Commit Graph

4510 Commits

Author SHA1 Message Date
Joshua Colp 62084eb2a4 Add SIPREFERRINGCONTEXT and SIPREFERREDBYHDR variables when a transfer takes place. (issue #8378 reported by jcovert)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@72354 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-27 23:13:09 +00:00
Joshua Colp 4897378ff1 Merged revisions 72331 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r72331 | file | 2007-06-27 18:58:02 -0400 (Wed, 27 Jun 2007) | 2 lines

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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2007-06-27 22:58:53 +00:00
Matthew Fredrickson 7f233a8dc2 Let's NOT create a deadlock scenario here
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@72304 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-27 21:44:13 +00:00
Kevin P. Fleming 86fcb89338 Merged revisions 72205 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r72205 | kpfleming | 2007-06-27 14:13:21 -0500 (Wed, 27 Jun 2007) | 2 lines

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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2007-06-27 19:13:54 +00:00
Jason Parker 8bf745194f Merged revisions 72125 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r72125 | qwell | 2007-06-27 12:10:32 -0500 (Wed, 27 Jun 2007) | 4 lines

Don't modify a variable that we don't want modified.  Make a copy of it instead.

Issue 10029, patch by phsultan with slight modifications by me (to remove needless casts).

Note: chan_jingle in trunk does not appear to have the same bug.

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2007-06-27 17:14:31 +00:00
Christian Richter bc478d6c41 Merged revisions 72042 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r72042 | crichter | 2007-06-27 09:58:06 +0200 (Mi, 27 Jun 2007) | 13 lines

Merged revisions 72040-72041 via svnmerge from 
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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2007-06-27 08:08:58 +00:00
Joshua Colp 1961b57705 Add rtpdest option to SIP CHANNEL() dialplan function to return the IP address and port that RTP (be it audio/video/text) is going to.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@71988 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-26 23:31:23 +00:00
Joshua Colp d77301b8cd Tweak CLI command completion and some help text. (issue #10049 reported by IgorG)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@71613 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-25 15:35:10 +00:00
Joshua Colp b7cd8648b5 Merged revisions 71576 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r71576 | file | 2007-06-25 10:13:45 -0400 (Mon, 25 Jun 2007) | 2 lines

Build a peer as well when hash323 is enabled in users.conf (issue #9599 reported by asagage)

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2007-06-25 14:15:05 +00:00
Joshua Colp b8d64002b2 Merged revisions 71522 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r71522 | file | 2007-06-25 09:03:03 -0400 (Mon, 25 Jun 2007) | 2 lines

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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2007-06-25 13:04:36 +00:00
Christian Richter 1e6b6d2df5 Merged revisions 71214 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r71214 | crichter | 2007-06-23 00:44:42 +0200 (Sa, 23 Jun 2007) | 9 lines

Merged revisions 70341 via svnmerge from 
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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2007-06-25 09:46:18 +00:00
Christian Richter d72da1523d Merged revisions 71123 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r71123 | crichter | 2007-06-22 17:38:08 +0200 (Fr, 22 Jun 2007) | 9 lines

Merged revisions 70672 via svnmerge from 
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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2007-06-25 09:44:35 +00:00
Christian Richter 71f6d61265 Merged revisions 71122 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r71122 | crichter | 2007-06-22 17:34:31 +0200 (Fr, 22 Jun 2007) | 9 lines

Merged revisions 70342 via svnmerge from 
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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2007-06-25 09:42:15 +00:00
Christian Richter 829f58b228 Merged revisions 71121 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r71121 | crichter | 2007-06-22 17:32:54 +0200 (Fr, 22 Jun 2007) | 9 lines

Merged revisions 70311 via svnmerge from 
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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2007-06-25 09:37:49 +00:00
Christian Richter e57fa9fa82 Merged revisions 71120 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r71120 | crichter | 2007-06-22 17:30:08 +0200 (Fr, 22 Jun 2007) | 9 lines

Merged revisions 69887 via svnmerge from 
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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2007-06-25 09:34:05 +00:00
Christian Richter f804c7ef5f Merged revisions 71118 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r71118 | crichter | 2007-06-22 17:27:53 +0200 (Fr, 22 Jun 2007) | 9 lines

Merged revisions 69053 via svnmerge from 
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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2007-06-25 09:06:56 +00:00
Christian Richter ebd9ba0456 Merged revisions 71106 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r71106 | crichter | 2007-06-22 17:22:06 +0200 (Fr, 22 Jun 2007) | 9 lines

Merged revisions 68887 via svnmerge from 
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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2007-06-25 08:57:06 +00:00
Christian Richter a378c3f852 Merged revisions 71096 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r71096 | crichter | 2007-06-22 17:17:04 +0200 (Fr, 22 Jun 2007) | 9 lines

Merged revisions 68874 via svnmerge from 
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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2007-06-25 08:53:47 +00:00
Joshua Colp 76455dda03 Merged revisions 71430 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r71430 | file | 2007-06-24 21:10:06 -0400 (Sun, 24 Jun 2007) | 10 lines

Merged revisions 71414 via svnmerge from 
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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2007-06-25 01:11:47 +00:00
Joshua Colp 8bab6c1f69 Merged revisions 71412 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r71412 | file | 2007-06-24 20:49:21 -0400 (Sun, 24 Jun 2007) | 2 lines

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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2007-06-25 00:52:03 +00:00
Russell Bryant 8fd5e05635 Merged revisions 71003 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r71003 | russell | 2007-06-21 22:14:41 -0500 (Thu, 21 Jun 2007) | 3 lines

Fix a small typo which ... well ... completely broke chan_iax2.  oops!
(issue #9937, patch by me)

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2007-06-22 03:15:34 +00:00
Russell Bryant 992c8ec757 Merged revisions 70883 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r70883 | russell | 2007-06-21 16:14:53 -0500 (Thu, 21 Jun 2007) | 3 lines

Put the thread reading from the socket back in the idle list if it deferred the
processing of a full frame to another thread

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@70887 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-21 21:16:36 +00:00
Russell Bryant 82151ac23f Merged revisions 70866 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r70866 | russell | 2007-06-21 16:07:04 -0500 (Thu, 21 Jun 2007) | 5 lines

If a full frame is received while one of the iax2 threads is in the middle
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)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@70877 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-21 21:12:03 +00:00
Matthew Fredrickson 8da2de3a6c Remove deprecated function call
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@70769 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-21 18:03:46 +00:00
Joshua Colp 18f4920227 Merged revisions 70552 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r70552 | file | 2007-06-20 18:22:20 -0400 (Wed, 20 Jun 2007) | 10 lines

Merged revisions 70551 via svnmerge from 
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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2007-06-20 22:24:47 +00:00
Jason Parker 8ccfdf3c80 Merged revisions 70494 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r70494 | qwell | 2007-06-20 15:53:16 -0500 (Wed, 20 Jun 2007) | 4 lines

Make sure we clear the previously dialed number if it did not exist.

Issue 9958.

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2007-06-20 20:54:08 +00:00
Russell Bryant a89e4c5083 Merged revisions 70397 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r70397 | russell | 2007-06-20 13:46:49 -0500 (Wed, 20 Jun 2007) | 13 lines

Merged revisions 70396 via svnmerge from 
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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2007-06-20 18:48:34 +00:00
Olle Johansson 3aafb233f2 Show TDD status in "zap show channels"
Inspired by work at Omnitor in Sweden


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@70310 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-20 14:33:34 +00:00
Russell Bryant 3957ce9215 Merged revisions 70084 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r70084 | russell | 2007-06-19 14:13:45 -0500 (Tue, 19 Jun 2007) | 3 lines

Only attempt to queue a hangup on the owner channel if it actually exists.
(issue #9795, patch from zandbelt)

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2007-06-19 19:15:03 +00:00
Steve Murphy c6ed12405f These changes were submitted via bug 6683, to allow CID detection in India, with carriers that do Polarity/DTMF CID signalling.
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2007-06-19 17:07:28 +00:00
Russell Bryant 238b7a54cc Merged revisions 69944 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r69944 | russell | 2007-06-19 10:22:36 -0500 (Tue, 19 Jun 2007) | 10 lines

Fix a crash that could occur when handing device state changes.
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)

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2007-06-19 15:27:16 +00:00
Matthew Fredrickson f408a5405a Add support for setting nature of address, presentation, and other related SS7 number options (#10000)
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2007-06-19 15:14:23 +00:00
Tilghman Lesher a67890d7a9 Merged revisions 69796 via svnmerge from
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r69796 | tilghman | 2007-06-18 14:48:17 -0500 (Mon, 18 Jun 2007) | 2 lines

Issue 10005 - Segfault with missing arguments, plus fix a missing define for SIP INFO channels

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2007-06-18 19:52:56 +00:00
Joshua Colp 9ed0563f17 Merged revisions 69794 via svnmerge from
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r69794 | file | 2007-06-18 15:00:50 -0400 (Mon, 18 Jun 2007) | 2 lines

Don't count RTP timeout when involved in a T38 fax session. (issue #9222 reported by ivoc)

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2007-06-18 19:02:45 +00:00
Joshua Colp 59bc48bd05 Merged revisions 69775 via svnmerge from
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r69775 | file | 2007-06-18 14:18:12 -0400 (Mon, 18 Jun 2007) | 10 lines

Merged revisions 69765 via svnmerge from 
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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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2007-06-18 18:19:54 +00:00
Joshua Colp 1dbfbe6d71 Merged revisions 69668 via svnmerge from
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r69668 | file | 2007-06-18 12:04:55 -0400 (Mon, 18 Jun 2007) | 2 lines

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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2007-06-18 16:06:17 +00:00
Joshua Colp 3e4980da79 Merged revisions 69661 via svnmerge from
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r69661 | file | 2007-06-18 11:46:32 -0400 (Mon, 18 Jun 2007) | 2 lines

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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2007-06-18 15:48:05 +00:00
Joshua Colp 981a94f023 Merged revisions 69625 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r69625 | file | 2007-06-18 09:55:00 -0400 (Mon, 18 Jun 2007) | 2 lines

