[asterisk-users] SIP failover between Sip Providers

Dinesh Nair dinesh at alphaque.com
Wed Apr 18 00:21:00 MST 2007

On Wed, 18 Apr 2007 09:04:22 +0200, Knud Müller wrote:

> I 
> think it can be done by using the dialplan and the database to store the 
> statistical information but maybe there is an easier way that integrates 
> better with asterisk!?

i dont think you'd even need a database with statistics. just have all
calls sent to provider A with an automatic failover to provider B if the
call can't be completed through A. you'd need to go look at the DIALSTATUS
variable for that.

Regards,                           /\_/\   "All dogs go to heaven."
dinesh at alphaque.com                (0 0)   http://www.openmalaysiablog.com/
| for a in past present future; do                                        |
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