Russell Bryant
5b544349d9
Merged revisions 73768 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r73768 | russell | 2007-07-06 18:01:22 -0500 (Fri, 06 Jul 2007) | 4 lines
If a sip_pvt struct has already registered an extension state callback,
remove the old one before adding a new one. If this isn't done, Asterisk
will crash. (issue #10120 )
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@73769 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Russell Bryant
1e588db75a
Merged revisions 73678 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r73678 | russell | 2007-07-06 10:55:41 -0500 (Fri, 06 Jul 2007) | 7 lines
(closes issue #10125 )
Reported by: makoto
Patches submitted by: makoto
This fixes a crash in chan_sip that happens when the bindaddr setting is not
valid on Asterisk startup, gets fixed, and then a reload gets issued.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@73679 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Mark Michelson
757160d5c9
Merged revisions 73674 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r73674 | mmichelson | 2007-07-06 10:26:40 -0500 (Fri, 06 Jul 2007) | 5 lines
Fixed a bug wherein agents get stuck busy. (issue 9618, reported by jiddings, patched by moi)
closes issue #9618
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@73675 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Russell Bryant
be09062a6a
Fix a crash in chan_sip. Don't try to stop the monitor thread if it was never
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started. (closes issue #10124 , reported by gzero, fixed by me)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@73598 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Russell Bryant
cf2c26a948
copy from the correct buffer when deferring a full frame
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(related to issue #9937 )
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@73555 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Russell Bryant
be1502faa3
* Store the call number that a thread is processing without the full frame bit
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set to ease debugging
* When deferring a full frame for processing, stick it into the queue for the
thread that is processing frames for that call, not the one that read the
current frame and is about to go back into the idle list
(related to issue #9937 )
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@73551 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Kevin P. Fleming
3b8be36363
Merged revisions 73547 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r73547 | kpfleming | 2007-07-05 17:11:51 -0500 (Thu, 05 Jul 2007) | 2 lines
we shouldn't allow G.723.1 endpoints to use VAD, just like we don't support it for G.729
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@73548 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Joshua Colp
9b753a0649
Merged revisions 73466 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r73466 | file | 2007-07-05 16:15:18 -0300 (Thu, 05 Jul 2007) | 2 lines
Copy language information to the dialog structure when calling a peer for situations where a PBX may be started on the dialed channel. (issue #10121 reported by clegall_proformatique)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@73467 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Russell Bryant
0041898603
Make this module build for me in dev-mode
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@73398 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Joshua Colp
f912c6ba71
Merged revisions 73318 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r73318 | file | 2007-07-05 10:26:02 -0300 (Thu, 05 Jul 2007) | 2 lines
Actually check to make sure a PBX was started on one of the Local channels instead of blindly assuming it was. (issue #10112 reported by makoto)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@73319 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Christian Richter
a95d9718c9
Merged revisions 73252 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r73252 | crichter | 2007-07-04 16:50:58 +0200 (Mi, 04 Jul 2007) | 1 line
bchannel configurations like echocancel and volume control, need to be setuped on inbound calls too.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@73253 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Christian Richter
17be3d5858
Merged revisions 73207 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r73207 | crichter | 2007-07-04 10:20:54 +0200 (Mi, 04 Jul 2007) | 1 line
bad bug in overlapdial case, we called start_pbx multiple times, because the state wasn't changed..
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@73208 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Christian Richter
1e392965f4
Merged revisions 73004 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r73004 | crichter | 2007-07-03 10:04:35 +0200 (Di, 03 Jul 2007) | 1 line
fixed issue, that misdn_l2l1_check could only be called from mISDN Source channels.. #9449
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@73005 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Christian Richter
07a72b546f
Merged revisions 72585 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r72585 | crichter | 2007-06-29 15:08:26 +0200 (Fr, 29 Jun 2007) | 1 line
check if the bchannel stack id is already used, if so don't use it a second time. Also added a release_chan lock, so that the same chan_list object cannot be freed twice. chan_misdn does not crash anymore on heavy load with these changes.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@72852 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Christian Richter
8fc4089947
Merged revisions 72099 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r72099 | crichter | 2007-06-27 15:22:37 +0200 (Mi, 27 Jun 2007) | 1 line
simplified generation for dummy bchannels, also we mark them as dummies, so they are not used later as real-bchannels, optimized the RESTART mechanisms, we block a channel now on cause:44, and send out a RESTART automatically, then on reception of RESTART_ACKNOWLEDGE we unblock the channel again.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@72851 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Christian Richter
9dc0104a12
Merged revisions 72087 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r72087 | crichter | 2007-06-27 11:26:53 +0200 (Mi, 27 Jun 2007) | 1 line
simplified channel finding and locking a lot. removed unnecessary #ifdefed areas.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@72850 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Luigi Rizzo
475e73e27d
Add -Wdeclaration-after-statement to AST_DEVMODE flags to catch
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variable declarations in the middle of a block.
Fix the few instances of the above spotted out by the compiler.
All of this has been already done or is not applicable in trunk,
so the merge of this change will be blocked.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@72462 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Joshua Colp
5055793871
Make payload IDs for iLBC/Speex match to our list. Since these are dynamic payloads the other side shouldn't care. (issue #9426 reported by irroot)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@72331 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Kevin P. Fleming
39236d2728
use the proper type for storing group number bits so that if someone specifies 'group=42' it will actually work instead of being silently ignored
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@72205 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Jason Parker
7508fce25b
Don't modify a variable that we don't want modified. Make a copy of it instead.
