[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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to use qualify times to route calls
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 ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to use qualify times to route calls
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of sean darcy
Sent: Wednesday, March 02, 2011 4:29 PM
To: asterisk-users@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.
--
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Re: [asterisk-users] how to use qualify times to route calls
On 3/03/11 11:34 AM, Danny Nicholas wrote: getprov.agi does sip show peers and gets the qualify time from status. The low value is returned in the variable BESTPROV. If you're going to do that, you could probably knock something up with the SIPPEER function - SIPPEER(status). -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to use qualify times to route calls
On 03/02/2011 05:34 PM, Danny Nicholas wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of sean darcy
Sent: Wednesday, March 02, 2011 4:29 PM
To: asterisk-users@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.
That would be a great idea, but would stretch my limits.
I'll try qualify=30 and qualifyfreq=20 to start.
sean
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Re: [asterisk-users] how to use qualify times to route calls
On Wed, 2 Mar 2011, sean darcy wrote: That would be a great idea, but would stretch my limits. Isn't that what makes it fun? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
