Patches: pbx.c-specificity.patch uploaded by Nick Lewis (license 657) Tested by: Nick_Lewis git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@285710 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Brett Bryant	93d5f26afd	Merged revisions 285638 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r285638 | bbryant | 2010-09-09 13:20:17 -0400 (Thu, 09 Sep 2010) | 7 lines Fixes an issue with MOH where it doesn't recover cleanly when it can't play a file and would just stop, instead of continuing to find the next playable file in the MOH class. (closes issue #17807) Reported by: kshumard Review: https://reviewboard.asterisk.org/r/910/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@285639 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	1f0690e96b	Merged revisions 285566 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r285566 | dvossel | 2010-09-08 17:07:31 -0500 (Wed, 08 Sep 2010) | 2 lines In retrans_pkt, do not unlock pvt until the end of the function on a transmit failure. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@285567 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	2fefbcf3e2	Fixes interoperability problems with session timer behavior in Asterisk. ... CHANGES: 1. Never put "timer" in "Require" header. This is not to our benefit and RFC 4028 section 7.1 even warns against it. It is possible for one endpoint to perform session-timer refreshes while the other endpoint does not support them. If in this case the end point performing the refreshing puts "timer" in the Require field during a refresh, the dialog will likely get terminated by the other end. 2. Change the behavior of 'session-timer=accept' in sip.conf (which is the default behavior of Asterisk with no session timer configuration specified) to only run session-timers as result of an incoming INVITE request if the INVITE contains an "Session-Expires" header... Asterisk is currently treating having the "timer" option in the "Supported" header as a request for session timers by the UAC. I do not agree with this. Session timers should only be negotiated in "accept" mode when the incoming INVITE supplies a "Session-Expires" header, otherwise RFC 4028 says we should treat a request containing no "Session-Expires" header as a session with no expiration. Below I have outlined some situations and what Asterisk's behavior is. The table reflects the behavior changes implemented by this patch. SITUATIONS: -Asterisk as UAS 1. Incoming INVITE: NO "Session-Expires" 2. Incoming INVITE: HAS "Session-Expires" -Asterisk as UAC 3. Outgoing INVITE: NO "Session-Expires". 200 Ok Response HAS "Session-Expires" header 4. Outgoing INVITE: NO "Session-Expires". 200 Ok Response NO "Session-Expires" header 5. Outgoing INVITE: HAS "Session-Expires". Active - Asterisk will have an active refresh timer regardless if the other endpoint does. Inactive - Asterisk does not have an active refresh timer regardless if the other endpoint does. XXXXXXX - Not possible for mode. ______________________________________ |SITUATIONS | 'session-timer' MODES | |___________|________________________| | | originate | accept | |-----------|------------|-----------| |1. | Active | Inactive | |2. | Active | Active | |3. | XXXXXXXX | Active | |4. | XXXXXXXX | Inactive | |5. | Active | XXXXXXXX | -------------------------------------- (closes issue #17005) Reported by: alexrecarey git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@285563 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Brett Bryant	68d3fea001	Fixes a bug with MeetMe where after announcing the amount of time left in a conference, if music on hold was playing, it doesn't restart. ... (closes issue #17408) Reported by: sysreq Patches: asterisk-issue-17408_fixed.patch uploaded by sysreq (license 1009) Tested by: sysreq git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@285532 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jason Parker	878abe8339	Follow coding guidelines in moh rescan fix. Also fix the documentation that got me in trouble. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@285529 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jason Parker	2a6af6063e	Fixes issue where moh files were no longer rescanned during a reload. ... (closes issue #16744) Reported by: pj Patches: 16744-reload.diff uploaded by qwell (license 4) Tested by: qwell git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@285526 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	0924686514	Merged revisions 285365 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r285365 | tilghman | 2010-09-07 15:30:22 -0500 (Tue, 07 Sep 2010) | 9 lines Catch invalid extensions at the parser, instead of making the core deal with them. (closes issue #17794) Reported by: PavelL Patches: 20100820__issue17794__1.6.2.diff.txt uploaded by tilghman (license 14) 20100820__issue17794__1.4.diff.txt uploaded by tilghman (license 14) Tested by: PavelL ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@285366 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	3ca871af91	Merged revisions 285266 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r285266 | tilghman | 2010-09-07 14:04:50 -0500 (Tue, 07 Sep 2010) | 4 lines Use poll, if indicated to do so, in the ast_poll2 implementation. This fixes the unit tests on FreeBSD 8.0. