Joshua Colp
067876afbc
Until we can do media level parsing for sendrecv/etc just use the first value found. This crept up when a phone was offered audio+video and returned an inactive video stream. chan_sip thought the phone said to put the person on hold but that was totally wrong. (issue #9319 reported by benbrown)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@59081 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-03-21 03:25:48 +00:00
Joshua Colp
bfeb3bcf18
Copy len variable as well, should fix remaining IAX2 DTMF issues.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@59076 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-03-20 20:42:46 +00:00
Christian Richter
30165b1bf2
Merged revisions 58849-58850,59062-59063 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r58849 | crichter | 2007-03-13 12:58:16 +0100 (Di, 13 Mär 2007) | 1 line
added method standard_dec for dialing out on groups, to avoid conflicts, which caused issues with some ISDN providers
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r58850 | crichter | 2007-03-13 13:58:32 +0100 (Di, 13 Mär 2007) | 1 line
fixed the crypt_keys stuff
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r59062 | crichter | 2007-03-20 10:18:06 +0100 (Di, 20 Mär 2007) | 1 line
avoid sending a disconnect when we already received one.
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r59063 | crichter | 2007-03-20 10:23:22 +0100 (Di, 20 Mär 2007) | 1 line
modified a loglevel
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2007-03-20 13:16:06 +00:00
Olle Johansson
156c178143
Issue #9313 , Asterisk crash on SIP return code 0 (reported by qwerty1979) (ASA-2007-011)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@59037 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-03-18 20:37:06 +00:00
Joshua Colp
6548086af0
Don't assume that the pvt structure will still exist after calling schedule_delivery as it may not. (issue #9278 reported by fmachado)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58923 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-03-15 18:13:21 +00:00
Russell Bryant
4fc7d081d3
Some people like to put "limitonpeer" instead of "limitonpeers" in their
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configuration. While we're at it, support "limitonpeerz" and
"limitonpeerssssss". (inspired by issue #9172 )
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58906 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-03-14 19:18:08 +00:00
Olle Johansson
0eda92609d
Merged revisions 58847 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r58847 | oej | 2007-03-13 12:45:52 +0100 (Tue, 13 Mar 2007) | 2 lines
Issue #9229 - No port in request URI on register to non default SIP ports (neelakantan)
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2007-03-13 11:49:35 +00:00
Olle Johansson
f546453573
Don't hangup the call on OK or errors on MESSAGE and INFO
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inside of a dialog (like video update requests).
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58845 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-03-13 10:03:03 +00:00
Olle Johansson
83bb6718fc
Issue #9251 - Clear From URI from user attributes (tgrman)
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2007-03-13 09:12:16 +00:00
Christian Richter
5923e48d7f
Merged revisions 57034,57523,57753,58558 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r57034 | crichter | 2007-02-28 17:09:27 +0100 (Mi, 28 Feb 2007) | 1 line
fixed bugs.digium.com bugs: #9157 and bugs.beronet.com bugs: #302 , #303 , #304
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r57523 | crichter | 2007-03-02 19:32:51 +0100 (Fr, 02 Mar 2007) | 1 line
fixed typo
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r57753 | crichter | 2007-03-04 11:39:50 +0100 (So, 04 Mar 2007) | 1 line
fixed another place where the out_cause was hardcoded to 16
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r58558 | crichter | 2007-03-09 15:43:58 +0100 (Fr, 09 Mar 2007) | 1 line
we can free channel 31 as well, since we can occupy it
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2007-03-12 13:08:06 +00:00
Christian Richter
160cfaf1c6
added UU transceiving and corect handling for rdnis
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58825 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-03-12 12:43:24 +00:00
Joshua Colp
fa866efb5c
Add matchexterniplocally setting which only substitutes your externip/externhost setting if it matches the localnet setting. I know of at least two people who need opposite settings, so I made it an option! (issue #8821 reported by kokoskarokoska)
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2007-03-12 00:51:16 +00:00
Russell Bryant
8ae4e264d1
Fix a few more places in chan_iax2 where the ast_frame used for receiving a
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frame was not properly initialized.
- Interpolating a frame when the jitterbuffer is in use
- decrypting a frame when IAX2 encryption is on
- frames in an IAX2 trunk
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2007-03-10 18:11:11 +00:00
Joshua Colp
b9124234e9
Merged revisions 58579 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r58579 | file | 2007-03-09 15:46:43 -0500 (Fri, 09 Mar 2007) | 2 lines
If we are unable to lookup the host in a c line we have to abort, otherwise the previous data is gone and we will (potentially) have no data when all is said and done.
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2007-03-09 20:49:47 +00:00
Russell Bryant
2f54661cf8
Add a missing break statement so that handling the above event does not
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incorrectly destroy the channel. (issue #9242 , andrew)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58510 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-03-08 22:06:54 +00:00
Russell Bryant
5603ca69eb
If we receive ZT_EVENT_REMOVED, destroy the specified channel.
