[asterisk-bugs] [Asterisk 0017561]: MOH stops working after a few seconds and P2P bridging restarts despite still on hold!

Asterisk Bug Tracker noreply at bugs.digium.com
Mon Jun 28 11:48:44 CDT 2010


A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=17561 
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Reported By:                alexr1
Assigned To:                
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Project:                    Asterisk
Issue ID:                   17561
Category:                   Core/RTP
Reproducibility:            random
Severity:                   minor
Priority:                   normal
Status:                     new
Asterisk Version:           1.4.33.1 
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:             2010-06-28 11:11 CDT
Last Modified:              2010-06-28 11:48 CDT
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Summary:                    MOH stops working after a few seconds and P2P
bridging restarts despite still on hold!
Description: 
Asterisk 1.4.33.1

Linksys SPA962 (02102) dials mobile phone 0411111111. Mobile phone
answers, SPA962 puts mobile on hold.

Music may or may not be heard. If music is NOT heard, Asterisk says

Started music on hold, class 'default', on SIP/XXXXtel-0000003b
Packet2Packet bridging SIP/02101-0000003a and SIP/XXXXtel-0000003b

If music IS heard, then it will show:
Started music on hold, class 'default', on SIP/XXXXtel-0000003b
<music is heard for about 5 seconds>
Packet2Packet bridging SIP/02101-0000003a and SIP/XXXXtel-0000003b

As soon as Packet2Packet bridging is seen (and mind you, the call is STILL
on hold), I start seeing errors like this:
RTP Transmission error of packet to 0.0.0.0:0: Invalid argument

It seems like there is something wrong with rtp.c?. In call queues, the
music on hold plays perfectly fine.

In Additional Information I have provided a CLI output (minus the RTP
transmission errors, as they don't show up in the CLI, only in the log file
which was since cleared).

I have another Asterisk box running 1.4.33.1 (not identical configuration
though), and I just can't seem to reproduce the problem on that box, only
on the first one.

Also in the additional information is where I have forced the phone's
codec to something different from the VSP XXXtel so that no packet2packet
bridging can take place. This results in normal, flawless operation of hold
music. From this, I'm sure this has something to do with packet2packet
bridging + moh.
====================================================================== 

---------------------------------------------------------------------- 
 (0123961) alexr1 (reporter) - 2010-06-28 11:48
 https://issues.asterisk.org/view.php?id=17561#c123961 
---------------------------------------------------------------------- 
Unfortunately it is a production server that has over 1000 SIP endpoints
registering and being used continuously. The amount of extra surplus
information in the debug log would drown out the actual problem, making it
difficult to isolate which lines relate to the call in question, and impede
on the privacy of users of the system (without a lot of editing).

Hopefully someone else will notice this problem on a less-used system that
can provide the appropriate "isolated" debugging logs. 

This problem did not exist in 1.4.25 which was used with identical
configuration before being upgraded to 1.4.33.1 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-06-28 11:48 alexr1         Note Added: 0123961                          
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