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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2007-06-18 13:57:33 +00:00
Russell Bryant b179e2155f Convert uses of strdup() to ast_strdup()
(issue #9983, eliel)


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2007-06-14 23:01:01 +00:00
Kevin P. Fleming 5fdba27ea2 Merged revisions 69392 via svnmerge from
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r69392 | kpfleming | 2007-06-14 16:50:40 -0500 (Thu, 14 Jun 2007) | 2 lines

use ast_localtime() in every place localtime_r() was being used

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2007-06-14 22:09:20 +00:00
Russell Bryant 055d82cbce Add a massive set of changes for converting to use the ast_debug() macro.
(issue #9957, patches from mvanbaak, caio1982, critch, and dimas)


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2007-06-14 19:39:12 +00:00
Matthew Fredrickson 60b029aa82 Clean up debug messages a little bit for ss7 linkset debugging
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@69308 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-14 16:41:57 +00:00
Joshua Colp 6a35e6cfb6 Merged revisions 69221 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r69221 | file | 2007-06-13 17:17:28 -0400 (Wed, 13 Jun 2007) | 2 lines

Let's make chan_iax2 media only native transfers actually work. (issue #9376 reported by simone cittadini)

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2007-06-13 21:20:32 +00:00
Russell Bryant 8e11d6c147 Merged revisions 69183 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r69183 | russell | 2007-06-13 14:57:38 -0500 (Wed, 13 Jun 2007) | 9 lines

Move the logic for destroying a call when no response is received to a BYE
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)

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2007-06-13 20:03:03 +00:00
Joshua Colp d127d4b7e3 Merged revisions 69184 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r69184 | file | 2007-06-13 15:58:59 -0400 (Wed, 13 Jun 2007) | 2 lines

Add TXMEDIA to list so that it is properly displayed during iax2 packet output.

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2007-06-13 20:00:56 +00:00
Jason Parker 0c4a3e24b6 Completely remove callback mode and all references to it from chan_agent.
Issue 9969, patch by eliel.


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2007-06-13 19:05:25 +00:00
Jason Parker 63535ada60 Fixes for ast_strlen_zero() janitor project.
Issue 9968, patch by eliel.


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2007-06-13 17:06:53 +00:00
Russell Bryant 156d6338b2 Merged revisions 69071 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r69071 | russell | 2007-06-13 11:56:16 -0500 (Wed, 13 Jun 2007) | 3 lines

Clarify a bit of logic.  This doesn't change behavior in any way, but it is
helpful when following the logic to debug problems like 9235.

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2007-06-13 16:59:42 +00:00
Russell Bryant 65880d7680 Merged revisions 69069 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r69069 | russell | 2007-06-13 11:29:12 -0500 (Wed, 13 Jun 2007) | 3 lines

Fix a place where a chan_iax2 pvt struct was accessed without the lock held.
This issue was reported to me via email by Dmitry Mishchenko.  Thanks!

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2007-06-13 16:29:45 +00:00
Joshua Colp 0b75ad2db9 Merged revisions 69014 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r69014 | file | 2007-06-12 15:36:29 -0400 (Tue, 12 Jun 2007) | 2 lines

Change the full frame dropping log message to debug to avoid future bug reports.

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2007-06-12 19:38:17 +00:00
Joshua Colp 74cda06deb Merged revisions 69012 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r69012 | file | 2007-06-12 15:26:38 -0400 (Tue, 12 Jun 2007) | 2 lines

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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2007-06-12 19:29:12 +00:00
Russell Bryant 9e0458e9f1 Completely remove all of the code related to jumping to priority n + 101. yay!
(issue #9926, caio1982)


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2007-06-12 15:58:28 +00:00
Joshua Colp f519db04e4 Remove old stuff from the AgentCallbackLogin days and only autocomplete agents in the agent logoff CLI command that are logged in. (issue #9952 reported by eliel)
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2007-06-12 13:34:57 +00:00
Christian Richter c2b4e6cef6 Merged revisions 68733 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r68733 | crichter | 2007-06-11 18:57:59 +0200 (Mo, 11 Jun 2007) | 9 lines

Merged revisions 68732 via svnmerge from 
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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2007-06-11 17:05:48 +00:00
Christian Richter cb42eae091 Merged revisions 68644 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r68644 | crichter | 2007-06-11 12:29:18 +0200 (Mo, 11 Jun 2007) | 9 lines

Merged revisions 68631 via svnmerge from 
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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2007-06-11 11:40:21 +00:00
Jason Parker 85683836eb Fix compile errors in chan_misdn.c
Reported by d1mas in #asterisk-bugs on IRC.


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2007-06-09 01:06:40 +00:00
Kevin P. Fleming a08922bb71 Merged revisions 68450 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r68450 | kpfleming | 2007-06-08 10:52:47 -0500 (Fri, 08 Jun 2007) | 2 lines

actually remember the type/subclass of full frames that are in process

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2007-06-08 16:03:01 +00:00
Russell Bryant 1d57ccb6f7 Fix a bunch of doxygen errors and document more things
(issue #9842, snuffy)


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2007-06-07 23:07:25 +00:00
Russell Bryant d2a78872cc Fix a couple of places that got missed in the conversion to using the new API
call for creating detached threads.
(issue #9915, reported by elguro, fixed by me)


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2007-06-07 22:38:59 +00:00
Kevin P. Fleming 2c9173986a Merged revisions 68313 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r68313 | kpfleming | 2007-06-07 17:14:35 -0500 (Thu, 07 Jun 2007) | 6 lines

some improvements to the IAX2 full frame dropping logic recently added:

- 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)


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2007-06-07 22:18:38 +00:00
Russell Bryant e911f41166 Merged revisions 68280 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r68280 | russell | 2007-06-07 16:16:07 -0500 (Thu, 07 Jun 2007) | 4 lines

Fix loading persistent queue members when using realtime configuration for queues.
Also, remove an unneeded leading slash for the astdb family.
(issue #9911, patch by atis)

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2007-06-07 21:17:52 +00:00
Jason Parker 4db176fdc2 Merged revisions 68249 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r68249 | qwell | 2007-06-07 15:25:18 -0500 (Thu, 07 Jun 2007) | 4 lines

Fix an issue with newer phones which require packets be padded out to the correct length.

Issue 9887, patch by DEA.

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2007-06-07 20:25:56 +00:00
Olle Johansson bfc8e1b17d Merged revisions 68196 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r68196 | oej | 2007-06-07 21:46:10 +0200 (Thu, 07 Jun 2007) | 2 lines

Disable chan_features by default in menuselect

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2007-06-07 19:49:15 +00:00
Olle Johansson 1f9d98016e - Doxygen updates
- Adding docs on flags to be able to clean up a bit


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2007-06-07 19:45:32 +00:00
Joshua Colp f2cc861bcf Merged revisions 67941 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r67941 | file | 2007-06-06 20:10:48 -0400 (Wed, 06 Jun 2007) | 10 lines

Merged revisions 67938 via svnmerge from 
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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2007-06-07 00:12:21 +00:00
Matthew Fredrickson 46e054c1af Allow overlapdialing directions to be configurable. Bug #8554
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2007-06-06 23:12:58 +00:00
Dwayne M. Hubbard 611c9ed353 added CLI 'iax2 unregister <peername>' for issue 9812, thanks eliel
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2007-06-06 22:35:35 +00:00
Russell Bryant 6aec360466 Remove our little joke that was making fun of email disclaimers which nobody
else seemed to think was very funny.  Oh well ... :)


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2007-06-06 22:27:18 +00:00
Tilghman Lesher 9d05ff8ed5 Issue 9869 - replace malloc and memset with ast_calloc, and other coding guidelines changes
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2007-06-06 21:20:11 +00:00
Russell Bryant 033a3df22a Merged revisions 67862 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r67862 | russell | 2007-06-06 16:14:46 -0500 (Wed, 06 Jun 2007) | 4 lines

Fix a crash when doing call pickups with SIP phones.  The code unlocked the
channel when it should not have.
(issue #9652, reported by corruptor, fixed by me)

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2007-06-06 21:16:18 +00:00
Russell Bryant 7d35df9d98 Make another small tweak ... mantis/svn testing
(issue #9828)


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2007-06-06 18:47:58 +00:00
Joshua Colp ca529e58e6 When SS7 is enabled add w/SS7 to the end. (issue #9893 reported by Mike Anikienko)
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2007-06-06 12:23:59 +00:00
Russell Bryant d8fa6b0312 Merged revisions 67457 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r67457 | russell | 2007-06-05 14:48:02 -0500 (Tue, 05 Jun 2007) | 2 lines

Suppress a bunch of debug output unless option_debug is on

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2007-06-05 19:50:04 +00:00
Jason Parker 48ac1e8a86 Merged revisions 67421 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r67421 | qwell | 2007-06-05 13:18:24 -0500 (Tue, 05 Jun 2007) | 4 lines

Correctly update date/time on devices throughout the life of the device, instead of just at registration.

Issue 9152, yet another patch by DEA.

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2007-06-05 18:19:53 +00:00
Dwayne M. Hubbard c78c8ec9dc changed #if DEBUG to #ifdef DEBUG to fix make failure when configured with --enable-dev-mode
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2007-06-05 17:24:46 +00:00
Russell Bryant bbd0541f98 Improve the way that the zaptel channel name is created by using the Asterisk
strings API and by only allocating space on the stack


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2007-06-05 17:11:32 +00:00
Christian Richter 27b334c66e Merged revisions 67334 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r67334 | crichter | 2007-06-05 18:14:07 +0200 (Di, 05 Jun 2007) | 9 lines

Merged revisions 67307 via svnmerge from 
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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2007-06-05 16:21:15 +00:00
Christian Richter f852627fbc Merged revisions 67329 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r67329 | crichter | 2007-06-05 18:11:57 +0200 (Di, 05 Jun 2007) | 9 lines

Merged revisions 67306 via svnmerge from 
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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2007-06-05 16:19:20 +00:00
Joshua Colp 72cd0474f6 Merged revisions 67304 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r67304 | file | 2007-06-05 12:22:30 -0300 (Tue, 05 Jun 2007) | 2 lines

Only muck with the thread structure if an idle one was found/created.