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Issue 10029, patch by phsultan with slight modifications by me (to remove needless casts).
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@72125 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Christian Richter
2294156c9d
Merged revisions 72040-72041 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r72040 | crichter | 2007-06-27 09:49:27 +0200 (Mi, 27 Jun 2007) | 1 line
for inbound TE calls, we setup the bchannel when we get the CONNECT_ACKNOWLEDGE, to make sure mISDN has everything ready. removed some #if 0 areas which weren't used anymore.
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r72041 | crichter | 2007-06-27 09:54:30 +0200 (Mi, 27 Jun 2007) | 1 line
isdn_lib.c didn't compile
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@72042 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Joshua Colp
bd8be32993
Build a peer as well when hash323 is enabled in users.conf (issue #9599 reported by asagage)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@71576 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Joshua Colp
e973bf0ba9
Minor tweak for queueing up the unhold frame... this will teach me to do bugs while half asleep. (issue #10046 reported by dimas)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@71522 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Joshua Colp
a6895cb26d
Merged revisions 71414 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r71414 | file | 2007-06-24 21:02:49 -0400 (Sun, 24 Jun 2007) | 2 lines
Ignore other URIs after the first in a 300 Multiple Choice response. (issue #10041 reported by homesick)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@71430 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Joshua Colp
2886f912d6
Check to make sure the channel pointer is present before queueing up an unhold frame on it. (issue #10046 reported by dimas)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@71412 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Christian Richter
d0bc56b953
Merged revisions 70341 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r70341 | crichter | 2007-06-20 17:29:09 +0200 (Mi, 20 Jun 2007) | 1 line
fixed a bug that was introduced by copy and paste in the last commit ..bchannels weren't cleaned properly.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@71214 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Christian Richter
795e189aa2
Merged revisions 70672 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r70672 | crichter | 2007-06-21 15:11:29 +0200 (Do, 21 Jun 2007) | 1 line
we activate the bchannels in TE mode on incoming calls only when we want to connect the call.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@71123 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Christian Richter
e1791cab97
Merged revisions 70342 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r70342 | crichter | 2007-06-20 17:42:39 +0200 (Mi, 20 Jun 2007) | 1 line
forgot one place ..
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@71122 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Christian Richter
9b17b11c08
Merged revisions 70311 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r70311 | crichter | 2007-06-20 16:47:59 +0200 (Mi, 20 Jun 2007) | 1 line
on receiption of cause:44 we mark the channel as in use and inform the user about the situation, we need to test the RESTART stuff then. Also shuffled the empty_chan_in_stack function after the bchannel cleaning functions, to avoid race conditions.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@71121 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Christian Richter
933ebc4141
Merged revisions 69887 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r69887 | crichter | 2007-06-19 15:23:04 +0200 (Di, 19 Jun 2007) | 1 line
when we send out a SETUP, but get no response, we should cleanup everything after reception of a hangup.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@71120 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Christian Richter
82fe97efd5
Merged revisions 69053 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r69053 | crichter | 2007-06-13 11:55:54 +0200 (Mi, 13 Jun 2007) | 1 line
restart indicator 0x80 is correct, at least that's what libpri does.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@71118 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Christian Richter
0847f57595
Merged revisions 68887 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r68887 | crichter | 2007-06-12 10:35:22 +0200 (Di, 12 Jun 2007) | 1 line
if the bridged partner is mISDN too we should not send dtmf tones, they are transmitted inband always
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@71106 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Christian Richter
84434f5f2f
Merged revisions 68874 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r68874 | crichter | 2007-06-12 09:48:52 +0200 (Di, 12 Jun 2007) | 1 line
if we have already some digits, we just stop the tones.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@71096 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Russell Bryant
5fb99cbf73
Fix a small typo which ... well ... completely broke chan_iax2. oops!
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(issue #9937 , patch by me)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@71003 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Russell Bryant
dbe2d8eeb6
Put the thread reading from the socket back in the idle list if it deferred the
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processing of a full frame to another thread
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@70883 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Russell Bryant
6e7332a507
If a full frame is received while one of the iax2 threads is in the middle
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of handling a full frame for the same call, queue it up for processing by that
same thread later instead of dropping it.
(issue #9937 , patch by me)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@70866 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Joshua Colp
f1c32710a8
Merged revisions 70551 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r70551 | file | 2007-06-20 18:20:16 -0400 (Wed, 20 Jun 2007) | 2 lines
Don't overwrite the configured username setting upon a REGISTER. (issue #8565 reported by jsmith)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@70552 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Jason Parker
60bf4f35c6
Make sure we clear the previously dialed number if it did not exist.
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Issue 9958.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@70494 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Russell Bryant
85f7646a2a
Merged revisions 70396 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r70396 | russell | 2007-06-20 13:45:38 -0500 (Wed, 20 Jun 2007) | 5 lines
Fix a problem where an established call would not be properly disconnected
when a PRI disconnect is received depending on which cause code was received.
(issue #9588 , original patch by softins, updated patch from jtexter3, and some
additional feedback from mhardeman)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@70397 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Russell Bryant
a0a33670bd
Only attempt to queue a hangup on the owner channel if it actually exists.
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(issue #9795 , patch from zandbelt)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@70084 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Russell Bryant
a2084d8ab8
Fix a crash that could occur when handing device state changes.
