On Dec 18, 2008, at 7:22 AM, Julien Chavanton wrote: > I have a concern with Dial command, I want to enable a secondary > route with a remote partner, if the first route fails then we use > the second one : > > > Solution1: it will try both (there will be 2 simultanious actives > calls ringing) this is not clean when calling an endusers > > exten => _X.,1,Dial(SIP/${ext...@remote-sip1,5) > exten => _X.,1,Dial(SIP/${ext...@remote-sip2,5) > > > > Solution2: it will wait until 5 seconds of timeout (on answer) and > then try the second alternative "n" > > exten => _X.,1,Dial(SIP/${ext...@remote-sip1,5) > exten => _X.,n,Dial(SIP/${ext...@remote-sip2,5) > > the problem is we can not select what timeout represents, timeout on > ACK from INVITE would be perfect I think (1 second for example), > timeout for answer ? this is to hard to predict, some mobile phone > can ring for 30 seconds, etc.
You should look at the configurable T1 timers in sip.conf, which allow you to specify the retransmit intervals. I think this will do what you want, but it is a very dangerous setting that can lead to significant unintended consequences. First, do some reading on what the T1 timer does - Google can help there. ;--------------------------- SIP timers ---------------------------------------------------- ; These timers are used primarily in INVITE transactions. ; The default for Timer T1 is 500 ms or the measured run-trip time between ; Asterisk and the device if you have qualify=yes for the device. ; ;t1min=100 ; Minimum roundtrip time for messages to monitored hosts ; Defaults to 100 ms ;timert1=500 ; Default T1 timer ; Defaults to 500 ms or the measured round-trip ; time to a peer (qualify=yes). ;timerb=32000 ; Call setup timer. If a provisional response is not received ; in this amount of time, the call will autocongest ; Defaults to 64*timert1 JT --- John Todd email:jt...@digium.com Digium, Inc. | Asterisk Open Source Community Director 445 Jan Davis Drive NW - Huntsville AL 35806 - USA direct: +1-256-428-6083 http://www.digium.com/ _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users