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2010-12-09Merged revisions 297965 via svnmerge from Terry Wilson
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r297965 | twilson | 2010-12-09 16:18:19 -0600 (Thu, 09 Dec 2010) | 28 lines Merged revisions 297960 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r297960 | twilson | 2010-12-09 16:10:31 -0600 (Thu, 09 Dec 2010) | 21 lines Merged revisions 297959 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r297959 | twilson | 2010-12-09 16:00:30 -0600 (Thu, 09 Dec 2010) | 14 lines Ignore spurious REGISTER requests If a REGISTER request with a Call-ID matching an existing transaction is received it was possible that the REGISTER request would overwrite the initreq of the private structure. This info is used to generate messages for other responses in the transaction. This patch ignores REGISTER requests that match non-REGISTER transactions. (closes issue #18051) Reported by: eeman Tested by: twilson Review: https://reviewboard.asterisk.org/r/1050/ ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@297972 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-12-06Merged revisions 297607 via svnmerge from Jeff Peeler
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r297607 | jpeeler | 2010-12-06 16:06:37 -0600 (Mon, 06 Dec 2010) | 25 lines Merged revisions 297605 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r297605 | jpeeler | 2010-12-06 16:03:04 -0600 (Mon, 06 Dec 2010) | 18 lines Merged revisions 297603 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r297603 | jpeeler | 2010-12-06 15:57:15 -0600 (Mon, 06 Dec 2010) | 12 lines Improve handling of REGISTER requests with multiple contact headers. The changes here attempt to more strictly follow RFC 3261 section 10.3. Basically the following will now cause a 400 Bad Response to be returned, if: - multiple Contact headers are present with one set to expire all bindings ("*") - wildcard parameter is specified for Contact without Expires header or Expires header is not set to zero. ABE-2442 ABE-2443 ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@297608 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-12-01Merged revisions 297075 via svnmerge from Jeff Peeler
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r297075 | jpeeler | 2010-12-01 11:53:13 -0600 (Wed, 01 Dec 2010) | 37 lines Merged revisions 297073 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r297073 | jpeeler | 2010-12-01 11:52:46 -0600 (Wed, 01 Dec 2010) | 30 lines Merged revisions 297072 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r297072 | jpeeler | 2010-12-01 11:50:09 -0600 (Wed, 01 Dec 2010) | 23 lines Fix not stopping MOH when transfered local channel queue member is answered. The problem here is only present when local channels are used with the MOH passthru option as well as no optimization (/nm). I will describe the slightly bizarre scenario that was used to test, where phones B and C are queue members: Phone A dials into a queue with two members using local channels and the above options. Phone B answers. Phone A blind transfers phone B into the same queue. Phone A hangs up. Phone C answers, but phone B didn't stop playing MOH. In this scenario, the unhold frame that should have gotten to phone B never arrived due to the masquerade from the blind transfer. This is usually fine since app_queue manages the starting and stopping of MOH. However, with the passthrough option enabled when app_queue attempts to stop MOH it tries to do so on the local channel rather than the real channel. The easiest solution was to just make sure to send an unhold frame during the transfer since it wouldn't make sense to have MOH playing after a transfer anyway. This only modifies SIP transfers, but the other transfers did not seem to be a problem. If DTMF based transfers were a problem it might be okay to add ast_moh_stop to finishup, but I didn't want to have to add that unless required. ABE-2624 ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@297076 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-29Merged revisions 296628 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r296628 | russell | 2010-11-29 15:26:44 -0600 (Mon, 29 Nov 2010) | 6 lines Complete some error handling in transmit_publish() in chan_sip.c. This error handling block caught my eye. It was missing a couple of things, but it should be safe now. Thanks to mmichelson for the quick peer review on IRC. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@296630 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-26Merged revisions 296352 via svnmerge from Brad Watkins
