asterisk users - Jan 2006 - instant fallback to zap in case of sip/iax/xyz-failure

i would like to carry some oversea pstn-destinations via sip to providers
like stanaphone, however, in case of a network-failure or if the provider
is not available, i want to fallback to the zap-channels so the call is
carried out to the pstn directly.
the usual approach would be to check the dialstatus(e.g.NOANSWER).
however, asterisk tries >60seconds to reach that peer(even when the ip
i'm
sending the call too is a dead end(no host)). i could limit a call by
setting a timeout but this limit would also apply if a final destination
doesn't pick up within the timeout.
so basically, when i send a call via a sip-channel, i would like to know
the network-status of the foreign host immediately(at least within 5
seconds) so i can reroute the call without having to wait for a host that
is probably dead...

this seems to be possible with iax and CHANUNAVAIL,
(http://bugs.digium.com/bug_view_advanced_page.php?bug_id=3360&history=1),
though i haven't tried it.
also i _need_ to use sip, iax (currently) is not an option.

is there any mechanism in asterisk that allows to get the vital sip-status
of a foreign host?! thanks for your input!!! ;-)

regards
christian

My experience is that when an iax or sip channel is unavailable for some 
reason it fails right away despite whatever timeout I have set for the 
call.  In these cases the caller doesn't even realize that the call has 
failed over to the next carrier.

   exten => s,n(dial1),Dial(${VOIPJET}/${ARG1}|90,T)
   exten => s,n,GotoIf($[${DIALSTATUS} = CHANUNAVAIL]?dial2)
   exten => s,n,Macro(rhangup)
   exten => s,dial1+101,GotoIf($[${DIALSTATUS} = BUSY]?s-BUSY|1)
   exten => s,n(dial2),Dial(IAX2/user@teliax-out/${ARG1}|90,T)
   exten => s,n,GotoIf($[${DIALSTATUS} = CHANUNAVAIL]?dial3)

MARK.

Christian Benke wrote:
> i would like to carry some oversea pstn-destinations via sip to providers
> like stanaphone, however, in case of a network-failure or if the provider
> is not available, i want to fallback to the zap-channels so the call is
> carried out to the pstn directly.
> the usual approach would be to check the dialstatus(e.g.NOANSWER).
> however, asterisk tries >60seconds to reach that peer(even when the ip
i'm
> sending the call too is a dead end(no host)). i could limit a call by
> setting a timeout but this limit would also apply if a final destination
> doesn't pick up within the timeout.
> so basically, when i send a call via a sip-channel, i would like to know
> the network-status of the foreign host immediately(at least within 5
> seconds) so i can reroute the call without having to wait for a host that
> is probably dead...
>
> this seems to be possible with iax and CHANUNAVAIL,
>
(http://bugs.digium.com/bug_view_advanced_page.php?bug_id=3360&history=1),
> though i haven't tried it.
> also i _need_ to use sip, iax (currently) is not an option.
>
> is there any mechanism in asterisk that allows to get the vital sip-status
> of a foreign host?! thanks for your input!!! ;-)
>
> regards
> christian
>
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