Commit Graph

4570 Commits

Author SHA1 Message Date
Olle Johansson
cb0fddc905 Merged revisions 56125 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@56126 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-22 10:46:09 +00:00
Russell Bryant
a6cbe5d651 Merged revisions 56055 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@55953 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-21 20:26:43 +00:00
Joshua Colp
93671917f9 Merged revisions 55914 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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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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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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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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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2007-02-20 23:12:55 +00:00
Joshua Colp
ab96ee237b Merged revisions 55670 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r55670 | file | 2007-02-20 17:47:00 -0500 (Tue, 20 Feb 2007) | 10 lines

Merged revisions 55669 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r55669 | file | 2007-02-20 17:39:14 -0500 (Tue, 20 Feb 2007) | 2 lines

Defer clearing callback information if channels are up until they are hung up. This ensures the hangup process goes smoothly and no channels get hung in limbo. (issue #8088 reported by kebl0155)

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2007-02-20 22:49:53 +00:00
Jason Parker
ae47fc4541 Merged revisions 55555 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r55555 | qwell | 2007-02-20 10:53:45 -0600 (Tue, 20 Feb 2007) | 4 lines

No need to cast nor free with strdupa (thanks file)

55555!

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2007-02-20 16:56:58 +00:00
Doug Bailey
c6dd460655 Merged revisions 55397 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r55397 | dbailey | 2007-02-19 08:52:59 -0600 (Mon, 19 Feb 2007) | 3 lines

Changed iax2 process thread to detached to correct memory leak due to left over thread context on thread exit.  
Modified module unload process to avoid deadlocks on pthread cancels

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@55409 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-19 15:24:46 +00:00
Joshua Colp
8ebf3659aa Update chan_h323 to new set_rtp_peer definition.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@55351 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-18 19:09:29 +00:00
Kevin P. Fleming
74f31a51bc add -Wundef to the --enable-dev-mode flags, so that mistyped macro names in #if expressions will be caught
convert various #if expressions to #ifdef for macros that may not be defined (and where the value is not important)
Note: two of these changes are in bison generated files which is going to be inconvenient when they are regenerated


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@55329 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-18 15:03:42 +00:00
Jason Parker
0e5f3cf25e Merged revisions 55217 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r55217 | qwell | 2007-02-17 11:10:09 -0600 (Sat, 17 Feb 2007) | 4 lines

Fix an issue where callerid would not be displayed on some phones.

Issue 8995, initial patch and research done by wedhorn

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@55218 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-17 17:11:07 +00:00
Joshua Colp
354c03c4e6 Update chan_jingle to new definition of set_rtp_peer.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@55088 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-17 01:37:29 +00:00
Joshua Colp
977fb01cdd Merged revisions 55086 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r55086 | file | 2007-02-16 20:16:59 -0500 (Fri, 16 Feb 2007) | 10 lines

Merged revisions 55073 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r55073 | file | 2007-02-16 20:09:50 -0500 (Fri, 16 Feb 2007) | 2 lines

Allow chan_sip to handle attended transfers from a SIP phone that is sitting behind chan_agent. Yes folks, all it took was one line of code. (issue #8784 reported by pzieba)

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2007-02-17 01:22:01 +00:00
Joshua Colp
4331761b5e Merged revisions 55002 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r55002 | file | 2007-02-16 17:18:46 -0500 (Fri, 16 Feb 2007) | 10 lines

Merged revisions 54999 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r54999 | file | 2007-02-16 17:13:45 -0500 (Fri, 16 Feb 2007) | 2 lines

Do not send indications through ast_indicate in chan_agent but instead go directly to the technology. This way when indications are emulated they happen on the Agent channel and do not screw up formats on the channels. (issue #8439 reported by punkgode)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@55003 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-16 22:20:55 +00:00
Olle Johansson
cfb3f84979 Formatting, whitespace fixes
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@54862 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-16 14:31:18 +00:00
Olle Johansson
ba32ee49d0 Adding Realtime Text support (T.140) to Asterisk
T.140/RFC 2793 is a live communication channel, originally
created for IP based text phones for hearing impaired. 
Feels very much like the old Unix talk application.

This code is developed and disclaimed by John Martin of Aupix, UK.
Tested for interoperability by myself and Omnitor in Sweden,
the company that wrote most of the specifications.

A big thank you to everyone involved in this.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@54838 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-16 13:35:44 +00:00
Olle Johansson
84d1cf37fe Merged revisions 54787 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r54787 | oej | 2007-02-16 13:06:23 +0100 (Fri, 16 Feb 2007) | 2 lines

Issue #7541 - Handle multipart attachments to SIP messages - even if boundary is quoted.

