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2007-05-15XXX-XXX-XXX appears to be the standard ANSI pointcode formatmattf1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@64455 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-14Only print the SS7 UP once. Not every time we get the test messages on the mattf1-2/+3
line. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@64384 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-14Merged revisions 64324 via svnmerge from oej1-17/+17
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: http://svn.digium.com/svn/asterisk/trunk@64325 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-14Merged revisions 64306 via svnmerge from russell1-0/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r64306 | russell | 2007-05-14 14:13:00 -0500 (Mon, 14 May 2007) | 3 lines Properly handle AST_CONTROL_PROGRESS by just ignoring it. An unknown indication will trigger an error and cause sounds to stop, which in this case, is ringing. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@64322 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-14If no port is specified in the outboundproxy setting then use the standard ↵file1-2/+1
SIP port. (issue #9665 reported by tootai) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@64274 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-14Merged revisions 64193 via svnmerge from murf1-2/+14
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r64193 | murf | 2007-05-14 07:58:42 -0600 (Mon, 14 May 2007) | 1 line As per 9570, worrisome CDR warnings have been removed, that are either not helpful, or not relevant. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@64208 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-13Improve handling network errors on transmission to hosts that don't reply or ↵oej1-29/+72
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: http://svn.digium.com/svn/asterisk/trunk@64142 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-12Merged revisions 64114 via svnmerge from file1-5/+5
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: http://svn.digium.com/svn/asterisk/trunk@64115 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-12Merged revisions 64086 via svnmerge from file1-2/+3
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: http://svn.digium.com/svn/asterisk/trunk@64087 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-12Merged revisions 64044 via svnmerge from file1-0/+2
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: http://svn.digium.com/svn/asterisk/trunk@64045 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-11Add/fix support for Redial, Speeddial, and Messages buttons.qwell1-49/+130
Combined effort by DEA and mvanbaak. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@64030 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-10Merged revisions 63830 via svnmerge from qwell1-1/+2
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r63830 | qwell | 2007-05-10 18:15:37 -0500 (Thu, 10 May 2007) | 12 lines Merged revisions 63828 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r63828 | qwell | 2007-05-10 18:14:55 -0500 (Thu, 10 May 2007) | 4 lines Fix an issue with trying to kill a thread before it gets created. Issue 9709, patch by nic_bellamy. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@63832 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-10Merged revisions 63749 via svnmerge from oej1-0/+4
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: http://svn.digium.com/svn/asterisk/trunk@63751 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-09Merged revisions 63654 via svnmerge from mattf1-16/+18
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r63654 | mattf | 2007-05-09 12:25:21 -0500 (Wed, 09 May 2007) | 10 lines Merged revisions 63653 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r63653 | mattf | 2007-05-09 12:20:20 -0500 (Wed, 09 May 2007) | 2 lines Make sure we only create a DSP if it's requested on SUB_REAL ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@63655 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-09Merged revisions 63611 via svnmerge from file1-5/+8
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: http://svn.digium.com/svn/asterisk/trunk@63613 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-09Merged revisions 63532 via svnmerge from oej1-2/+1
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: http://svn.digium.com/svn/asterisk/trunk@63533 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-08I noted this on the dev list but got no response, so I just did it myself.russell1-2/+6
Lock the call features when being used in chan_sip. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@63447 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-07Adding external referenses for doxygenoej1-0/+2
See http://www.asterisk.org/doxygen/trunk/extref.html git-svn-id: http://svn.digium.com/svn/asterisk/trunk@63230 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-07Adding external referenceoej1-0/+3
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@63229 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-07Doxyfication... There's a shortage of comments in this file...oej1-7/+7
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@63228 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-06Lock iax2 pvt structure when passing off to the AMI function, and make sure ↵file1-1/+4
