|Summary:||ASTERISK-19004: RTP timeout won't work with locally bridged SIP channels|
|Reporter:||Andrey Solovyev (corruptor)||Labels:|
|Date Opened:||2011-12-12 00:38:13.000-0600||Date Closed:||2012-01-24 10:23:52.000-0600|
|Description:||It's simple to reproduce.|
I have rtptimeout=30 in [general] and I have two sip peers 1001 and 1002
1002 calls 1001 with simple Dial application without any options and MixMonitor.
So we get Locally bridging SIP/1001-00000008 and SIP/1002-00000009
Disconnect 1001 or 1002 from the network. Asterisk won't see any RTP from the phone and won't hangup the call after 30 seconds.
If I do channel request hangup SIP/1001-00000008 I get this notice.
NOTICE chan_sip.c: Disconnecting call 'SIP/1001-00000008' for lack of RTP activity in 233 seconds
This seems to be unexpected behaviour.
|Comments:||By: Stefan Schmidt (schmidts) 2011-12-15 07:59:02.028-0600|
could you please try to reproduce this with asterisk 10. Rtp timeout is completly handled different there.
By: Andrey Solovyev (corruptor) 2011-12-20 03:40:19.790-0600
asterisk 10 successfully disconnects call in the same situation.