Russell Bryant
d6f2b914e8
Merged revisions 67020 via svnmerge from
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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".
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(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
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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
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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
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r66768 | file | 2007-05-31 12:14:48 -0400 (Thu, 31 May 2007) | 10 lines
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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!
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(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
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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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2007-05-29 19:35:43 +00:00
Olle Johansson
6d6c525b10
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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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2007-05-29 19:17:49 +00:00
Olle Johansson
0b67a7d80a
Merged revisions 66414 via svnmerge from
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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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2007-05-29 16:19:53 +00:00
Olle Johansson
f4e81d7a54
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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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2007-05-29 16:17:17 +00:00
Olle Johansson
36f15091bb
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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
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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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2007-05-24 15:29:10 +00:00
Olle Johansson
bd2ae8b587
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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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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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
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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
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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
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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
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- 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
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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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2007-05-23 21:01:18 +00:00
Kevin P. Fleming
5dc23536ec
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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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2007-05-23 20:53:10 +00:00
Kevin P. Fleming
04da47df24
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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
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r65677 | kpfleming | 2007-05-23 16:07:59 -0400 (Wed, 23 May 2007) | 10 lines
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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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2007-05-23 15:37:51 +00:00
Russell Bryant
5558323d0c
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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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2007-05-23 13:10:28 +00:00
Russell Bryant
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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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2007-05-22 18:41:56 +00:00
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
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r65342 | crichter | 2007-05-22 10:12:20 +0200 (Di, 22 Mai 2007) | 9 lines
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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
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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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2007-05-18 18:18:59 +00:00
Olle Johansson
1bc7fdeb6b
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r65076 | oej | 2007-05-18 17:18:13 +0200 (Fri, 18 May 2007) | 13 lines
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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
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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
https://origsvn.digium.com/svn/asterisk/branches/1.2
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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@65040 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-18 12:43:59 +00:00
Christian Richter
37f024bb90
Merged revisions 64904 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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
https://origsvn.digium.com/svn/asterisk/branches/1.2
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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@65006 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-18 11:17:16 +00:00
Olle Johansson
491827a28a
Merged revisions 64974 via svnmerge from
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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@64975 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-18 10:41:31 +00:00
Olle Johansson
ac343d43c8
Makeup, darling.
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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
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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
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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@64963 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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