[asterisk-bugs] [Asterisk 0016868]: One way audio after placing call on hold and resuming

Asterisk Bug Tracker noreply at bugs.digium.com
Thu Nov 4 14:03:04 CDT 2010


The following issue requires your FEEDBACK. 
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https://issues.asterisk.org/view.php?id=16868 
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Reported By:                jordankirby
Assigned To:                
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Project:                    Asterisk
Issue ID:                   16868
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.6.2.2 
JIRA:                       SWP-938 
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:             2010-02-19 09:21 CST
Last Modified:              2010-11-04 14:03 CDT
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Summary:                    One way audio after placing call on hold and
resuming
Description: 
This fault occurs on 1.6.1.11, 1.6.2.0, 1.6.2.2, 1.6.2.3-RC2, SVN-247894.

sip.conf: directrtpsetup=yes, nat=yes
Both extensions: canreinvite=yes, nat=yes
Phones are on the same LAN and behind NAT.

Server is in a separate location also behind NAT. All standard internal
and external calls work fine.

Problem:

Extension A calls extension B. Extension A puts the call on hold,
extension B gets played music as expected. When extension A resumes the
call extension B can't hear extension A. 

This seems to be because Asterisk sends the external IP address of
extension B to extension A in the SDP when the call is resumed.
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Relationships       ID      Summary
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related to          0013545 Channel re-invited on destination ringi...
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---------------------------------------------------------------------- 
 (0128630) lmadsen (administrator) - 2010-11-04 14:03
 https://issues.asterisk.org/view.php?id=16868#c128630 
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Do you have externip set? It sounds like you might.

Do you have localnet set? If not, that is likely the problem.

Sounds like it might be a configuration issue in sip.conf. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-11-04 14:03 lmadsen        Note Added: 0128630                          
2010-11-04 14:03 lmadsen        Status                   acknowledged =>
feedback
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