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/[EMAIL PROTECTED]/${ARG1}|90,T)
exten => s,n,GotoIf($[${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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