[asterisk-dev] Strange Asterisk Behaviour - Stuck channels

[John Lange]

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
www.johnlange.ca