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When the state of a device changes, the device state thread tells the extension
state handling code that it changed. Then, the extension state code calls the
callback in chan_sip so that it can update subscriptions to that extension.
A pointer to a sip_pvt structure is passed to this function as the call which
needs a NOTIFY sent. However, there was no locking done to ensure that the pvt
struct didn't disappear during this process.
(issue #9946 , reported by tdonahue, patch by me, patch updated to trunk to use
the sip_pvt lock wrappers by eliel)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@69944 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Tilghman Lesher
759dc00599
Issue 10005 - Segfault with missing arguments, plus fix a missing define for SIP INFO channels
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@69796 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Joshua Colp
a50bc6e3a8
Don't count RTP timeout when involved in a T38 fax session. (issue #9222 reported by ivoc)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@69794 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Joshua Colp
f95038d97e
Merged revisions 69765 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r69765 | file | 2007-06-18 14:13:03 -0400 (Mon, 18 Jun 2007) | 2 lines
Set the peer name on the dialog to the one configured in sip.conf and NOT the username to be used for authentication attempts. (issue #9967 reported by achauvin)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@69775 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Joshua Colp
009f6f9112
Don't defer the BYE till later on a transfer when the transfer itself goes kaboom and has no hope of working.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@69668 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Joshua Colp
3e29d89200
Few minor transfer tweaks. We can't unlock something we never locked, and better handle a specific scenario with doing an attended transfer between two non-bridged calls.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@69661 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Joshua Colp
2c9ffadffb
Fix issue where it would be possible for the negotiated codecs to get set back to nothing. (issue #9992 reported by yehavi)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@69625 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Kevin P. Fleming
ae82d97c6d
use ast_localtime() in every place localtime_r() was being used
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@69392 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Joshua Colp
b86b8229c8
Whoops...
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@69222 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Joshua Colp
9604f04ed5
Let's make chan_iax2 media only native transfers actually work. (issue #9376 reported by simone cittadini)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@69221 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Joshua Colp
7572eb866a
Add TXMEDIA to list so that it is properly displayed during iax2 packet output.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@69184 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Russell Bryant
93f3abb3e8
Move the logic for destroying a call when no response is received to a BYE
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outside of the block that checks for FLAG_FATAL to be set. This flag is only
set when the packet is transmitted with the reliability set to XMIT_CRITICAL
when the original packet is transmitted. A BYE is always sent with it set
to XMIT_RELIABLE, meaning this code could never be encountered. This resulted
in seeing some SIP channels that would never go away with the last packet
sent being a BYE.
(part of issue #9235 , patch from jcmoore)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@69183 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Russell Bryant
f56c3be8ad
Clarify a bit of logic. This doesn't change behavior in any way, but it is
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helpful when following the logic to debug problems like 9235.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@69071 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Russell Bryant
0072ea9933
Fix a place where a chan_iax2 pvt struct was accessed without the lock held.
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This issue was reported to me via email by Dmitry Mishchenko. Thanks!
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@69069 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Joshua Colp
200248f687
Change the full frame dropping log message to debug to avoid future bug reports.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@69014 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Joshua Colp
9d7dae818f
Schedule the sending of a PING packet a second later than previously so that it does not collide with the LAGRQ.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@69012 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Christian Richter
48f8dc5340
Merged revisions 68732 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r68732 | crichter | 2007-06-11 18:49:00 +0200 (Mo, 11 Jun 2007) | 1 line
added check for NULL Pointer when calling misdn_new. Asterisk does not allow us to create channels anymore when stop gracefully is used :). also modified the restart_indicator to 0
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@68733 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Christian Richter
7d491a7c24
Merged revisions 68631 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r68631 | crichter | 2007-06-11 11:18:01 +0200 (Mo, 11 Jun 2007) | 1 line
fixed problem that the dummybc chanels had no lock, checking for the lock now. Also fixed the channel restart stuff, we can now specify and restart particular channels too.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@68644 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Kevin P. Fleming
f4b7a2c9cb
actually remember the type/subclass of full frames that are in process
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@68450 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Kevin P. Fleming
e816700d6e
some improvements to the IAX2 full frame dropping logic recently added:
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- use inaddrcmp(), since we have it
- output the type of frame and subclass being dropped, and the type/subclass that is already being processed (which caused the drop)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@68313 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Russell Bryant
e7cb53e8b2
Fix loading persistent queue members when using realtime configuration for queues.
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Also, remove an unneeded leading slash for the astdb family.
(issue #9911 , patch by atis)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@68280 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Jason Parker
51f1d28d2d
Fix an issue with newer phones which require packets be padded out to the correct length.
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Issue 9887, patch by DEA.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@68249 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Olle Johansson
2d491b0eae
Disable chan_features by default in menuselect
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@68196 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Joshua Colp
b58a48d672
Merged revisions 67938 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r67938 | file | 2007-06-06 20:09:13 -0400 (Wed, 06 Jun 2007) | 2 lines
Only notify the devicestate system of a peer state change when the peer is built from the config file. (issue #9900 reported by arkadia)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@67941 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Russell Bryant
6595debbc5
Fix a crash when doing call pickups with SIP phones. The code unlocked the
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channel when it should not have.
(issue #9652 , reported by corruptor, fixed by me)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@67862 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Russell Bryant
83744c244a
Suppress a bunch of debug output unless option_debug is on
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@67457 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Jason Parker
4a7ae2ebde
Correctly update date/time on devices throughout the life of the device, instead of just at registration.
