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2009-05-14Merged revisions 194496 via svnmerge from mmichelson1-4/+16
https://origsvn.digium.com/svn/asterisk/trunk ................ r194496 | mmichelson | 2009-05-14 17:20:51 -0500 (Thu, 14 May 2009) | 30 lines Merged revisions 194484 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r194484 | mmichelson | 2009-05-14 17:17:55 -0500 (Thu, 14 May 2009) | 24 lines Fix a race condition where a reinvite could trigger a 482 response. The loop detection/spiral detection code in chan_sip used the owner channel's state as a criterion for determining if the incoming INVITE is a looped request. The problem with this is that the INVITE-handling code happens in a different thread than the thread that marks the owner channel as being up. As a result, if a reinvite were to come in very quickly, say from another Asterisk on the same LAN, it was possible for the reinvite to arrive before the owner channel had been set to the up state. This patch corrects the problem by using the invitestate of the sip_pvt instead, since that can be guaranteed to be set correctly by the time the reinvite arrives. Since there is a switch statement further in the INVITE-handling code, the AST_STATE_RINGING state also checks the invitestate of the sip_pvt in case we should actually be treating the channel as if it were up already. (closes issue #12215) Reported by: jpyle Patches: 12215_confirmed.patch uploaded by mmichelson (license 60) Tested by: lmadsen ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@194510 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-12Merged revisions 193954 via svnmerge from mmichelson1-11/+6
https://origsvn.digium.com/svn/asterisk/trunk ........ r193954 | mmichelson | 2009-05-12 15:28:13 -0500 (Tue, 12 May 2009) | 18 lines Update spiral support in trunk and 1.6.X to match what is in 1.4. In 1.4, a SIP spiral is treated the same way as a call forward. This works much better than what is currently in trunk and 1.6.X. The code in trunk and 1.6.X did not create a new call to the recipient of the spiral, instead trying to continue the same call. In addition to just being plain wrong, this also had the side effect of only being able to spiral calls to other SIP channels. With this in place, as long as call forwards are honored, SIP spirals will work properly. This means that it will work for outbound calls made by the Queue, Dial, and Page applications. For originated calls and spool calls, however, the spiral will not work properly until a generic call forward mechanism is introduced into Asterisk. (relates to issue #13630) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@193962 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-11Merged revisions 193614 via svnmerge from rmudgett1-2/+2
https://origsvn.digium.com/svn/asterisk/trunk ................ r193614 | rmudgett | 2009-05-11 14:11:29 -0500 (Mon, 11 May 2009) | 19 lines Merged revisions 193613 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r193613 | rmudgett | 2009-05-11 14:09:00 -0500 (Mon, 11 May 2009) | 12 lines Sent wrong message to clear a call we started if the other end has not responed yet. In the state MISDN_CALLING (i.e. SETUP was sent but no answer has arrived yet), it is not allowed to clear the call with RELEASE_COMPLETE. It must be cleared with DISCONNECT. A RELEASE_COMPLETE is only allowed as an answer to a SETUP. (See Q.931 ch. 5.3.2, 5.3.2.a, 5.3.2.b) Patches: chan-misdn-ccstate7.patch uploaded by customer. JIRA ABE-1862 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@193617 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-08Merged revisions 193387 via svnmerge from dvossel1-0/+16
https://origsvn.digium.com/svn/asterisk/trunk ........ r193387 | dvossel | 2009-05-08 15:32:51 -0500 (Fri, 08 May 2009) | 7 lines TCP not matching valid peer. find_peer() does not find a valid peer when using pvt->recv as the sockaddr_in argument. Because of the way TCP works, the port number in pvt->recv is not what we're looking for at all. There is currently only one place that find_peer searches for a peer using the sockaddr_in argument. If the peer is not found after using pvt->recv (works for UDP since the port number will be correct), a temp sockaddr_in struct is made using the Contact header in the sip_request. This has the correct port number in it. Review: http://reviewboard.digium.com/r/236/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@193390 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-08Merged revisions 193263 via svnmerge from dvossel1-1/+3
