As I read it you can do it like this: >From http://www.voip-info.org/wiki/view/Asterisk+sip+rtptimeout Exten => s,1,noop (set rtptimeout so we can have 2 timeouts on a dial) Exten => s,n,Set(rtptimeout=60) Exten => s,n,Dial(SIP/peer1,60) Exten => s,n,Dial(SIP/peer2,60)
Haven't tested this. -----Original Message----- From: [email protected] [mailto:[email protected]] On Behalf Of Klaus Darilion Sent: Friday, January 18, 2013 9:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] rtptimeout: how to detect it in dialplan? Hi! I want to forward a call to another destination if the outgoing call leg has an rtptimeout. But as far as I see there is no way to find out if the hangup was due to a rtp timeout or any other reason. I thought that HANGUPCAUSE or DIALSTATUS would be set, but they aren't. Are there any means to detect an rtp timeout in extensions.conf? Thanks Klaus -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