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Issue 9152, yet another patch by DEA.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@67421 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Christian Richter
619cdb016a
Merged revisions 67307 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r67307 | crichter | 2007-06-05 17:42:03 +0200 (Di, 05 Jun 2007) | 1 line
briding is a bool, fixed copy and paste issue.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@67334 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Christian Richter
b0dd31cb52
Merged revisions 67306 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r67306 | crichter | 2007-06-05 17:39:43 +0200 (Di, 05 Jun 2007) | 1 line
simplified the EVENT_SETUP handling in the cb_events function a lot. Commented the different possibilities a bit and made functions of shared code. When the dialed extension does not exist in the extensions.conf we'll jump into the 'i' extension if this does exist, else we disconnect the call with the cause:1 = No Route to Destination.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@67329 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Joshua Colp
24922406f4
Only muck with the thread structure if an idle one was found/created.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@67304 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Kevin P. Fleming
5a5d549b1d
ensure that a burst of full frames (AST_FRAME_DTMF being the prime example) will not be processed out of order... this is a brute force fix, but seems to be the safest fix for now (thanks to the Digium PQ department for finding this bug)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@67270 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Christian Richter
9f749257ef
Merged revisions 67209 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r67209 | crichter | 2007-06-05 12:05:45 +0200 (Di, 05 Jun 2007) | 1 line
added possibility to deactivate bridging per port
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@67210 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Russell Bryant
3b5bd316ed
Fix up a bunch of places where the iax2 pvt structure can disappear and the
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code did not account for it and crashes.
(issues #9642 , #9569 , #9666 , probably others ... based on the work by
stevedavies and mihai, with additional changes from me)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@67158 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Jason Parker
0b7607ad55
Fix for skinny keepalives.
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If there is no traffic from the phone for (keep_alive * 1100) ms (arbitrarily
adding 10% for network issues, etc), unregister the device.
Issue 8394, patch by DEA.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@67156 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Jason Parker
6b150d7b9c
Fixes for dtmf/dialing with mgcp (similar to the recent fix for chan_skinny)
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Issue 9855, patch by DEA.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@67121 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Russell Bryant
885d846739
Add comments for two functions that get called with the appropriate call locked,
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but perform operations that could result in the pvt structure getting destroyed
before returning again, causing numerous seg faults all over the module.
(inspired by issues #9642 , #9569 , and #9666 , and the work done by stevedavies
and mihai)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@67119 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Joshua Colp
01456184f9
Better handle SIP devices that say they have SDP content... but really don't. (issue #9398 reported by mthomasslo)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@67068 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Russell Bryant
0b7719e9a1
Resolve a deadlock in chan_iax2. When handling an implicit ACK to a frame that
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was marked as the final transmission for a call, don't call iax2_destroy() for
that call while the global frame queue is still locked. There is a very nice
explanation of the deadlock in the report.
(issue #9663 , thorough report and patch from stevedavies, additional positive
test reports from mihai and joff_oconnell)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@67020 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Russell Bryant
0ecb2ecf42
Changes to the way DTMF is handled in the core broke dialing in chan_skinny.
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This patch makes chan_skinny usable again. I did not end up testing this,
but there are multiple positive test reports listed in the bug report.
(issue #9596 , reported by pj, testing by pj and mvanbaak, and the fix was
written by DEA)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@66881 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Joshua Colp
c7112015ba
Merged revisions 66764 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r66764 | file | 2007-05-31 12:12:39 -0400 (Thu, 31 May 2007) | 2 lines
It is now possible for this path of execution to have the frame pointer be NULL, therefore we need to check for it before trying to access it. (issue #9836 reported by barthpbx)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@66768 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Joshua Colp
612f61a9b2
Silly me for having out of date source! Oh well... I'm still leaving my comment.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@66639 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Joshua Colp
3d8d697e34
When calling some peer/host that may not exist/reply back... don't keep the dialog in memory for all of eternity.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@66637 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Joshua Colp
7905fca326
Change how channel names are generated a bit. (issue #9825 reported by eldadran)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@66602 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Olle Johansson
332eabcc07
Properly handle 408 request timeout - according to the RFC, the dialog dies if a request in a dialog gets this response.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@66503 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Olle Johansson
8d06f379fe
Don't issue hangup on hangup on hangup on hangup (for jcmoore)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@66474 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Olle Johansson
9f15005143
Don't reset hangupcause if we already have one
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@66414 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Olle Johansson
ff9e2751c6
Tracking down hanging channels, killing them one by one. Issue #9235 and related
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@66404 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Olle Johansson
bab6473879
Merged revisions 66349 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r66349 | oej | 2007-05-29 09:53:14 +0200 (Tue, 29 May 2007) | 2 lines
Issue #9802 - Change inuse counter on CANCEL
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@66363 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Joshua Colp
3a5a649415
Make the usedistinctiveringdetection option work again. (issue #9823 reported by premeau)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@66312 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Jason Parker
a8ab3e5738
I don't know what this was trying to do, but it's clearly incorrect.
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Issues 9808 and 9809.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@66244 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Kevin P. Fleming
5f9897ba56
handle the GNUTLS library properly in the configure script and build system
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don't build in OSP support unless we have found and are allowed to use SSL support
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@66157 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Kevin P. Fleming
0b14e93d00
oops, use #ifdef instead of #if
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65967 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Kevin P. Fleming
e49f14e6cb
don't reference GnuTLS headers and functions unless the configure script found it
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65966 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Olle Johansson
4eab565b25
Issue 7672 - fix by zandbelt - Asterisk core dump since the GnuTLS interface did not support multithreading correctly.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65901 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Olle Johansson
e8addde3f1
Issue 8193 - NAT issues with gtalk/STUN. Patch by phsultan. Thanks!
