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20110520
2006 Feb 14
2
audio cuts out
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