asterisk users - Mar 2011 - how to use qualify times to route calls

I'm using 1.8.3, and have 2 sip providers. Both are set with 
qualify=yes. Each of them sometimes have qualify times 10+ times the 
other. For instance, one will be at 10-15ms, the other at 200ms.

Is there a way I can route an outgoing call to the provider with the 
lower qualify time?

sean

On 3/03/11 11:29 AM, sean darcy wrote:
> I'm using 1.8.3, and have 2 sip providers. Both are set with
> qualify=yes. Each of them sometimes have qualify times 10+ times the
> other. For instance, one will be at 10-15ms, the other at 200ms.
>
> Is there a way I can route an outgoing call to the provider with the
> lower qualify time?
Traditionally you'd use a value you consider to be good enough for calls 
and set qualify to that.  I.E. if you think 30ms is ok then set 
qualify=30 and then just route via the first then the second depending 
on status.

-- 
Cheers,

Matt Riddell
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-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of sean darcy
Sent: Wednesday, March 02, 2011 4:29 PM
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] how to use qualify times to route calls

I'm using 1.8.3, and have 2 sip providers. Both are set with 
qualify=yes. Each of them sometimes have qualify times 10+ times the 
other. For instance, one will be at 10-15ms, the other at 200ms.

Is there a way I can route an outgoing call to the provider with the 
lower qualify time?

sean

You could do a context using an AGI to do a "sip show peers" and
select the
provider from that.  Something like this

[pick_prov]
exten => s,1,AGI(getprov.agi)
exten => s,n,Dial(SIP/${EXTEN}@${BESTPROV},30,m)

getprov.agi does "sip show peers" and gets the qualify time from
status.
The low value is returned in the variable BESTPROV.

Should be about 50 lines of PERL or PHP.