Guido Hecken
2006-Sep-19 07:18 UTC
[asterisk-users] fast SIP failover (outgoing sIP requests) wi th 1.2
> -----Urspr?ngliche Nachricht----- > Von: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] > Gesendet: Dienstag, 19. September 2006 16:03 > An: asterisk-users@lists.digium.com > Betreff: [asterisk-users] fast SIP failover (outgoing sIP requests) with1.2> > 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 > klausHi, 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 K?nigswinter Germany fon +49(2244) 870663 fax +49(2244) 870664 mobil +49(179) 1267353 web http://www.gwsnettech.de mailto:guido.hecken@gwsnettech.de
Klaus Darilion
2006-Sep-19 07:44 UTC
[asterisk-users] fast SIP failover (outgoing sIP requests) wi th 1.2
Guido Hecken wrote:>> -----Urspr?ngliche Nachricht----- >> Von: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] >> 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 ... >Hi! I've tried it but apparently chanisavail does not work with "non-local" SIP peers. thanks klaus