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(issue #7256 , tzafrir)
Also, update the configure script to make sure that we don't try to build
chan_zap if the installed version of zaptel does not include ZT_EVENT_REMOVED.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58320 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-03-08 01:01:46 +00:00
Russell Bryant
f34c30860e
(This bug was reported to me by Kinsey Moore)
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Merged revisions 58242 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r58242 | russell | 2007-03-07 12:17:07 -0600 (Wed, 07 Mar 2007) | 7 lines
Fix a problem where the Asterisk channel name could be that of the wrong IAX2
user for a call. This is because the first step of choosing this name is to
look for an IAX2 peer that happens to have the same IP/port number that this
call is coming from and assuming that is it. However, this is not always
correct. So, I have made it change this name after authentication happens
since at that point, we have an exact match.
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2007-03-07 18:19:19 +00:00
Joshua Colp
240355d78b
Ensure we have (or should have) at least one matching codec before attempting early bridge SDP seeding. (issue #9221 reported by marcelbarbulescu)
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2007-03-07 17:52:58 +00:00
Steve Murphy
81ef9a4f18
Merged revisions 58115 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r58115 | murf | 2007-03-06 15:52:52 -0700 (Tue, 06 Mar 2007) | 1 line
Fix for 9220: Eyebeam cannot renew subscriptions for presence info. Reason: re-SUBSCRIBE requests don't include Accept headers, which the rfc says are optional (to put it tersely), (it uses MAY), and luckily, the sip_pvt struct has the format info stored, so we simply leave it if the format is set, and the accept header null.
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2007-03-06 23:10:14 +00:00
Olle Johansson
275abf4e08
Merged revisions 58052 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r58052 | oej | 2007-03-06 21:33:21 +0100 (Tue, 06 Mar 2007) | 2 lines
Change error message to proper message
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58053 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-03-06 20:37:07 +00:00
Russell Bryant
905238839b
Return an error of transmit_response is called without a session.
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(issue #9002 )
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58023 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-03-06 18:01:20 +00:00
Joshua Colp
94c9dcda60
Since chan_iax2 does not support reception of DTMF with duration ensure that it is set to 0 on the frame. (issue #8521 reported by gdhgdh)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@57914 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-03-05 19:19:07 +00:00
Joshua Colp
53b9bc89c0
Merged revisions 57475 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r57475 | file | 2007-03-02 12:02:46 -0500 (Fri, 02 Mar 2007) | 2 lines
If a SIP message comes in and goes to a method handler that requires additional values that may not be present then send back an error.
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2007-03-02 17:06:52 +00:00
Joshua Colp
219f4b4913
Merged revisions 57317 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r57317 | file | 2007-03-01 17:19:32 -0500 (Thu, 01 Mar 2007) | 2 lines
Don't even attempt to optimize things when a proxy channel is involved. It will just explode in weird and unexplaineable ways. (issue #9175 reported by clegall_proformatique)
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2007-03-01 22:21:44 +00:00
Joshua Colp
43f841cadf
Merged revisions 57092 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r57092 | file | 2007-02-28 13:55:45 -0500 (Wed, 28 Feb 2007) | 2 lines
Fix a few more issues with the agent logoff CLI command. (issue #9123 reported by arbrandes)
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2007-02-28 18:57:52 +00:00
Russell Bryant
83ce3720bc
Restore the behavior of Asterisk 1.2 where if a device was not specified in
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alsa.conf, then we just use the system default, instead of creating our own
default of hw:0,0. (issue #9139 )
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@56888 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-26 20:42:21 +00:00
Russell Bryant
dd6b58edcd
Fix a crash in my last change to iax2_indicate(). (issue #9150 )
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@56847 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-26 20:04:13 +00:00
Russell Bryant
268fc8a9cc
Do more complete locking of the chan_iax2_pvt struct in the indicate callback.
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(Problem brought up by Ben Smithurst on the asterisk-dev list)
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2007-02-26 16:51:18 +00:00
Jason Parker
a9e4e0eefa
Make sure to set a speeddials parent on creation.
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Don't crash if hold is pressed when no call is active.
Don't return in places that we shouldn't..
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@56569 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-24 02:02:53 +00:00
Russell Bryant
76a643978e
Merged revisions 56406 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r56406 | russell | 2007-02-23 14:17:56 -0600 (Fri, 23 Feb 2007) | 4 lines
Don't destroy mutexes before unregistering all of the entry points from the core.
Also, fix a potential memory leak from not destroying the locks for all of the
possible call numbers (about 32k of them).
