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path: root/channels/chan_sip.c
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2008-06-10Fix issue where session timer headers were present when they should not have ↵Joshua Colp
been. (closes issue #12706) Reported by: falves11 Patches: chan_sip.c.diff uploaded by rjain (license 226) Tested by: falves11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121503 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-10Merged revisions 121495 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r121495 | file | 2008-06-10 10:34:27 -0300 (Tue, 10 Jun 2008) | 4 lines If we are destroying a dialog only set the MWI dialog pointer on the related peer to NULL if it is the dialog currently being destroyed. (closes issue #12828) Reported by: ramonpeek ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121496 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-09Expand RQ_INTEGER type out to multiple types, one for each precisionTilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121367 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-09Add storage of the useragent in the realtime database.Tilghman Lesher
(Closes AST-38) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121334 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-06Added a facility for sending arbitrary SIP notify commands from AMI.Tilghman Lesher
(closes issue #12562) Reported by: michael-fig Patches: 20080515__bug12562.diff.txt uploaded by Corydon76 (license 14) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121042 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-06Merged revisions 120959 via svnmerge from Jeff Peeler
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r120959 | jpeeler | 2008-06-06 13:29:14 -0500 (Fri, 06 Jun 2008) | 1 line add another LOW_MEMORY define I forgot ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@120960 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-06Merged revisions 120908 via svnmerge from Jeff Peeler
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r120908 | jpeeler | 2008-06-06 13:05:15 -0500 (Fri, 06 Jun 2008) | 1 line only define thread storage variable if necessary for LOW_MEMORY ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@120909 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-06Merged revisions 120863,120885 via svnmerge from Jeff Peeler
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r120863 | jpeeler | 2008-06-06 10:33:15 -0500 (Fri, 06 Jun 2008) | 3 lines This fixes a crash when LOW_MEMORY is turned on. Two allocations of the ast_rtp struct that were previously allocated on the stack have been modified to use thread local storage instead. ........ r120885 | jpeeler | 2008-06-06 11:39:20 -0500 (Fri, 06 Jun 2008) | 2 lines Correction to commmit 120863, make sure proper destructor function is called as well define two thread storage local variables. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@120906 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-05Merge the adaptive realtime branch, which will make adding new required fieldsTilghman Lesher
to realtime less painful in the future. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@120789 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-05This patch adds more detailed statistics for RTP channels, and provides an ↵Brett Bryant
API call to access it, including maximums, minimums, standard deviatinos, and normal deviations. Currently this is implemented for chan_sip, but could be added to the func_channel_read callbacks for the CHANNEL function for any channel that uses RTP. (closes issue #10590) Reported by: gasparz Patches: chan_sip_c.diff uploaded by gasparz (license 219) rtp_c.diff uploaded by gasparz (license 219) rtp_h.diff uploaded by gasparz (license 219) audioqos-trunk.diff uploaded by snuffy (license 35) rtpqos-trunk-r119891.diff uploaded by sergee (license 138) Tested by: jsmith, gasparz, snuffy, marsosa, chappell, sergee git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@120635 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-03Merged revisions 119926 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r119926 | file | 2008-06-03 11:46:24 -0300 (Tue, 03 Jun 2008) | 2 lines Treat ECONNREFUSED as an error that will stop further retransmissions. (issue #AST-58, patch from Switchvox) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@119927 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-05-28Merged revisions 118646 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r118646 | file | 2008-05-28 11:23:34 -0300 (Wed, 28 May 2008) | 4 lines Add an option to use the source IP address of RTP as the destination IP address of UDPTL when a specific option is enabled. If the remote side is properly configured (ports forwarded) then UDPTL will flow. (closes issue #10417) Reported by: cstadlmann ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@118647 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-05-27Merged revisions 118558 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r118558 | file | 2008-05-27 16:32:38 -0300 (Tue, 27 May 2008) | 4 lines Fix an issue where codec preferences were not set on dialogs that were not authenticated via a user or peer and allow framing to work without rtpmap in the SDP. (closes issue #12501) Reported by: slimey ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@118560 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-05-25Merged revisions 118251 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r118251 | tilghman | 2008-05-25 11:02:04 -0500 (Sun, 25 May 2008) | 12 lines Realtime flag affects construction in multiple ways, so consulting whether rtcachefriends was set was done too soon (needed to be done inside build_peer, not just as a flag to build_peer). Also, fullcontact needed to be reconstructed, because realtime separates the embedded ';' into multiple fields. (closes issue #12722) Reported by: barthpbx Patches: 20080525__bug12722.diff.txt uploaded by Corydon76 (license 14) Tested by: barthpbx (Much of the discussion happened on #asterisk-dev for diagnosing this issue) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@118252 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-05-22- revert change to ast_queue_hangup and create ast_queue_hangup_with_causeMichiel van Baak