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2007-06-05 15:24:01 +00:00
Russell Bryant 05169b004d Doxygenify the comments for new members of the iax2_thread struct
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2007-06-05 14:55:09 +00:00
Kevin P. Fleming a236af6137 Merged revisions 67270 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r67270 | kpfleming | 2007-06-05 09:35:52 -0500 (Tue, 05 Jun 2007) | 3 lines

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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2007-06-05 14:45:48 +00:00
Christian Richter 8e4dbdd5e1 Merged revisions 67210 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r67210 | crichter | 2007-06-05 12:25:32 +0200 (Di, 05 Jun 2007) | 9 lines

Merged revisions 67209 via svnmerge from 
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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2007-06-05 11:48:25 +00:00
Russell Bryant 71544542a0 Merged revisions 67158 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r67158 | russell | 2007-06-04 18:31:40 -0500 (Mon, 04 Jun 2007) | 5 lines

Fix up a bunch of places where the iax2 pvt structure can disappear and the
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)

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2007-06-04 23:32:26 +00:00
Jason Parker 95327fdb2f Merged revisions 67156 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r67156 | qwell | 2007-06-04 18:26:28 -0500 (Mon, 04 Jun 2007) | 6 lines

Fix for skinny keepalives.
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.

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2007-06-04 23:29:22 +00:00
Jason Parker 8611e870bb Merged revisions 67121 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r67121 | qwell | 2007-06-04 17:36:57 -0500 (Mon, 04 Jun 2007) | 4 lines

Fixes for dtmf/dialing with mgcp (similar to the recent fix for chan_skinny)

Issue 9855, patch by DEA.

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2007-06-04 22:39:10 +00:00
Russell Bryant 6857957000 Merged revisions 67119 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r67119 | russell | 2007-06-04 17:28:55 -0500 (Mon, 04 Jun 2007) | 6 lines

Add comments for two functions that get called with the appropriate call locked,
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)

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2007-06-04 22:29:43 +00:00
Russell Bryant c7fec92508 Remove a leftover unlock and lock of the iax2 pvt struct lock that was left
over from my attempt at putting pvt structs in a hash table.  It can cause
seg faults, and has no reason to stay.
(issue #9642, pointed out by stevedavies)


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2007-06-04 21:45:24 +00:00
Joshua Colp e3492b9511 Merged revisions 67068 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r67068 | file | 2007-06-04 15:31:09 -0400 (Mon, 04 Jun 2007) | 2 lines

Better handle SIP devices that say they have SDP content... but really don't. (issue #9398 reported by mthomasslo)

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2007-06-04 19:32:08 +00:00
Russell Bryant d6f2b914e8 Merged revisions 67020 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r67020 | russell | 2007-06-04 10:47:40 -0500 (Mon, 04 Jun 2007) | 7 lines

Resolve a deadlock in chan_iax2.  When handling an implicit ACK to a frame that
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)

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2007-06-04 15:50:46 +00:00
Russell Bryant 93f2be2675 Fix a couple of places where "tos" was used instead of "cos".
(issue #9540, patch by IgorG)


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2007-06-04 15:14:52 +00:00
Tilghman Lesher ce9ec91897 ast_calloc janitor (Inspired by issue 9860)
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2007-06-03 06:10:27 +00:00
Russell Bryant c9cf12b675 Remove 80 bytes in the iax2_registry struct that weren't being used
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2007-06-01 23:02:57 +00:00
Russell Bryant 0f409e66cf Merged revisions 66881 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r66881 | russell | 2007-06-01 14:41:30 -0500 (Fri, 01 Jun 2007) | 6 lines

Changes to the way DTMF is handled in the core broke dialing in chan_skinny.
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)

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2007-06-01 19:42:27 +00:00
Russell Bryant 3d2b58751f To satisfy some legal concerns, add an option for chan_sip to include a
disclaimer along with SIP messages in the header, X-Disclaimer.  This is off by
default.  Also, the text of the disclaimer can be customized in sip.conf.


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2007-05-31 19:41:03 +00:00
Joshua Colp 9f79587d47 Merged revisions 66768 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r66768 | file | 2007-05-31 12:14:48 -0400 (Thu, 31 May 2007) | 10 lines

Merged revisions 66764 via svnmerge from 
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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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2007-05-31 16:18:14 +00:00
Olle Johansson 0b2db74e5a Issue #9842 - Doxygen updates by snuffy. Thanks!
(Committed from Media Plaza in Utrecht, Netherlands - Open Source VoIP conference)


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2007-05-31 10:26:55 +00:00
Joshua Colp c103a8d298 This concludes my tweaking of things.
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2007-05-30 16:11:57 +00:00
Tilghman Lesher 1236ed3214 Issue 9477 - Improve menuselect labels
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2007-05-30 05:17:09 +00:00
Olle Johansson 4c1068c136 oops. Thanks Terry.
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2007-05-29 19:53:40 +00:00
Olle Johansson ee3a0af16a Merged revisions 66503 via svnmerge from
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r66503 | oej | 2007-05-29 21:32:57 +0200 (Tue, 29 May 2007) | 2 lines

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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Olle Johansson 6d6c525b10 Merged revisions 66474 via svnmerge from
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r66474 | oej | 2007-05-29 21:02:04 +0200 (Tue, 29 May 2007) | 2 lines

Don't issue hangup on hangup on hangup on hangup (for jcmoore)

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r66414 | oej | 2007-05-29 18:07:44 +0200 (Tue, 29 May 2007) | 2 lines

Don't reset hangupcause if we already have one

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Olle Johansson f4e81d7a54 Merged revisions 66404 via svnmerge from
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r66404 | oej | 2007-05-29 18:02:50 +0200 (Tue, 29 May 2007) | 2 lines

Tracking down hanging channels, killing them one by one. Issue #9235 and related

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Olle Johansson 36f15091bb Merged revisions 66363 via svnmerge from
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r66363 | oej | 2007-05-29 11:41:40 +0200 (Tue, 29 May 2007) | 10 lines

Merged revisions 66349 via svnmerge from 
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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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2007-05-29 10:02:31 +00:00
Joshua Colp 4d03b4f268 Don't try to unregister a peer using the sip unregister CLI command if they are not registered. (issue #9811 reported by eliel)
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2007-05-28 23:28:52 +00:00
Joshua Colp 39e9b3112c Due to the way stringfields work the value of the url pointer will always be non-NULL so we have to use ast_strlen_zero to make sure it is not empty. (issue #9821 reported by pj)
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2007-05-28 23:24:04 +00:00
Kevin P. Fleming f371a4c756 more minor fixes
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2007-05-25 15:07:26 +00:00
Kevin P. Fleming c74518e3ff Merged revisions 66157 via svnmerge from
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r66157 | kpfleming | 2007-05-25 10:28:46 -0400 (Fri, 25 May 2007) | 3 lines

handle the GNUTLS library properly in the configure script and build system
don't build in OSP support unless we have found and are allowed to use SSL support

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2007-05-25 14:37:55 +00:00
Kevin P. Fleming 4573b36af1 use the OpenSSL AES implementation if it's available (unless configured not to)
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2007-05-24 22:07:50 +00:00
Kevin P. Fleming 287bcc6127 Merged revisions 65965-65967 via svnmerge from
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r65965 | kpfleming | 2007-05-24 14:24:55 -0400 (Thu, 24 May 2007) | 2 lines

don't use uninitialized variables

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r65966 | kpfleming | 2007-05-24 14:25:21 -0400 (Thu, 24 May 2007) | 2 lines

don't reference GnuTLS headers and functions unless the configure script found it

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r65967 | kpfleming | 2007-05-24 14:28:48 -0400 (Thu, 24 May 2007) | 2 lines

oops, use #ifdef instead of #if

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2007-05-24 19:05:42 +00:00
Russell Bryant 4b3a3fb14c Add a new API call for creating detached threads. Then, go replace all of the
places in the code where the same block of code for creating detached threads
was replicated.  (patch from bbryant)


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2007-05-24 18:30:19 +00:00
Olle Johansson b4f7d35240 Merged revisions 65901 via svnmerge from
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r65901 | oej | 2007-05-24 17:26:10 +0200 (Thu, 24 May 2007) | 2 lines

Issue 7672 - fix by zandbelt - Asterisk core dump since the GnuTLS interface did not support multithreading correctly.

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Olle Johansson bd2ae8b587 Merged revisions 65892 via svnmerge from
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r65892 | oej | 2007-05-24 17:20:54 +0200 (Thu, 24 May 2007) | 2 lines

Issue 8193 - NAT issues with gtalk/STUN. Patch by phsultan. Thanks!

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2007-05-24 15:23:04 +00:00
Olle Johansson 45bb9430b9 Merged revisions 65857 via svnmerge from
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r65857 | oej | 2007-05-24 17:05:10 +0200 (Thu, 24 May 2007) | 2 lines

Issue 7686, fix by phsultan, NAT issues when calling from gtalk to SIP over nat.

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2007-05-24 15:21:39 +00:00
Olle Johansson d3c7be548d Merged revisions 65841 via svnmerge from
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r65841 | oej | 2007-05-24 16:48:55 +0200 (Thu, 24 May 2007) | 2 lines

Issue #8536 - Caller ID not set in CDR for jingle

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2007-05-24 14:52:01 +00:00
Joshua Colp 202fbe363a Merged revisions 65839 via svnmerge from
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r65839 | file | 2007-05-24 10:42:12 -0400 (Thu, 24 May 2007) | 10 lines

Merged revisions 65837 via svnmerge from 
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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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2007-05-24 14:43:49 +00:00
Olle Johansson 9b95428cce Issue #8409 and accidentally a fix to chan_sip that wasn't supposed to be there
but is still ok... Sorry. Lack of Tea, really.