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r296352 | marquis | 2010-11-26 13:19:02 -0500 (Fri, 26 Nov 2010) | 12 lines Fix reloading of peer when a user is requested. Prevent peer reloading from causing multiple MWI subscriptions to be created when using realtime. This had the effect of sending one NOTIFY for every time a sip peer made a call, in one case eventually overwhelming the phone and causing it to reboot. (closes issue #18342) Reported by: nivek Patches: issue0018342p1.patch uploaded by nivek (license 636) Tested by: nivek Review: https://reviewboard.asterisk.org/r/1029/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@296353 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-19Merged revisions 295673 via svnmerge from Terry Wilson
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r295673 | twilson | 2010-11-19 14:06:10 -0800 (Fri, 19 Nov 2010) | 22 lines Merged revisions 295672 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r295672 | twilson | 2010-11-19 13:55:48 -0800 (Fri, 19 Nov 2010) | 15 lines Merged revisions 295628 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r295628 | twilson | 2010-11-19 12:53:36 -0800 (Fri, 19 Nov 2010) | 8 lines Discard responses with more than one Via This is not a perfect solution as headers that are joined via commas are not detected. This is a parsing issue that to fix "correctly" would necessitate a new SIP parser. Review: https://reviewboard.asterisk.org/r/1019/ ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@295674 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-11Merged revisions 294734 via svnmerge from Jeff Peeler
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r294734 | jpeeler | 2010-11-11 15:58:25 -0600 (Thu, 11 Nov 2010) | 32 lines Merged revisions 294733 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r294733 | jpeeler | 2010-11-11 15:57:22 -0600 (Thu, 11 Nov 2010) | 25 lines Merged revisions 294688 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r294688 | jpeeler | 2010-11-11 15:12:27 -0600 (Thu, 11 Nov 2010) | 18 lines Fix problem with qualify option packets for realtime peers never stopping. The option packets not only never stopped, but if a realtime peer was not in the peer list multiple options dialogs could accumulate over time. This scenario has the potential to progress to the point of saturating a link just from options packets. The fix was to ensure that the poke scheduler checks to see if a peer is in the peer list before continuing to poke. The reason a peer must be in the peer list to be able to properly manage an options dialog is because otherwise the call pointer is lost when the peer is regenerated from the database, which is how existing qualify dialogs are detected. (closes issue #16382) (closes issue #17779) Reported by: lftsy Patches: bug16382-3.patch uploaded by jpeeler (license 325) Tested by: zerohalo ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294735 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-08Merged revisions 294243 via svnmerge from Matthew Nicholson
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r294243 | mnicholson | 2010-11-08 14:56:30 -0600 (Mon, 08 Nov 2010) | 15 lines Merged revisions 294242 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r294242 | mnicholson | 2010-11-08 14:50:21 -0600 (Mon, 08 Nov 2010) | 8 lines Go off hold when we get an empty reinvite telling us to. (closes issue 0014448) Reported by: frawd (closes issue #17878) Reported by: frawd ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294244 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-05Merged revisions 294084 via svnmerge from Brett Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294084 | bbryant | 2010-11-05 18:03:11 -0400 (Fri, 05 Nov 2010) | 9 lines Fixed deadlock avoidance issues while locking channel when adding the Max-Forwards header to a request. (closes issue #17949) (closes issue #18200) Reported by: bwg Review: https://reviewboard.asterisk.org/r/997/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294086 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-05Perform proper handling of forked outbound INVITE requests.David Vossel
RFC3261 section 12 about dialog creation says an INVITE transaction results in an established dialog once it receives the 200 OK response. It is possible to receive multiple differing 200 OK responses for a single outbound INVITE Request, and this should result in establishing multiple dialogs. This patch allows for all differing 200 OK responses to an INVITE request to establish a separate dialog, but only the first dialog is kept. All other resulting dialogs from the initial request are immediately ACKed and then immediately terminated with a BYE request. Review: https://reviewboard.asterisk.org/r/946/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294083 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-05Merged revisions 293924 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r293924 | dvossel | 2010-11-04 16:39:51 -0500 (Thu, 04 Nov 2010) | 4 lines Fixes ringback tone on sip semi-attended transfer. ABE-2168 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294046 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-04Merged revisions 293887 via svnmerge from Paul Belanger
