[asterisk-users] Tracing audio problems

[Avi Miller]

Hey guys,

I need some assistance in tracking down the cause of audio problems that 
are occurring at two of my sites:

Both sites run Asterisk 1.2.10 and use Sangoma A101u PRI cards. Both 
sites are reporting that audio in calls is "dropping out" during words, 
so that the other caller (i.e. the remote user) can only hear bits of 
the words.

This used to only happen on Asterisk-to-Asterisk calls via IAX2 (using 
g729) so I assumed it was latency or bandwidth problems on the 
inter-office network. However, the network is hardly used and my 
round-trip times are sub 100ms according to iax2 show peers (with 
qualify=yes).

Then, thinking it might be g729 issues, I changed the entire system to 
only use alaw and the problem persists.

Does anyone have any suggestions on where to look next? My users are 
getting increasingly annoyed and I'm quickly running out of ideas.

Thanks,
Avi

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