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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@54534 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r54481 | file | 2007-02-14 16:07:23 -0500 (Wed, 14 Feb 2007) | 2 lines
Forward begin DTMF frames as well as end. (issue #9068 reported by mhardeman)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@54466 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@54465 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@54464 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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@54379 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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caused by outbound proxy patch.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@54372 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@54348 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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produces core every time you try to load codec_g722.
Fixed. ;-)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@54325 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r54290 | file | 2007-02-13 20:09:40 -0500 (Tue, 13 Feb 2007) | 2 lines
Add G722 to ast_best_codec. If anyone disagrees with it's placement, feel free to change it. (issue #9045 reported by gork)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@54291 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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pretty cool things.
First, you can get the device state of anything in the dialplan:
NoOp(SIP/mypeer has state ${DEVSTATE(SIP/mypeer)})
NoOp(The conference room 1234 has state ${DEVSTATE(MeetMe:1234)})
Most importantly, this allows you to create custom device states so you can
control phone lamps directly from the dialplan.
Set(DEVSTATE(Custom:mycustomlamp)=BUSY)
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exten => mycustomlamp,hint,Custom:mycustomlamp
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@54261 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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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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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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r54218 | russell | 2007-02-13 14:56:50 -0600 (Tue, 13 Feb 2007) | 3 lines
Fix the documentation on the return values from device state provider
registration and deletion.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@54149 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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line.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@54148 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Simplify printing out the warranty and license
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@54147 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r54103 | russell | 2007-02-12 13:17:08 -0600 (Mon, 12 Feb 2007) | 2 lines
Change ast_set_state_callback() to ast_dial_set_state_callback()
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r54066 | russell | 2007-02-12 11:58:43 -0600 (Mon, 12 Feb 2007) | 4 lines
- Add the ability to register a callback to monitor state changes in an
asynchronous dial operation.
- Rename the various references to "status" to "state" in the dial API
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r54026 | file | 2007-02-12 11:34:45 -0500 (Mon, 12 Feb 2007) | 2 lines
Make the --without-oss argument work. (issue #9026 reported by puzzled)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r54002 | russell | 2007-02-12 10:38:39 -0500 (Mon, 12 Feb 2007) | 2 lines
Fix a typo where "vmpassword" should be "vmsecret"
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release....
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53953 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53933 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53932 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53914 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r53880 | pcadach | 2007-02-10 01:08:55 -0800 (Сбт, 10 Фев 2007) | 1 line
Much simpler than previous one ;-)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r53879 | pcadach | 2007-02-10 01:07:11 -0800 (Сбт, 10 Фев 2007) | 1 line
Provide correct DTMF duration
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r53878 | pcadach | 2007-02-10 01:04:47 -0800 (Сбт, 10 Фев 2007) | 1 line
Bring deprecated 'debug channel <x|all>' command back
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r53850 | kpfleming | 2007-02-10 00:06:08 -0600 (Sat, 10 Feb 2007) | 3 lines
don't display the --with-imap message unless --with-imap was specified without a path
use '-n' instead of '! -z' for tests
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53851 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r53818 | russell | 2007-02-09 18:41:57 -0600 (Fri, 09 Feb 2007) | 2 lines
Change some text to properly state "On Hold", which was already done in trunk.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53819 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53817 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53785 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r53783 | russell | 2007-02-09 18:15:50 -0600 (Fri, 09 Feb 2007) | 4 lines
When the Echo() application receives the digit '#', echo that back as well.
Since we already sent the BEGIN frame for that digit, it makes sense to send
the END as well.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53784 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r53749 | file | 2007-02-09 14:33:31 -0500 (Fri, 09 Feb 2007) | 2 lines
Temporarily change musicclass on channel to one specified in Dial so that the 'm' option functions properly. (issue #8969 reported by christianbee)
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seanbright)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r53715 | kpfleming | 2007-02-09 10:42:22 -0600 (Fri, 09 Feb 2007) | 2 lines
clarify the fact that voicemail IMAP storage cannot be built against a distro's binary c-client library package (at least not at this time)
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read/write as well!
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HTTP server. (issue #9013 reported by Jynger)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r53601 | file | 2007-02-08 12:54:32 -0500 (Thu, 08 Feb 2007) | 2 lines
Fix timeout issue when utterance is longer then timeout itself.
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actually parse the busy-limit option from sip.conf, instead of ignoring it
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53577 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r53532 | tilghman | 2007-02-08 07:47:54 -0600 (Thu, 08 Feb 2007) | 2 lines
Issue 9007 - Mutex not released on early return
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r53530 | tilghman | 2007-02-08 07:40:02 -0600 (Thu, 08 Feb 2007) | 10 lines
Merged revisions 53529 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r53529 | tilghman | 2007-02-08 07:36:10 -0600 (Thu, 08 Feb 2007) | 2 lines
Issue 9003 - If fullname is empty, quote() passes back "\""
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r53497 | russell | 2007-02-07 17:52:45 -0600 (Wed, 07 Feb 2007) | 6 lines
When building libdb1.a, put the additional flags needed at the beginning of
ASTCFLAGS, instead of at the end. This way, we ensure that we find the local
headers first before accidentally trying to use headers that exist in
locations specified in the ASTCFLAGS passed from the main Makefile.
(issue #8637, ovi)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53498 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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