[asterisk-bugs] [Asterisk 0014850]: One Way audio on incoming calls from SIP provider trunk

Tue Apr 14 15:38:26 CDT 2009

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=14850 
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Reported By:                BlargMaN
Assigned To:                file
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Project:                    Asterisk
Issue ID:                   14850
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   block
Priority:                   normal
Status:                     assigned
Target Version:             1.6.1.0
Asterisk Version:           1.6.1.0-rc4 
Regression:                 Yes 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2009-04-07 17:22 CDT
Last Modified:              2009-04-14 15:38 CDT
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Summary:                    One Way audio on incoming calls from SIP provider
trunk
Description: 
in 1.6.1-rc1 i have perfectly working sip trunking...  in rc4 i have
outgoing working just fine, but incoming calls on the sip trunk are one
way...  i.e. They can hear me, but I can't hear them...
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 (0103245) file (administrator) - 2009-04-14 15:38
 http://bugs.digium.com/view.php?id=14850#c103245 
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Per discussion on IRC this was narrowed down to rc3, and it seems to be
that the gateway does not accept the reinvite. 

Issue History 
Date Modified    Username       Field                    Change               
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2009-04-14 15:38 file           Note Added: 0103245                          
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