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2007-05-24 14:41:43 +00:00
Kevin P. Fleming 0ec502099f Yes Virginia, there is a reason why we have stringfields in the sip_pvt structure...
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2007-05-24 11:38:20 +00:00
Christian Richter e03f5a8e88 Merged revisions 65768 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r65768 | crichter | 2007-05-24 11:37:32 +0200 (Do, 24 Mai 2007) | 9 lines

Merged revisions 65767 via svnmerge from 
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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2007-05-24 09:51:01 +00:00
Russell Bryant d1ba4f90de - Remove debug variable that was only used in one place
- convert string handling to the ast_str API
 - Convert strdup() to ast_strdup() and check the result
 - Minor formatting changes


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2007-05-24 03:28:39 +00:00
Mark Spencer 04e45cfda3 Add SendURL support
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2007-05-24 02:23:08 +00:00
Kevin P. Fleming 48895d885d Merged revisions 65685 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r65685 | kpfleming | 2007-05-23 16:59:19 -0400 (Wed, 23 May 2007) | 2 lines

start the delayed PBX when receive voice or video full frames as well, and comment this delayed-PBX activity

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Kevin P. Fleming 5dc23536ec Merged revisions 65683 via svnmerge from
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r65683 | kpfleming | 2007-05-23 16:51:56 -0400 (Wed, 23 May 2007) | 10 lines

Merged revisions 65682 via svnmerge from 
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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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Kevin P. Fleming 04da47df24 Merged revisions 65679-65680 via svnmerge from
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r65679 | kpfleming | 2007-05-23 16:30:24 -0400 (Wed, 23 May 2007) | 2 lines

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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r65680 | kpfleming | 2007-05-23 16:35:50 -0400 (Wed, 23 May 2007) | 2 lines

clear the 'delay PBX' flag when we are ready to start the PBX

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2007-05-23 20:39:42 +00:00
Kevin P. Fleming 61a53d859e Merged revisions 65677 via svnmerge from
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r65677 | kpfleming | 2007-05-23 16:07:59 -0400 (Wed, 23 May 2007) | 10 lines

Merged revisions 65676 via svnmerge from 
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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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2007-05-23 20:08:59 +00:00
Matthew Fredrickson cc016de696 Make sure we get the cause code in the REL
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Russell Bryant 5558323d0c Merged revisions 65589 via svnmerge from
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r65589 | russell | 2007-05-23 08:07:13 -0500 (Wed, 23 May 2007) | 11 lines

Merged revisions 65588 via svnmerge from 
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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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Russell Bryant d3ee017682 Merged revisions 65501 via svnmerge from
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r65501 | russell | 2007-05-22 13:40:38 -0500 (Tue, 22 May 2007) | 3 lines

List res_smdi as a dependency for app_voicemail and chan_zap
(Thanks to mnicholson for pointing it out)

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Joshua Colp a3e603cd0f Use ast_strlen_zero where possible. (issue #9774 reported by eliel)
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2007-05-22 15:08:08 +00:00
Christian Richter 47362f0906 Merged revisions 65342 via svnmerge from
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r65342 | crichter | 2007-05-22 10:12:20 +0200 (Di, 22 Mai 2007) | 9 lines

Merged revisions 65328 via svnmerge from 
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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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2007-05-22 08:21:59 +00:00
Olle Johansson a3f9350ec2 Related to issue #9235 btw.
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r65123 | oej | 2007-05-18 20:16:09 +0200 (Fri, 18 May 2007) | 10 lines

Merged revisions 65122 via svnmerge from 
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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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Olle Johansson 1bc7fdeb6b Merged revisions 65076 via svnmerge from
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r65076 | oej | 2007-05-18 17:18:13 +0200 (Fri, 18 May 2007) | 13 lines

Merged revisions 65075 via svnmerge from 
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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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Christian Richter b3231b9db6 Merged revisions 65039 via svnmerge from
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r65039 | crichter | 2007-05-18 14:40:46 +0200 (Fr, 18 Mai 2007) | 9 lines

Merged revisions 65007 via svnmerge from 
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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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Christian Richter 37f024bb90 Merged revisions 64904 via svnmerge from
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r64904 | crichter | 2007-05-18 10:58:51 +0200 (Fr, 18 Mai 2007) | 9 lines

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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2007-05-18 11:17:16 +00:00
Olle Johansson 491827a28a Merged revisions 64974 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r64974 | oej | 2007-05-18 12:37:44 +0200 (Fri, 18 May 2007) | 2 lines

Issue 9487 - stop media flows at hangup of call

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2007-05-18 10:41:31 +00:00
Olle Johansson ac343d43c8 Makeup, darling.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@64973 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-18 10:28:56 +00:00
Christian Richter c282567843 Merged revisions 64515 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r64515 | crichter | 2007-05-16 10:44:51 +0200 (Mi, 16 Mai 2007) | 9 lines

Merged revisions 64513 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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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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2007-05-18 10:03:38 +00:00
Christian Richter 2ce08e259c Merged revisions 63534 via svnmerge from
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r63534 | crichter | 2007-05-09 15:17:10 +0200 (Mi, 09 Mai 2007) | 17 lines

Merged revisions 62945,63402,63519 via svnmerge from 
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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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2007-05-18 10:00:21 +00:00
Christian Richter 0f39ea1a98 Merged revisions 62912 via svnmerge from
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r62912 | crichter | 2007-05-03 16:36:32 +0200 (Do, 03 Mai 2007) | 17 lines

Merged revisions 61357,61770,62885 via svnmerge from 
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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2007-05-18 09:50:33 +00:00
Christian Richter 2cdaf67ae0 Merged revisions 59774 via svnmerge from
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r59774 | crichter | 2007-04-03 09:20:27 +0200 (Di, 03 Apr 2007) | 17 lines

Merged revisions 59623-59624,59639 via svnmerge from 
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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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2007-05-18 09:47:19 +00:00
Christian Richter b05f83410a Merged revisions 59254 via svnmerge from
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r59254 | crichter | 2007-03-27 17:00:10 +0200 (Di, 27 Mär 2007) | 9 lines

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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2007-05-18 09:42:42 +00:00
Christian Richter e6cad524f8 Merged revisions 59064 via svnmerge from
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r59064 | crichter | 2007-03-20 14:16:06 +0100 (Di, 20 Mär 2007) | 21 lines

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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2007-05-18 09:36:16 +00:00
Christian Richter ecfbebafe9 Merged revisions 58825-58826 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r58825 | crichter | 2007-03-12 13:43:24 +0100 (Mo, 12 Mär 2007) | 1 line

added UU transceiving and corect handling for rdnis
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r58826 | crichter | 2007-03-12 14:08:06 +0100 (Mo, 12 Mär 2007) | 21 lines

Merged revisions 57034,57523,57753,58558 via svnmerge from 
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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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2007-05-18 09:31:27 +00:00
Olle Johansson f3ec447a23 Another fix for the support for recordings controlled by INFO-packets
We still lack a setting to enable/disable this per peer


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2007-05-18 08:49:34 +00:00
Joshua Colp a769766c53 Merged revisions 64754 via svnmerge from
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r64754 | file | 2007-05-17 12:10:12 -0400 (Thu, 17 May 2007) | 2 lines

Even more direct RTP setup fixes! Don't allow a codec that isn't supported to creep into the SDP of either side. (issue #9446 reported by marcelbarbulescu)

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2007-05-17 16:11:26 +00:00
Olle Johansson d83dcae6b1 Below patches with some re-structuring for trunk
---
Merged revisions 64602 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r64602 | oej | 2007-05-16 12:38:18 +0200 (Wed, 16 May 2007) | 2 lines

Issue #9681 - Handle www-auth on BYE

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2007-05-16 10:58:22 +00:00
Olle Johansson c472b899ef Merged revisions 64578 via svnmerge from
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r64578 | oej | 2007-05-16 12:05:47 +0200 (Wed, 16 May 2007) | 2 lines

Final part of issue #9483 - fixing transfer() of sip calls in the dial plan (twilson)

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2007-05-16 10:09:42 +00:00
Olle Johansson c352f7b0d5 Merged revisions 64543 via svnmerge from
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r64543 | oej | 2007-05-16 11:12:34 +0200 (Wed, 16 May 2007) | 10 lines

Merged revisions 64535 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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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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2007-05-16 10:02:06 +00:00
Olle Johansson 09aec2f622 Merged revisions 64516 via svnmerge from
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r64516 | oej | 2007-05-16 10:46:18 +0200 (Wed, 16 May 2007) | 17 lines

Merged following patch with a lot of changes for 1.4
------

Merged revisions 64514 via svnmerge from 
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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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2007-05-16 08:51:39 +00:00
Olle Johansson 7532b0bc4b Issue #9304 - Update help text to match functionality. Patch by kshumard with changes by oej
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@64498 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-16 07:58:43 +00:00
Olle Johansson 90bad9d2f5 Issue #6789 - Marquis - Add option to support regexten removal when host becomes unreachable
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2007-05-16 07:35:56 +00:00
Matthew Fredrickson 9b5454232c XXX-XXX-XXX appears to be the standard ANSI pointcode format
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@64455 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-15 20:45:20 +00:00
Matthew Fredrickson 79f553a4ab Only print the SS7 UP once. Not every time we get the test messages on the
line.


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2007-05-14 22:25:56 +00:00
Olle Johansson fa2622cb1d Merged revisions 64324 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r64324 | oej | 2007-05-14 21:26:50 +0200 (Mon, 14 May 2007) | 2 lines

Change -2 to XMIT_ERROR to clarify a bit more

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2007-05-14 19:35:58 +00:00
Russell Bryant ee7217c011 Merged revisions 64306 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r64306 | russell | 2007-05-14 14:13:00 -0500 (Mon, 14 May 2007) | 3 lines

Properly handle AST_CONTROL_PROGRESS by just ignoring it.  An unknown indication
will trigger an error and cause sounds to stop, which in this case, is ringing.

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2007-05-14 19:21:31 +00:00
Joshua Colp 4fbb449444 If no port is specified in the outboundproxy setting then use the standard SIP port. (issue #9665 reported by tootai)
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2007-05-14 18:21:30 +00:00
Steve Murphy a76ff39c0e Merged revisions 64193 via svnmerge from
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r64193 | murf | 2007-05-14 07:58:42 -0600 (Mon, 14 May 2007) | 1 line

As per 9570, worrisome CDR warnings have been removed, that are either not helpful, or not relevant.
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2007-05-14 14:13:45 +00:00
Olle Johansson fc057b1890 Improve handling network errors on transmission to hosts that don't reply or are unreachable
With this code, the call will fail as soon as we get a network error. This may happen on
first xmit or a later one, so the retransmit code handles this too.