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@285267 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Brett Bryant	a5c986a4e4	Merged revisions 285194 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r285194 | bbryant | 2010-09-07 13:45:41 -0400 (Tue, 07 Sep 2010) | 10 lines Fixes voicemail.conf issues where mailboxes with passwords that don't precede a comma would throw unnecessary error messages. (closes issue #15726) Reported by: 298 Patches: M15726.diff uploaded by junky (license 177) Tested by: junky Review: [full review board URL with trailing slash] ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@285196 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	7c50101e0b	Merged revisions 285088 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r285088 | tilghman | 2010-09-06 01:54:18 -0500 (Mon, 06 Sep 2010) | 2 lines Silly convenience script for BSD platforms. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@285089 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Brett Bryant	baf689788e	This is a patch provided for issue #17935 to add the ActionID to the IAXregistry AMI response. ... (closes issue #17935) Reported by: alexkuklin Patches: iaxshowreg uploaded by alexkuklin (license 1115) Tested by: alexkuklin git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@284958 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	00bb0abd8e	Merged revisions 284881 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r284881 | twilson | 2010-09-03 11:10:23 -0500 (Fri, 03 Sep 2010) | 5 lines Properly detect when a sound file doesn't exist ast_fileexists returns -1 for error and 0 for a non-existant file. The existing code treated missing files as though they existed. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@284897 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Brett Bryant	5f7150b890	Merged revisions 284777 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r284777 | bbryant | 2010-09-02 16:25:03 -0400 (Thu, 02 Sep 2010) | 7 lines Fixes a bug in manager.c where the default configuration values weren't reset when the manager configuration was reloaded. (closes issue #17917) Reported by: lmadsen Review: https://reviewboard.asterisk.org/r/883/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@284778 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	3325966e1d	Merged revisions 284703 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r284703 | dvossel | 2010-09-02 11:47:15 -0500 (Thu, 02 Sep 2010) | 7 lines Removed relatedpeer code from sip_autodestruct Handling of the relatedpeer structure associated with a sip_pvt should be done during the final sip_destruction function, not in sip_autodestruct. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@284704 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	270a11e92b	Fixing build. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@284665 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	4e046ccf08	Don't reset queue stats on a module reload. ... (closes issue #17535) Reported by: raarts Patches: 20100819__issue17535.diff.txt uploaded by tilghman (license 14) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@284631 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	5e12dc60f2	Failed to rerun bootstrap.sh after last commit ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@284595 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	59f700a9b2	Merged revisions 284478 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r284478 | tilghman | 2010-09-01 13:49:11 -0500 (Wed, 01 Sep 2010) | 11 lines Ensure that all areas that previously used select(2) now use poll(2), with implementations that need poll(2) implemented with select(2) safe against 1024-bit overflows. This is a followup to the fix for the pthread timer in 1.6.2 and beyond, fixing a potential crash bug in all supported releases. (closes issue #17678) Reported by: russell Branch: https://origsvn.digium.com/svn/asterisk/team/tilghman/ast_select Review: https://reviewboard.asterisk.org/r/824/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@284593 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	f70cd032b7	Don't warn on floats and timestamps ... (closes issue #17082) Reported by: coolmig git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@284472 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	58b8667476	Merged revisions 284393 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r284393 | tilghman | 2010-08-31 15:13:21 -0500 (Tue, 31 Aug 2010) | 7 lines Don't send a devstate change on poke_noanswer if the state did not change. (closes issue #17741) Reported by: schmidts Patches: chan_sip.c.patch uploaded by schmidts (license 1077) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@284399 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Leif Madsen	928ef055fe	Merged revisions 284316 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r284316 | lmadsen | 2010-08-31 13:57:59 -0500 (Tue, 31 Aug 2010) | 7 lines Update say.conf.sample to match the rules in say.c (closes issue #17835) Reported by: RoadKill Patches: say.conf.sample.patch.rules uploaded by RoadKill (license 933) Tested by: RoadKill ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@284317 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	7f48d17174	Fix 3 coding errors: ... 1) After we close FD, we should not be trying to write to it. 2) Call _exit(0), not exit(0), to avoid running shutdown routines in a child. 