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path: root/channels/chan_sip.c
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2007-05-18Another fix for the support for recordings controlled by INFO-packetsOlle Johansson
We still lack a setting to enable/disable this per peer git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@64903 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-17Merged revisions 64754 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r64754 | file | 2007-05-17 12:10:12 -0400 (Thu, 17 May 2007) | 2 lines Even more direct RTP setup fixes! Don't allow a codec that isn't supported to creep into the SDP of either side. (issue #9446 reported by marcelbarbulescu) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@64755 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-16Below patches with some re-structuring for trunkOlle Johansson
--- Merged revisions 64602 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r64602 | oej | 2007-05-16 12:38:18 +0200 (Wed, 16 May 2007) | 2 lines Issue #9681 - Handle www-auth on BYE ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@64604 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-16Merged revisions 64578 via svnmerge from Olle Johansson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r64578 | oej | 2007-05-16 12:05:47 +0200 (Wed, 16 May 2007) | 2 lines Final part of issue #9483 - fixing transfer() of sip calls in the dial plan (twilson) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@64587 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-16Merged revisions 64543 via svnmerge from Olle Johansson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r64543 | oej | 2007-05-16 11:12:34 +0200 (Wed, 16 May 2007) | 10 lines Merged revisions 64535 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r64535 | oej | 2007-05-16 11:08:22 +0200 (Wed, 16 May 2007) | 2 lines Support SIP uri's starting with SIP: and sip: (reported by Tony Mountfield on the mailing list. Thanks!) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@64570 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-16Merged revisions 64516 via svnmerge from Olle Johansson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r64516 | oej | 2007-05-16 10:46:18 +0200 (Wed, 16 May 2007) | 17 lines Merged following patch with a lot of changes for 1.4 ------ Merged revisions 64514 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r64514 | oej | 2007-05-16 10:25:56 +0200 (Wed, 16 May 2007) | 6 lines Issue #9726 - rlister - Better logging for ACL denials While at it, also added better logging and handling of peers that are not supposed to register. My patch, stole the issue report from Russell. My apologies, Russell :-) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@64517 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-16Issue #9304 - Update help text to match functionality. Patch by kshumard ↵Olle Johansson
with changes by oej git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@64498 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-16Issue #6789 - Marquis - Add option to support regexten removal when host ↵Olle Johansson
becomes unreachable git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@64497 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-14Merged revisions 64324 via svnmerge from Olle Johansson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r64324 | oej | 2007-05-14 21:26:50 +0200 (Mon, 14 May 2007) | 2 lines Change -2 to XMIT_ERROR to clarify a bit more ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@64325 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-14If no port is specified in the outboundproxy setting then use the standard ↵Joshua Colp
SIP port. (issue #9665 reported by tootai) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@64274 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-13Improve handling network errors on transmission to hosts that don't reply or ↵Olle Johansson
are unreachable With this code, the call will fail as soon as we get a network error. This may happen on first xmit or a later one, so the retransmit code handles this too. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@64142 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-12Merged revisions 64114 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r64114 | file | 2007-05-12 18:27:04 -0400 (Sat, 12 May 2007) | 2 lines This concludes my final adventure with bitmasks and the onhold flag. Would anyone care for some peanuts? ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@64115 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-12Merged revisions 64086 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r64086 | file | 2007-05-12 17:10:45 -0400 (Sat, 12 May 2007) | 2 lines Tweak hold flags some more. They can be of three states when active: active, inactive, one direction. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@64087 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-12Merged revisions 64044 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r64044 | file | 2007-05-12 12:32:15 -0400 (Sat, 12 May 2007) | 2 lines Ensure the onhold flag is set no matter what when being put on hold. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@64045 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-10Merged revisions 63749 via svnmerge from Olle Johansson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r63749 | oej | 2007-05-10 22:46:41 +0200 (Thu, 10 May 2007) | 12 lines Merged revisions 63748 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r63748 | oej | 2007-05-10 22:38:54 +0200 (Thu, 10 May 2007) | 4 lines Do not allocate SIP pvt's for PEERs we can not reach. This was seen as a lot of dialogs being created then immediately destroyed at reload/restart of the SIP channel. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@63751 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-09Merged revisions 63611 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r63611 | file | 2007-05-09 12:54:56 -0400 (Wed, 09 May 2007) | 10 lines Merged revisions 63610 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r63610 | file | 2007-05-09 12:51:03 -0400 (Wed, 09 May 2007) | 2 lines Properly handle hints that point to multiple devices in chan_sip. Why chan_sip is even doing this I have no idea but I would rather not go into a rant. (issue #9536 reported by rlister) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@63613 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-09Merged revisions 63532 via svnmerge from Olle Johansson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r63532 | oej | 2007-05-09 15:04:14 +0200 (Wed, 09 May 2007) | 2 lines Don't retransmit 200 OK's on ignore status. (Reported on asterisk-users) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@63533 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-08I noted this on the dev list but got no response, so I just did it myself.Russell Bryant