https://origsvn.digium.com/svn/asterisk/trunk ................ r193263 | dvossel | 2009-05-08 09:52:19 -0500 (Fri, 08 May 2009) | 15 lines Merged revisions 193262 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r193262 | dvossel | 2009-05-08 09:51:09 -0500 (Fri, 08 May 2009) | 9 lines "misdn show config" segfaults asterisk, if no MSN lists (closes issue #14976) Reported by: alecdavis Patches: misdn_config.diff.txt uploaded by alecdavis (license 585) Tested by: alecdavis, FabienToune ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@193266 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-07Merged revisions 193077 via svnmerge from rmudgett1-8/+14
https://origsvn.digium.com/svn/asterisk/trunk ................ r193077 | rmudgett | 2009-05-07 17:24:04 -0500 (Thu, 07 May 2009) | 12 lines Merged revisions 193050 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r193050 | rmudgett | 2009-05-07 17:17:06 -0500 (Thu, 07 May 2009) | 5 lines Give a more helpful message when an incoming call's dialed extension does not match. Added the dialed extension and context to the chan_misdn messages warning that the dialed number cannot be matched in the dialplan. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@193080 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-07Merged revisions 192938 via svnmerge from tilghman1-6/+5
https://origsvn.digium.com/svn/asterisk/trunk ........ r192938 | tilghman | 2009-05-07 12:13:36 -0500 (Thu, 07 May 2009) | 6 lines Send DTMF frame before playing back audio. (closes issue #14858) Reported by: barryf Patches: 20090507__bug14858.diff.txt uploaded by tilghman (license 14) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@192942 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-07Merged revisions 192933 via svnmerge from tilghman1-0/+10
https://origsvn.digium.com/svn/asterisk/trunk ................ r192933 | tilghman | 2009-05-07 11:43:56 -0500 (Thu, 07 May 2009) | 17 lines Merged revisions 192932 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r192932 | tilghman | 2009-05-07 11:29:08 -0500 (Thu, 07 May 2009) | 10 lines Eliminate repetition of fullcontact during reconstruction. If the fullcontact field appears in both the sippeers and the sipregs table, then during reconstruction of the field, it will otherwise be doubled. (closes issue #14754) Reported by: Alexei Gradinari Patches: 20090506__bug14754.diff.txt uploaded by tilghman (license 14) Tested by: lmadsen ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@192936 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-06Merged revisions 190946 via svnmerge from mattf1-1/+3
https://origsvn.digium.com/svn/asterisk/trunk ........ r190946 | mattf | 2009-04-28 17:05:05 -0500 (Tue, 28 Apr 2009) | 1 line Make sure that we do not clear the down flag on the BRI during PTMP link transients. Also refix SS7 audio that the early media patch broke. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@192813 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-06Merged revisions 192808 via svnmerge from file1-0/+3
https://origsvn.digium.com/svn/asterisk/trunk ........ r192808 | file | 2009-05-06 14:38:51 -0300 (Wed, 06 May 2009) | 10 lines Fix a bug where a timer would be created but not acknowledged. This scenario crept up if chan_iax2 was loaded with no configuration file present. It would create a timer and tell it to go at an interval but the thread that normally acknowledges it would not be created because no configuration file was present. The timer will now be closed if no configuration file is present. (closes issue #15014) Reported by: madkins ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@192810 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-06Merged revisions 192634 via svnmerge from file1-2/+4
https://origsvn.digium.com/svn/asterisk/trunk ................ r192634 | file | 2009-05-06 10:34:35 -0300 (Wed, 06 May 2009) | 14 lines Merged revisions 192633 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r192633 | file | 2009-05-06 10:30:51 -0300 (Wed, 06 May 2009) | 7 lines Update some old logic to stop both begin and end DTMF frames from reaching the core if rfc2833 is not enabled. (closes issue #15036) Reported by: dimas Patches: v1-15036.patch uploaded by dimas (license 88) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@192637 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-05Merged revisions 192387 via svnmerge from file1-0/+5
