[Asterisk-Users] audio cuts out

[trixter aka Bret McDanel]

has anyone experienced a problem where RTP audio cuts out when doing
30-40 concurrent channels via sip?

The box is freebsd 6, asterisk 1.2.4, voip only (no libpri, no zaptel -
not even a timing source)

The box has plenty of bandwidth, when a call to the same box is iax2 it
works, but when its sip a call gets connected a few frames of audio are
passed and then silence.

When the box is completly idle sip does not experience this problem, it
is only when there are a few concurrent calls.  


-- 
Trixter http://www.0xdecafbad.com     Bret McDanel
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378
http://www.sacaug.org/ Sacramento Asterisk Users Group
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