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65892 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Olle Johansson
abc8683ce7
Issue 7686, fix by phsultan, NAT issues when calling from gtalk to SIP over nat.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65857 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Olle Johansson
e4f9cc15e6
Issue #8536 - Caller ID not set in CDR for jingle
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65841 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Joshua Colp
0df2a42f96
Merged revisions 65837 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r65837 | file | 2007-05-24 10:40:38 -0400 (Thu, 24 May 2007) | 2 lines
Allow RFC2833 to be negotiated when an INVITE comes in without SDP and is not matched to a user or peer. (issue #9546 reported by mcrawford)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65839 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Olle Johansson
6cfe6a550e
Issue 8409 - phsultan - Fix "login" as component to jabber server.
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...and, by accident, fix a bug in chan_sip for stopping a loop on retransmits
of BYE requests.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65836 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Christian Richter
9f54cd55af
Merged revisions 65767 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r65767 | crichter | 2007-05-24 11:19:58 +0200 (Do, 24 Mai 2007) | 1 line
we should only activate the generator in chan_misdn, when asterisk hask not yet taken the call (WAITING4DIGS state). Alerting audio will be generated fomr asterisk for example.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65768 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Kevin P. Fleming
e1518f42ae
start the delayed PBX when receive voice or video full frames as well, and comment this delayed-PBX activity
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65685 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Kevin P. Fleming
ca6b421be4
Merged revisions 65682 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r65682 | kpfleming | 2007-05-23 16:46:22 -0400 (Wed, 23 May 2007) | 2 lines
ensure that variables are set on a newly created channel before we start a PBX on it
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65683 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Kevin P. Fleming
b89faf596b
clear the 'delay PBX' flag when we are ready to start the PBX
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65680 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Kevin P. Fleming
09dc4253d8
don't start a PBX on a new incoming IAX2 channel until we have some sort of response to our ACCEPT (ACK or anything else)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65679 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Kevin P. Fleming
f608d90283
Merged revisions 65676 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r65676 | kpfleming | 2007-05-23 16:06:13 -0400 (Wed, 23 May 2007) | 2 lines
if we are going to set variables on a newly created channel, it should be done *before* we start the PBX on it
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65677 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Russell Bryant
7398856cea
Merged revisions 65588 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r65588 | russell | 2007-05-23 08:06:17 -0500 (Wed, 23 May 2007) | 3 lines
Revert revision 62417 as someone reported problems with it to Mark. This was
related to issue #9588 .
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65589 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Russell Bryant
c556cc222c
List res_smdi as a dependency for app_voicemail and chan_zap
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(Thanks to mnicholson for pointing it out)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65501 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Christian Richter
e7355ec53b
Merged revisions 65328 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r65328 | crichter | 2007-05-22 09:46:39 +0200 (Di, 22 Mai 2007) | 1 line
we stop the tones only when we're in the pre-call phase, otherwise e.g. when in CONNECTED state we should not stop tones when we receive an Information Message
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65342 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Olle Johansson
4483fa12e8
Merged revisions 65122 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r65122 | oej | 2007-05-18 20:10:46 +0200 (Fri, 18 May 2007) | 2 lines
Not getting an ACK to a 200 OK in the initial invite is critical to the call.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65123 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Olle Johansson
7fe3608300
Merged revisions 65075 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r65075 | oej | 2007-05-18 17:12:09 +0200 (Fri, 18 May 2007) | 5 lines
Issue 9235 - part of the problem, maybe not all. Please retry with this patch (and no
other patch) if you have problems with hanging SIP channels. Thank you.
A special Thank You to WeBRainstorm that gave me access to his system.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65076 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Olle Johansson
50f79ba4b2
- Adding support for putting calls OFF hold with a re-invite with blank SDP. This was a bug found while doing tests at SIPit in Antwerp.
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- In order to not duplicate code, I restructured some of the code for putting calls on/off hold.
Thanks DEA for reminding me. This fix has been asleep in the videocaps branch until now.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@65041 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Christian Richter
012fe116fa
Merged revisions 65007 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r65007 | crichter | 2007-05-18 13:23:11 +0200 (Fr, 18 Mai 2007) | 1 line
fixed a warning regarding Keypad encoding. encode the IE sending_complete at the right position.
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18 years ago
Olle Johansson
73d0ba053b
Issue 9487 - stop media flows at hangup of call
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18 years ago
Christian Richter
d682a74e26
Merged revisions 64902 via svnmerge from
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r64902 | crichter | 2007-05-18 10:24:08 +0200 (Fr, 18 Mai 2007) | 1 line
we *need* to send a PROCEEDING when sending_complete is set, even if need_more_infos is requested.
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18 years ago
Joshua Colp
7a8ca54257
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)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@64754 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Olle Johansson
4ae20ba8e4
Fix auth on BYE. (Different patch than for 1.2)
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18 years ago
Olle Johansson
5acc63d688
Issue #9681 - Handle www-auth on BYE
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18 years ago
Olle Johansson
374b52f717
Final part of issue #9483 - fixing transfer() of sip calls in the dial plan (twilson)
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18 years ago
Olle Johansson
70ad556544
Issue #9439 - properly handle username parameters in SIP uri.
