> -----Ursprüngliche Nachricht----- > Von: Klaus Darilion [mailto:[EMAIL PROTECTED] > Gesendet: Dienstag, 19. September 2006 16:03 > An: asterisk-users@lists.digium.com > Betreff: [asterisk-users] fast SIP failover (outgoing sIP requests) with 1.2 > > Hi! > > I have the following problem: I route calls from one office to the other > office via SIP, but if for any reason this SIP call fails, I want a > backup route via the PSTN. > > Thus, I use: > > > exten => _[1-9].,4,Dial(SIP/${enumresult},90) > exten => _[1-9].,5,GotoIf($["${DIALSTATUS}" = "CHANUNAVAIL"]?103:6) > exten => _[1-9].,6,GotoIf($["${DIALSTATUS}" = "CONGESTION"]?103:7) > exten => _[1-9].,7,Hangup > exten => _[1-9].,103,Dial(ZAP/g1/${EXTEN},90) > > The problem is, if the SIP server at the remote office is down, thus no > responses to the INVITE, it takes 64 seconds to timeout. > > Is there a method to shorten this interval - e.g. if there is no > response within 10 seconds give up - without changing the hardcoded > retransmission value (6) in chan_sip ? > > regards > klaus
Hi, maybe I'm wrong, but what about using the ChanisAvail function? We did something like this in a customer installation: exten => _XXX.,1,Set(LANGUAGE()=de) exten => _XXX.,2,ChanisAvail(CAPI/ISDN3/${EXTEN},s) exten => _XXX.,3,Dial(CAPI/ISDN3/${EXTEN}/b,60,tT) exten => _XXX.,4,Congestion exten => _XXX.,103,ChanisAvail(CAPI/ISDN2/${EXTEN},s) exten => _XXX.,104,Dial(CAPI/ISDN2/${EXTEN}/b,60,tT) exten => _XXX.,105,Congestion Hope, it helps ... Guido gwsNetTech Guido Hecken Quirrenbacher Str. 36 53639 Königswinter Germany fon +49(2244) 870663 fax +49(2244) 870664 mobil +49(179) 1267353 web http://www.gwsnettech.de mailto:[EMAIL PROTECTED] _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users