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r293887 | pabelanger | 2010-11-04 09:27:54 -0400 (Thu, 04 Nov 2010) | 8 lines Do not output port in IPaddress for AMI sippeers. (closes issue #18248) Reported by: orn Patches: ami_sippeers.patch uploaded by pabelanger (license 224) Tested by: orn ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@293888 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-03Merged revisions 293803 via svnmerge from Terry Wilson
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r293803 | twilson | 2010-11-03 11:05:14 -0700 (Wed, 03 Nov 2010) | 25 lines Avoid valgrind warnings for ast_rtp_instance_get_xxx_address The documentation for ast_rtp_instance_get_(local/remote)_address stated that they returned 0 for success and -1 on failure. Instead, they returned 0 if the address structure passed in was already equivalent to the address instance local/remote address or 1 otherwise. 90% of the calls to these functions completely ignored the return address and passed in an uninitialized struct, which would make valgrind complain even though the operation was technically safe. This patch fixes the documentation and converts the get_xxx_address functions to void since all they really do is copy the address and cannot fail. Additionally two new functions (ast_rtp_instance_get_and_cmp_(local/remote)_address) are created for the 3 times where the return value was actually checked. The get_and_cmp_local_address function is currently unused, but exists for the sake of symmetry. The only functional change as a result of this change is that we will not do an ast_sockaddr_cmp() on (mostly uninitialized) addresses before doing the ast_sockaddr_copy() in the get_*_address functions. So, even though it is an API change, it shouldn't have a noticeable change in behavior. Review: https://reviewboard.asterisk.org/r/995/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@293809 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-11-02Merged revisions 293724 via svnmerge from Jeff Peeler
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r293724 | jpeeler | 2010-11-02 18:09:06 -0500 (Tue, 02 Nov 2010) | 22 lines Merged revisions 293723 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r293723 | jpeeler | 2010-11-02 18:07:13 -0500 (Tue, 02 Nov 2010) | 15 lines Merged revisions 293722 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r293722 | jpeeler | 2010-11-02 18:02:51 -0500 (Tue, 02 Nov 2010) | 8 lines Add enabled/disabled information for rtautoclear sip show settings output. When setting to zero/"no", the numeric default was shown making it not obvious the disabled setting was respected. (closes issue #18123) Reported by: zerohalo ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@293725 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-10-29Merged revisions 293305 via svnmerge from Jeff Peeler
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r293305 | jpeeler | 2010-10-29 16:48:38 -0500 (Fri, 29 Oct 2010) | 9 lines Modify sip_setoption to not complain about unknown options. This now behaves just like the other setoption callbacks. For the curious the offending option for the reporter was AST_OPTION_CHANNEL_WRITE which was getting passed due to a fix for chan_local in 286189. (closes issue #17985) Reported by: globalnetinc ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@293306 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-10-22Merged revisions 292787 via svnmerge from Leif Madsen
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r292787 | lmadsen | 2010-10-22 16:28:43 -0500 (Fri, 22 Oct 2010) | 21 lines Merged revisions 292786 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r292786 | lmadsen | 2010-10-22 16:16:12 -0500 (Fri, 22 Oct 2010) | 13 lines Update the LDIF file for LDAP. The LDIF file asterisk.ldif was quite a bit out of date from the asterisk.ldap-schema file, so I've now updated that to be in sync. The asterisk.ldif file being out of sync was a problem on my systems where I was doing an ldapadd to import the schema into the LDAP database, and the existing file would cause problems and ERROR messages when registering. Additional documention has been added based on feedback in the issue I'm closing. (closes issue #13861) Reported by: scramatte Patches: ldap-update.txt uploaded by lmadsen (license 10) Tested by: lmadsen, jcovert, suretec, rgenthner ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@292788 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-10-19Merged revisions 292309 via svnmerge from Terry Wilson