https://origsvn.digium.com/svn/asterisk/trunk ........ r192387 | file | 2009-05-05 11:22:47 -0300 (Tue, 05 May 2009) | 10 lines Fix a bug with setting t38pt_udptl at the user or peer level. If an incoming call authenticated as a user or peer and t38pt_udptl was not set to yes in general then no UDPTL session would be present and any T38 related things would fail. This commit changes it so that if after authenticating T38 is enabled but no UDPTL session is present one will be created. (issue AST-215) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@192402 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-04Merged revisions 192214 via svnmerge from dvossel1-0/+5
https://origsvn.digium.com/svn/asterisk/trunk ................ r192214 | dvossel | 2009-05-04 17:44:51 -0500 (Mon, 04 May 2009) | 17 lines Merged revisions 192213 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r192213 | dvossel | 2009-05-04 17:37:31 -0500 (Mon, 04 May 2009) | 11 lines global mohinterpret setting is ignored mohinterpret and mohsuggest global variables were not copied over during build_users and build_peers. (closes issue #14728) Reported by: dimas Patches: v1-14728.patch uploaded by dimas (license 88) Tested by: dimas, dvossel ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@192217 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-01Merged revisions 191560 via svnmerge from tilghman1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r191560 | tilghman | 2009-05-01 15:01:21 -0500 (Fri, 01 May 2009) | 13 lines Merged revisions 191559 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r191559 | tilghman | 2009-05-01 15:00:23 -0500 (Fri, 01 May 2009) | 6 lines SIP Response 410 maps to cause code 22 (or 23), not 1. (closes issue #14993) Reported by: BigJimmy Patches: causepatch uploaded by BigJimmy (license 371) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@191563 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-01Merged revisions 191494 via svnmerge from tilghman1-3/+3
https://origsvn.digium.com/svn/asterisk/trunk ........ r191494 | tilghman | 2009-05-01 13:18:00 -0500 (Fri, 01 May 2009) | 4 lines Set debug message back to DEBUG level. (closes issue #15007) Reported by: hulber ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@191554 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-29Merged revisions 191219 via svnmerge from tilghman2-1/+7
https://origsvn.digium.com/svn/asterisk/trunk ........ r191219 | tilghman | 2009-04-29 18:06:56 -0500 (Wed, 29 Apr 2009) | 2 lines Make H.323 compile with FDLEAK detection code enabled ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@191224 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-23Merged revisions 190371 via svnmerge from russell1-1/+0
https://origsvn.digium.com/svn/asterisk/trunk ........ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@190383 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-23Merged revisions 190287 via svnmerge from file1-3/+0
https://origsvn.digium.com/svn/asterisk/trunk ................ r190287 | file | 2009-04-23 16:15:30 -0300 (Thu, 23 Apr 2009) | 13 lines Merged revisions 190286 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r190286 | file | 2009-04-23 16:13:18 -0300 (Thu, 23 Apr 2009) | 6 lines Fix a bug in chan_local glare hangup detection. If both sides of a Local channel were hung up at around the same time it was possible for one thread to destroy the local private structure and have the other thread immediately try to remove the already freed structure from the local channel list. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@190297 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-22Merged revisions 189993 via svnmerge from jpeeler3-24/+31
https://origsvn.digium.com/svn/asterisk/trunk ........ r189993 | jpeeler | 2009-04-22 14:23:49 -0500 (Wed, 22 Apr 2009) | 18 lines Make chan_h323 respect packetization settings and fix small reload issue. Previously, packetization settings were ignored and now they are not. A new config option 'autoframing' has been added to mirror the way chan_sip handles it. Turning on the autoframing option (available both as a global option or per peer) overrides the local settings with the remote packetization settings. Testing was performed with varying packetization levels with the following codecs: ulaw, alaw, gsm, and g729. Also, an unrelated config reload issue has been fixed in the case of the config file not changing. (closes issue #12415) Reported by: pj Patches: 2009012200_h323packetization.diff.txt uploaded by mvanbaak (license 7), modified by me ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@189997 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-22Merged revisions 189911 via svnmerge from tilghman1-3/+3
