we are experiencing 30 second delay before voice is heard after answer
when we ran wireshark it showed the problem
between frames 634 (where the softphone answers)
and 1366. Between those frames, asterisk receives RTP packets from
both the softphone and the sip carrier, but doesn't forward them to each
other....
Then we see a bunch of packets sent at once from the asterisk server,
until frame 1491, where things get normal, and asterisk starts
forwarding packets as it should.
Does anyone have any ideas ? We are running the latest version of asterisk
Thx in advance.
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