it exists. (issue #9674 reported by arabe) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@63182 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-05- Adding some missing spacesoej1-19/+32
- Correcting error messages - Disabling code that doesn't do anything - Making sure we always respond to this request, happily git-svn-id: http://svn.digium.com/svn/asterisk/trunk@63136 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-04a small upgrade to the coding standard, and an update to the code that ↵murf1-1/+1
triggered the upgrade. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@63048 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-04Added a small bit of code to support the SNOM 360's Record button. Made the ↵murf1-0/+32
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: http://svn.digium.com/svn/asterisk/trunk@63046 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-04Add the new ChannelUpdate event to inform manager clients about the PVT ID ↵oej2-0/+27
and some other channel driver data that is needed to follow the call through the PBX. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@63032 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-03Merged revisions 62989 via svnmerge from file1-0/+2
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: http://svn.digium.com/svn/asterisk/trunk@62990 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-02Merged revisions 62692 via svnmerge from tilghman1-0/+7
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r62692 | tilghman | 2007-05-02 12:43:48 -0500 (Wed, 02 May 2007) | 12 lines Merged revisions 62691 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r62691 | tilghman | 2007-05-02 12:38:16 -0500 (Wed, 02 May 2007) | 4 lines Issue 9638 - if a text frame is sent with no terminating NULL through a bridged IAX connection, the remote end will receive garbage characters tacked onto the end. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@62693 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-02Merged revisions 62689 via svnmerge from murf1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r62689 | murf | 2007-05-02 11:10:50 -0600 (Wed, 02 May 2007) | 1 line a)In chan_zap, set the clid, src fields in channel_alloc call. b)in the channel_alloc func, set the cid_num and name fields from the arglist[blush]. c) don't update the channel app & app data fields if you are in the 'h' extension. d)the load_module func in cdr_radius needs to return DECLINE, SUCCESS. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@62690 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-02Update the device state functionality of chan_local such that it will returnrussell1-2/+14
NOT_INUSE or INUSE when Local channels are in use as opposed to just UNKNOWN. It will still return INVALID if the extension doesn't exist at all. (issue #8048, patch from tim_ringenbach) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@62673 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-02Add a small message that we're doing something. On my systems, there's a longoej1-0/+1
dead period with a non-responsive CLI after I issue "load chan_sip.so" git-svn-id: http://svn.digium.com/svn/asterisk/trunk@62656 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-02More username body parts to fix... If working, this needs to be backported ↵oej1-14/+29
to 1.2, 1.4. But first, some serious SIP testing :-) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@62655 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-02Handle sip:username;parameter=12345@example.com;parameter=1234 URI's properlyoej1-12/+27
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@62640 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-02Merged revisions 62624 via svnmerge from oej1-1/+0
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: http://svn.digium.com/svn/asterisk/trunk@62639 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-30Add support for setting the CoS for VLAN traffic (802.1p) in Linux. Therussell5-53/+74
file doc/qos.tex has been updated to document the new functionality. (issue #9540, patch submitted by IgorG) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@62457 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-30Merged revisions 62419 via svnmerge from russell1-31/+16
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r62419 | russell | 2007-04-30 10:58:28 -0500 (Mon, 30 Apr 2007) | 12 lines Merged revisions 62417 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r62417 | russell | 2007-04-30 10:57:26 -0500 (Mon, 30 Apr 2007) | 4 lines This patch fixes an issue where depending on the cause code, when the network sends a PRI disconnect, the call may not be properly hung up. (issue #9588, reported and patched by softins) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@62422 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-30Don't crash when invalid arguments are provided to the CHANNEL() functionrussell1-2/+8
for a SIP channel. (issue #9619, reported by jtodd, original patch by Corydon76, committed patch slightly modified by me) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@62416 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-29Merged revisions 62331 via svnmerge from russell1-2/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r62331 | russell | 2007-04-29 00:50:37 -0500 (Sun, 29 Apr 2007) | 3 lines Fix a bug that made the "language" setting in zapata.conf not functional. (issue #9626, reported and fixed by sergee) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@62332 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-28Reformat some of iax2.h and convert comments to doxygen formatrussell1-142/+172
