[Asterisk-Users] Help Debugging Dropped Call Audio - Add'l Info

Matt Roth mroth at imminc.com
Thu Dec 22 14:04:52 MST 2005


Martin,

Please follow the "Steps to Reproduce" in my bug report and post your 
results back to the list.  If you're comfortable with adding the 
ast_log() statements to the ast_read() and ast_write() functions located 
in channel.c, it would really help to show that we are experiencing the 
same problem.  MONITOR_CONSTANT_DELAY is usually not defined, so be 
careful to add the ast_log() statements to the appropriate code block.  
Specifically, the code block between the "#ifndef 
MONITOR_CONSTANT_DELAY" and "#else" preprocessor directives in 
ast_read() and ast_write().

If you feel uncomfortable mixing down to a GSM format WAV file, use the 
following options to soxmix instead:

#soxmix -v 1.0 -t ul LEG-IN.PCM -t ul LEG-OUT.PCM MIXED.WAV

This will create a standard WAV file from the PCM legs.  I believe this 
does very little other than mixing the legs together and adding a header 
to the mixed file.  The mixed file should be a few dozen bytes (the size 
of the header) larger than one of the leg files.  There should be no 
compression and no compression artifacts, but you'll have a file that 
you can listen to.

Alternately, you can mix the individual legs to standard WAV files with sox:

#sox -v 1.0 -t ul LEG-IN.PCM LEG-IN.WAV
#sox -v 1.0 -t ul LEG-OUT.PCM LEG-OUT.WAV

Thanks alot for your response.  Determining if this is a problem that is 
isolated to certain hardware/software configurations or if it is a 
general problem with Asterisk is a huge step towards resolving it.

Sincerely,

Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer

Martin Joseph wrote:

> List users,
> Huh, I have noticed this type of popping on an SIP to SIP connection 
> using ulaw also,  but I figured it was just me.  I am running * 1.21.
>
> I am kind of a newbie to asterisk, so how should I go about 
> documenting this in your opinion.
>
> I don't think mixing down with sox is acceptable as that introduces 
> another potential source of noise?
>
> I'd love to help fix this, or pin it down, but would need some more 
> direction to do so...
>
> Marty
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



More information about the asterisk-users mailing list