Lock the call features when being used in chan_sip. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@63447 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-05- Adding some missing spacesOlle Johansson
- Correcting error messages - Disabling code that doesn't do anything - Making sure we always respond to this request, happily git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@63136 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-04a small upgrade to the coding standard, and an update to the code that ↵Steve Murphy
triggered the upgrade. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@63048 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-04Added a small bit of code to support the SNOM 360's Record button. Made the ↵Steve Murphy
find_feature func in res_features.c public, so I could use it to find the automon dial sequence as configured by the user. When the INFO packet has a Record: header with on/off, the sequence is sent as consecutive DTMF frames on the phone's channel, triggering the automon functionality. The user has to configure the automon in features.conf, and set up his dialplan accordingly. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@63046 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-04Add the new ChannelUpdate event to inform manager clients about the PVT ID ↵Olle Johansson
and some other channel driver data that is needed to follow the call through the PBX. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@63032 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-03Merged revisions 62989 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r62989 | file | 2007-05-03 13:44:00 -0300 (Thu, 03 May 2007) | 10 lines Merged revisions 62987 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r62987 | file | 2007-05-03 13:42:19 -0300 (Thu, 03 May 2007) | 2 lines When a peer is seeded or built tell the devicestate core to update it's status. This is easier then having chan_sip load before pbx_config. (issue #9658 reported by dlynes) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@62990 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-02Add a small message that we're doing something. On my systems, there's a longOlle Johansson
dead period with a non-responsive CLI after I issue "load chan_sip.so" git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@62656 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-02More username body parts to fix... If working, this needs to be backported ↵Olle Johansson
to 1.2, 1.4. But first, some serious SIP testing :-) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@62655 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-02Handle sip:username;parameter=12345@example.com;parameter=1234 URI's properlyOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@62640 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-02Merged revisions 62624 via svnmerge from Olle Johansson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r62624 | oej | 2007-05-02 08:15:43 +0200 (Wed, 02 May 2007) | 2 lines Don't unlock a channel that we already know does not exist (propably isue 8228) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@62639 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-30Add support for setting the CoS for VLAN traffic (802.1p) in Linux. TheRussell Bryant
file doc/qos.tex has been updated to document the new functionality. (issue #9540, patch submitted by IgorG) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@62457 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-30Don't crash when invalid arguments are provided to the CHANNEL() functionRussell Bryant
for a SIP channel. (issue #9619, reported by jtodd, original patch by Corydon76, committed patch slightly modified by me) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@62416 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-28Merge changes from team/russell/eventsRussell Bryant
This set of changes introduces a new generic event API for use within Asterisk. I am still working on a way for events to be shared between servers, but this part is ready and can already be used inside of Asterisk. This set of changes introduces the first use of the API, as well. I have restructured the way that MWI (message waiting indication) is handled. It is now event based instead of polling based. For example, if there are a bunch of SIP phones subscribed to mailboxes, then chan_sip will not have to constantly poll the mailboxes for changes. app_voicemail will generate events when changes occur. See UPGRADE.txt and CHANGES for some more information on the effects of these changes from the user perspective. For developer information, see the text in include/asterisk/event.h. As always, additional feedback is welcome on the asterisk-dev mailing list. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@62292 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-27Issue #9545 Autocomplete for "sip unregister" cli command. (eliel) Thanks!Olle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@62141 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-27Merged revisions 62137 via svnmerge from Olle Johansson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r62137 | oej | 2007-04-27 16:04:07 +0200 (Fri, 27 Apr 2007) | 12 lines Merged revisions 62126 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r62126 | oej | 2007-04-27 15:57:45 +0200 (Fri, 27 Apr 2007) | 4 lines Issue #7351 - SIP Cancel fails due to the wrong contact uri. Reported by PPYY, failed to fix by OEJ final fix by wojtekka - THANKS!!!! THis was a hard one to catch. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@62140 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-24Merged revisions 61772 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r61772 | file | 2007-04-24 12:07:02 -0400 (Tue, 24 Apr 2007) | 10 lines Merged revisions 61771 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r61771 | file | 2007-04-24 12:05:06 -0400 (Tue, 24 Apr 2007) | 2 lines Allow RFC2833 to be sent in the response SDP when an INVITE comes in without SDP. (issue #9546 reported by mcrawford) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61773 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-20Use the last line in the SDP, even if it has no CRLF. Remember Jon Postel :-)Olle Johansson
