Olle Johansson
32495f91f0
Add support for outbound proxy for peers and [general]
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This replaces the older, broken, implementation where a setting in
[general] did not do anything and the [peer] part was broken.
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2007-02-11 19:42:55 +00:00
Paul Cadach
012e99016c
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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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2007-02-10 09:23:09 +00:00
Russell Bryant
5715b49c30
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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
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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
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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
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r53358 | file | 2007-02-07 10:43:39 -0500 (Wed, 07 Feb 2007) | 10 lines
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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
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r53324 | crichter | 2007-02-07 10:22:44 +0100 (Mi, 07 Feb 2007) | 9 lines
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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
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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
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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
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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.
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2007-02-02 18:21:46 +00:00
Olle Johansson
cfe66e6b26
Patch based on this patch with small changes for trunk...
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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
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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
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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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2007-02-01 22:26:11 +00:00
Joshua Colp
bf66c620c3
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r53097 | file | 2007-02-01 15:54:28 -0600 (Thu, 01 Feb 2007) | 10 lines
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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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2007-02-01 21:56:23 +00:00
Olle Johansson
544f414c0d
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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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2007-02-01 21:17:08 +00:00
Olle Johansson
0b84b386b9
Implementing "busy-limit".
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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.
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2007-02-01 20:43:49 +00:00
Olle Johansson
34eaa61700
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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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2007-02-01 20:33:59 +00:00
Olle Johansson
38b87ec4b7
Signal HOLD status to phones that subscribe for status.
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2007-02-01 19:04:47 +00:00
Joshua Colp
e88dda8ca9
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r53064 | file | 2007-02-01 11:37:44 -0600 (Thu, 01 Feb 2007) | 2 lines
Fix silly logic. We really want to write UDPTL frames out when the call is up.
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2007-02-01 17:42:08 +00:00
Paul Cadach
e5f184ee43
Oops -- Merged revisions 53057 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r53057 | pcadach | 2007-02-01 03:07:41 -0800 (Чтв, 01 Фев 2007) | 1 line
chan_h323 is very stable, so let it built by default
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2007-02-01 11:16:00 +00:00
Russell Bryant
b233892198
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r53046 | russell | 2007-01-31 15:32:08 -0600 (Wed, 31 Jan 2007) | 11 lines
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r53045 | russell | 2007-01-31 15:25:11 -0600 (Wed, 31 Jan 2007) | 3 lines
Fix a bunch of places where pthread_attr_init() was called, but
pthread_attr_destroy() was not.
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2007-01-31 21:35:15 +00:00
Russell Bryant
d11f8b7ccd
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r52952 | russell | 2007-01-30 13:33:12 -0600 (Tue, 30 Jan 2007) | 5 lines
Only set the DTMF flag on the rtp structure if the DTMF mode is actually
RFC2833, not just that it is not INFO. This makes it get set for inband DTMF
as well, which is not valid.
(issue #8936 )
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2007-01-30 19:36:28 +00:00
Joshua Colp
1fc144435d
Use provided variable for name instead of one in the structure since the structure was just allocated and will be NULL. (issue #8938 reported by st41ker)
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2007-01-30 15:39:09 +00:00
Russell Bryant
116952bba1
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r52763 | russell | 2007-01-29 18:15:50 -0600 (Mon, 29 Jan 2007) | 13 lines
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r52762 | russell | 2007-01-29 18:15:06 -0600 (Mon, 29 Jan 2007) | 5 lines
Fix the extraction of the timestamp from video frames. It was using the
mapping for a mini-frame instead of a video-frame, which caused it to
get invalid data.
(issue #8795 , mihai)
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2007-01-30 00:16:38 +00:00
Joshua Colp
8e755c25ca
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r52370 | file | 2007-01-26 19:08:18 -0500 (Fri, 26 Jan 2007) | 10 lines
Merged revisions 52360 via svnmerge from
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r52360 | file | 2007-01-26 19:03:23 -0500 (Fri, 26 Jan 2007) | 2 lines
Make the last context entry read in the dominant one. (issue #8918 reported by pj)
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2007-01-27 00:10:19 +00:00
Joshua Colp
300f980223
Use atomic operation functions for use/ringing/hold manipulation.
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2007-01-25 18:10:18 +00:00
Joshua Colp
b8d6cbcd3f
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r52210 | file | 2007-01-25 12:49:39 -0500 (Thu, 25 Jan 2007) | 2 lines
Drop out variables I accidentally put in.
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2007-01-25 17:51:35 +00:00
Joshua Colp
afb9151e19
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r52208 | file | 2007-01-25 12:14:53 -0500 (Thu, 25 Jan 2007) | 2 lines
Decrement onHold count if we are hung up on and still on hold. (issue #8909 reported by alexh42)
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2007-01-25 17:17:56 +00:00
Joshua Colp
2396e24e65
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r52016 | file | 2007-01-24 12:59:55 -0500 (Wed, 24 Jan 2007) | 2 lines
Fix changing channel formats when joint capability changes and there are no audio formats... I didn't break it originally! (issue #8535 reported by ivoc)
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2007-01-24 18:04:47 +00:00
Olle Johansson
273f1d78c7
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r51931 | oej | 2007-01-24 10:30:21 +0100 (Wed, 24 Jan 2007) | 3 lines
Show capabilities *and* preference in general settings in "sip show settings"
(reported by Clona/Telio - Thanks!)
