[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. 
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https://issues.asterisk.org/view.php?id=18779 
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Reported By:                bhvictor
Assigned To:                
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Project:                    Asterisk
Issue ID:                   18779
Category:                   Channels/General
Reproducibility:            random
Severity:                   minor
Priority:                   normal
Status:                     feedback
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:              
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Date Submitted:             2011-02-09 15:45 CST
Last Modified:              2011-02-14 11:40 CST
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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.

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---------------------------------------------------------------------- 
 (0131936) bhvictor (reporter) - 2011-02-14 11:40
 https://issues.asterisk.org/view.php?id=18779#c131936 
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We're running 1.4.30. We have strict customer requirements to meet, so each
upgrade is pretty major for us, and this version has so far proved stable.

We're dependent on customers reporting the dropping issue, but I'll update
when they do.

Here is the relevant dialplan snippet, I have tried to simplify it a bit
for readability:

[macro-page-with-recording]
; ARG1 = originating caller ID
; ARG2 = zone to page
exten => s,1,Set(callerid=${ARG1})
exten => s,n,Set(zone=${ARG2})
exten => s,n,Set(num-times=2)

exten =>
do-page,1,Set(page-recording=tmp/${zone}-page-message-${RAND(100000,999999)}.wav)
exten => do-page,n,Noop(${callerid} is paging a recorded message
${num-times} time(s) to zone ${ARG2}...)
exten => do-page,n,Macro(system-page-fetch-regs,${callerid},${ARG2})    ;
${regs} set in macro
exten => do-page,n,Record(${page-recording})
exten => do-page,n,Hangup                                               ;
they pressed #, so goto h

exten =>
h,1,DeadAGI(pagefile,${callerid},${page-recording},${num-times},${zone},${regs})

[system-do-page-with-recording]                             ; called from
AGI script
exten => s,1,SIPAddHeader(Alert-Info: Ring Answer)          ; force phones
offhook
exten => s,n,Page(${regs})                                  ; regs set in
AGI script 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2011-02-14 11:40 bhvictor       Note Added: 0131936                          
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