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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