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2007-05-13 19:20:36 +00:00
Joshua Colp b71e691b3e Merged revisions 64114 via svnmerge from
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r64114 | file | 2007-05-12 18:27:04 -0400 (Sat, 12 May 2007) | 2 lines

This concludes my final adventure with bitmasks and the onhold flag. Would anyone care for some peanuts?

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2007-05-12 22:28:09 +00:00
Joshua Colp 38e951cfda Merged revisions 64086 via svnmerge from
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r64086 | file | 2007-05-12 17:10:45 -0400 (Sat, 12 May 2007) | 2 lines

Tweak hold flags some more. They can be of three states when active: active, inactive, one direction.

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2007-05-12 21:12:18 +00:00
Joshua Colp 82a30356da Merged revisions 64044 via svnmerge from
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r64044 | file | 2007-05-12 12:32:15 -0400 (Sat, 12 May 2007) | 2 lines

Ensure the onhold flag is set no matter what when being put on hold.

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2007-05-12 16:33:34 +00:00
Jason Parker 60f6a72123 Add/fix support for Redial, Speeddial, and Messages buttons.
Combined effort by DEA and mvanbaak.


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2007-05-11 22:52:36 +00:00
Jason Parker 2768946cf5 Merged revisions 63830 via svnmerge from
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r63830 | qwell | 2007-05-10 18:15:37 -0500 (Thu, 10 May 2007) | 12 lines

Merged revisions 63828 via svnmerge from 
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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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2007-05-10 23:16:33 +00:00
Olle Johansson aa320037d2 Merged revisions 63749 via svnmerge from
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r63749 | oej | 2007-05-10 22:46:41 +0200 (Thu, 10 May 2007) | 12 lines

Merged revisions 63748 via svnmerge from 
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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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2007-05-10 20:51:59 +00:00
Matthew Fredrickson e2ca869abd Merged revisions 63654 via svnmerge from
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r63654 | mattf | 2007-05-09 12:25:21 -0500 (Wed, 09 May 2007) | 10 lines

Merged revisions 63653 via svnmerge from 
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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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2007-05-09 17:35:54 +00:00
Joshua Colp 7e10164e20 Merged revisions 63611 via svnmerge from
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r63611 | file | 2007-05-09 12:54:56 -0400 (Wed, 09 May 2007) | 10 lines

Merged revisions 63610 via svnmerge from 
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r63610 | file | 2007-05-09 12:51:03 -0400 (Wed, 09 May 2007) | 2 lines

Properly handle hints that point to multiple devices in chan_sip. Why chan_sip is even doing this I have no idea but I would rather not go into a rant. (issue #9536 reported by rlister)

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2007-05-09 16:56:45 +00:00
Olle Johansson c358b18a5a Merged revisions 63532 via svnmerge from
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r63532 | oej | 2007-05-09 15:04:14 +0200 (Wed, 09 May 2007) | 2 lines

Don't retransmit 200 OK's on ignore status. (Reported on asterisk-users)

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2007-05-09 13:07:44 +00:00
Russell Bryant 314c874d7d I noted this on the dev list but got no response, so I just did it myself.
Lock the call features when being used in chan_sip.


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2007-05-08 16:41:35 +00:00
Olle Johansson a39f95b94f Adding external referenses for doxygen
See http://www.asterisk.org/doxygen/trunk/extref.html


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2007-05-07 18:25:56 +00:00
Olle Johansson 33214f76a7 Adding external reference
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2007-05-07 18:18:43 +00:00
Olle Johansson 238bcd6c45 Doxyfication... There's a shortage of comments in this file...
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2007-05-07 18:17:18 +00:00
Joshua Colp 28f4727e75 Lock iax2 pvt structure when passing off to the AMI function, and make sure it exists. (issue #9674 reported by arabe)
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2007-05-06 20:09:18 +00:00
Olle Johansson d326d84ae0 - Adding some missing spaces
- Correcting error messages
- Disabling code that doesn't do anything
- Making sure we always respond to this request, happily


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2007-05-05 08:05:38 +00:00
Steve Murphy 02337303ef a small upgrade to the coding standard, and an update to the code that triggered the upgrade.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@63048 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-04 17:49:20 +00:00
Steve Murphy 3ee0077f04 Added a small bit of code to support the SNOM 360's Record button. Made the find_feature func in res_features.c public, so I could use it to find the automon dial sequence as configured by the user. When the INFO packet has a Record: header with on/off, the sequence is sent as consecutive DTMF frames on the phone's channel, triggering the automon functionality. The user has to configure the automon in features.conf, and set up his dialplan accordingly.
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2007-05-04 16:37:23 +00:00
Olle Johansson 1b15d8852d Add the new ChannelUpdate event to inform manager clients about the PVT ID and some other channel driver data that
is needed to follow the call through the PBX.


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2007-05-04 13:56:25 +00:00
Joshua Colp 81cade7a4c Merged revisions 62989 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r62989 | file | 2007-05-03 13:44:00 -0300 (Thu, 03 May 2007) | 10 lines

Merged revisions 62987 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r62987 | file | 2007-05-03 13:42:19 -0300 (Thu, 03 May 2007) | 2 lines

When a peer is seeded or built tell the devicestate core to update it's status. This is easier then having chan_sip load before pbx_config. (issue #9658 reported by dlynes)

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

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r62692 | tilghman | 2007-05-02 12:43:48 -0500 (Wed, 02 May 2007) | 12 lines

Merged revisions 62691 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r62691 | tilghman | 2007-05-02 12:38:16 -0500 (Wed, 02 May 2007) | 4 lines

Issue 9638 - if a text frame is sent with no terminating NULL through a bridged
IAX connection, the remote end will receive garbage characters tacked onto the
end.

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

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r62689 | murf | 2007-05-02 11:10:50 -0600 (Wed, 02 May 2007) | 1 line

a)In chan_zap, set the clid, src fields in channel_alloc call. b)in the channel_alloc func, set the cid_num and name fields from the arglist[blush]. c) don't update the channel app & app data fields if you are in the 'h' extension. d)the load_module func in cdr_radius needs to return DECLINE, SUCCESS.
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2007-05-02 17:24:03 +00:00
Russell Bryant 3d409eb793 Update the device state functionality of chan_local such that it will return
NOT_INUSE or INUSE when Local channels are in use as opposed to just UNKNOWN.
It will still return INVALID if the extension doesn't exist at all.
(issue #8048, patch from tim_ringenbach)


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2007-05-02 15:46:49 +00:00
Olle Johansson e1ec3f917c Add a small message that we're doing something. On my systems, there's a long
dead period with a non-responsive CLI after I issue "load chan_sip.so"


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2007-05-02 12:12:02 +00:00
Olle Johansson 1d51b2e161 More username body parts to fix... If working, this needs to be backported to 1.2, 1.4.
But first, some serious SIP testing :-)


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

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r62624 | oej | 2007-05-02 08:15:43 +0200 (Wed, 02 May 2007) | 2 lines

Don't unlock a channel that we already know does not exist (propably isue 8228)

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2007-05-02 09:35:14 +00:00
Russell Bryant b419fc1134 Add support for setting the CoS for VLAN traffic (802.1p) in Linux. The
file doc/qos.tex has been updated to document the new functionality.
(issue #9540, patch submitted by IgorG)


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2007-04-30 16:16:26 +00:00
Russell Bryant a91f9b138d Merged revisions 62419 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r62419 | russell | 2007-04-30 10:58:28 -0500 (Mon, 30 Apr 2007) | 12 lines

Merged revisions 62417 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r62417 | russell | 2007-04-30 10:57:26 -0500 (Mon, 30 Apr 2007) | 4 lines

This patch fixes an issue where depending on the cause code, when the network
sends a PRI disconnect, the call may not be properly hung up.
(issue #9588, reported and patched by softins)

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2007-04-30 15:59:27 +00:00
Russell Bryant 5cb08adc7a Don't crash when invalid arguments are provided to the CHANNEL() function
for a SIP channel.
(issue #9619, reported by jtodd, original patch by Corydon76, committed patch
 slightly modified by me)


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2007-04-30 15:37:23 +00:00
Russell Bryant 0efe511879 Merged revisions 62331 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r62331 | russell | 2007-04-29 00:50:37 -0500 (Sun, 29 Apr 2007) | 3 lines

Fix a bug that made the "language" setting in zapata.conf not
functional.  (issue #9626, reported and fixed by sergee)

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2007-04-29 05:51:18 +00:00
Russell Bryant 7d2102c081 Reformat some of iax2.h and convert comments to doxygen format
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2007-04-28 21:26:00 +00:00
Russell Bryant b6b1bf3213 Merge changes from team/russell/events
This set of changes introduces a new generic event API for use within Asterisk.
I am still working on a way for events to be shared between servers, but this
part is ready and can already be used inside of Asterisk.

This set of changes introduces the first use of the API, as well.  I have
restructured the way that MWI (message waiting indication) is handled.  It is
now event based instead of polling based.  For example, if there are a bunch
of SIP phones subscribed to mailboxes, then chan_sip will not have to
constantly poll the mailboxes for changes.  app_voicemail will generate events
when changes occur.

See UPGRADE.txt and CHANGES for some more information on the effects of these
changes from the user perspective.  For developer information, see the text in
include/asterisk/event.h.

As always, additional feedback is welcome on the asterisk-dev mailing list.


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

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r62218 | russell | 2007-04-27 16:10:51 -0500 (Fri, 27 Apr 2007) | 11 lines

Fix a weird problem where when a caller talking to someone sitting behind an
agent channel sent a digit, the digit would be played to the agent for forever.
This is because chan_agent always returned -1 from its send_digit_begin and _end
callbacks.  This non-zero return value indicates to the Asterisk core that it
would like an inband DTMF generator put on the channel.  However, this is the
wrong thing to do.  It should *always* return 0, instead.  When the digit begin
and end functions are called on the proxied channel, the underlying channel
will indicate whether inband DTMF is needed or not, and the generator will be
put on that one, and not the Agent channel.
(issue #9615, #9616, reported by jiddings and BigJimmy, and fixed by me)

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

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r62137 | oej | 2007-04-27 16:04:07 +0200 (Fri, 27 Apr 2007) | 12 lines

Merged revisions 62126 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r62126 | oej | 2007-04-27 15:57:45 +0200 (Fri, 27 Apr 2007) | 4 lines

Issue #7351 - SIP Cancel fails due to the wrong contact uri. Reported by PPYY, failed to fix by OEJ
final fix by wojtekka - THANKS!!!! THis was a hard one to catch.