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r292309 | twilson | 2010-10-19 12:27:32 -0700 (Tue, 19 Oct 2010) | 10 lines Add sip show peer info about crypto and remove dated comment This patch adds information about the encryption setting to 'sip show peers' and removes an out-of-date comment from res_srtp.c and instead directs users to the proper documentation. (closes issue #18140) Reported by: chodorenko ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@292310 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-10-15Merged revisions 291942 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r291942 | dvossel | 2010-10-15 15:12:04 -0500 (Fri, 15 Oct 2010) | 8 lines Fixes peer's host port information being lost on sip reload. (closes issue #18135) Reported by: lmadsen Patches: crazy_ports_v2.diff uploaded by dvossel (license 671) Tested by: lmadsen ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@291943 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-10-14Merged revisions 291758 via svnmerge from Paul Belanger
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r291758 | pabelanger | 2010-10-14 11:15:12 -0400 (Thu, 14 Oct 2010) | 11 lines Add the ability for ast_find_ourip to return IPv4, IPv6 or both. While testing chan_gtalk I noticed jabber was using my IPv6 address and not IPv4. When using bindaddr=0.0.0.0 it is possible for ast_find_ourip() to return both IPv6 and IPv4 results. Adding a family parameter gives you the ablility to choose. Since jabber/gtalk/h323 do not support IPv6, we should only return IPv4 results. Review: https://reviewboard.asterisk.org/r/973/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@291760 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-10-13Merged revisions 291394 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r291394 | russell | 2010-10-13 10:46:39 -0500 (Wed, 13 Oct 2010) | 20 lines Merged revisions 291393 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r291393 | russell | 2010-10-13 10:29:21 -0500 (Wed, 13 Oct 2010) | 13 lines Merged revisions 291392 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r291392 | russell | 2010-10-13 10:23:19 -0500 (Wed, 13 Oct 2010) | 6 lines Lock pvt so pvt->owner can't disappear when queueing up a frame. This fixes a crash due to a hangup race condition. ABE-2601 ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@291395 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-10-11Add todo comment about handle_incoming() calling assumption.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@291115 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-10-11Merged revisions 291112-291113 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r291112 | rmudgett | 2010-10-11 13:48:15 -0500 (Mon, 11 Oct 2010) | 20 lines Merged revisions 291110-291111 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r291110 | rmudgett | 2010-10-11 13:34:22 -0500 (Mon, 11 Oct 2010) | 9 lines Merged revisions 291109 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r291109 | rmudgett | 2010-10-11 13:29:43 -0500 (Mon, 11 Oct 2010) | 1 line Add missing unlock to an exception condition in reload_config(). ........ ................ r291111 | rmudgett | 2010-10-11 13:39:06 -0500 (Mon, 11 Oct 2010) | 1 line Make exit from handle_request_do() consistent. ................ ................ r291113 | rmudgett | 2010-10-11 13:51:13 -0500 (Mon, 11 Oct 2010) | 1 line Move declaration closer to where now used. ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@291114 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-10-02Merged revisions 289840 via svnmerge from Jeff Peeler
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r289840 | jpeeler | 2010-10-01 21:43:45 -0500 (Fri, 01 Oct 2010) | 29 lines Merged revisions 289798 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r289798 | jpeeler | 2010-10-01 18:01:31 -0500 (Fri, 01 Oct 2010) | 22 lines Merged revisions 289797 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r289797 | jpeeler | 2010-10-01 17:58:38 -0500 (Fri, 01 Oct 2010) | 15 lines Change RFC2833 DTMF event duration on end to report actual elapsed time. The scenario here is with a non P2P early media session. The reported time length of DTMF presses are coming up short when sending to the remote side. Currently the event duration is a running total that is incremented when sending continuation packets. These continuation packets are only triggered upon incoming media from the remote side, which means that the running total probably is not going to end up matching the actual length of time Asterisk received DTMF. This patch changes the end event duration to be lengthened if it is detected that the end event is going to come up short. Review: https://reviewboard.asterisk.org/r/957/ ABE-2476 ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@289841 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-10-01Merged revisions 289701 via svnmerge from Jeff Peeler
