summaryrefslogtreecommitdiff
path: root/channels/chan_sip.c
AgeCommit message (Collapse)Author
2008-08-20Fix output of sipshowpeer manager response.Jason Parker
(closes issue #13346) Reported by: srt Patches: 13346_malformed_sip_show_peer_response.diff uploaded by srt (license 378) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@139210 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-08-20Merged revisions 139015 via svnmerge from Mark Michelson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r139015 | mmichelson | 2008-08-20 10:37:56 -0500 (Wed, 20 Aug 2008) | 6 lines sip_read should properly handle a NULL return from sip_rtp_read. (closes issue #13257) Reported by: travishein ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@139016 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-08-19Let it compile now, too (woops)Sean Bright
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@138780 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-08-19And remove code we don't need anymore.Sean Bright
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@138779 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-08-19While we're at it, make this machine parseable too.Sean Bright
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@138778 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-08-18Change event header to RegistrationTime to be more consistent (and avoidSean Bright
breaking existing frameworks). Pointed out by Laureano on #asterisk-dev. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@138775 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-08-15Merged revisions 138258 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r138258 | tilghman | 2008-08-15 17:33:42 -0500 (Fri, 15 Aug 2008) | 8 lines More fixes for realtime peers. (closes issue #12921) Reported by: Nuitari Patches: 20080804__bug12921.diff.txt uploaded by Corydon76 (license 14) 20080815__bug12921.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@138260 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-08-15regseconds is actually stored as the epoch time, not registration lengthTilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@138086 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-08-14Make sure we set the socket port, so we don't try to use <ip address>:0.Jason Parker
(closes issue #13255) Reported by: falves11 Patches: 13255-socketport.diff uploaded by qwell (license 4) Tested by: falves11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@137812 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-08-13Correctly end locally ended calls.Jason Parker
(closes issue #12170) Reported by: pj Patches: 20080702__issue12170_clear_pendinginvite.diff uploaded by bbryant (license 36) Tested by: bbryant, pabelanger git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@137532 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-08-09More RSW merges. This should do it for the channels/ dir.Sean Bright
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@136917 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-08-01Picky, picky, buildbotTilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@135128 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-08-01SIP should use the transport type set in the Moved Temporarily for the nextTilghman Lesher
invite. (closes issue #11843) Reported by: pestermann Patches: 20080723__issue11843_302_ignores_transport_16branch.diff uploaded by bbryant (license 36) 20080723__issue11843_302_ignores_transport_trunk.diff uploaded by bbryant (license 36) Tested by: pabelanger git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@135126 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-30Fix the parsing of the "reason" parameter in theMark Michelson
Diversion: header. (closes issue #13195) Reported by: woodsfsg git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@134556 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-30Move implementation of an attended-transfer-complete sound from one channelTilghman Lesher
driver into a common place for multiple channel drivers. (closes issue #13152) Reported by: caio1982 Patches: atxfer_complete_sound3.diff uploaded by caio1982 (license 22) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@134401 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-25Deprecate *_device_state_* APIs in favor of *_devstate_* APIsTilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@133860 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-25Merged revisions 133572 via svnmerge from Mark Michelson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r133572 | mmichelson | 2008-07-25 09:40:10 -0500 (Fri, 25 Jul 2008) | 7 lines We need to make sure to null-terminate the "name" portion of SIP URI parameters so that there are no bogus comparisons. Thanks to bbryant for pointing this out. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@133573 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-25Minor coding guidelines tweaks ...Russell Bryant
- Use ast_strlen_zero in one place - check for successful string comparison the way most of Asterisk code does it git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@133568 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-24Merged revisions 133488 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r133488 | tilghman | 2008-07-24 16:17:55 -0500 (Thu, 24 Jul 2008) | 3 lines Fix rtautoclear and rtcachefriends (Closes issue #12707) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@133509 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-23Fix issue where tcp in sip is enabled by default, despite what it says in ↵Brett Bryant
the config sample file. Also fix "sip show settings" for tcp connections. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@133197 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-23Well, the content of a channel variable may be longer than the size of a ↵Olle Johansson
pointer... Thanks, eliel! Reported by: eliel Patches: chan_siptrunk.SIPPEER.patch uploaded by eliel (license 64) (closes issue #13135) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@132823 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-22Merged revisions 132777 via svnmerge from Mark Michelson
https://origsvn.digium.com/svn/asterisk/trunk ........ Allow Spiraled INVITEs to work correctly within Asterisk. Prior to this change, a spiraled INVITE would cause a 482 Loop Detected to be sent to the caller. With this change, if a potential loop is detected, the Request-URI is inspected to see if it has changed from what was originally received. If pedantic mode is on, then this inspection is fully RFC 3261 compliant. If pedantic mode is not on, then a string comparison is used to test the equality of the two R-URIs. This has been tested by using OpenSER to rewrite the R-URI and send the INVITE back to Asterisk. (closes issue #7403) Reported by: stephen_dredge Modified: branches/1.4/channels/chan_sip.c ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@132795 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-22Merged revisions 132645 via svnmerge from Olle Johansson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r132645 | oej | 2008-07-22 22:10:26 +0200 (Tis, 22 Jul 2008) | 9 lines The most common question on the #asterisk iRC channel and on mailing lists seems to be in regards to an error message when retransmit fails. This is frequently misunderstood as a failure of Asterisk, not a failure of the network to reach the other party. This document tries to assist the Asterisk user in sorting out these issues by explaining the logic and pointing at some possible causes. Hopefully, we will get other questions now :-) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@132703 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-21Fix bug where ast_parse_arg would inadvertantly enable sip tcp when parsing ↵Brett Bryant
