Just a thought I had on this.. Why not setup a sip peer entry in
sip.conf with a qualify statement in it and send the call to the peer
entry? That way asterisk will know if the peer is alive or not and I
would think it would skip that particular peer accordingly.. that or
it's really late and I've misunderstood what qualify is for.. but it
seems to want to work how I imagine it would.. and from a small test I
just tried it appears to do just that.. but who knows..
Adi Linden wrote:
I have a similar problem. I asked the same question in a message to the
list a few days ago titles "IAX outgoing redundancy". It would seem
app_dial would need to have some code added to it to have two different
kind of timeouts, one an answer timeout (which is the current timeout in
the Dial() command) and the other a "ringing" timeout (which would be a
timeout to confirm the call has been passed on to the PSTN
successfully).
I don't know if "ringing" timeout is how I would describe it. More like a
network timeout. I can describe another scenario, based on what I
originally posted:
exten => _9737XXXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten => _9737XXXX,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
If I physically pull the netowrk plug on 172.17.99.5 Asterisk will just
sit there for about a minute before continuing to 172.17.99.6. And the
same scenario holds true for IAX instead of SIP.
Adi
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