https://origsvn.digium.com/svn/asterisk/trunk ........ r189911 | tilghman | 2009-04-22 11:01:30 -0500 (Wed, 22 Apr 2009) | 7 lines Do not continue to receive DTMF, when the channel is hungup and about to be destroyed. (closes issue #14858) Reported by: barryf Patches: 20090421__bug14858.diff.txt uploaded by tilghman (license 14) Tested by: barryf ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@189914 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-21Merged revisions 189771 via svnmerge from dvossel1-7/+14
https://origsvn.digium.com/svn/asterisk/trunk ........ r189771 | dvossel | 2009-04-21 15:28:37 -0500 (Tue, 21 Apr 2009) | 11 lines Fixes segfault when switching UDP to TCP in sip.conf after reload. If transport in sip.conf is switched from UDP to TCP, Asterisk segfaults right after issuing a sip reload. The problem is the socket type is changed to TCP but the fd may still be present for UDP. Later, when the TCP session should be created or set using an existing one, it isn't because the old file descriptor is still present. Now every time transport is changed during a sip.conf reload, the file descriptor is set to -1, signifying it must be created or found. (closes issue #14727) Reported by: pj Tested by: dvossel Review: http://reviewboard.digium.com/r/229/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@189775 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-20Merged revisions 189350 via svnmerge from file1-1/+2
https://origsvn.digium.com/svn/asterisk/trunk ........ r189350 | file | 2009-04-20 14:05:15 -0300 (Mon, 20 Apr 2009) | 10 lines Fix a bug with non-UDP connections that caused dialogs to not get freed. This issue crept up because of a reference count issue on non-UDP based dialogs. The dialog reference count was increased when transmitting a packet reliably but never decreased. This caused the dialog structure to hang around despite being unlinked from the dialogs container. (closes issue #14919) Reported by: vrban ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@189353 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-18Merged revisions 188647 via svnmerge from dvossel1-0/+12
https://origsvn.digium.com/svn/asterisk/trunk ................ r188647 | dvossel | 2009-04-15 17:10:04 -0500 (Wed, 15 Apr 2009) | 18 lines Merged revisions 188646 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r188646 | dvossel | 2009-04-15 17:08:40 -0500 (Wed, 15 Apr 2009) | 12 lines National prefix inserted even when caller ID not available 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 callers number even if its restricted (empty) causing the caller id to be the national prefix rather than blank. (closes issue #13207) Reported by: shawkris Patches: national_prefix.diff uploaded by dvossel (license 671) Review: http://reviewboard.digium.com/r/220/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@189208 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-18Merged revisions 189204 via svnmerge from dvossel1-51/+63
https://origsvn.digium.com/svn/asterisk/trunk ................ r189204 | dvossel | 2009-04-17 20:28:45 -0500 (Fri, 17 Apr 2009) | 18 lines Merged revisions 189203 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r189203 | dvossel | 2009-04-17 20:27:19 -0500 (Fri, 17 Apr 2009) | 12 lines Fixed autologoff in agents.conf not working when agent logs in via AgentLogin app An agent logs in by calling an extension that calls the AgentLogin app. In agents.conf ackcall=always is set, so when they get a call they have the choice to either acknowledge it or ignore it. autologoff=10 is set as well, so if the agent ignores the call over 10sec one may assume that the agent should be logged out (and in this case hungup on as well), but this was not happening. (closes issue #14091) Reported by: evandro Patches: autologoff.diff uploaded by dvossel (license 671) Review: http://reviewboard.digium.com/r/225/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@189207 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-17Merged revisions 189137 via svnmerge from rmudgett2-3/+7
https://origsvn.digium.com/svn/asterisk/trunk ................ r189137 | rmudgett | 2009-04-17 16:48:10 -0500 (Fri, 17 Apr 2009) | 17 lines Merged revisions 188833,189134 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r188833 | rmudgett | 2009-04-16 16:37:58 -0500 (Thu, 16 Apr 2009) | 4 lines Only disable mISDN DSP if Asterisk DSP is enabled. Leave jitter setting alone. JIRA ABE-1835 ........ r189134 | rmudgett | 2009-04-17 16:27:55 -0500 (Fri, 17 Apr 2009) | 4 lines Modifed/added some debug messages. JIRA ABE-1835 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@189140 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-17Merged revisions 189097 via svnmerge from mmichelson1-5/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r189097 | mmichelson | 2009-04-17 15:20:23 -0500 (Fri, 17 Apr 2009) | 13 lines Prevent a crash when SIP blonde transferring an unbridged call. If one attempts to use the attended transfer button on a SIP phone to transfer an unbridged call (such as a call to an IVR) but hangs up while the target of the transfer is still ringing, we need to not crash. The problem was that ast_hangup was called from outside the channel thread. AST-211 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@189105 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-17Merged revisions 188947 via svnmerge from file1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r188947 | file | 2009-04-17 11:44:56 -0300 (Fri, 17 Apr 2009) | 22 lines Merged revisions 188946 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r188946 | file | 2009-04-17 11:41:25 -0300 (Fri, 17 Apr 2009) | 15 lines Fix a bug where a value used to create the channel name was bogus. This commit fixes the scenario where an incoming call is authenticated using a peer entry. Previously the channel name was created using either the username setting from the sip.conf entry or the IP address that the call came from. Now the channel name will be created using the peer name itself. This commit will not change the way the channel name is generated for users or friends. (closes issue #14256) Reported by: Nick_Lewis Patches: chan_sip.c-chname.patch uploaded by Nick (license 657) Tested by: Nick_Lewis, file ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@188950 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-17Merged revisions 188938 via svnmerge from file1-0/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r188938 | file | 2009-04-17 11:26:53 -0300 (Fri, 17 Apr 2009) | 11 lines Merged revisions 188937 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r188937 | file | 2009-04-17 11:25:57 -0300 (Fri, 17 Apr 2009) | 4 lines Fix a situation where the DAHDI channel private structure lock was not unlocked when it should have been. (issue AST-210) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@188941 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-16Merged revisions 188836 via svnmerge from tilghman1-5/+8
https://origsvn.digium.com/svn/asterisk/trunk ................ r188836 | tilghman | 2009-04-16 16:57:37 -0500 (Thu, 16 Apr 2009) | 14 lines Merged revisions 188835 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r188835 | tilghman | 2009-04-16 16:41:13 -0500 (Thu, 16 Apr 2009) | 7 lines Only update realtime, if global option rtupdate != false (closes issue #14885) Reported by: deepesh Patches: 20090413__bug14885.diff.txt uploaded by tilghman (license 14) Tested by: deepesh ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@188839 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-14Merged revisions 188247 via svnmerge from file1-5/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r188247 | file | 2009-04-14 10:14:21 -0300 (Tue, 14 Apr 2009) | 7 lines Fix a bug with the change I made yesterday to outbound proxy support. Per discussion with oej on IRC we need the actual IP address, not the outbound proxy IP address, in the sa field. Upon further inspection this should make the behaviour of all other uses of the outbound proxy in the code. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@188259 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-13Merged revisions 188067 via svnmerge from file1-0/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r188067 | file | 2009-04-13 13:28:06 -0300 (Mon, 13 Apr 2009) | 10 lines Fix a bug where using an outbound proxy would cause the local address to be 127.0.0.1. Copy the outbound proxy IP address into the SIP dialog structure as the IP address we will be sending to. This has to be done because the logic that determines what local IP address to use in the SIP messages is not aware of an outbound proxy being in place. It only knows what IP address we are sending to. (closes issue #12006) Reported by: mnicholson ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@188070 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-10Merged revisions 187906 via svnmerge from jpeeler1-6/+7
https://origsvn.digium.com/svn/asterisk/trunk ........ r187906 | jpeeler | 2009-04-10 15:26:46 -0500 (Fri, 10 Apr 2009) | 12 lines Fix module embedding for chan_h323. Include libchanh323.a in the modules.link file so that all the symbols can be resolved at link time. (closes issue #11966) Reported by: dome Patches: issue_11966.patch uploaded by kpfleming (license 421) Tested by: jpeeler ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@187916 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-10Merged revisions 187674 via svnmerge from tilghman1-2/+2