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r289701 | jpeeler | 2010-10-01 11:22:19 -0500 (Fri, 01 Oct 2010) | 28 lines Merged revisions 289700 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r289700 | jpeeler | 2010-10-01 11:21:04 -0500 (Fri, 01 Oct 2010) | 21 lines Merged revisions 289699 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r289699 | jpeeler | 2010-10-01 11:20:00 -0500 (Fri, 01 Oct 2010) | 14 lines Ensure user portion of SIP URI matches dialplan when using encoded characters. This commit takes a simliar approach to 288112 and checks the dialplan to determine the proper action for an incoming contact header as to whether or not it should be decoded or not. sip_new was blindly always decoding the extension, which also caused the outgoing contact header to be incorrect as well as failing to match the encoded extension in the dialplan. (closes issue #17892) Reported by: wdoekes Patches: bug17892-1.patch uploaded by jpeeler (license 325) Tested by: wdoekes ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@289702 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-10-01don't iterate through all dialogs to find and delete old subscribesStefan Schmidt
On every incoming subscribe there is a iteration through all dialogs to find old subscribes and delete them. This is slow and not RFC conform. This was only needed in 1.2 cause a subscribe was not deleted when a dialog was destroyed, after 1.4 a subscribe get removed when its dialog is destroyed. Review: https://reviewboard.asterisk.org/r/901/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@289623 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-30Merged revisions 289554 via svnmerge from Matthew Nicholson
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r289554 | mnicholson | 2010-09-30 14:53:10 -0500 (Thu, 30 Sep 2010) | 11 lines Merged revisions 289553 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r289553 | mnicholson | 2010-09-30 14:51:27 -0500 (Thu, 30 Sep 2010) | 4 lines Properly handle channel allocation failures duing invites with replaces. ABE-2588 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@289555 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-28Merged revisions 289054-289055 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r289054 | rmudgett | 2010-09-27 19:32:18 -0500 (Mon, 27 Sep 2010) | 1 line Break up long ast_manager_event_multichan() event lines. ........ r289055 | rmudgett | 2010-09-27 19:35:25 -0500 (Mon, 27 Sep 2010) | 1 line Revert stuff not ready for commit in -r289054. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@289056 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-27For an INVITE transaction, treat all 2XX responses the same as a 200.David Vossel
ABE-2305 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@289023 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-27Formatting fixesOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@288993 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-27Merged revisions 288961 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r288961 | tilghman | 2010-09-27 13:37:41 -0500 (Mon, 27 Sep 2010) | 5 lines Still build SIP, even if res_crypto cannot be built (use, not depend). (closes issue #18062) Reported by: a user on the mailing list ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@288962 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-24Merged revisions 288852 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r288852 | dvossel | 2010-09-24 12:58:57 -0500 (Fri, 24 Sep 2010) | 5 lines Append Retry-After header on 500 error response to Re-INVITE according to RFC3261 section 14.2. ABE-2301 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@288853 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-24Merged revisions 288821 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r288821 | dvossel | 2010-09-24 12:05:12 -0500 (Fri, 24 Sep 2010) | 4 lines Inspect Require header on BYE transaction according to RFC3261 section 8.2.2.3. ABE-2293 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@288822 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-22Merged revisions 288418 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r288418 | dvossel | 2010-09-22 12:49:56 -0500 (Wed, 22 Sep 2010) | 18 lines Merged revisions 288417 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r288417 | dvossel | 2010-09-22 12:49:05 -0500 (Wed, 22 Sep 2010) | 11 lines Merged revisions 288416 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r288416 | dvossel | 2010-09-22 12:48:15 -0500 (Wed, 22 Sep 2010) | 5 lines RFC3261 section 12.2 explicitly says out of order requests are responded with a 500 Server Internal Error response. ABE-2458 ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@288419 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-22Merged revisions 288345 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r288345 | dvossel | 2010-09-22 11:59:14 -0500 (Wed, 22 Sep 2010) | 16 lines Merged revisions 288344 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r288344 | dvossel | 2010-09-22 11:53:28 -0500 (Wed, 22 Sep 2010) | 9 lines Merged revisions 288343 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r288343 | dvossel | 2010-09-22 11:49:56 -0500 (Wed, 22 Sep 2010) | 2 lines During check_pendings, if the dialog is terminated with a CANCEL, change the invitestate to INV_CANCEL like in sip_hangup. ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@288346 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-21Merged revisions 288159 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r288159 | tilghman | 2010-09-21 17:57:22 -0500 (Tue, 21 Sep 2010) | 29 lines Merged revisions 288113 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r288113 | tilghman | 2010-09-21 16:59:46 -0500 (Tue, 21 Sep 2010) | 22 lines Merged revisions 288112 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r288112 | tilghman | 2010-09-21 16:58:13 -0500 (Tue, 21 Sep 2010) | 15 lines Try both the encoded and unencoded subscription URI for a match in hints. When a phone sends an encoded URI for a subscription, the URI is not matched with the actual hint that is in decoded format. For example, if we have an extension with a hint that is named: "#5601" or "*5601", the subscription will work fine if the phone subscribes with an already decoded URI, but when it's decoded like "%255601" or "%2A5601", Asterisk is unable to match it with the correct hint. (closes issue #17785) Reported by: ramonpeek Patches: 20100831__issue17785.diff.txt uploaded by tilghman (license 14) Tested by: ramonpeek ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@288160 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-21Instead of iterate through all dialogs, add two separte container for ↵Stefan Schmidt
needdestroy and rtptimeout adding two dialog container, one for dialogs which need destroy, another for rtptimeout checks. both container will be checked on every loop of do_monitor instead of iterate through all dialogs. (closes issue #17912) Reported by: schmidts Tested by: schmidts Review: https://reviewboard.asterisk.org/r/917/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@288063 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-21Merged revisions 287929 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r287929 | dvossel | 2010-09-21 13:32:12 -0500 (Tue, 21 Sep 2010) | 4 lines Send a "415 Unsupported Media Type" after failure to process sdp due to unknown Content-Encoding header. ABE-2258 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@287930 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-21Merged revisions 287895 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r287895 | russell | 2010-09-21 10:43:33 -0500 (Tue, 21 Sep 2010) | 10 lines Don't use ast_strdupa() from within the arguments to a function. (closes issue #17902) Reported by: afried Patches: issue_17902.rev1.txt uploaded by russell (license 2) Tested by: russell Review: https://reviewboard.asterisk.org/r/927/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@287896 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-21Merged revisions 287893 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r287893 | tilghman | 2010-09-21 10:24:47 -0500 (Tue, 21 Sep 2010) | 9 lines Anonymous callerid needs a "sip:" uri prefix. (closes issue #17981) Reported by: avalentin Patches: sip-anonymous-aastra.patch uploaded by avalentin (license 1107) (plus an additional fix by me) Tested by: avalentin ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@287894 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-20Merged revisions 287645 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r287645 | dvossel | 2010-09-20 16:34:15 -0500 (Mon, 20 Sep 2010) | 9 lines Fixes issue with registrations not working properly with pedantic=yes. (closes issue #18017) Reported by: schmidts Patches: issues_18017_v1.diff uploaded by dvossel (license 671) Tested by: schmidts ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@287646 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-16We do not handle AST_CAUSE_INTERWORKING which we set on a lot of incomingOlle Johansson
SIP messages. Adding error based on RFC 3398 recommendations. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@287087 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-15Merged revisions 286931 via svnmerge from Jeff Peeler