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18 years ago
Olle Johansson
bf1a15b9bf
Merged revisions 64535 via svnmerge from
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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!)
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18 years ago
Olle Johansson
56af259505
Merged following patch with a lot of changes for 1.4
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Merged revisions 64514 via svnmerge from
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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 :-)
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18 years ago
Christian Richter
d17174cfc1
Merged revisions 64513 via svnmerge from
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r64513 | crichter | 2007-05-16 10:23:42 +0200 (Mi, 16 Mai 2007) | 1 line
in the case immediate=yes, we directly jump into the dialplan, where people can use PlayTones to indicate a Dialtone, so we don't need to to that by ourself. also we should not do a dialtone_indicate for incoming calls on a TE port in overlapdialmode.
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18 years ago
Olle Johansson
1f2afa0ff1
Change -2 to XMIT_ERROR to clarify a bit more
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18 years ago
Russell Bryant
b340fcd7d0
Properly handle AST_CONTROL_PROGRESS by just ignoring it. An unknown indication
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will trigger an error and cause sounds to stop, which in this case, is ringing.
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18 years ago
Olle Johansson
e041d57175
Handle network errors, like host or network unreachable, in a better way. This means that
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calls to hosts or qualify (OPTION) messages will fail quicker if the TCP/IP stack tells us
that there is an issue.
Since this is an unconnected UDP socket, we will not get error messages directly
in most cases, but maybe on the second and third try.
This is already implemented in trunk.
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18 years ago
Steve Murphy
8c635fb2c1
As per 9570, worrisome CDR warnings have been removed, that are either not helpful, or not relevant.
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18 years ago
Joshua Colp
9a73c07fb1
This concludes my final adventure with bitmasks and the onhold flag. Would anyone care for some peanuts?
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18 years ago
Joshua Colp
699aa6ad74
Tweak hold flags some more. They can be of three states when active: active, inactive, one direction.
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18 years ago
Joshua Colp
026000bae8
Ensure the onhold flag is set no matter what when being put on hold.
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18 years ago
Jason Parker
c2c91c7fef
Merged revisions 63828 via svnmerge from
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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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18 years ago
Olle Johansson
ca1ae5e81a
Merged revisions 63748 via svnmerge from
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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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18 years ago
Joshua Colp
67a0bbaa83
Do not prematurely go on hold if sendonly was not actually set.
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18 years ago
Matthew Fredrickson
a181c6a777
Merged revisions 63653 via svnmerge from
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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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18 years ago
Joshua Colp
c83c0072bb
Merged revisions 63610 via svnmerge from
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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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18 years ago
Christian Richter
3e63c9d542
Merged revisions 62945,63402,63519 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r62945 | crichter | 2007-05-03 17:39:21 +0200 (Do, 03 Mai 2007) | 1 line
when we're in state WAITING4DIGS, we use the asterisk tone-generator which prods us, so we can't just return -1 in misdn_write in this case. Added a MISDN_KEYPAD channel variable, and fixed the sending of keypad. this enables us to modify the call forward parameters in the switch.
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r63402 | crichter | 2007-05-08 17:07:37 +0200 (Di, 08 Mai 2007) | 1 line
added application misdn_check_l2l1 which tries to pull up the L1/L2 on all ports that have the layers down in a group. It waits then for a timeout. This helps for scenarios where multiple PMP BRIs are grouped together, or where a provider has a faulty PTP Implementation, that looses the L2 after a while.
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r63519 | crichter | 2007-05-09 13:26:16 +0200 (Mi, 09 Mai 2007) | 1 line
release_chan frees ch, so we should never touch ch after release_chan, this may cause segfaults.
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18 years ago
Olle Johansson
d30faa1dc2
Don't retransmit 200 OK's on ignore status. (Reported on asterisk-users)
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18 years ago
Joshua Colp
b50b92ab77
Minor backport of revision 59083 in trunk. Don't queue an unhold frame up if the call was never on hold to begin with.
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18 years ago
Joshua Colp
95de3fbf0c
Merged revisions 62987 via svnmerge from
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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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18 years ago
Christian Richter
460e677ea6
Merged revisions 61357,61770,62885 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r61357 | crichter | 2007-04-11 14:05:57 +0200 (Mi, 11 Apr 2007) | 1 line
some fixes for PMP Hold/Retrieve, it should work now, when briding=no
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r61770 | crichter | 2007-04-24 15:50:05 +0200 (Di, 24 Apr 2007) | 1 line
added lock for sending messages to avoid double sending. shuffled some empty_chans after the cb_event calls, this avoids that a release_complete from a quite different call releases a fresh created setup by accident.
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r62885 | crichter | 2007-05-03 15:59:00 +0200 (Do, 03 Mai 2007) | 1 line
fixed the problem that misdn_write did not return -1 when called with 0 samples in a frame this resultet in a deadlock in some circumstances, when the call ended because of a busy extension. added encoding of keypad.
