[asterisk-bugs] [Asterisk 0017666]: Direct RTP failures

Asterisk Bug Tracker noreply at bugs.digium.com
Thu Jul 29 14:28:08 CDT 2010


The following issue requires your FEEDBACK. 
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https://issues.asterisk.org/view.php?id=17666 
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Reported By:                digitalc
Assigned To:                
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Project:                    Asterisk
Issue ID:                   17666
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.6.2.9 
JIRA:                       SWP-1869 
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2010-07-16 18:29 CDT
Last Modified:              2010-07-29 14:28 CDT
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Summary:                    Direct RTP failures
Description: 
Related to bug https://issues.asterisk.org/view.php?id=14244

This bug is written from the perspective of Asterisk B, as that is there
the problem seems to be.

A call comes from OpenSER to Asterisk A.
Asterisk A calls Asterisk B and reinvites audio, apparently successfully.
The caller dials an extension.  Asterisk B calls an IP phone with
directmedia enabled.
The caller can hear the dialed party, but the dialed party cannot hear the
caller.

It appears that Asterisk B neglects to tell Asterisk A where to send the
audio to once the IP phone has been answered.


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Relationships       ID      Summary
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related to          0014244 No Audio on Call Transfer (Invite not b...
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---------------------------------------------------------------------- 
 (0125297) lmadsen (administrator) - 2010-07-29 14:28
 https://issues.asterisk.org/view.php?id=17666#c125297 
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You're probably onto something here, but we really do need to see the
configuration here or we're going to spend hours attempting to reproduce
your particular issue. If you can provide the necessary information to
reproduce this then we can move this forward.

Also a SIP debug of the call setup is likely useful as well. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-07-29 14:28 lmadsen        Note Added: 0125297                          
2010-07-29 14:28 lmadsen        Status                   acknowledged =>
feedback
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