This code exists in 1.2 and 1.4 but was removed from trunk for some unknown reason. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61689 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-13added CLI 'sip unregister <peer>' for issue 9326. thanks elielDwayne M. Hubbard
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61660 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-13Merged revisions 61648 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r61648 | file | 2007-04-13 13:19:53 -0400 (Fri, 13 Apr 2007) | 2 lines For those very verbose SIP implementations that attach tons of info to the Contact header... let's increase our variable sizes. (issue #9535 reported by jeffg) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61649 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-13Merged revisions 61641 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r61641 | file | 2007-04-13 12:32:03 -0400 (Fri, 13 Apr 2007) | 2 lines Don't assume the callid of a dialog will be set, as in some circumstances it may not. (issue #9534 reported by tecnoxarxa) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61642 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-12Don't treat a host lookup as failed if sipregs is not in use when doing a ↵Joshua Colp
realtime lookup. (issue #9255 reported by sergee) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61618 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-11Merged revisions 61477 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r61477 | russell | 2007-04-11 11:05:29 -0500 (Wed, 11 Apr 2007) | 13 lines Merged revisions 61476 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r61476 | russell | 2007-04-11 11:01:25 -0500 (Wed, 11 Apr 2007) | 5 lines If someone sets the "useragent" option in sip.conf to be empty, then don't add the User-Agent header at all. It is an optional header, anyway. Also, the bug report says that some of Japan's SIP providers don't allow it for some weird reason. (issue #9488, reported by makoto, fixed by me) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61478 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-11Merged revisions 61427 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r61427 | russell | 2007-04-11 10:09:39 -0500 (Wed, 11 Apr 2007) | 14 lines Merged revisions 61426 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r61426 | russell | 2007-04-11 10:05:36 -0500 (Wed, 11 Apr 2007) | 6 lines Fix a bug with switching between host=dynamic and using specific hosts for peers. The code would only reset the peer's address when it is dynamic if it was a new peer structure. Now, it will also reset the address if it was already in the peer list, but before the reload, it was not dynamic. (issue #9515, reported by caio1982, fixed by me) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61428 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-11Merged revisions 61377 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r61377 | russell | 2007-04-11 09:04:44 -0500 (Wed, 11 Apr 2007) | 13 lines Merged revisions 61376 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r61376 | russell | 2007-04-11 09:02:54 -0500 (Wed, 11 Apr 2007) | 5 lines Remove the attempt at reporting configuration errors in sip.conf. This can cause a bunch of improper messages when using realtime. I give up. As oej tried to convince me when I put this in, there is just no easy way to do it. (inspired by a message on the -dev list) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61379 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-11Remove duplicate prototype declaration. (issue #9517 reported by junky)Joshua Colp
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61375 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-10Merged revisions 60989 via svnmerge from Steve Murphy
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r60989 | murf | 2007-04-09 12:32:07 -0600 (Mon, 09 Apr 2007) | 1 line This is a big improvement over the current CDR fixes. It may still need refinement, but this won't have as many folks bothered. This also adds the mods from 1.4/r.61136; ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61152 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-09Merged revisions 61072 via svnmerge from Olle Johansson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r61072 | oej | 2007-04-09 21:58:17 +0200 (Mon, 09 Apr 2007) | 11 lines Merged revisions 61038 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r61038 | oej | 2007-04-09 21:38:59 +0200 (Mon, 09 Apr 2007) | 3 lines - Don't send ActionID before Response: header. - Don't use a blank in an AMI header ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61073 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-09use "ChannelType" in events to indicate which channel driver that generates ↵Olle Johansson
the event. This replaces "ChannelDriver" and "Channel", previously used to indicate channel driver. ChannelType is more in line with "core show channeltypes" git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@60987 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-09Make RTP session ID and session version generation random. (issue #9456 ↵Joshua Colp
reported by tjardick) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@60898 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-09Add counter for sip show registry CLI command. (issue #9352 reported by junky)Joshua Colp
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@60796 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-06Use the same parameter to the two "Registry" AMI events - ChannelDriverOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@60531 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-05Merged revisions 60214 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r60214 | file | 2007-04-05 08:55:02 -0400 (Thu, 05 Apr 2007) | 10 lines Merged revisions 60213 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r60213 | file | 2007-04-05 08:52:50 -0400 (Thu, 05 Apr 2007) | 2 lines Only unlock our pvt and net locks if we are actually going to try to lock the owner again. (issue #9472 reported by zoa) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@60215 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-04Merged revisions 60112 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r60112 | russell | 2007-04-04 11:49:45 -0500 (Wed, 04 Apr 2007) | 3 lines Add a Content-Length of 0 to the response built by transmit_response_with_unsupported(). (issue #9454, reported by makoto, fixed by me) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@60114 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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