3) Use endian, not processor, detection to ensure bytes are written in the correct order. (closes issue #15706) Reported by: modelnine Patches: asterisk-1.6.1.1-festival-debug.patch uploaded by modelnine (license 865) Tested by: gmartinez git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@284280 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	9253b159fd	Merged revisions 283960 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r283960 | dvossel | 2010-08-27 17:17:26 -0500 (Fri, 27 Aug 2010) | 8 lines Parse all "Accept" headers for SIP SUBSCRIBE requests. (closes issue #17758) Reported by: ibc Patches: multiple_accept_headers_1.4.diff uploaded by dvossel (license 671) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@284002 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jason Parker	e8361949d4	Merged revisions 283880 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r283880 | qwell | 2010-08-27 15:29:11 -0500 (Fri, 27 Aug 2010) | 8 lines Fix issue with decoding ^-escaped characters in realtime. (closes issue #17790) Reported by: denzs Patches: 17790-chunky.diff uploaded by qwell (license 4) Tested by: qwell, denzs ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@283881 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	015d89b40a	Merged revisions 283690 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r283690 | dvossel | 2010-08-26 10:22:28 -0500 (Thu, 26 Aug 2010) | 19 lines Fixed how Asterisk destroys a dialog on channel hangup before invite receives a response. If an ast_channel with a SIP tech pvt hangs up before the sip dialog gets a response to its outgoing INVITE, Asterisk used to pretend_ack the INVITE. This is not rfc compliant and results in confusion at the other endpoint. sip_pretend_ack will ack and remove all the packets in the retransmit queue. This means that the INVITE will stop retransmitting, and that any response to that INVITE that comes after the pretend_ack occurs will be ignored. Instead of faking any sort of acknowledgement for an outgoing INVITE during an internal hangup, we should let the protocol stack process the INVITE transaction and terminate the dialog properly. This is achieved by setting the PENDING_BYE flag. When this flag is used, once the dialog proceeds to an escapable state the transaction will either be canceled with a SIP_CANCEL or completed followed immediately by a BYE. Attempting to do this any other way is incorrect. If the endpoint is not responding to the INVITE request, the INVITE must continue to be retransmitted until it times out which will result in the dialog being destroyed. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@283691 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	8aa02d419f	Add to and from tags to NOTIFY dialog-info xml body so pickup can occur. ... When pedantic mode is used, the dialog-info xml generated during a ringing event must contain the to and from tag values. Otherwise if a pickup occurs using INVITE with replaces, Astrisk will not be able to locate the subscription. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@283594 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	4d36037db1	Asterisk will not advertise session timers are supported when 'session-timers=refuse' is used. ... Asterisk now dynamically builds the "Supported" header depending on what is enabled/disabled in sip.conf. Session timers used to always be advertised as being supported even when they were disabled in the configuration. This caused problems with some end points. (issue #17005) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@283558 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	967a8a11d8	Merged revisions 283380 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r283380 | dvossel | 2010-08-24 11:01:51 -0500 (Tue, 24 Aug 2010) | 11 lines This fix makes sure the ast_channel hangs up correctly when the dialog's PENDING_BYE flag is set. When the pending bye flag is used, it is possible that the dialog will terminate and leave the sip_pvt->owner channel up. This is because we never hangup the ast_channel after sending the SIP_BYE request. When we receive the response for the SIP_BYE we set need_destroy which we would expect to destroy the dialog on the next do_monitor loop, but this is not the case. The dialog will only be destroyed once the owner is hungup even with the need_destroy flag set. This patch sets the softhangup flag on the ast_channel when a SIP_BYE request is sent as a result of the pending bye flag. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@283381 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	727cf54d50	CDR drivers depend upon res_odbc, not directly on the ODBC libraries ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@283318 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	db355da802	Merged revisions 283123 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r283123 | rmudgett | 2010-08-20 11:46:22 -0500 (Fri, 20 Aug 2010) | 9 lines Merged revision 278274 from https://origsvn.digium.com/svn/asterisk/trunk .......... r278274 | rmudgett | 2010-07-20 17:38:13 -0500 (Tue, 20 Jul 2010) | 1 line Reference correct struct member for unlikely event PRI_EVENT_CONFIG_ERR. .......... ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@283124 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	bdda8674c6	Merged revisions 283048 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r283048 | rmudgett | 2010-08-20 10:24:36 -0500 (Fri, 20 Aug 2010) | 22 lines Q931 - Sending PROGRESS after sending ALERTING is a protocol error The PRI layer in chan_dadhi will check if a PROGRESS message has already been sent, and not allow sending another (although that is technically allowed by the Q931 spec), however it does not protect against sending an ALERTING and then sending a PROGRESS message, which is a violation of the specification. Most switches don't seem to care too deeply about this, but some do, and will disconnect the call when receiving this invalid sequence. Protocol specification reference: T-REC-Q.931-199805-I page 223, "Figure A.5/Q.931 -- Overview protocol control (network side) point-point (sheet 3 of 8)" (closes issue #17874) Reported by: nic_bellamy Patches: asterisk-1.4-r282537_no-progress-after-alerting.patch uploaded by nic bellamy (license 299) asterisk-1.6.2-r282537_no-progress-after-alerting.patch uploaded by nic bellamy (license 299) asterisk-trunk-r282537_no-progress-after-alerting.patch uploaded by nic bellamy (license 299) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@283049 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	c74e4511de	Merged revisions 282893 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r282893 | dvossel | 2010-08-19 16:03:24 -0500 (Thu, 19 Aug 2010) | 11 lines tos_sip option was not being set correctly When tos_sip is used, the tos of the sip socket is only set correctly if the socket binding changes on a reload. If the binding stays the same but the TOS changes, the new tos value would not take into effect. This patch fixes that. (closes issue #17712) Reported by: nickb ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@282894 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	c63a2b3fa7	fixes sip peer memory leaks in the peer_by_ip table ... (issue #17798) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@282890 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	b3bede4744	Merged revisions 277944 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r277944 | pabelanger | 2010-07-19 15:56:07 -0500 (Mon, 19 Jul 2010) | 16 lines Regression with T.38 negotiation Prior to 1.4.26.3 T.38 negotiation worked properly, in the case of the reporter. (issue #16852) Reported by: cfc (closes issue #16705) Reported by: mpiazzatnetbug Patches: issue16705_2.diff uploaded by ebroad (license 878) Tested by: vrban, ebroad, c0rnoTa, samdell3 Review: https://reviewboard.asterisk.org/r/754/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@282859 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	0faad5f43b	Merged revisions 282729 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r282729 | twilson | 2010-08-18 21:12:55 -0500 (Wed, 18 Aug 2010) | 2 lines Add some documentation about codec negotiation to sip.conf ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@282730 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	5338d346b0	fixes crash with notifycid ... (closes issue #17868) Reported by: francesco_r Patches: issue_17868.diff uploaded by dvossel (license 671) Tested by: francesco_r git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@282668 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	943034663a	Don't warn on callerid when completely text, instead of numeric with localdialplan prefixes. ... (closes issue #16770) Reported by: jamicque Patches: 20100413__issue16770.diff.txt uploaded by tilghman (license 14) 20100811__issue16770.diff.txt uploaded by tilghman (license 14) Tested by: jamicque git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@282607 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	93927c17ef	fixes no default transport for temp peer creation in chan_sip ... (closes issue #17829) Reported by: falves11 Patches: issue_17829.rev1.txt uploaded by russell (license 2) issue_17829.diff uploaded by dvossel (license 671) Tested by: falves11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@282576 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Leif Madsen	4c653e4cfa	Add information about creating sounds files using ... the sounds tools publically available so that others can create their own sounds prompts using the same tools we use to generate sounds releases. This allows people creating their own prompts to sound consistent with the prompts available from the open source project. SWP-595 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@282469 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	79d5f46e1c	Merged revisions 282430 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r282430 | twilson | 2010-08-16 12:06:37 -0500 (Mon, 16 Aug 2010) | 16 lines Send a SRCCHANGE indication when we masquerade Masquerading a channel means that the src of the audio is potentially changing, so send a SRCCHANGE so that RTP-based media streams can get a new SSRC generated to reflect the change. Original patch by addix (along with lots of testing--thanks!). (closes issue #17007) Reported by: addix Patches: 1001-reset-SSRC-original-channel.diff uploaded by addix (license 1006) srcchange.diff uploaded by twilson (license 396) Tested by: addix, twilson Review: https://reviewboard.asterisk.org/r/862/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@282467 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	a95d1e73f2	only do magic pickup when notifycid is enabled ... A new way of doing BLF pickup was introduced into 1.6.2. This feature adds a call-id value into the XML of a SIP_NOTIFY message sent to alert a subscriber that a device is ringing. This option should only be enabled when the new 'notifycid' option is set... but this was not the case. Instead the call-id value was included for every RINGING Notify message, which caused a regression for people who used other methods for call pickup. (closes issue #17633) Reported by: urosh Patches: chan_sip.txt uploaded by urosh (license ) blf_cid_issue.diff uploaded by dvossel (license 671) Tested by: dvossel, urosh, okrief, alecdavis git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@282235 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago