[asterisk-bugs] [Asterisk 0014850]: One Way audio on incoming calls from SIP provider trunk
Asterisk Bug Tracker
noreply at bugs.digium.com
Thu Apr 16 15:35:48 CDT 2009
The following issue has been UPDATED.
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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: closed
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:
Resolution: no change required
Fixed in Version:
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Date Submitted: 2009-04-07 17:22 CDT
Last Modified: 2009-04-16 15:35 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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(0103332) file (administrator) - 2009-04-16 15:35
http://bugs.digium.com/view.php?id=14850#c103332
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After a ton of discussion with the great reporter this was isolated to
revision 182527 and seems to have actually exposed an issue with the
version of CCM in use. It seems to require audio to flow before a reinvite
can be issued. Due to the changes present in 182527 this no longer happens
causing the bug to show itself. The reporter has been given a one line
change for their environment.
Issue History
Date Modified Username Field Change
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2009-04-16 15:35 file Note Added: 0103332
2009-04-16 15:35 file Status feedback => resolved
2009-04-16 15:35 file Resolution open => no change
required
2009-04-16 15:35 file Status resolved => closed
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