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

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r62038 | file | 2007-04-26 12:33:52 -0400 (Thu, 26 Apr 2007) | 10 lines

Merged revisions 62037 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r62037 | file | 2007-04-26 12:30:57 -0400 (Thu, 26 Apr 2007) | 2 lines

Revert previous fix for when the IAX2 channel goes funky (that's the technical term). This is causing legit calls to be prematurely hung up. (issue #9600 reported by justdave)

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

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r61914 | kpfleming | 2007-04-25 17:29:53 -0500 (Wed, 25 Apr 2007) | 10 lines

Merged revisions 61913 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r61913 | kpfleming | 2007-04-25 17:24:59 -0500 (Wed, 25 Apr 2007) | 2 lines

handle a very bizarre race condition with channels being redirected before a simple switch can be started on them (issue #9286)

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

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r61870 | russell | 2007-04-25 16:59:07 -0500 (Wed, 25 Apr 2007) | 10 lines

Merged revisions 61866 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r61866 | russell | 2007-04-25 16:55:23 -0500 (Wed, 25 Apr 2007) | 2 lines

If the callerid= option is specified, but empty, clear any previous data.

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

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r61863 | russell | 2007-04-25 16:13:15 -0500 (Wed, 25 Apr 2007) | 10 lines

Merged revisions 61862 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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

Ensure that callerid settings are reset on a reload.

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

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r61799 | russell | 2007-04-25 11:22:07 -0500 (Wed, 25 Apr 2007) | 11 lines

Merged revisions 61798 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r61798 | russell | 2007-04-25 11:20:38 -0500 (Wed, 25 Apr 2007) | 3 lines

Fix a typo where cid_num got copied instead of cid_ani.  
(issue #9587, reported and patched by xrg)

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

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r61772 | file | 2007-04-24 12:07:02 -0400 (Tue, 24 Apr 2007) | 10 lines

Merged revisions 61771 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r61771 | file | 2007-04-24 12:05:06 -0400 (Tue, 24 Apr 2007) | 2 lines

Allow RFC2833 to be sent in the response SDP when an INVITE comes in without SDP. (issue #9546 reported by mcrawford)

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2007-04-24 16:10:10 +00:00
Russell Bryant b94378b5e6 Merge changes from team/russell/iax2_osp
This set of changes adds OSP support to chan_iax2.  However, I have modified
the patch a bit from what was submitted.  You now use the CHANNEL() function
to get and set the OSP token for IAX2.

(issue #8531, reported by and original patch by homesick, patch updated by me)


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2007-04-20 21:12:53 +00:00
Olle Johansson 49af71c100 Use the last line in the SDP, even if it has no CRLF. Remember Jon Postel :-)
This code exists in 1.2 and 1.4 but was removed from trunk for some unknown reason.


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

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r61648 | file | 2007-04-13 13:19:53 -0400 (Fri, 13 Apr 2007) | 2 lines

For those very verbose SIP implementations that attach tons of info to the Contact header... let's increase our variable sizes. (issue #9535 reported by jeffg)

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

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

A fix for chan_oss that resulted from the CDR changes; it helps to use the right info.
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2007-04-13 17:11:53 +00:00
Joshua Colp 80ec0b13ba Merged revisions 61641 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r61641 | file | 2007-04-13 12:32:03 -0400 (Fri, 13 Apr 2007) | 2 lines

Don't assume the callid of a dialog will be set, as in some circumstances it may not. (issue #9534 reported by tecnoxarxa)

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

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r61477 | russell | 2007-04-11 11:05:29 -0500 (Wed, 11 Apr 2007) | 13 lines

Merged revisions 61476 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r61476 | russell | 2007-04-11 11:01:25 -0500 (Wed, 11 Apr 2007) | 5 lines

If someone sets the "useragent" option in sip.conf to be empty, then don't add
the User-Agent header at all.  It is an optional header, anyway.  Also, the bug
report says that some of Japan's SIP providers don't allow it for some weird
reason.  (issue #9488, reported by makoto, fixed by me)

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2007-04-11 16:06:37 +00:00
Nadi Sarrar 9978dc647b Merged revisions 61342,61372-61373,61443 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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

AOCD's are now exported to asterisk channel variables.

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

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

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

Export AOCD variables on misdn_hangup.

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

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

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2007-04-11 15:48:54 +00:00
Russell Bryant 6b033eea04 Merged revisions 61427 via svnmerge from
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r61427 | russell | 2007-04-11 10:09:39 -0500 (Wed, 11 Apr 2007) | 14 lines

Merged revisions 61426 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r61426 | russell | 2007-04-11 10:05:36 -0500 (Wed, 11 Apr 2007) | 6 lines

Fix a bug with switching between host=dynamic and using specific hosts for
peers.  The code would only reset the peer's address when it is dynamic if
it was a new peer structure.  Now, it will also reset the address if it was
already in the peer list, but before the reload, it was not dynamic.
(issue #9515, reported by caio1982, fixed by me)

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2007-04-11 15:13:12 +00:00
Russell Bryant e34c67d308 Merged revisions 61377 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r61377 | russell | 2007-04-11 09:04:44 -0500 (Wed, 11 Apr 2007) | 13 lines

Merged revisions 61376 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r61376 | russell | 2007-04-11 09:02:54 -0500 (Wed, 11 Apr 2007) | 5 lines

Remove the attempt at reporting configuration errors in sip.conf.  This can
cause a bunch of improper messages when using realtime.  I give up.  As oej
tried to convince me when I put this in, there is just no easy way to do it.
(inspired by a message on the -dev list)

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2007-04-11 14:13:08 +00:00
Joshua Colp 9fff461080 Remove duplicate prototype declaration. (issue #9517 reported by junky)
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2007-04-11 14:01:53 +00:00
Doug Bailey 7c9e46507b Added zapata.conf parameter "cid_rxgain" to allow the user to adjust the gain bump used during CID acquisition.
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2007-04-10 22:02:53 +00:00
Steve Murphy 28b5fb02bd updated ast_channel_alloc() call to include the 4 extra args everyone got. Not much info there, as the config file evidently does not allow amaflags, or accountcode settings; and the pvt's exten doesn't sound like what we need in the cdr, either.
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2007-04-10 16:07:10 +00:00
Nadi Sarrar 0220ec0511 Merged revisions 61183 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r61183 | nadi | 2007-04-10 14:43:40 +0200 (Di, 10 Apr 2007) | 10 lines

Merged revisions 61170 via svnmerge from 
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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2007-04-10 12:47:57 +00:00
Steve Murphy ecaf781933 Merged revisions 60989 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r60989 | murf | 2007-04-09 12:32:07 -0600 (Mon, 09 Apr 2007) | 1 line

This is a big improvement over the current CDR fixes. It may still need refinement, but this won't have as many folks bothered.
This also adds the mods from 1.4/r.61136;
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2007-04-10 05:41:34 +00:00
Olle Johansson b52f774850 Merged revisions 61072 via svnmerge from
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r61072 | oej | 2007-04-09 21:58:17 +0200 (Mon, 09 Apr 2007) | 11 lines

Merged revisions 61038 via svnmerge from 
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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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2007-04-09 20:01:28 +00:00
Olle Johansson 4aef0155d6 use "ChannelType" in events to indicate which channel driver that generates the event. This replaces
"ChannelDriver" and "Channel", previously used to indicate channel driver. ChannelType is more
in line with "core show channeltypes"


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2007-04-09 18:22:43 +00:00
Joshua Colp a4bef3bb3a Make RTP session ID and session version generation random. (issue #9456 reported by tjardick)
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2007-04-09 12:33:49 +00:00
Tilghman Lesher c6c64af72d Merged revisions 60847 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r60847 | tilghman | 2007-04-08 21:42:48 -0500 (Sun, 08 Apr 2007) | 10 lines

Merged revisions 60846 via svnmerge from 
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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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2007-04-09 02:44:37 +00:00
Joshua Colp ed75ded048 Add counter for sip show registry CLI command. (issue #9352 reported by junky)
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2007-04-09 00:47:06 +00:00
Olle Johansson 5bc2aa8ab1 Use the same parameter to the two "Registry" AMI events - ChannelDriver
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2007-04-06 19:26:01 +00:00
Matthew Fredrickson bc0310e94e Merged revisions 60485 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r60485 | mattf | 2007-04-06 13:21:52 -0500 (Fri, 06 Apr 2007) | 2 lines

Make sure we check the faxdetect option before doing fax processing

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2007-04-06 18:26:24 +00:00
Matthew Fredrickson 1031bbba01 Merged revisions 60459 via svnmerge from
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r60459 | mattf | 2007-04-06 12:32:31 -0500 (Fri, 06 Apr 2007) | 10 lines

Merged revisions 60456 via svnmerge from 
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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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2007-04-06 18:25:38 +00:00
Joshua Colp 95a7dc0509 Merged revisions 60214 via svnmerge from
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r60214 | file | 2007-04-05 08:55:02 -0400 (Thu, 05 Apr 2007) | 10 lines

Merged revisions 60213 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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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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2007-04-05 12:57:35 +00:00
Russell Bryant d1588ce2d5 Merged revisions 60112 via svnmerge from
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r60112 | russell | 2007-04-04 11:49:45 -0500 (Wed, 04 Apr 2007) | 3 lines

Add a Content-Length of 0 to the response built by transmit_response_with_unsupported().
(issue #9454, reported by makoto, fixed by me)

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2007-04-04 16:50:31 +00:00
Russell Bryant bb53ef9d32 Merged revisions 60088 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r60088 | russell | 2007-04-04 11:39:04 -0500 (Wed, 04 Apr 2007) | 12 lines