a tcpbindaddr if it was disabled. (closes issue #13117) Reported by: pj git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@132468 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-15Merged revisions 130959 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r130959 | tilghman | 2008-07-15 12:19:13 -0500 (Tue, 15 Jul 2008) | 8 lines astman_send_error does not need a newline appended -- the API takes care of that for us. (closes issue #13068) Reported by: gknispel_proformatique Patches: asterisk_1_4_astman_send.patch uploaded by gknispel (license 261) asterisk_trunk_astman_send.patch uploaded by gknispel (license 261) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@131044 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-15Additional option for videosupport (always) that disables the optimization toTilghman Lesher
fail to setup video RTP if the two endpoints will not support it. This assists with call files and certain transfers to ensure that if two video phones are ever connected, they will always share a video feed. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@130951 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-11Missed one. Formatting only.Sean Bright
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@130167 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-11Janitor patch to change uses of sizeof to ARRAY_LENBrett Bryant
(closes issue #13054) Reported by: pabelanger Patches: ARRAY_LEN.patch2 uploaded by pabelanger (license 224) Tested by: seanbright git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@130129 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-10A couple more minor text changesSean Bright
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@129642 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-10Remove extraneous \n. Pointed out by eliel on #asterisk-dev.Sean Bright
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@129638 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-08Merged revisions 129149 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r129149 | tilghman | 2008-07-08 15:27:47 -0500 (Tue, 08 Jul 2008) | 8 lines Cause SIP to return a 480 instead of a 404 when a sip peer exists, but is not registered. (closes issue #12885) Reported by: ibc Patches: 20080701__bug12885__2.diff.txt uploaded by Corydon76 (license 14) Tested by: ibc ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@129152 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-08Janitor project to convert sizeof to ARRAY_LEN macro.Brett Bryant
(closes issue #13002) Reported by: caio1982 Patches: janitor_arraylen5.diff uploaded by caio1982 (license 22) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@129045 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-08Merged revisions 128950 via svnmerge from Olle Johansson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r128950 | oej | 2008-07-08 11:52:21 +0200 (Tis, 08 Jul 2008) | 11 lines Don't hangup the call if we can't resolve the Contact if there's a proxy route set for the call. ---- This comment was added a while ago and today it hit me badly. /* OEJ: Possible issue that may need a check: If we have a proxy route between us and the device, should we care about resolving the contact or should we just send it? */ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@128951 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-08Merged revisions 128912 via svnmerge from Olle Johansson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r128912 | oej | 2008-07-08 11:06:08 +0200 (Tis, 08 Jul 2008) | 7 lines Fix issues where repeated messages where ignored, but retransmitted reliably instead of unreliably. Reported by: johan Patches: 12746.txt uploaded by oej (license 306) Tested by: johan (issue #12746) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@128927 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-07As pointed out on the -dev list, actually use the result of find_peer() so thatRussell Bryant
a peer reference is not leaked. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@128564 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-06- Adding alias "udpbindaddr" for the UDP port to comply with "tcpbindaddr" ↵Olle Johansson
and "tlsbindaddr". Note: I don't think we can start properly without UDP port open, that needs to be tested. - Removing "bindport" from configuration example, not needed to mention this any more I suggest we deprecate "bindaddr" and "bindport" in trunk (for 1.6.1) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@128525 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-06- Fixing issues with "sip show settings"Olle Johansson
- Adding IP address for TCP and/or TLS too if auto-domain is enabled and binding to a different IP address - Fixing documentation in sip.conf.sample git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@128524 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-06- Remove unused variable "expiry"Olle Johansson
- Set global_outboundproxy.force at reload. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@128491 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-06More doxygen comments.Olle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@128485 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-06- Formatting changesOlle Johansson
- Doxygen changes - Replacing a doxygen description that was copied from another function git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@128455 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-06Adding note about incorrect manager registration...Olle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@128454 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-06Remove comments that doesn't make sense. The deprecation of type=user will ↵Olle Johansson
come at a later stage, as indicated by previous commit message git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@128381 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-06Fix severe problem with my previous commit of "kill-the-user". Russell saw a ↵Olle Johansson
problem with this code, but not the correct problem. Thanks, anyway! ;-) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@128379 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-06Disabling code used by dumpdb with #ifdef, since I believe we might use it ↵Olle Johansson
sometime in the future, but also want to avoid compiler warnings now. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@128344 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-06Removing the CLI dumpdb command (see asterisk-dev discussion and decision)Olle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@128343 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-05Adding a few remindersOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@128294 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-05Adding doxygen comments to missing parts, moving some #defineOlle Johansson
...trying to get my head around the thoughts behind the TCP/TLS stuff and figure out what needs to be done to make it useful... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@128290 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-05Adding TCP and TLS to "sip show settings". Olle Johansson
TLS needs to have one configuration per configured domain at some point. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@128287 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-05Add some comments...Olle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@128284 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-07-05Set tls setting to default in reload_config.Olle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@128274 65c4cc65-6c06-0410-ace0-fbb531ad65f3