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@62295 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-28Merge changes from team/russell/eventsrussell4-115/+218
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: http://svn.digium.com/svn/asterisk/trunk@62292 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-27Merged revisions 62218 via svnmerge from russell1-4/+2
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r62218 | russell | 2007-04-27 16:10:51 -0500 (Fri, 27 Apr 2007) | 11 lines Fix a weird problem where when a caller talking to someone sitting behind an agent channel sent a digit, the digit would be played to the agent for forever. This is because chan_agent always returned -1 from its send_digit_begin and _end callbacks. This non-zero return value indicates to the Asterisk core that it would like an inband DTMF generator put on the channel. However, this is the wrong thing to do. It should *always* return 0, instead. When the digit begin and end functions are called on the proxied channel, the underlying channel will indicate whether inband DTMF is needed or not, and the generator will be put on that one, and not the Agent channel. (issue #9615, #9616, reported by jiddings and BigJimmy, and fixed by me) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@62219 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-27Issue #9545 Autocomplete for "sip unregister" cli command. (eliel) Thanks!oej1-3/+35
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@62141 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-27Merged revisions 62137 via svnmerge from oej1-0/+2
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: http://svn.digium.com/svn/asterisk/trunk@62140 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-26Merged revisions 62038 via svnmerge from file1-2/+2
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r62038 | file | 2007-04-26 12:33:52 -0400 (Thu, 26 Apr 2007) | 10 lines Merged revisions 62037 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r62037 | file | 2007-04-26 12:30:57 -0400 (Thu, 26 Apr 2007) | 2 lines Revert previous fix for when the IAX2 channel goes funky (that's the technical term). This is causing legit calls to be prematurely hung up. (issue #9600 reported by justdave) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@62039 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-25Merged revisions 61914 via svnmerge from kpfleming1-0/+9
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r61914 | kpfleming | 2007-04-25 17:29:53 -0500 (Wed, 25 Apr 2007) | 10 lines Merged revisions 61913 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r61913 | kpfleming | 2007-04-25 17:24:59 -0500 (Wed, 25 Apr 2007) | 2 lines handle a very bizarre race condition with channels being redirected before a simple switch can be started on them (issue #9286) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@61915 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-25Merged revisions 61870 via svnmerge from russell1-0/+24
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r61870 | russell | 2007-04-25 16:59:07 -0500 (Wed, 25 Apr 2007) | 10 lines Merged revisions 61866 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r61866 | russell | 2007-04-25 16:55:23 -0500 (Wed, 25 Apr 2007) | 2 lines If the callerid= option is specified, but empty, clear any previous data. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@61876 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-25Merged revisions 61863 via svnmerge from russell1-20/+38
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r61863 | russell | 2007-04-25 16:13:15 -0500 (Wed, 25 Apr 2007) | 10 lines Merged revisions 61862 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r61862 | russell | 2007-04-25 16:06:22 -0500 (Wed, 25 Apr 2007) | 2 lines Ensure that callerid settings are reset on a reload. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@61864 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-25Merged revisions 61799 via svnmerge from russell1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r61799 | russell | 2007-04-25 11:22:07 -0500 (Wed, 25 Apr 2007) | 11 lines Merged revisions 61798 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r61798 | russell | 2007-04-25 11:20:38 -0500 (Wed, 25 Apr 2007) | 3 lines Fix a typo where cid_num got copied instead of cid_ani. (issue #9587, reported and patched by xrg) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@61800 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-24removed #if 0 block from chan_zap restart_monitor()dhubbard1-7/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@61784 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-24Merged revisions 61772 via svnmerge from file1-0/+2
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: http://svn.digium.com/svn/asterisk/trunk@61773 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-20Merge changes from team/russell/iax2_osprussell4-0/+136
This set of changes adds OSP support to chan_iax2. However, I have modified the patch a bit from what was submitted. You now use the CHANNEL() function to get and set the OSP token for IAX2. (issue #8531, reported by and original patch by homesick, patch updated by me) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@61702 f38db490-d61c-443f-a65b-d21fe96a405b
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