[asterisk-users] fast SIP failover (outgoing sIP requests) with 1.2

[Klaus Darilion]

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