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18 years ago
Tilghman Lesher
8bcfcfca88
Merged revisions 62691 via svnmerge from
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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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18 years ago
Steve Murphy
55f4eb3e3d
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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18 years ago
Olle Johansson
d5fda03428
Don't unlock a channel that we already know does not exist (propably isue 8228)
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18 years ago
Russell Bryant
3595c5fed1
Merged revisions 62417 via svnmerge from
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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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18 years ago
Russell Bryant
8936804377
Fix a bug that made the "language" setting in zapata.conf not
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functional. (issue #9626 , reported and fixed by sergee)
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18 years ago
Russell Bryant
6b76a95f09
Fix a weird problem where when a caller talking to someone sitting behind an
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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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18 years ago
Olle Johansson
34f9e0c4dd
Merged revisions 62126 via svnmerge from
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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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18 years ago
Olle Johansson
b0121ca8a3
Issue #9608 - fix some annoying DEBUG messages not controlled by option_debug (DEA). Thanks!
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18 years ago
Joshua Colp
0525347958
Merged revisions 62037 via svnmerge from
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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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18 years ago
Kevin P. Fleming
0e525813a8
Merged revisions 61913 via svnmerge from
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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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18 years ago
Russell Bryant
5574da0c12
Merged revisions 61866 via svnmerge from
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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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18 years ago
Russell Bryant
e6aa1145ed
Merged revisions 61862 via svnmerge from
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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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18 years ago
Russell Bryant
77c52fd8bf
Merged revisions 61798 via svnmerge from
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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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18 years ago
Dwayne M. Hubbard
db55615090
Merged revisions 61777 via svnmerge from
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r61777 | dhubbard | 2007-04-24 13:20:31 -0500 (Tue, 24 Apr 2007) | 1 line
removed #if 0 block from chan_phone, chan_zap, and chan_modem restart_monitor()
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18 years ago
Joshua Colp
b271179161
Merged revisions 61771 via svnmerge from
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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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18 years ago
Olle Johansson
32be6621f6
Merged revisions 61685 via svnmerge from
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r61685 | oej | 2007-04-19 09:56:21 +0200 (Thu, 19 Apr 2007) | 3 lines
Send NOTIFY to Contact: in SUBSCRIBE
- as reported by Intertex and Citel. Fixed during SIPit 20 in Antwerp.
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18 years ago
Olle Johansson
dc6d309bdb
#9483 , half of patch by twilson to solve 302 redirect issues
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18 years ago
Joshua Colp
eb631c00a3
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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18 years ago
Steve Murphy
8a386a9816
A fix for chan_oss that resulted from the CDR changes; it helps to use the right info.
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18 years ago
Joshua Colp
c06e063e1a
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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18 years ago
Russell Bryant
5858778f40
Merged revisions 61476 via svnmerge from
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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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18 years ago
Nadi Sarrar
3bd81c31fb
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/branches/1.4@61443 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Russell Bryant
a0d75effd5
Merged revisions 61426 via svnmerge from
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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/branches/1.4@61427 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Russell Bryant
9ce476a89c
Merged revisions 61376 via svnmerge from
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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/branches/1.4@61377 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Nadi Sarrar
4e2ab7de64
Export AOCD variables on misdn_hangup.
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18 years ago
Nadi Sarrar
b31fe9be83
Ignore facility messages in case we don't have a corresponding channel object.
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18 years ago
Nadi Sarrar
5e08746b28
AOCD's are now exported to asterisk channel variables.
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18 years ago
Nadi Sarrar
50d606bbf8
Merged revisions 61170 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r61170 | nadi | 2007-04-10 14:31:45 +0200 (Di, 10 Apr 2007) | 2 lines
msns config parameter defaults to '*'
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18 years ago
Olle Johansson
3e5c59f235
Merged revisions 61038 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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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
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18 years ago
Steve Murphy
7d5a79a0b9
This is a big improvement over the current CDR fixes. It may still need refinement, but this won't have as many folks bothered.
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18 years ago
Tilghman Lesher
1d3fbc214a
Merged revisions 60846 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r60846 | tilghman | 2007-04-08 21:37:18 -0500 (Sun, 08 Apr 2007) | 2 lines
Bug 9505 - If the return value for local_queue_frame is set, then p->lock is no longer valid.
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18 years ago
Tilghman Lesher
dc54d451dc
Off by one error, resulting in a crash (Issue 9500)
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18 years ago
Matthew Fredrickson
e7e3929740
Make sure we check the faxdetect option before doing fax processing
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@60485 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Matthew Fredrickson
96a8eadf29
Merged revisions 60456 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r60456 | mattf | 2007-04-06 12:03:15 -0500 (Fri, 06 Apr 2007) | 2 lines
There should only be one code path for doing DTMF conditionals on channels. This fixes it.
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18 years ago
Joshua Colp
a85156a737
Merged revisions 60213 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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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)
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18 years ago
Russell Bryant
0b94c7c27c
Add a Content-Length of 0 to the response built by transmit_response_with_unsupported().
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(issue #9454 , reported by makoto, fixed by me)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@60112 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Russell Bryant
3e058dab84
Merged revisions 60083 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r60083 | russell | 2007-04-04 11:37:04 -0500 (Wed, 04 Apr 2007) | 4 lines
Fix the return value of handle_common_options() so that it always properly
indicates whether it handled the option or not.
(issue #9455 , reported by Netview, fixed by me)
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18 years ago
Russell Bryant
dff65e38f4
Merged revisions 59938 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r59938 | russell | 2007-04-03 14:15:04 -0500 (Tue, 03 Apr 2007) | 4 lines
Don't attempt to report configuration errors in build_user(). oej pointed out
that for a "friend" entry, this won't work, because all user options are valid
for peers, but not the other way around.
