Re: [asterisk-users] SIP call control via RTCP

2014-06-10 Thread Jan Gaida
Hello, I have found here http://www.voip-info.org/wiki/view/Asterisk+RTCP that there has been a patch for RTCP of Asterisk 1.4. Does this mean that starting with Asterisk version 1.6 RTCP call control is working correctly? Kind regards Jan Gaida On Mon, May 12, 2014 at 2:40 PM, Jan Gaida

[asterisk-users] SIP call control via RTCP

2014-05-12 Thread Jan Gaida
Hello, We are using Asterisk 1.4 as call distribution system with simple queues for SIP calls. With high load (4000 calls/hour) some calls remain in queue forever (until queue's max wait time) in spite of being hung up already by the caller. It seems that when a BYE is lost, Asterisk has no

Re: [asterisk-users] SIP call control via RTCP

2014-05-12 Thread Matt Behrens
On May 12, 2014, at 5:02 AM, Jan Gaida jan.ga...@grupoamper.com wrote: We are using Asterisk 1.4 as call distribution system with simple queues for SIP calls. With high load (4000 calls/hour) some calls remain in queue forever (until queue's max wait time) in spite of being hung up

Re: [asterisk-users] SIP call control via RTCP

2014-05-12 Thread Jan Gaida
Thank you. Yes, that should work. But if I understand it correctly, only if there's no silence detection activated. Otherwise, when silence is detected no RTP would be send, so that rtptimeout would hang up a still active call. I there no option to use RTCP? Not even in Asterisk 11? Regards On