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2007-02-16 12:14:53 +00:00
Olle Johansson
fef74f1574 Add callgroup and pickupgroup to SIPPEER function. (thanks ramon)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@54598 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-15 15:52:35 +00:00
Olle Johansson
1f52d1cc81 Issue #7443 - amdtech - Optionally SIP registrations in another
realtime family. 


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@54574 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-15 12:10:55 +00:00
Matthew Fredrickson
daa5f504ec Merged revisions 54375 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r54375 | mattf | 2007-02-14 10:56:40 -0600 (Wed, 14 Feb 2007) | 10 lines

Merged revisions 54373 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r54373 | mattf | 2007-02-14 10:25:49 -0600 (Wed, 14 Feb 2007) | 2 lines

When handling glare on a PRI, move the requested channel rather than hang up the old one. Fix for 8957 and 9011.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@54404 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-14 17:14:42 +00:00
Olle Johansson
276b570c3e Issue #9060 - host= parameter in sip.conf stopped working
caused by outbound proxy patch.


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2007-02-14 16:20:48 +00:00
Olle Johansson
0653be0c33 Add port number to SIPPEER dialplan function
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@54348 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-14 15:27:49 +00:00
Russell Bryant
83856d4683 Merged revisions 54204 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r54204 | russell | 2007-02-13 13:42:00 -0600 (Tue, 13 Feb 2007) | 5 lines

If we fail to create the SIP socket, then return -1 from reload_config() so
that load_module() will return AST_MODULE_LOAD_DECLINE.  Otherwise, the console
will just get spammed with error messages every time chan_sip tries to send a
message.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@54260 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-13 21:57:31 +00:00
Russell Bryant
9e99a51802 Merged revisions 54235 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r54235 | russell | 2007-02-13 15:31:22 -0600 (Tue, 13 Feb 2007) | 2 lines

Remove a couple of leftover debug messages

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2007-02-13 21:33:03 +00:00
Olle Johansson
1295d40d77 Be careful with debug messages in trunk, they tend to stay around for release....
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53953 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-11 20:49:38 +00:00
Olle Johansson
6e139adc56 Small fix in outbound proxy support.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53933 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-11 20:04:49 +00:00
Olle Johansson
32495f91f0 Add support for outbound proxy for peers and [general]
This replaces the older, broken, implementation where a setting in
[general] did not do anything and the [peer] part was broken.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53932 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-11 19:42:55 +00:00
Paul Cadach
012e99016c Merged revisions 53881 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r53881 | pcadach | 2007-02-10 01:09:49 -0800 (Сбт, 10 Фев 2007) | 1 line

Fix VLDTMF reception
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53885 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-10 09:23:09 +00:00
Russell Bryant
5715b49c30 Merged revisions 53810 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r53810 | russell | 2007-02-09 18:35:09 -0600 (Fri, 09 Feb 2007) | 24 lines

Merge team/russell/sla_rewrite

This is a completely new implementation of the SLA functionality introduced in
Asterisk 1.4.  It is now functional and ready for testing.  However, I will be
adding some additional features over the next week, as well.

For information on how to set this up, see configs/sla.conf.sample 
and doc/sla.txt.

In addition to the changes in app_meetme.c for the SLA implementation itself,
this merge brings in various other changes:

chan_sip:
 - Add the ability to indicate HOLD state in NOTIFY messages.
 - Queue HOLD and UNHOLD control frames even if the channel is not bridged to
   another channel.

linkedlists.h:
 - Add support for rwlock based linked lists.

dial.c:
 - Add the ability to run ast_dial_start() without a reference channel to
   inherit information from.

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2007-02-10 00:40:57 +00:00
Russell Bryant
ac4090fce0 add another dependency
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2007-02-10 00:20:57 +00:00
Kevin P. Fleming
e758ef09b3 Merged revisions 53779-53781 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r53779 | kpfleming | 2007-02-09 17:51:29 -0600 (Fri, 09 Feb 2007) | 2 lines

fix awk scripts to work when both MODULEINFO and MAKEOPTS are present in a source file

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r53780 | kpfleming | 2007-02-09 17:51:41 -0600 (Fri, 09 Feb 2007) | 2 lines

add some inter-module dependencies

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r53781 | kpfleming | 2007-02-09 17:52:44 -0600 (Fri, 09 Feb 2007) | 2 lines

another dependency

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2007-02-09 23:53:51 +00:00
Jason Parker
cee4bd43dc Rename this instance of "busy limit" to "busy level" as well
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2007-02-08 17:19:27 +00:00
Kevin P. Fleming
44c6630e4d rename busy-limit to busy-level, since it is not a limit
actually parse the busy-limit option from sip.conf, instead of ignoring it


