[asterisk-users] fast SIP failover (outgoing sIP requests) wi
th 1.2
Guido Hecken
guido.hecken at gwsnettech.de
Tue Sep 19 07:28:23 MST 2006
> -----Ursprüngliche Nachricht-----
> Von: Klaus Darilion [mailto:klaus.mailinglists at pernau.at]
> Gesendet: Dienstag, 19. September 2006 16:03
> An: asterisk-users at 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
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web http://www.gwsnettech.de
mailto:guido.hecken at gwsnettech.de
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