[asterisk-users] SIP failover between Sip Providers

[Knud Müller]

Dinesh Nair wrote:

>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.
>
>  
>
The disadvantage of that solution is, that I'll always try to make a 
connection with a provider for that I know by experience it wouldn't 
work. In the failover case the time between starting to dial and the 
first ring gets longer. If I know that Provider A fails 60% of Calls 
then I don't need to start with a but can start with b directly.

-- 
Knud A. Müller