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2007-01-24 09:42:31 +00:00
Joshua Colp
ee3ab150f6
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r51788 | file | 2007-01-23 17:46:31 -0500 (Tue, 23 Jan 2007) | 2 lines
Update channel drivers to use module referencing so that unloading them while in use will not result in crashes. (issue #8897 reported by junky)
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2007-01-23 22:59:55 +00:00
Olle Johansson
ef4db783c6
Issue #8817 - Registry corruption when packet retransmits fail. (tootai, patchy by oej)
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2007-01-23 15:36:01 +00:00
Paul Cadach
cd61f69a83
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r51615 | pcadach | 2007-01-22 22:51:51 -0800 (Пнд, 22 Янв 2007) | 1 line
Do not abort Asterisk startup if h323 configuration file not found (reported by mithraen)
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2007-01-23 06:56:26 +00:00
Joshua Colp
34df128519
Add SRV Lookup support on outbound calls to chan_iax2. It's listed in the RFC so we might want to support it and please don't hurt me Marko ... (issue #7812 reported by drorlb)
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2007-01-23 03:15:04 +00:00
Joshua Colp
5a25c156c6
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r51558 | file | 2007-01-22 22:00:12 -0500 (Mon, 22 Jan 2007) | 2 lines
Only change audio formats on the channel if we have an audio format to change to. (issue #8535 reported by ivoc)
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2007-01-23 03:02:09 +00:00
Olle Johansson
1a5dfca2a1
Remove (to quote Rizzo) "useless" variable.
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2007-01-22 19:00:25 +00:00
Russell Bryant
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r51328 | russell | 2007-01-19 13:08:25 -0600 (Fri, 19 Jan 2007) | 5 lines
Fix VLDTMF support in chan_gtalk. AST_FRAME_DTMF and AST_FRAME_DTMF_END are
actually the same thing. So, a digit would have been interpreted incorrectly
here. Since the channel driver will always have the begin and end callbacks
called for a digit, only support the button-down and button-up messages.
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2007-01-19 19:09:04 +00:00
Russell Bryant
dcca8f345f
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r51311 | russell | 2007-01-19 11:49:38 -0600 (Fri, 19 Jan 2007) | 23 lines
Merge the changes from the /team/group/vldtmf_fixup branch.
The main bug being addressed here is a problem introduced when two SIP
channels using SIP INFO dtmf have their media directly bridged. So, when a
DTMF END frame comes into Asterisk from an incoming INFO message, Asterisk
would try to emulate a digit of some length by first sending a DTMF BEGIN
frame and sending a DTMF END later timed off of incoming audio. However,
since there was no audio coming in, the DTMF_END was never generated. This
caused DTMF based features to no longer work.
To fix this, the core now knows when a channel doesn't care about DTMF BEGIN
frames (such as a SIP channel sending INFO dtmf). If this is the case, then
Asterisk will not emulate a digit of some length, and will instead just pass
through the single DTMF END event.
Channel drivers also now get passed the length of the digit to their digit_end
callback. This improves SIP INFO support even further by enabling us to put
the real digit duration in the INFO message instead of a hard coded 250ms.
Also, for an incoming INFO message, the duration is read from the frame and
passed into the core instead of just getting ignored.
(issue #8597 , maybe others...)
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2007-01-19 18:06:03 +00:00
Luigi Rizzo
a52fa7e5ad
sizeof() is compatible with format %d so don't be too
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picky on printf formats.
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2007-01-19 17:45:46 +00:00
Luigi Rizzo
455ea1c99e
remove variable declaration in the middle of a block
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51308 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-01-19 17:38:28 +00:00
Luigi Rizzo
0a8d59d88a
whoops, fix a cut&paste error...
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51298 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-01-19 16:35:37 +00:00
Luigi Rizzo
03fc21712a
slight change to the initialization of a structure,
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also using '\0' to make it clear we need a (char)0
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51297 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-01-19 16:32:05 +00:00
Luigi Rizzo
3fefb9d516
include "asterisk/zapata.h" to get the zaptel headers.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51295 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-01-19 16:28:24 +00:00
Luigi Rizzo
dc31eaa025
include "asterisk/zapata.h" instead of looking
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directly for the zaptel.h and tonezone.h
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51292 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-01-19 16:16:19 +00:00
Dwayne M. Hubbard
18c23101a5
Merged revisions 51274 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r51274 | dhubbard | 2007-01-18 18:17:32 -0600 (Thu, 18 Jan 2007) | 3 lines
chan_zap compiles without libpri after committing 7877 patch
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51275 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-01-19 00:28:25 +00:00
Dwayne M. Hubbard
33243c6624
Merged revisions 51272 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r51272 | dhubbard | 2007-01-18 17:56:49 -0600 (Thu, 18 Jan 2007) | 11 lines
Merged revisions 51271 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r51271 | dhubbard | 2007-01-18 17:47:10 -0600 (Thu, 18 Jan 2007) | 3 lines
issue 7877: chan_zap module reload does not use default/initialized values on subsequent loads. Reset configuration variables to default values prior to parsing configuration file.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51273 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-01-19 00:06:35 +00:00
Joshua Colp
1a06a58250
Merged revisions 51243 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r51243 | file | 2007-01-18 13:36:35 -0500 (Thu, 18 Jan 2007) | 2 lines
Copy MOH settings when calling a peer so that if they put someone on hold or get put on hold themselves they get the right music class. (issue #8840 reported by mdu113)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51244 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-01-18 18:39:21 +00:00
Paul Cadach
ae286e1843
Update ast_append_ha() usage
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51221 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-01-18 06:59:22 +00:00