https://origsvn.digium.com/svn/asterisk/trunk ........ r187674 | tilghman | 2009-04-10 10:59:40 -0500 (Fri, 10 Apr 2009) | 4 lines Ensure pvt is not NULL before dereferencing it. (closes issue #14784) Reported by: pj ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@187679 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-09Merge revision 187488 from trunk.mmichelson1-6/+19
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@187564 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-09Merged revisions 185846 via svnmerge from dvossel1-13/+30
https://origsvn.digium.com/svn/asterisk/trunk ................ r185846 | dvossel | 2009-04-01 14:03:32 -0500 (Wed, 01 Apr 2009) | 16 lines Merged revisions 185845 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185845 | dvossel | 2009-04-01 14:02:00 -0500 (Wed, 01 Apr 2009) | 10 lines Fixes issue with dropped calles due to re-Invite glare and re-Invites never executing after a 491 Acknowledgement for 491 responses were never being processed because it didn't match our pending invite's seqno. Since the ACK was never processed, the 491 frame would continue to be retransmitted until eventually the call was dropped due to max retries. Now during a pending invite, if we receive another invite, we send an 491 and hold on to that glare invite's seqno in the "glareinvite" variable for that sip_pvt struct. When ACK's are received, we first check to see if it is in response to our pending invite, if not we check to see if it is in response to a glare invite. In this case, it is in response to the glare invite and must be dealt with or the call is dropped. I've changed the wait time for resending the re-Invite after receving a 491 response to comply with RFC 3261. Before this patch the scheduled re-Invite would only change a flag indicating that the re-Invite should be sent out, now it actually sends it out as well. (closes issue #12013) Reported by: alx Review: http://reviewboard.digium.com/r/213/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@187531 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-09Merged revisions 187381 via svnmerge from tilghman1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r187381 | tilghman | 2009-04-09 12:20:49 -0500 (Thu, 09 Apr 2009) | 4 lines Allow '/' in username portion of register; this is a regression. (closes issue #14668) Reported by: Netview ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@187391 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-09Merged revisions 187363 via svnmerge from tilghman1-1/+3
https://origsvn.digium.com/svn/asterisk/trunk ................ r187363 | tilghman | 2009-04-09 11:39:43 -0500 (Thu, 09 Apr 2009) | 10 lines Merged revisions 187362 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r187362 | tilghman | 2009-04-09 11:38:37 -0500 (Thu, 09 Apr 2009) | 3 lines Permit zero-length text messages in SIP. (Related to an issue posted to the -users list, subject "AEL2, BASE64_DECODE and hexadecimal") ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@187366 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-08Merged revisions 186928 via svnmerge from russell1-28/+13
https://origsvn.digium.com/svn/asterisk/trunk ........ r186928 | russell | 2009-04-08 07:35:57 -0500 (Wed, 08 Apr 2009) | 13 lines Update some comments and resolve potential memory corruption in chan_sip. While browsing chan_sip the other day, I noticed this dangerous code in dialog_needdestroy(). This function is an ao2_callback. It is absolutely _not_ okay to unlock the container from within this function. It's also not clear why it was useful. Given that it could cause memory corruption, I have removed it. There was also a TODO comment left describing a potential implementation of an improvement to the needdestroy handling. I'm not convinced that what was described is the best choice here, so I have briefly described the way that this function is used today that could be improved. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@186929 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-08Merged revisions 186899 via svnmerge from tilghman1-0/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r186899 | tilghman | 2009-04-08 00:06:22 -0500 (Wed, 08 Apr 2009) | 2 lines Add lastms to the require API call. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@186901 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-08Merged revisions 186837 via svnmerge from mmichelson1-3/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r186837 | mmichelson | 2009-04-07 19:01:49 -0500 (Tue, 07 Apr 2009) | 7 lines Fix bad merge from fix for issue 13867. (closes issue #14686) Reported by: davidw ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@186840 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-03Merged revisions 186461 via svnmerge from kpfleming1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r186461 | kpfleming | 2009-04-03 15:20:01 -0500 (Fri, 03 Apr 2009) | 11 lines Merged revisions 186458 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186458 | kpfleming | 2009-04-03 15:19:20 -0500 (Fri, 03 Apr 2009) | 5 lines Fix a bug where DAHDI/Zaptel channels would not properly switch formats when requested Don't offer AST_FORMAT_SLINEAR on DAHDI/Zaptel channels... while it could provide a slight performance benefit, the translation core in Asterisk has some flaws when a channel driver offers multiple raw formats. this fix is much simpler than fixing the translation core to solve that issue (although that will be done later). ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@186469 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-02Merged revisions 186101 via svnmerge from kpfleming1-0/+2
https://origsvn.digium.com/svn/asterisk/trunk ................ r186101 | kpfleming | 2009-04-02 12:26:07 -0500 (Thu, 02 Apr 2009) | 9 lines Merged revisions 186081 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186081 | kpfleming | 2009-04-02 12:21:29 -0500 (Thu, 02 Apr 2009) | 3 lines ensure that the buffer passed to DAHDI_SET_BUFINFO is fully initialized ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@186111 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-02Merged revisions 186060 via svnmerge from tilghman1-8/+107
https://origsvn.digium.com/svn/asterisk/trunk ................ r186060 | tilghman | 2009-04-02 12:10:28 -0500 (Thu, 02 Apr 2009) | 16 lines Merged revisions 186059 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r186059 | tilghman | 2009-04-02 12:09:13 -0500 (Thu, 02 Apr 2009) | 9 lines Merged revisions 186056 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r186056 | tilghman | 2009-04-02 12:02:18 -0500 (Thu, 02 Apr 2009) | 2 lines Fix for AST-2009-003 ........ ................ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@186063 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-02Merged revisions 185953 via svnmerge from kpfleming1-11/+20
https://origsvn.digium.com/svn/asterisk/trunk ................ r185953 | kpfleming | 2009-04-02 08:51:44 -0500 (Thu, 02 Apr 2009) | 11 lines Merged revisions 185952 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185952 | kpfleming | 2009-04-02 08:43:43 -0500 (Thu, 02 Apr 2009) | 5 lines the DAHDI_GETCONF, DAHDI_SETCONF and DAHDI_GET_PARAMS ioctls were recently corrected to show that they do, in fact, read data from userspace as part of their work. due to this fix, valgrind now reports a number of cases where chan_dahdi passed an uninitialized (or partially) buffer to these ioctls, which could lead to unexpected behavior. this patch corrects chan_dahdi to ensure that buffers passed to these ioctls are always fully initialized. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@185957 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-31Merged revisions 185363 via svnmerge from dbrooks1-7/+7
https://origsvn.digium.com/svn/asterisk/trunk ................ r185363 | dbrooks | 2009-03-31 11:46:57 -0500 (Tue, 31 Mar 2009) | 44 lines Merged revisions 185362 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185362 | dbrooks | 2009-03-31 11:37:12 -0500 (Tue, 31 Mar 2009) | 35 lines Fix incorrect parsing in chan_gtalk when xmpp contains extra whitespaces To drill into the xmpp to find the capabilities between channels, chan_gtalk calls iks_child() and iks_next(). iks_child() and iks_next() are functions in the iksemel xml parsing library that traverse xml nodes. The bug here is that both iks_child() and iks_next() will return the next iks_struct node *regardless* of type. chan_gtalk expects the next node to be of type IKS_TAG, which in most cases, it is, but in this case (a call being made from the Empathy IM client), there exists iks_struct nodes which are not IKS_TAG data (they are extraneous whitespaces), and chan_gtalk doesn't handle that case, so capabilities don't match, and a call cannot be made. iks_first_tag() and iks_next_tag(), on the other hand, will not return the very next iks_struct, but will check to see if the next iks_struct is of type IKS_TAG. If it isn't, it will be skipped, and the next struct of type IKS_TAG it finds will be returned. This assures that chan_gtalk will find the iks_struct it is looking for. This fix simply