Commit Graph

6848 Commits

Author SHA1 Message Date
Jason Parker
2768946cf5 Merged revisions 63830 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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

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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.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@63832 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-10 23:16:33 +00:00
Olle Johansson
aa320037d2 Merged revisions 63749 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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

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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.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@63751 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-10 20:51:59 +00:00
Matthew Fredrickson
e2ca869abd Merged revisions 63654 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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

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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

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@63655 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-09 17:35:54 +00:00
Joshua Colp
7e10164e20 Merged revisions 63611 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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

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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)

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2007-05-09 16:56:45 +00:00
Olle Johansson
c358b18a5a Merged revisions 63532 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@63533 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-09 13:07:44 +00:00
Russell Bryant
314c874d7d I noted this on the dev list but got no response, so I just did it myself.
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
2007-05-08 16:41:35 +00:00
Olle Johansson
a39f95b94f Adding external referenses for doxygen
See http://www.asterisk.org/doxygen/trunk/extref.html


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2007-05-07 18:25:56 +00:00
Olle Johansson
33214f76a7 Adding external reference
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@63229 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-07 18:18:43 +00:00
Olle Johansson
238bcd6c45 Doxyfication... There's a shortage of comments in this file...
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@63228 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-07 18:17:18 +00:00
Joshua Colp
28f4727e75 Lock iax2 pvt structure when passing off to the AMI function, and make sure it exists. (issue #9674 reported by arabe)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@63182 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-06 20:09:18 +00:00
Olle Johansson
d326d84ae0 - Adding some missing spaces
- 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
2007-05-05 08:05:38 +00:00
Steve Murphy
02337303ef a small upgrade to the coding standard, and an update to the code that triggered the upgrade.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@63048 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-04 17:49:20 +00:00
Steve Murphy
3ee0077f04 Added a small bit of code to support the SNOM 360's Record button. Made the 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
2007-05-04 16:37:23 +00:00
Olle Johansson
1b15d8852d Add the new ChannelUpdate event to inform manager clients about the PVT ID 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
2007-05-04 13:56:25 +00:00
Joshua Colp
81cade7a4c Merged revisions 62989 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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

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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)

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2007-05-03 16:45:39 +00:00
Tilghman Lesher
121561076e Merged revisions 62692 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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

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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.

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2007-05-02 17:49:36 +00:00
Steve Murphy
fe7068a51b Merged revisions 62689 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@62690 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-02 17:24:03 +00:00
Russell Bryant
3d409eb793 Update the device state functionality of chan_local such that it will return
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: https://origsvn.digium.com/svn/asterisk/trunk@62673 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-02 15:46:49 +00:00
Olle Johansson
e1ec3f917c Add a small message that we're doing something. On my systems, there's a long
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
2007-05-02 12:12:02 +00:00
Olle Johansson
1d51b2e161 More username body parts to fix... If working, this needs to be backported to 1.2, 1.4.
But first, some serious SIP testing :-)


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2007-05-02 12:00:03 +00:00
Olle Johansson
8fee67c83b Handle sip:username;parameter=12345@example.com;parameter=1234 URI's properly
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@62640 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-02 09:41:03 +00:00
Olle Johansson
daefa6a8b4 Merged revisions 62624 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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)

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2007-05-02 09:35:14 +00:00
Russell Bryant
b419fc1134 Add support for setting the CoS for VLAN traffic (802.1p) in Linux. The
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
2007-04-30 16:16:26 +00:00
Russell Bryant
a91f9b138d Merged revisions 62419 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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

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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)

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2007-04-30 15:59:27 +00:00
Russell Bryant
5cb08adc7a Don't crash when invalid arguments are provided to the CHANNEL() function
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
2007-04-30 15:37:23 +00:00
Russell Bryant
0efe511879 Merged revisions 62331 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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)

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2007-04-29 05:51:18 +00:00
Russell Bryant
7d2102c081 Reformat some of iax2.h and convert comments to doxygen format
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@62295 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-28 21:26:00 +00:00
Russell Bryant
b6b1bf3213 Merge changes from team/russell/events
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.