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2007-02-08 16:41:23 +00:00
Joshua Colp
bd41ac0ae8 Merged revisions 53358 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r53358 | file | 2007-02-07 10:43:39 -0500 (Wed, 07 Feb 2007) | 10 lines

Merged revisions 53357 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r53357 | file | 2007-02-07 10:38:48 -0500 (Wed, 07 Feb 2007) | 2 lines

Fix a few potential memory leaks with realtime users and peers. (issue #8999 reported by bsmithurst)

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2007-02-07 15:46:30 +00:00
Christian Richter
a96e62d9d2 Merged revisions 53324 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r53324 | crichter | 2007-02-07 10:22:44 +0100 (Mi, 07 Feb 2007) | 9 lines

Merged revisions 52843 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r52843 | crichter | 2007-01-30 15:38:08 +0100 (Di, 30 Jan 2007) | 1 line

fixed some possible segfaults. also fixed an very important bug which occurs on high load (when calls are very fast generated)
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2007-02-07 09:51:06 +00:00
Olle Johansson
17af1bd4c8 Merged revisions 53143 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r53143 | oej | 2007-02-05 01:18:34 +0100 (Mon, 05 Feb 2007) | 3 lines

Add some comments on queue system behaviour and how it affects the
SIP channel

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2007-02-05 00:30:03 +00:00
Joshua Colp
014feba426 Merged revisions 53138 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r53138 | file | 2007-02-03 15:05:02 -0600 (Sat, 03 Feb 2007) | 2 lines

Make SIPDtmfMode application work with recent capability changes, and also fix an RTP stack issue when the auto option was used. (issue #8972 reported by mdu113)

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2007-02-03 21:06:36 +00:00
Paul Cadach
3e5e0f5f00 Merged revisions 53131 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r53131 | pcadach | 2007-02-03 02:02:55 -0800 (Сбт, 03 Фев 2007) | 1 line

Remove quote from H.323 vendor string because due to compatibilities with
Nortel Meridian CS1000 reported at www.voip-info.org
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2007-02-03 10:12:20 +00:00
Joshua Colp
ce8a7c3d9c Add onHold value to sip show inuse as well.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53125 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-02 18:21:46 +00:00
Olle Johansson
cfe66e6b26 Patch based on this patch with small changes for trunk...
Merged revisions 53109 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r53109 | oej | 2007-02-02 01:24:03 +0100 (Fri, 02 Feb 2007) | 4 lines

Disable the direct p2p RTP call setup in SIP. You can enable it in sip.conf, but it is now
considered experimental until we solve the AST_CONTROL_ANSWER with payload and videocaps
stuff.

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2007-02-02 00:26:25 +00:00
Joshua Colp
44a9af3576 Merged revisions 53104 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r53104 | file | 2007-02-01 16:24:32 -0600 (Thu, 01 Feb 2007) | 10 lines

Merged revisions 53103 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r53103 | file | 2007-02-01 16:21:56 -0600 (Thu, 01 Feb 2007) | 2 lines

Copy noncodeccapability over to the joint variable so that telephone-event will get transmitted in the sent INVITE.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53105 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-01 22:26:11 +00:00
Joshua Colp
bf66c620c3 Merged revisions 53097 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r53097 | file | 2007-02-01 15:54:28 -0600 (Thu, 01 Feb 2007) | 10 lines

Merged revisions 53095 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r53095 | file | 2007-02-01 15:47:11 -0600 (Thu, 01 Feb 2007) | 2 lines

Don't negotiate RFC2833 when not configured to do so. (issue #8799 reported by mdu113) 

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53098 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-01 21:56:23 +00:00
Olle Johansson
544f414c0d Merged revisions 53085 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r53085 | oej | 2007-02-01 22:05:34 +0100 (Thu, 01 Feb 2007) | 4 lines

- Clean INC_COUNT flag when we decrement call counter
- If it's still set at time of dialog destruction, make sure we decrement the device call counter properly
  before we destroy the dialog

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53092 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-01 21:17:08 +00:00
Olle Johansson
0b84b386b9 Implementing "busy-limit".
If you set call limit and busy limit, chan_sip will indicate BUSY for a device
that has reached the busy limit and allow calls up to the call limit, allowing
for call transfers (that generate a new call).

If you only set call limit, chan_sip will not indicate BUSY until that limit
is filled. 

This affects SIP subscriptions, call queues and manager applications.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53082 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-01 20:43:49 +00:00
Olle Johansson
34eaa61700 Merged revisions 53079 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r53079 | oej | 2007-02-01 21:28:54 +0100 (Thu, 01 Feb 2007) | 2 lines

Cleaning up the devicestate callback function

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53080 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-01 20:33:59 +00:00