[asterisk-bugs] [Asterisk 0018779]: Streaming audio file through Local channel to a few SIP devices randomly loses audio

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
====================================================================== 
https://issues.asterisk.org/view.php?id=18779 
====================================================================== 
Reported By:                bhvictor
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   18779
Category:                   Channels/General
Reproducibility:            random
Severity:                   major
Priority:                   normal
Status:                     new
Asterisk Version:           Older 1.4 - please test a newer version 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
====================================================================== 
Date Submitted:             2011-02-09 15:45 CST
Last Modified:              2011-02-10 12:21 CST
====================================================================== 
Summary:                    Streaming audio file through Local channel to a few
SIP devices randomly loses audio
Description: 
We have a paging feature which allows one to record a message (with
Record), and then have that message sent twice in a row as a Page to a five
SIP devices, using a Local channel setup in a callfile.

About 99% of the time it works perfectly. The other 1% or so (about once
per day), the audio cuts out on the page at a random place. Unfortunately,
this situation occurs randomly each day. All users associated with the five
devices report the same cutoff point when there's a failure.

The logs seems to indicate everything's fine, except we always see:
    WARNING[3292] file.c: Unexpected control subclass '-1'
This shows up for all pre-recorded pages, even the successful ones.

Thanks for any help.

====================================================================== 

---------------------------------------------------------------------- 
 (0131797) lmadsen (administrator) - 2011-02-10 12:21
 https://issues.asterisk.org/view.php?id=18779#c131797 
---------------------------------------------------------------------- 
Can you provide the dialplan you're using, along with the exact version of
Asterisk you're using?

Can you provide a recording of the successful and unsuccessful call
recordings so we can hear what it is doing? A pcap capture with the RTP may
be of some use as well (including the SIP messages). 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2011-02-10 12:21 lmadsen        Note Added: 0131797                          
======================================================================