[asterisk-dev] Strange Asterisk Behaviour - Stuck channels
john at johnlange.ca
Wed Jul 30 15:19:07 CDT 2008
We do not use that setting but note that rtptimeout terminates a call if
there is no RTP activity for a given number of seconds.
What we have is lots of RTP activity but no SIP activity so I would
assume rtptimeout wouldn't have any effect.
Also, keep in mind that when a second call is placed to the device,
Asterisk sends two RTP streams on the same port which results in the
garbled audio I mentioned.
This perhaps implies something more significant is wrong.
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