0a4f7d0352
Issue #7443 - amdtech - Optionally SIP registrations in another
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realtime family.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@54574 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-15 12:10:55 +00:00
89b2a62b82
Clean up the I/O context handler.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@54552 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-15 02:11:34 +00:00
9e8f015acd
Few more code clean ups.
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2007-02-15 02:02:30 +00:00
3cc71d5fb0
Clean up app_milliwatt code.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@54528 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-15 01:52:10 +00:00
0e24af1090
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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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2007-02-14 21:10:53 +00:00
d1ed177de1
Show version in "core show settings"
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2007-02-14 20:45:23 +00:00
c05d59a018
Updates and re-organization to make it easier to digest this information
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2007-02-14 20:31:10 +00:00
18b268814d
New CLI command "Core show settings" to list some core settings
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@54464 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-14 20:22:20 +00:00
21a2f91b3e
Merged revisions 54375 via svnmerge from
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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
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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: http://svn.digium.com/svn/asterisk/trunk@54404 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-14 17:14:42 +00:00
40ad6c100d
Make documentation match the source code.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@54379 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-14 17:02:16 +00:00
d32f177a29
Issue #9060 - host= parameter in sip.conf stopped working
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caused by outbound proxy patch.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@54372 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-14 16:20:48 +00:00
554875989e
Add port number to SIPPEER dialplan function
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@54348 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-14 15:27:49 +00:00
81d103d263
I don't know how it worked earlier, but valgrind
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produces core every time you try to load codec_g722.
Fixed. ;-)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@54325 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-14 08:34:10 +00:00
4debc4eead
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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: http://svn.digium.com/svn/asterisk/trunk@54291 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-14 01:12:21 +00:00
bb6564f8e7
This introduces a new dialplan function, DEVSTATE, which allows you to do some
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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
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2007-02-13 22:02:20 +00:00
76f1766c9f
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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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2007-02-13 21:57:31 +00:00
20050f395d
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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
4b62a35dfb
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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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2007-02-13 20:57:41 +00:00
2387cd4cdf
Use spaces instead of tabs in the help text for a CLI command
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@54149 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-13 06:10:11 +00:00
71bcc4064f
Simplify WELCOME_MESSAGE to be a single function call instead of one for each
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line.
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2007-02-13 06:02:57 +00:00
32e9fe9768
- Constify the format string passed to ast_cli()
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- Simplify printing out the warranty and license
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@54147 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-13 05:57:52 +00:00
fa69a1b414
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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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2007-02-12 19:18:33 +00:00
f065afb987
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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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2007-02-12 18:01:15 +00:00
a563a68e9f
Blocked revisions 54026 via svnmerge
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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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2007-02-12 16:40:54 +00:00
3c410b1cf2
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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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2007-02-12 15:48:28 +00:00
b83ba7c021
Simplify a small bit of logic.
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2007-02-12 15:40:23 +00:00
15abba63d9
Formatting fixes
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2007-02-12 02:44:33 +00:00
44942cb5c1
Be careful with debug messages in trunk, they tend to stay around for release....
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@53953 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-11 20:49:38 +00:00
53793ff5f0
Small fix in outbound proxy support.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@53933 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-11 20:04:49 +00:00
5ba4828ffc
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.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@53932 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-11 19:42:55 +00:00
6fc0c809bb
Fix debug handling in acl.c
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2007-02-11 19:15:55 +00:00
9e58b235fc
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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
d616dd92d9
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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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2007-02-10 09:22:15 +00:00
8be4fd12ab
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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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2007-02-10 09:21:22 +00:00
bb5299f19a
Blocked revisions 53878 via svnmerge
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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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2007-02-10 09:19:58 +00:00
65019aebdf
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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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2007-02-10 06:14:55 +00:00
512ca565ba
Blocked revisions 53818 via svnmerge
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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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2007-02-10 00:42:19 +00:00
be94f38009
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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
0fbac396a5
add another dependency
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2007-02-10 00:20:57 +00:00
26566c5864
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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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2007-02-10 00:16:22 +00:00
f56e86c1e1
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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
ff82f3cb25
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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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2007-02-09 19:39:26 +00:00
3397dfdc03
Clean up documentation of Queue application. (issue #9022 reported by seanbright)
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2007-02-09 17:14:10 +00:00
3b057a16a8
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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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2007-02-09 16:43:36 +00:00
eb2358c03c
I'm crazy so I think I'll change the musiconhold classes linked list to read/write as well!
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2007-02-09 01:57:48 +00:00
1e80cfd061
It is with pleasure that I announce the return of rawman support through the HTTP server. (issue #9013 reported by Jynger)
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2007-02-08 20:39:41 +00:00
e617d4e0d4
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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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2007-02-08 17:56:53 +00:00
b9c22bbbec
Rename this instance of "busy limit" to "busy level" as well
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2007-02-08 17:19:27 +00:00
4f15869b21
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
31d7b2af85
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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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2007-02-08 13:50:33 +00:00
oej