Merged revisions 60083 via svnmerge from 
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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2007-04-04 16:40:01 +00:00
Russell Bryant b2ddaaf033 Add support for RTP packetization in chan_jingle and chan_gtalk.
(issue #9416, phsultan)


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2007-04-03 22:33:03 +00:00
Russell Bryant 11ab6db24b Merged revisions 59939 via svnmerge from
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r59939 | russell | 2007-04-03 14:16:53 -0500 (Tue, 03 Apr 2007) | 12 lines

Merged revisions 59938 via svnmerge from 
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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2007-04-03 19:17:55 +00:00
Russell Bryant 3f14c8b6fc Merged revisions 59936 via svnmerge from
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r59936 | russell | 2007-04-03 13:55:57 -0500 (Tue, 03 Apr 2007) | 11 lines

Merged revisions 59916 via svnmerge from 
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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2007-04-03 18:57:52 +00:00
Russell Bryant b908f9717a Remove a duplicate function prototype. (issue #9444, junky)
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2007-04-03 18:34:14 +00:00
Nadi Sarrar 4d1edcd4fd Merged revisions 59804 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r59804 | nadi | 2007-04-03 13:02:46 +0200 (Di, 03 Apr 2007) | 15 lines

Merged revisions 59788,59803 via svnmerge from 
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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2007-04-03 11:15:45 +00:00
Russell Bryant 592870b6ce This hashing code is still causing some random crashes on my system, and
probably others, too.  I don't really have time to work on it at the moment,
so I am just going to revert it for now.


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2007-04-02 17:40:26 +00:00
Matthew Fredrickson d6e7b047f0 Update to support multiple CIC groups and DPCs per linkset.
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2007-03-30 20:54:42 +00:00
Russell Bryant 5d4e6a480a Fix an issue with hashing iax2 pvt structures that caused random crashes on
systems under heavy load such as IAXtel.
(possibly related to issue #9403)


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2007-03-29 18:31:35 +00:00
Russell Bryant 5a1c6826ce Merged revisions 59341 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r59341 | russell | 2007-03-29 11:55:39 -0500 (Thu, 29 Mar 2007) | 8 lines

When the IAX2 read callback gets called, return NULL instead of a "null frame".
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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2007-03-29 16:56:35 +00:00
Dwayne M. Hubbard 6626c31f35 added filtering options to 'zap show channels' to implement functionality described in issue 6520
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2007-03-28 00:09:25 +00:00
Joshua Colp 04f1955781 Use better english. Renegotiate! Repeat after me: renegotiate.
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2007-03-27 20:11:11 +00:00
Russell Bryant 93e2d66f13 Merged revisions 59262 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r59262 | russell | 2007-03-27 13:17:47 -0500 (Tue, 27 Mar 2007) | 3 lines

Fix the check that ensures that the CHANNEL function's first argument is "rtpqos".
Thanks, Corydon.  :)

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2007-03-27 18:18:36 +00:00
Russell Bryant b5adacebf1 Merged revisions 59259 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r59259 | russell | 2007-03-27 13:05:46 -0500 (Tue, 27 Mar 2007) | 12 lines

Merged revisions 59258 via svnmerge from 
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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2007-03-27 18:08:59 +00:00
Russell Bryant 7c884d76ea Merged revisions 59256 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r59256 | russell | 2007-03-27 11:20:53 -0500 (Tue, 27 Mar 2007) | 4 lines

Convert the RTPQOS function to just be additional parameter of the CHANNEL
function.  This way, it will be possible for other RTP based channel drivers
to expose this information in the future.

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2007-03-27 16:25:02 +00:00
Tilghman Lesher 0e0600a446 Merged revisions 59227 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r59227 | tilghman | 2007-03-26 16:37:41 -0500 (Mon, 26 Mar 2007) | 2 lines

Change this to a single dp function to make oej happy.

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2007-03-26 21:44:59 +00:00
Russell Bryant 46b15992c7 Merged revisions 59209 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r59209 | russell | 2007-03-26 12:53:07 -0500 (Mon, 26 Mar 2007) | 1 line

Rename the new dialplan functions to match the variable name
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2007-03-26 17:57:50 +00:00
Russell Bryant 08e3a9bdc8 Merged revisions 59207 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r59207 | russell | 2007-03-26 12:45:55 -0500 (Mon, 26 Mar 2007) | 7 lines

The AUDIORTPQOS and VIDEORTPQOS variables are not fully functional in some
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)

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2007-03-26 17:51:27 +00:00
Matthew Fredrickson e4a7d1b35f Fix bug in which parameter type we are passing. This shouldn't be a problem
since both types are the same underneath.


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2007-03-26 16:48:29 +00:00
Matthew Fredrickson df42d78569 Small API related SS7 updates.
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2007-03-26 16:46:07 +00:00
Nadi Sarrar 24d8595d00 Merged revisions 59202 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r59202 | nadi | 2007-03-26 17:25:53 +0200 (Mo, 26 Mär 2007) | 4 lines

* mISDN >= 1.2 provides a dsp pipeline for i.e. echo cancellation modules, make chan_misdn use it.
* 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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2007-03-26 15:59:56 +00:00
Joshua Colp cc22f60f30 Merged revisions 59195 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r59195 | file | 2007-03-23 21:39:44 -0400 (Fri, 23 Mar 2007) | 10 lines

Merged revisions 59194 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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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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2007-03-24 01:42:11 +00:00
Kevin P. Fleming e8e9e5e23c Merged revisions 59182 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r59182 | kpfleming | 2007-03-22 16:40:01 -0700 (Thu, 22 Mar 2007) | 2 lines

don't allow string input to overrun the buffer to hold it (ASA-2007-010)

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2007-03-22 23:41:37 +00:00
Kevin P. Fleming a6d4f89769 Merged revisions 59180 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r59180 | kpfleming | 2007-03-22 16:34:22 -0700 (Thu, 22 Mar 2007) | 2 lines

remove variables that are no longer used (--enable-dev-mode is good, developers should be using it)

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2007-03-22 23:39:26 +00:00
Joshua Colp af9c17025f Minor tweak. Only queue up an unhold control frame if we are actually on hold. This would have shown itself when a call was initially being setup and the SDP data was being parsed in.
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2007-03-21 03:33:57 +00:00
Joshua Colp 1d5be2d1c7 Merged revisions 59081 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r59081 | file | 2007-03-20 23:25:48 -0400 (Tue, 20 Mar 2007) | 2 lines

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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2007-03-21 03:27:58 +00:00
Joshua Colp 2874a4317d Merged revisions 59076 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r59076 | file | 2007-03-20 16:42:46 -0400 (Tue, 20 Mar 2007) | 2 lines

Copy len variable as well, should fix remaining IAX2 DTMF issues.

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2007-03-20 20:44:58 +00:00
Olle Johansson dddb57b242 Merged revisions 59037 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r59037 | oej | 2007-03-18 21:37:06 +0100 (Sun, 18 Mar 2007) | 3 lines

Issue #9313, Asterisk crash on SIP return code 0 (reported by qwerty1979) (ASA-2007-011)


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2007-03-18 20:39:37 +00:00
Joshua Colp 4787adb4e9 Merged revisions 58923 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r58923 | file | 2007-03-15 15:13:21 -0300 (Thu, 15 Mar 2007) | 2 lines

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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2007-03-15 18:21:03 +00:00
Russell Bryant 1bc728ed4b Merged revisions 58906 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r58906 | russell | 2007-03-14 14:18:08 -0500 (Wed, 14 Mar 2007) | 4 lines

Some people like to put "limitonpeer" instead of "limitonpeers" in their
configuration.  While we're at it, support "limitonpeerz" and 
"limitonpeerssssss".  (inspired by issue #9172)

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2007-03-14 19:19:00 +00:00
Olle Johansson f9c3f60ab9 Merged revisions 58848 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r58848 | oej | 2007-03-13 12:49:35 +0100 (Tue, 13 Mar 2007) | 10 lines

Merged revisions 58847 via svnmerge from 
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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2007-03-14 16:59:35 +00:00
Olle Johansson b4490af8ac Merged revisions 58845 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r58845 | oej | 2007-03-13 11:03:03 +0100 (Tue, 13 Mar 2007) | 3 lines

Don't hangup the call on OK or errors on MESSAGE and INFO 
inside of a dialog (like video update requests).

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2007-03-13 10:14:13 +00:00
Olle Johansson 32f9227ca4 Merged revisions 58843 via svnmerge from
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r58843 | oej | 2007-03-13 10:12:16 +0100 (Tue, 13 Mar 2007) | 2 lines

Issue #9251 - Clear From URI from user attributes (tgrman)

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2007-03-13 09:15:17 +00:00
Olle Johansson d008232b27 Change URL to OpenH323 (thanks, Tzafrir!)
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2007-03-12 09:37:13 +00:00
Joshua Colp ea226e9d77 Merged revisions 58779 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r58779 | file | 2007-03-11 20:51:16 -0400 (Sun, 11 Mar 2007) | 2 lines

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 00:54:13 +00:00
Russell Bryant 5dc7f1d898 Merged revisions 58705 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r58705 | russell | 2007-03-10 12:11:11 -0600 (Sat, 10 Mar 2007) | 6 lines

Fix a few more places in chan_iax2 where the ast_frame used for receiving a
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 18:15:41 +00:00
Joshua Colp 5629ee0292 Merged revisions 58584 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r58584 | file | 2007-03-09 15:49:47 -0500 (Fri, 09 Mar 2007) | 10 lines

Merged revisions 58579 via svnmerge from 
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 20:53:37 +00:00
Russell Bryant d04a9e0bc0 Merged revisions 58510 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r58510 | russell | 2007-03-08 16:06:54 -0600 (Thu, 08 Mar 2007) | 3 lines

Add a missing break statement so that handling the above event does not
incorrectly destroy the channel. (issue #9242, andrew)

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2007-03-08 22:08:28 +00:00
Russell Bryant 9266a6b8d8 Merged revisions 58320 via svnmerge from
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r58320 | russell | 2007-03-07 19:01:46 -0600 (Wed, 07 Mar 2007) | 6 lines

If we receive ZT_EVENT_REMOVED, destroy the specified channel. 
(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.