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18 years ago
Russell Bryant
f8773a1e80
Merged revisions 59916 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r59916 | russell | 2007-04-03 13:43:54 -0500 (Tue, 03 Apr 2007) | 3 lines
Make chan_sip report when it encounters an unknown option.
(issue #9440 , reported by nightcrawler)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@59936 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Nadi Sarrar
911a8f501d
Merged revisions 59788,59803 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r59788 | nadi | 2007-04-03 11:37:00 +0200 (Di, 03 Apr 2007) | 2 lines
Use the new sysfs way of mISDN 1.2 to check if a port is NT or not.
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r59803 | nadi | 2007-04-03 12:40:58 +0200 (Di, 03 Apr 2007) | 2 lines
ptp is the 5th bit, not the 4th.
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18 years ago
Christian Richter
30ace959bb
Merged revisions 59623-59624,59639 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r59623 | crichter | 2007-04-02 09:12:24 +0200 (Mo, 02 Apr 2007) | 1 line
we can now make 30 channels on a PRI (before we forgot chan 31..)
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r59624 | crichter | 2007-04-02 09:25:54 +0200 (Mo, 02 Apr 2007) | 1 line
don't be verbose if no need
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r59639 | crichter | 2007-04-02 14:08:12 +0200 (Mo, 02 Apr 2007) | 1 line
added option which allows us to accept incoming SETUP Messages without automatically sending Proceeding or Setup Acknowledge, this is useful with some broken switches and if you want to Release incoming calls without previously having acknowledged them. The new option is noautorespond_on_setup=yes|no default is no, so we don't break the existing behaviour
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18 years ago
Russell Bryant
8acd1e8349
When the IAX2 read callback gets called, return NULL instead of a "null frame".
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This will cause Asterisk to hangup the call instead of keep trying whatever it
was doing. Under normal conditions, this function would *never* be called.
However, the author of this patch says an error will occur that will cause it
to get called every 100 thousand calls or so. When this does happen, it puts
the channel in a loop that eventually brings down the system. So, hangup up
the call is certainly a better alternative. (issue #8286 , john)
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18 years ago
Russell Bryant
a2fc7b9e36
Fix the check that ensures that the CHANNEL function's first argument is "rtpqos".
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Thanks, Corydon. :)
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18 years ago
Russell Bryant
8fdf6e3c47
Merged revisions 59258 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r59258 | russell | 2007-03-27 13:04:02 -0500 (Tue, 27 Mar 2007) | 4 lines
Fix the use of the "sourceaddress" option when "bindaddr" is set to 0.0.0.0
instead of having each interface explicitly listed.
(issue #7874 , patch by stevens)
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18 years ago
Russell Bryant
c0aca26f30
Convert the RTPQOS function to just be additional parameter of the CHANNEL
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function. This way, it will be possible for other RTP based channel drivers
to expose this information in the future.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@59256 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Christian Richter
10760fce2c
Merged revisions 59252 via svnmerge from
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r59252 | crichter | 2007-03-27 15:56:15 +0200 (Di, 27 Mär 2007) | 1 line
fixed #9355
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18 years ago
Tilghman Lesher
0220349ca4
Oops, this should be case insensitive
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18 years ago
Tilghman Lesher
ec028bd1b2
Change this to a single dp function to make oej happy.
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18 years ago
Russell Bryant
cd3238d543
Rename the new dialplan functions to match the variable name
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@59209 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Russell Bryant
fa97f6c381
The AUDIORTPQOS and VIDEORTPQOS variables are not fully functional in some
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because they get set in sip_hangup. So, there are common situations where
the variables will not be available in the dialplan at all. So, this patch
provides an alternate method for getting to this information by introducing
AUDIORTPQOS and VIDEORTPQOS dialplan functions.
(issue #9370 , patch by Corydon76, with some testing by blitzrage)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@59207 65c4cc65-6c06-0410-ace0-fbb531ad65f3
18 years ago
Nadi Sarrar
980b0bc785
* mISDN >= 1.2 provides a dsp pipeline for i.e. echo cancellation modules, make chan_misdn use it.
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* add a check for linux/mISDNdsp.h to configure.ac and update the autogenerated files: 'configure', 'autoconfig.h.in'
(the 'configure' script was not in sync with the latest configure.ac, so the diff is a bit bigger than expected).
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18 years ago
Joshua Colp
3b6f376425
Merged revisions 59194 via svnmerge from
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r59194 | file | 2007-03-23 21:35:49 -0400 (Fri, 23 Mar 2007) | 2 lines
Only try to handle a response if it has a response code. (ASA-2007-011)
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18 years ago
Kevin P. Fleming
cf58b2ba46
don't allow string input to overrun the buffer to hold it (ASA-2007-010)
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18 years ago
Kevin P. Fleming
34f55a9bf6
remove variables that are no longer used (--enable-dev-mode is good, developers should be using it)
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18 years ago
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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18 years ago
Joshua Colp
bfeb3bcf18
Copy len variable as well, should fix remaining IAX2 DTMF issues.
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18 years ago
Christian Richter
30165b1bf2
Merged revisions 58849-58850,59062-59063 via svnmerge from
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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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18 years ago
Olle Johansson
156c178143
Issue #9313 , Asterisk crash on SIP return code 0 (reported by qwerty1979) (ASA-2007-011)
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18 years ago
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
18 years ago
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 )
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18 years ago
Olle Johansson
0eda92609d
Merged revisions 58847 via svnmerge from
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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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18 years ago