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

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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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:                     acknowledged
Asterisk Version:           1.6.2.9 
JIRA:                        
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:              2010-07-20 10:09 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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 (0124740) lmadsen (administrator) - 2010-07-20 10:09
 https://issues.asterisk.org/view.php?id=17666#c124740 
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I think it'd also be beneficial to have the configuration for Asterisk A
and B so this can be reproduced in a lab. 

Issue History 
Date Modified    Username       Field                    Change               
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2010-07-20 10:09 lmadsen        Note Added: 0124740                          
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