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2007-03-08 01:06:00 +00:00
Russell Bryant f86aee41f7 Merged revisions 58243 via svnmerge from
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r58243 | russell | 2007-03-07 12:19:19 -0600 (Wed, 07 Mar 2007) | 17 lines

(This bug was reported to me by Kinsey Moore)

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 18:20:51 +00:00
Joshua Colp e7da006562 Merged revisions 58240 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r58240 | file | 2007-03-07 12:52:58 -0500 (Wed, 07 Mar 2007) | 2 lines

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 17:55:11 +00:00
Steve Murphy 5bb1d16f47 Merged revisions 58121 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r58121 | murf | 2007-03-06 16:10:14 -0700 (Tue, 06 Mar 2007) | 9 lines

Merged revisions 58115 via svnmerge from 
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 23:19:59 +00:00
Olle Johansson 52750892b0 Merged revisions 58053 via svnmerge from
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r58053 | oej | 2007-03-06 21:37:07 +0100 (Tue, 06 Mar 2007) | 10 lines

Merged revisions 58052 via svnmerge from 
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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2007-03-06 20:45:29 +00:00
Russell Bryant d7aa9b6e2d Merged revisions 58023 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r58023 | russell | 2007-03-06 12:01:20 -0600 (Tue, 06 Mar 2007) | 3 lines

Return an error of transmit_response is called without a session.
(issue #9002)

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2007-03-06 18:02:35 +00:00
Joshua Colp b7e47198da Add zap show version CLI command. This pulls the version/echo canceller in use directly using the ZT_GETVERSION ioctl. (issue #9094 reported by tootai)
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2007-03-05 20:13:51 +00:00
Joshua Colp 2adf4fa4e0 Merged revisions 57914 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r57914 | file | 2007-03-05 14:19:07 -0500 (Mon, 05 Mar 2007) | 2 lines

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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Joshua Colp e23433e50e Merged revisions 57477 via svnmerge from
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r57477 | file | 2007-03-02 12:06:52 -0500 (Fri, 02 Mar 2007) | 10 lines

Merged revisions 57475 via svnmerge from 
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 17:09:18 +00:00
Joshua Colp b3bc675402 Merged revisions 57318 via svnmerge from
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r57318 | file | 2007-03-01 17:21:44 -0500 (Thu, 01 Mar 2007) | 10 lines

Merged revisions 57317 via svnmerge from 
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 22:23:26 +00:00
Joshua Colp fd2db34f15 Merged revisions 57093 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r57093 | file | 2007-02-28 13:57:52 -0500 (Wed, 28 Feb 2007) | 10 lines

Merged revisions 57092 via svnmerge from 
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 19:00:24 +00:00
Tilghman Lesher a3da18c244 Issue 7789 - some telcos want the TON set based on the number, but without the NANP prefix removed
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2007-02-27 00:11:32 +00:00
Russell Bryant b3776415aa Merged revisions 56888 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r56888 | russell | 2007-02-26 14:42:21 -0600 (Mon, 26 Feb 2007) | 4 lines

Restore the behavior of Asterisk 1.2 where if a device was not specified in
alsa.conf, then we just use the system default, instead of creating our own
default of hw:0,0.  (issue #9139)

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2007-02-26 20:43:18 +00:00
Russell Bryant c617945b5c Merged revisions 56847 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r56847 | russell | 2007-02-26 14:04:13 -0600 (Mon, 26 Feb 2007) | 2 lines

Fix a crash in my last change to iax2_indicate(). (issue #9150)

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2007-02-26 20:04:53 +00:00
Russell Bryant 82179d1202 Merged revisions 56785 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r56785 | russell | 2007-02-26 10:51:18 -0600 (Mon, 26 Feb 2007) | 3 lines

Do more complete locking of the chan_iax2_pvt struct in the indicate callback.
(Problem brought up by Ben Smithurst on the asterisk-dev list)

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2007-02-26 16:54:17 +00:00
Russell Bryant 5873c9c90f There is no need to look in the iaxs array for the pvt struct when we already
have a pointer to it.


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2007-02-26 00:37:03 +00:00
Olle Johansson 75d387acbc Doxygen additions, corrections
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2007-02-24 20:29:41 +00:00
Olle Johansson e916cf45da Doxygen updates and corrections
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2007-02-24 19:49:11 +00:00
Olle Johansson bc01e39174 Creating new doxygen macro "\extref" to create page that lists
external libraries and URLs to these. Please help me add these
references.

We might want to create a similar macro "\linuxpackage" to list
the needed Linux packages in popular distributions.



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2007-02-24 19:27:50 +00:00
Jason Parker 97ab07a9e8 Allow a Skinny device to monitor a dialplan hint (w00t!).
See skinny.conf.sample for configuration example.


Note: Some devices (seen on 12SP+/30VIP) will lock up if they monitor too many hints.
This seems to be a hardware limitation - there isn't anything we can do about it.


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2007-02-24 02:23:43 +00:00
Jason Parker a23096985b Support devicestate requests.
Now you should be able to subscribe to a Skinny device/line.


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2007-02-24 02:06:44 +00:00
Jason Parker 639aee2ec7 Merged revisions 56569 via svnmerge from
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r56569 | qwell | 2007-02-23 20:02:53 -0600 (Fri, 23 Feb 2007) | 4 lines

Make sure to set a speeddials parent on creation.
Don't crash if hold is pressed when no call is active.
Don't return in places that we shouldn't..
Update softkey map when call is connected

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2007-02-24 02:04:34 +00:00
Russell Bryant b85a5ffbe3 Make the hashing function calculate something that makes more sense.
(Thanks to bmd on #asterisk-dev for pointing out my pointless math).


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2007-02-23 22:26:00 +00:00
Russell Bryant 9fb1404279 Merge team/russell/iax2_performance.
There is not a large amount of code here and the changes are not very invasive.
However, they should significantly improve performance of chan_iax2 under load.

IAX2 media frames only carry the *source* call number.  So, when one arrives,
the correct session that it is a part of has to be matched on IP address, port
number, and call number, instead of just a call number.  Had these frames
carried the *destination* call number, this would not be an issue, because that
would be a unique identifier that would make it easy to immediately identify
the correct session.

The way that chan_iax2 did this matching was extremely inefficient.  It starts
at the first available call number and traverses each call number sequentially,
locking and unlocking a mutex for each one, to try to match against it.  It
would do this regardless of whether the call number was in use or not.  So,
for a call with a local call number of 25000, every single incoming media
frame would require a traversal that required 25000 mutex lock and unlock
operations.  (Note that the max call number is about 32k).

I have introduced a hash table of active IAX2 calls to improve this lookup
process.  The hash is done on the IP address, port number, and call number.  
So, for the previous example, a few lock/unlock operations may be done versus 
25000 for each frame.


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2007-02-23 21:20:33 +00:00
Russell Bryant 70ce880338 Merged revisions 56407 via svnmerge from
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r56407 | russell | 2007-02-23 14:20:00 -0600 (Fri, 23 Feb 2007) | 12 lines

Merged revisions 56406 via svnmerge from 
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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2007-02-23 20:21:35 +00:00
Joshua Colp afc99294fa Merged revisions 56231 via svnmerge from
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r56231 | file | 2007-02-22 13:49:39 -0500 (Thu, 22 Feb 2007) | 10 lines

Merged revisions 56230 via svnmerge from 
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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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2007-02-22 18:53:22 +00:00
Olle Johansson cb0fddc905 Merged revisions 56125 via svnmerge from
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r56125 | oej | 2007-02-22 11:33:55 +0100 (Thu, 22 Feb 2007) | 2 lines

Move message from verbose to debug

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2007-02-22 10:46:09 +00:00
Russell Bryant a6cbe5d651 Merged revisions 56055 via svnmerge from
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r56055 | russell | 2007-02-21 19:24:10 -0600 (Wed, 21 Feb 2007) | 3 lines

Restructure a little bit of code to reduce nesting.  There is no functionality
change here.

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2007-02-22 01:26:22 +00:00
Russell Bryant 80fce39036 Merged revisions 56011 via svnmerge from
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r56011 | russell | 2007-02-21 18:57:36 -0600 (Wed, 21 Feb 2007) | 11 lines

Merged revisions 56010 via svnmerge from 
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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2007-02-22 00:59:17 +00:00
Jason Parker 28a6129af8 Merged revisions 55954 via svnmerge from
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r55954 | qwell | 2007-02-21 14:27:08 -0600 (Wed, 21 Feb 2007) | 4 lines

Fix locking issue, and accept "transport-accept" as a valid accept message.

This should solve issues 8970 and 8503.

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2007-02-21 20:30:54 +00:00
Joshua Colp cdf9cab49d Clarify in the doxygen docs abou RFC2833 compensation flag.
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2007-02-21 20:26:43 +00:00
Joshua Colp 93671917f9 Merged revisions 55914 via svnmerge from
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r55914 | file | 2007-02-21 12:18:19 -0500 (Wed, 21 Feb 2007) | 2 lines

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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2007-02-21 17:23:42 +00:00
Olle Johansson 4534fbb7bc Merged revisions 55834 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r55834 | oej | 2007-02-21 09:32:34 +0100 (Wed, 21 Feb 2007) | 2 lines

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/trunk@55835 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-21 08:39:15 +00:00
Jason Parker 8f28800765 Merged revisions 55799 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r55799 | qwell | 2007-02-20 20:01:36 -0600 (Tue, 20 Feb 2007) | 4 lines

Fix segfault when buddy couldn't be found.

Issue 7764, patch by sailer

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@55805 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-21 02:04:10 +00:00
Joshua Colp 22c1925696 Merged revisions 55717 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r55717 | file | 2007-02-20 18:57:03 -0500 (Tue, 20 Feb 2007) | 2 lines

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/trunk@55718 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-21 00:00:11 +00:00
Joshua Colp 62cb480f28 Merged revisions 55688 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r55688 | file | 2007-02-20 18:08:45 -0500 (Tue, 20 Feb 2007) | 2 lines

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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@55697 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-20 23:12:55 +00:00