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r286931 | jpeeler | 2010-09-15 14:22:15 -0500 (Wed, 15 Sep 2010) | 16 lines Add parking extension for non-default parking lots. This is a new feature that allows for parking to custom parking lots to be accessed directly, rather than with channel variables or by changing the default parking lot. The extension is set with the parkext option just as the default parking lot is done. Also, the manager action has been updated to optionally allow a specified parking lot. (closes issue #14882) Reported by: vmikhnevych Patches: patch_14882.txt uploaded by mnick (license 874) modified by me Review: https://reviewboard.asterisk.org/r/884/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@286939 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-15Merged revisions 286868 via svnmerge from Matthew Nicholson
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r286868 | mnicholson | 2010-09-15 08:05:52 -0500 (Wed, 15 Sep 2010) | 16 lines Set tohost to the domain specified in the configuration file instead of the IP address of the host we are calling. This fixes a regression introduced in r274783. (closes issue #17960) Reported by: adriavidal Patches: sip-tohost-fix1.diff uploaded by mnicholson (license 96) Tested by: mich, mnicholson, adriavidal (closes issue #17676) Reported by: outcast Patches: sip-tohost-fix1.diff uploaded by mnicholson (license 96) Tested by: mnicholson ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@286869 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-14Merged revisions 286834 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r286834 | dvossel | 2010-09-14 16:57:35 -0500 (Tue, 14 Sep 2010) | 2 lines Sets subscribed type for outgoing MWI subscriptions so correct Event header is used. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@286835 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-14Merged revisions 286758 via svnmerge from Matthew Nicholson
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r286758 | mnicholson | 2010-09-14 14:28:38 -0500 (Tue, 14 Sep 2010) | 27 lines Merged revisions 286757 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r286757 | mnicholson | 2010-09-14 14:27:28 -0500 (Tue, 14 Sep 2010) | 20 lines Merged revisions 286756 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r286756 | mnicholson | 2010-09-14 14:26:18 -0500 (Tue, 14 Sep 2010) | 13 lines Don't clear the username from a realtime database when a registration expires. Non-realtime chan_sip does not clear the username from memory when a registration expiries so realtime probably shouldn't either. (closes issue #17551) Reported by: ricardolandim Patches: reg-expiry-username-1.4-fix1.diff uploaded by mnicholson (license 96) reg-expiry-username-1.6.2-fix1.diff uploaded by mnicholson (license 96) reg-expiry-username-1.8-fix1.diff uploaded by mnicholson (license 96) reg-expiry-username-trunk-fix1.diff uploaded by mnicholson (license 96) Tested by: ricardolandim, mnicholson ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@286759 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-13Merged revisions 286457 via svnmerge from Jason Parker
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r286457 | qwell | 2010-09-13 14:40:05 -0500 (Mon, 13 Sep 2010) | 12 lines Merged revisions 286456 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r286456 | qwell | 2010-09-13 14:38:35 -0500 (Mon, 13 Sep 2010) | 5 lines Remove "Internal IP" from sip show settings, as it's not at all useful to display. (closes issue #17840) Reported by: oej ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@286458 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-11Formatting changes.Olle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@286271 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-08Merged revisions 285568 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r285568 | dvossel | 2010-09-08 17:14:19 -0500 (Wed, 08 Sep 2010) | 16 lines Merged revisions 285567 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r285567 | dvossel | 2010-09-08 17:11:28 -0500 (Wed, 08 Sep 2010) | 9 lines 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/trunk@285569 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-08Merged revisions 285564 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r285564 | dvossel | 2010-09-08 16:48:37 -0500 (Wed, 08 Sep 2010) | 60 lines Merged revisions 285563 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r285563 | dvossel | 2010-09-08 16:47:29 -0500 (Wed, 08 Sep 2010) | 54 lines 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/trunk@285565 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-09-07Merged revisions 285455 via svnmerge from Jason Parker
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r285455 | qwell | 2010-09-07 17:22:14 -0500 (Tue, 07 Sep 2010) | 8 lines Don't automatically add domains for wildcard bindaddrs. (closes issue #17832) Reported by: oej Patches: 17832-wildcard.diff uploaded by qwell (license 4) Tested by: qwell ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@285456 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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