changes all calls to iks_child() and iks_next() to become calls to iks_first_tag() and iks_next_tag(), which resolves the capability matching. The following is a payload listing from Empathy, which, due to the extraneous whitespace, will not be parsed correctly by iksemel: <iq from='dbrooksjab@235-22-24-10/Telepathy' to='astjab@235-22-24-10/asterisk' type='set' id='542757715704'> <session xmlns='http://www.google.com/session' initiator='dbrooksjab@235-22-24-10/Telepathy' type='initiate' id='1837267342'> <description xmlns='http://www.google.com/session/phone'> <payload-type clockrate='16000' name='speex' id='96'/> <payload-type clockrate='8000' name='PCMA' id='8'/> <payload-type clockrate='8000' name='PCMU' id='0'/> <payload-type clockrate='90000' name='MPA' id='97'/> <payload-type clockrate='16000' name='SIREN' id='98'/> <payload-type clockrate='8000' name='telephone-event' id='99'/> </description> </session> </iq> Review: http://reviewboard.digium.com/r/181/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@185428 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-30Merged revisions 185123 via svnmerge from rmudgett1-4/+5
https://origsvn.digium.com/svn/asterisk/trunk ................ r185123 | rmudgett | 2009-03-30 15:42:14 -0500 (Mon, 30 Mar 2009) | 9 lines Merged revisions 185121 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185121 | rmudgett | 2009-03-30 15:40:11 -0500 (Mon, 30 Mar 2009) | 1 line Update the channel allocation method documentation. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@185129 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-30Merged revisions 185122 via svnmerge from rmudgett1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r185122 | rmudgett | 2009-03-30 15:41:24 -0500 (Mon, 30 Mar 2009) | 26 lines Merged revisions 185120 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185120 | rmudgett | 2009-03-30 15:38:11 -0500 (Mon, 30 Mar 2009) | 19 lines Make chan_misdn BRI TE side normally defer channel selection to the NT side. Channel allocation collisions are not handled by chan_misdn very well. This patch simply avoids the problem for BRI only. For PRI, allocation collisions are still possible but less likely since there are simply more channels available and each end could use a different allocation strategy. misdn.conf options available: te_choose_channel - Use to force the TE side to allocate channels. method - Specify the channel allocation strategy. (closes issue #13488) Reported by: Christian_Pinedo Patches: isdn_lib.patch.txt uploaded by crich Tested by: crich, siepkes, festr ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@185128 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-30Merged revisions 184948 via svnmerge from file1-268/+235
https://origsvn.digium.com/svn/asterisk/trunk ................ r184948 | file | 2009-03-30 11:37:47 -0300 (Mon, 30 Mar 2009) | 21 lines Merged revisions 184947 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r184947 | file | 2009-03-30 11:35:47 -0300 (Mon, 30 Mar 2009) | 14 lines Improve our handling of T38 in the initial INVITE from a device. We now answer with matching media streams to what is requested. If an INVITE is received with both a T38 and RTP media stream this means we answer with both. For any outgoing calls created as a result of this inbound one no T38 is requested in the initial INVITE. Instead if we start receiving udptl packets we trigger a reinvite on the outbound side. (closes issue #12437) Reported by: marsosa Tested by: pinga-fogo, okrief, file, afu Review: http://reviewboard.digium.com/r/208/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@184951 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-30Merged revisions 184910 via svnmerge from russell1-0/+2
https://origsvn.digium.com/svn/asterisk/trunk ........ r184910 | russell | 2009-03-30 08:55:44 -0500 (Mon, 30 Mar 2009) | 4 lines Fix build error when chan_h323 is not being built. (reported by cai1982 in #asterisk-dev) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@184913 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-29Merged revisions 184838 via svnmerge from russell1-15/+10
https://origsvn.digium.com/svn/asterisk/trunk ........ r184838 | russell | 2009-03-29 00:32:04 -0500 (Sun, 29 Mar 2009) | 8 lines Simplify chan_h323 build to not require a second run of "make". (closes issue #14715) Reported by: jthurman Patches: h323-makefile-1.6.2.0-beta1.patch uploaded by jthurman (license 614) Tested by: tzafrir, russell ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.2@184839 f38db490-d61c-443f-a65b-d21fe96a405b
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