[Asterisk-Users] Multiple gateways for same dial pattern

[Matt Hess]

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/${EXTEN:1}@172.17.99.5)
>    exten => _9737XXXX,2,Dial(SIP/${EXTEN:1}@172.17.99.6)
>
>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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