Klaus Darilion
2006-Sep-19 07:03 UTC
[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
Damon Estep
2006-Sep-19 08:20 UTC
[asterisk-users] fast SIP failover (outgoing sIP requests) with 1.2
Try taking to 90 second timeout off
Change
exten => _[1-9].,4,Dial(SIP/${enumresult},90)
to
exten => _[1-9].,4,Dial(SIP/${enumresult})
a btter method is to set up each office as a unique peer with qualify = yes and
then add the peer name to the dial string, like dial(SIP/3035551212@peername)
if the peer is offline (qualify has failed) the unavaialbe status will come back
right away.
> -----Original Message-----
> From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-
> bounces@lists.digium.com] On Behalf Of Klaus Darilion
> Sent: Tuesday, September 19, 2006 8:45 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] fast SIP failover (outgoing sIP requests)
> with 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
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