[asterisk-dev] Strange Asterisk Behaviour - Stuck channels

John Lange 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.

John Lange

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