- make data member of the ast_frame struct a named union instead of a void Recently the ast_queue_hangup function got a new parameter, the hangupcause Feedback came in that this is no good and that instead a new function should be created. This I did. The hangupcause was stored in the seqno member of the ast_frame struct. This is not very elegant, and since there's already a data member that one should be used. Problem is, this member was a void *. Now it's a named union so it can hold a pointer, an uint32 and there's a padding in case someone wants to store another type in there in the future. This commit is so massive, because all ast_frame.data uses have to be altered to ast_frame.data.data Thanks russellb and kpfleming for the feedback. (closes issue #12674) Reported by: mvanbaak git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@117802 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-05-21Merged revisions 117574 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r117574 | file | 2008-05-21 16:38:28 -0300 (Wed, 21 May 2008) | 2 lines Apply the autoframing setting to dialogs that do not get matched against a user or peer. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@117575 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-05-18Remove duplicate colon on Reason headerRussell Bryant
(closes issue #12678) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@116919 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-05-16Try to fix attended transfers.Joshua Colp
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@116848 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-05-15Fixes a problem I was having with two SIP phones using Packet2Packet ↵Jeff Peeler
bridging dropping audio nearly immediately. The problem was that the lock on the SIP dialog was not being unlocked while the bridge was still active. (Related to issue #12566) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@116663 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-05-14Fix pedanticness.Joshua Colp
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@116471 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-05-14Don't add linefeed on received MESSAGEOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@116240 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-05-14Properly declare charset for text messages.Olle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@116239 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-05-14Adding spport for T.140 RED - Simple RTP redundancy to prevent packet loss ↵Olle Johansson
in text stream Work sponsored by Omnitor AB, Stockholm, Sweden (http://www.omnitor.se) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@116237 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-05-14Merged revisions 116230 via svnmerge from Olle Johansson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r116230 | oej | 2008-05-14 14:51:06 +0200 (Ons, 14 Maj 2008) | 3 lines Accept text messages even with Content-Type: text/plain;charset=Södermanländska ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@116234 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-05-14Add support for codec settings in originate via call file and manager.Olle Johansson
This is to enable video and text in originated calls. Development sponsored by Omnitor AB, Sweden. (http://www.omnitor.se) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@116229 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-05-14ReformattingOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@116223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-05-14Adding commentsOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@116222 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-05-09Adding support for "urgent" voicemail messages. Messages which areMark Michelson
marked "urgent" are considered to be higher priority than other messages and so they will be played before any other messages in a user's mailbox. There are two ways to leave an urgent message. 1. send the 'U' option to VoiceMail(). 2. Set review=yes in voicemail.conf. This will give instructions for a caller to mark a message as urgent after the message has been recorded. I have tested that this works correctly with file and ODBC storage, and James Rothenberger (who wrote initial support for this feature) has tested its use with IMAP storage. (closes issue #11817) Reported by: jaroth Based on branch http://svn.digium.com/svn/asterisk/team/jrothenberger/asterisk-urgent Tested by: putnopvut, jaroth git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@115588 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-05-08Merged revisions 115561 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115561 | russell | 2008-05-08 11:11:33 -0500 (Thu, 08 May 2008) | 3 lines Don't give up on attempting an outbound registration if we receive a 408 Timeout. (closes issue #12323) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@115562 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-05-07Remove redundant header getting.Joshua Colp
(closes issue #12597) Reported by: hooi git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@115507 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-05-06Change some NOTICE log messages to debug.Russell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@115344 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-05-05Merged revisions 115304 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115304 | russell | 2008-05-05 14:49:25 -0500 (Mon, 05 May 2008) | 5 lines Avoid putting opaque="" in Digest authentication. This patch came from switchvox. It fixes authentication with Primus in Canada, and has been in use for a very long time without causing problems with any other providers. (closes issue AST-36) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@115305 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-05-02Add attributes to various API calls, to help track down bugs (and remove a ↵Tilghman Lesher
deprecated function) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@115157 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-30Add support for specifying the registration expiry on a per registration ↵Joshua Colp