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2007-04-28 21:01:44 +00:00
Russell Bryant
c2468b4c32 Merged revisions 62218 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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)

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2007-04-27 21:11:46 +00:00
Olle Johansson
240bd841b0 Issue #9545 Autocomplete for "sip unregister" cli command. (eliel) Thanks!
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2007-04-27 14:40:28 +00:00
Olle Johansson
f9c592e50c Merged revisions 62137 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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

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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.


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2007-04-27 14:37:10 +00:00
Joshua Colp
14d8979b2f Merged revisions 62038 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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

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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)

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2007-04-26 16:35:14 +00:00
Kevin P. Fleming
ee95074173 Merged revisions 61914 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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

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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)

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2007-04-25 22:34:58 +00:00
Russell Bryant
dc7514a746 Merged revisions 61870 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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

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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.

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2007-04-25 22:01:37 +00:00
Russell Bryant
891a005706 Merged revisions 61863 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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

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r61862 | russell | 2007-04-25 16:06:22 -0500 (Wed, 25 Apr 2007) | 2 lines

Ensure that callerid settings are reset on a reload.

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2007-04-25 21:15:19 +00:00
Russell Bryant
f5d4a16cda Merged revisions 61799 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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

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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)

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2007-04-25 16:23:00 +00:00
Dwayne M. Hubbard
f3ab33014a removed #if 0 block from chan_zap restart_monitor()
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2007-04-24 19:08:28 +00:00
Joshua Colp
721f85d084 Merged revisions 61772 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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

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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)

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2007-04-24 16:10:10 +00:00
Russell Bryant
b94378b5e6 Merge changes from team/russell/iax2_osp
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: https://origsvn.digium.com/svn/asterisk/trunk@61702 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-20 21:12:53 +00:00
Olle Johansson
49af71c100 Use the last line in the SDP, even if it has no CRLF. Remember Jon Postel :-)
This code exists in 1.2 and 1.4 but was removed from trunk for some unknown reason.


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2007-04-20 08:41:24 +00:00
Dwayne M. Hubbard
34469a8707 added CLI 'sip unregister <peer>' for issue 9326. thanks eliel
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2007-04-13 21:23:10 +00:00
Joshua Colp
4f04ff8597 Merged revisions 61648 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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)

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2007-04-13 17:21:53 +00:00
Steve Murphy
901413c76c Merged revisions 61644 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r61644 | murf | 2007-04-13 11:01:02 -0600 (Fri, 13 Apr 2007) | 1 line

A fix for chan_oss that resulted from the CDR changes; it helps to use the right info.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61646 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-13 17:11:53 +00:00
Joshua Colp
80ec0b13ba Merged revisions 61641 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61642 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-13 16:35:33 +00:00
Joshua Colp
c4c2def716 Don't treat a host lookup as failed if sipregs is not in use when doing a realtime lookup. (issue #9255 reported by sergee)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61618 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-12 19:32:00 +00:00
Russell Bryant
3c0b24bda8 Merged revisions 61477 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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

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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)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61478 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-11 16:06:37 +00:00
Nadi Sarrar
9978dc647b Merged revisions 61342,61372-61373,61443 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r61342 | nadi | 2007-04-11 12:52:28 +0200 (Mi, 11 Apr 2007) | 2 lines

AOCD's are now exported to asterisk channel variables.

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r61372 | nadi | 2007-04-11 15:33:30 +0200 (Mi, 11 Apr 2007) | 2 lines

Ignore facility messages in case we don't have a corresponding channel object.

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r61373 | nadi | 2007-04-11 15:40:26 +0200 (Mi, 11 Apr 2007) | 2 lines

Export AOCD variables on misdn_hangup.

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r61443 | nadi | 2007-04-11 17:39:14 +0200 (Mi, 11 Apr 2007) | 2 lines

Don't export AOCD variables on misdn_hangup anymore, this was mainly a fix for trunk..

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-11 15:48:54 +00:00
Russell Bryant
6b033eea04 Merged revisions 61427 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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

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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)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61428 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-11 15:13:12 +00:00
Russell Bryant
e34c67d308 Merged revisions 61377 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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

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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)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61379 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-11 14:13:08 +00:00
Joshua Colp
9fff461080 Remove duplicate prototype declaration. (issue #9517 reported by junky)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61375 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-11 14:01:53 +00:00