[asterisk-bugs] [Asterisk 0012803]: [patch] one way only audio in MS-Office 2007 to asterisk calls

noreply at bugs.digium.com noreply at bugs.digium.com
Mon Jun 16 13:25:12 CDT 2008


A NOTE has been added to this issue. 
====================================================================== 
http://bugs.digium.com/view.php?id=12803 
====================================================================== 
Reported By:                lanzaandrea
Assigned To:                file
====================================================================== 
Project:                    Asterisk
Issue ID:                   12803
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     closed
Asterisk Version:           1.6.0-beta8 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
Resolution:                 fixed
Fixed in Version:           
====================================================================== 
Date Submitted:             06-06-2008 08:24 CDT
Last Modified:              06-16-2008 13:25 CDT
====================================================================== 
Summary:                    [patch] one way only audio in MS-Office 2007 to
asterisk calls
Description: 
When connectin an asterisk server 1.6.0beta8 to an MS Office Communication
server 2007 Mediation server role, you can place bidirectiona audio calls
when calling from an asterisk to a MS ocs client
doing the oppsite, asterisk phone speak and the ms ocs client listen, but
not the opposite.
The problem is the presence of a sip header 
application/sdp; charset=utf-8
sent by MS OCS unknown and unhandled by chan_sip


====================================================================== 

---------------------------------------------------------------------- 
 svnbot - 06-16-08 13:25  
---------------------------------------------------------------------- 
Repository: asterisk
Revision: 123107

_U  team/seanbright/resolve-shadow-warnings/
U   team/seanbright/resolve-shadow-warnings/UPGRADE.txt
A   team/seanbright/resolve-shadow-warnings/apps/app_fax.c
U   team/seanbright/resolve-shadow-warnings/cdr/cdr_tds.c
U   team/seanbright/resolve-shadow-warnings/channels/chan_iax2.c
U   team/seanbright/resolve-shadow-warnings/channels/chan_sip.c
U   team/seanbright/resolve-shadow-warnings/configs/modules.conf.sample
U   team/seanbright/resolve-shadow-warnings/funcs/func_channel.c
U   team/seanbright/resolve-shadow-warnings/include/asterisk/_private.h
U   team/seanbright/resolve-shadow-warnings/include/asterisk/config.h
U   team/seanbright/resolve-shadow-warnings/include/asterisk/timing.h
U   team/seanbright/resolve-shadow-warnings/main/asterisk.c
U   team/seanbright/resolve-shadow-warnings/main/channel.c
U   team/seanbright/resolve-shadow-warnings/main/timing.c
A   team/seanbright/resolve-shadow-warnings/res/res_timing_pthread.c

------------------------------------------------------------------------
r123107 | seanbright | 2008-06-16 13:25:06 -0500 (Mon, 16 Jun 2008) | 109
lines

Merged revisions
122766,122802,122834,122870,122920,122923,122926,122928,122977,123009,123041,123044,123076
via svnmerge from 
https://origsvn.digium.com/svn/asterisk/trunk

................
r122766 | tilghman | 2008-06-13 18:52:20 -0400 (Fri, 13 Jun 2008) | 2
lines

Document the input for ast_realtime_require_field()

................
r122802 | tilghman | 2008-06-15 11:21:16 -0400 (Sun, 15 Jun 2008) | 8
lines

Add some more IAX2-specific information about the channel to the CHANNEL()
function and begin the transition from SIPCHANINFO() to just using
CHANNEL().
(closes issue http://bugs.digium.com/view.php?id=12856)
 Reported by: mostyn
 Patches: 
       iax_and_sip_channel_info.patch uploaded by mostyn (license 398)
       (with some additional cleanup by me)

................
r122834 | seanbright | 2008-06-15 23:33:03 -0400 (Sun, 15 Jun 2008) | 1
line

Resurrected app_fax
................
r122870 | file | 2008-06-16 08:09:54 -0400 (Mon, 16 Jun 2008) | 14 lines

Merged revisions 122869 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r122869 | file | 2008-06-16 09:08:28 -0300 (Mon, 16 Jun 2008) | 6 lines

Don't send a BYE on a dialog that is already gone during a REFER.
(closes issue http://bugs.digium.com/view.php?id=12865)
Reported by: flefoll
Patches:
      chan_sip.c.br14.121495.patch-ALREADYGONE uploaded by flefoll
(license 244)

........

................
r122920 | file | 2008-06-16 08:32:02 -0400 (Mon, 16 Jun 2008) | 14 lines

Merged revisions 122919 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r122919 | file | 2008-06-16 09:31:09 -0300 (Mon, 16 Jun 2008) | 6 lines

Only compare the first 15 characters so that even if the charset is
specified we still accept it as SDP.
(closes issue http://bugs.digium.com/view.php?id=12803)
Reported by: lanzaandrea
Patches:
      chan_sip.c.diff uploaded by lanzaandrea (license 496)

........

................
r122923 | russell | 2008-06-16 08:48:11 -0400 (Mon, 16 Jun 2008) | 5 lines

 - Fix a typo in a timing API call
 - Convert the last part of channel.c over to use the timing API.  This
would
   not have made a difference when using the dahdi timing module.  I
noticed
   it when trying to use another timing source.  Oops.  :)

................
r122926 | russell | 2008-06-16 09:03:40 -0400 (Mon, 16 Jun 2008) | 4 lines

Add a "timing test" CLI command.  It opens a timer and configures it for
50 ticks per second, and then counts to see how many ticks it actually
gets in a second.

................
r122928 | russell | 2008-06-16 09:08:13 -0400 (Mon, 16 Jun 2008) | 11
lines

Merge res_timing_pthread.  This is a timing interface for Asterisk that
does not require DAHDI.  It's called "pthread" because it uses a pthread
API call in the timing thread for sleeping and ensuring we wake up at
an appropriate time.  I wasn't sure what else to call it.  :)

The timing API requires a file descriptor that can be polled on.  So,
when you open a timer, this module creates a pipe and returns the read
end of the pipe.  There is a background thread that wakes up every 10ms
and checks to see if any of the currently open timers need a 'tick' and
writes to the appropriate pipe.

................
r122977 | russell | 2008-06-16 09:31:36 -0400 (Mon, 16 Jun 2008) | 2 lines

Note that only one timing interface should get loaded.

................
r123009 | seanbright | 2008-06-16 11:25:03 -0400 (Mon, 16 Jun 2008) | 1
line

Coding guidelines stuff only.
................
r123041 | seanbright | 2008-06-16 12:29:18 -0400 (Mon, 16 Jun 2008) | 1
line

Remove some unused variables
................
r123044 | seanbright | 2008-06-16 13:14:11 -0400 (Mon, 16 Jun 2008) | 1
line

Convert to use stringfields.  Still some more work to do on config
load/reload.
................
r123076 | seanbright | 2008-06-16 13:33:10 -0400 (Mon, 16 Jun 2008) | 1
line

Last commit for a bit, minor cleanups and move the lock initialization.
................

------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=123107 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
06-16-08 13:25  svnbot         Note Added: 0088768                          
======================================================================




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