basis in the register line. This comes from a Switchvox patch. (issue AST-24) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114912 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-30Merged revisions 114890 via svnmerge from Olle Johansson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114890 | oej | 2008-04-30 18:23:17 +0200 (Ons, 30 Apr 2008) | 7 lines Don't crash on bad SIP replys. Fix created in Huntsville together with Mark M (putnopvut) (closes issue #12363) Reported by: jvandal Tested by: putnopvut, oej ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114899 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-26Unleak referenceTilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114692 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-26Add 'sip qualify peer <peer>' command (with AMI SIPqualifypeer)Tilghman Lesher
(closes issue #12524) Reported by: ctooley Patches: sip_qualify_peer.diff.2 uploaded by ctooley (license 136) some modifications for trunk by Corydon76 Tested by: Corydon76 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114683 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-24Pass the hangup cause all the way to the calling app/channel.Michiel van Baak
(closes issue #11328) Reported by: rain Patches: 20071207__pass_cause_in_hangup_control_frame.diff.txt uploaded by Corydon76 (license 14) brought up-to-date to trunk by me git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114637 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-24Hey look, it builds.Joshua Colp
(closes issue #12519) Reported by: falves11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114635 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-24Merged revisions 114632 via svnmerge from Mark Michelson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114632 | mmichelson | 2008-04-24 16:35:08 -0500 (Thu, 24 Apr 2008) | 11 lines Re-invite RTP during a masquerade so that, for instance, an AMI redirect of two channels which are natively bridged will preserve audio on both channels. This prevents a problem with Asterisk not re-inviting due to one of the channels having being a zombie. (closes issue #12513) Reported by: mneuhauser Patches: asterisk-1.4-114602_restore-RTP-on-fixup.patch uploaded by mneuhauser (license 425) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114633 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-24Merged revisions 114603 via svnmerge from Olle Johansson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114603 | oej | 2008-04-24 16:55:18 +0200 (Tor, 24 Apr 2008) | 3 lines Only have one max-forwards header in outbound REFERs. Discovered in the Asterisk SIP Masterclass in Orlando. Thanks Joe! ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114606 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-24Change a verbose message to debug.Russell Bryant
(closes issue #12514) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114604 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-23Merged revisions 114584 via svnmerge from Olle Johansson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114584 | oej | 2008-04-23 18:51:41 +0200 (Ons, 23 Apr 2008) | 2 lines Add 502 support for both directions, not only one... (see r114571) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114585 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-22Merged revisions 114571 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114571 | tilghman | 2008-04-22 18:51:44 -0500 (Tue, 22 Apr 2008) | 2 lines Treat a 502 just like a 503, when it comes to processing a response code ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114572 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-22Add support for authenticating on a NOTIFY request. This is useful for ↵Joshua Colp
phones that require it when sending them a special packet to get them to do something (such as reload their configuration). (closes issue #9896) Reported by: IgorG Patches: sipnotify-113980-v14.patch uploaded by IgorG (license 20) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114529 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-22Hopefully, this will resolve the issues that russellb had with this ↵Steve Murphy
log_show_lock(). I gathered the code that filled the string, and put it in a different func which I cryptically call "append_lock_information()". Now, both log_show_lock(), and handle_show_locks() both call this code to do the work. Tested, seems to work fine. Also, log_show_lock was modified to use the ast_str stuff, along with checking for successful ast_str creation, and freeing the ast_str obj when finished. A break was inserted to terminate the search for the lock; we should never see it twice. An example usage in chan_sip.c was created as a comment, for instructional purposes. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114520 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-21(closes issue #6113)Jeff Peeler
Reported by: oej Tested by: jpeeler This patch implements multiple parking lots for parked calls. The default parkinglot is used by default, however setting the channel variable PARKINGLOT in the dialplan will allow use of any other configured parkinglot. See configs/features.conf.sample for more details on setting up another non-default parkinglot. Also, one can (currently) set the default parkinglot to use in the driver configuration file via the parkinglot option. Patch initially written by oej, brought up to date and finalized by mvanbaak, and then stabilized and converted to astobj2 by me. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114487 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-21Merged revisions 114322 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114322 | file | 2008-04-21 11:39:32 -0300 (Mon, 21 Apr 2008) | 4 lines Only drop audio if we receive it without a progress indication. We allow other frames through such as DTMF because they may be needed to complete the call. (closes issue #12440) Reported by: aragon ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114323 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-18Merged revisions 114245 via svnmerge from Sean Bright
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114245 | seanbright | 2008-04-18 09:33:32 -0400 (Fri, 18 Apr 2008) | 1 line Only complete the SIP channel name once for 'sip show channel <channel>' ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114246 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-17Thanks to snuff for finding these omissionsSteve Murphy
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114201 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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