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@56407 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-23 20:20:00 +00:00
Joshua Colp
4565c1483c
Merged revisions 56230 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r56230 | file | 2007-02-22 13:44:24 -0500 (Thu, 22 Feb 2007) | 2 lines
Only change the original or clone channel if it's the channel behind the proxy channel, not if it's just a regular bridged channel.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@56231 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-22 18:49:39 +00:00
Olle Johansson
d67ef52159
Move message from verbose to debug
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@56125 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-22 10:33:55 +00:00
Russell Bryant
9a7d1cc182
Restructure a little bit of code to reduce nesting. There is no functionality
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change here.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@56055 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-22 01:24:10 +00:00
Russell Bryant
71921a8329
Merged revisions 56010 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r56010 | russell | 2007-02-21 18:53:25 -0600 (Wed, 21 Feb 2007) | 3 lines
If we receive a frame that is not in any of the negotiated formats, then drop
it. (potentially issue #8781 and SPD-12)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@56011 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-22 00:57:36 +00:00
Jason Parker
b937d7e8bc
Fix locking issue, and accept "transport-accept" as a valid accept message.
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This should solve issues 8970 and 8503.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@55954 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-21 20:27:08 +00:00
Joshua Colp
3d95841eb3
Add a flag that indicates whether a SIP dialog is an outgoing call or not. SIP_OUTGOING originally did it but it was repurposed to the direction of the last transaction, which can cause update_call_counter to falsely decrease the wrong counters. (please don't hurt me oej) (issue #8943 reported by mdu113)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@55914 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-21 17:18:19 +00:00
Olle Johansson
c0f5102378
Issue #8848 - Turn off lamp more quickly after transfer (decrement inuse early on transferer's call leg)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@55834 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-21 08:32:34 +00:00
Jason Parker
67421802de
Fix segfault when buddy couldn't be found.
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Issue 7764, patch by sailer
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2007-02-21 02:01:36 +00:00
Joshua Colp
30d30fef76
Return behavior I removed. I did not remember that you could just add a localnet entry to make it work.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@55717 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-20 23:57:03 +00:00
Joshua Colp
49749e44a2
Don't test our own address against the localnet settings. At least one person has had issues as a result of this from #7051 so I'm reversing it. (issue #8821 reported by kokoskarokoska)
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2007-02-20 23:08:45 +00:00
Joshua Colp
4f5689f95f
Merged revisions 55669 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r55669 | file | 2007-02-20 17:39:14 -0500 (Tue, 20 Feb 2007) | 2 lines
Defer clearing callback information if channels are up until they are hung up. This ensures the hangup process goes smoothly and no channels get hung in limbo. (issue #8088 reported by kebl0155)
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2007-02-20 22:47:00 +00:00
Jason Parker
e0b88aba12
No need to cast nor free with strdupa (thanks file)
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55555!
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2007-02-20 16:53:45 +00:00
Doug Bailey
c967342095
Changed iax2 process thread to detached to correct memory leak due to left over thread context on thread exit.
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Modified module unload process to avoid deadlocks on pthread cancels
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@55397 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-19 14:52:59 +00:00
Jason Parker
0ade592435
Fix an issue where callerid would not be displayed on some phones.
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Issue 8995, initial patch and research done by wedhorn
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2007-02-17 17:10:09 +00:00
Joshua Colp
7a4bed883e
Merged revisions 55073 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r55073 | file | 2007-02-16 20:09:50 -0500 (Fri, 16 Feb 2007) | 2 lines
Allow chan_sip to handle attended transfers from a SIP phone that is sitting behind chan_agent. Yes folks, all it took was one line of code. (issue #8784 reported by pzieba)
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2007-02-17 01:16:59 +00:00
Joshua Colp
b0df14b32c
Merged revisions 54999 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r54999 | file | 2007-02-16 17:13:45 -0500 (Fri, 16 Feb 2007) | 2 lines
Do not send indications through ast_indicate in chan_agent but instead go directly to the technology. This way when indications are emulated they happen on the Agent channel and do not screw up formats on the channels. (issue #8439 reported by punkgode)
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2007-02-16 22:18:46 +00:00
Olle Johansson
3ca445e34c
Issue #7541 - Handle multipart attachments to SIP messages - even if boundary is quoted.
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2007-02-16 12:06:23 +00:00
Matthew Fredrickson
91e22d1df2
Merged revisions 54373 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r54373 | mattf | 2007-02-14 10:25:49 -0600 (Wed, 14 Feb 2007) | 2 lines
When handling glare on a PRI, move the requested channel rather than hang up the old one. Fix for 8957 and 9011.
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2007-02-14 16:56:40 +00:00
Russell Bryant
7bcf1d913a
Remove a couple of leftover debug messages
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2007-02-13 21:31:22 +00:00
Russell Bryant
b100b69703
If we fail to create the SIP socket, then return -1 from reload_config() so
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that load_module() will return AST_MODULE_LOAD_DECLINE. Otherwise, the console
will just get spammed with error messages every time chan_sip tries to send a
message.
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2007-02-13 19:42:00 +00:00