[asterisk-users] PRI failover to SIP trunk

[Dave Fullerton]

Tzafrir Cohen wrote:
> On Thu, Jul 09, 2009 at 05:31:18PM -0400, Steve Totaro wrote:
> 
> You have a small typo:
> 
>> exten => _.,1,Dial(Zap,g1,${EXTEN})
>> exten => _.,2,Dial(SIP,Provider,${EXTEN})
> 
>   exten => _.,1,Dial(Zap/g1/${EXTEN})
>   exten => _.,2,Dial(SIP/Provider/${EXTEN})
> 
> ('/' instead of ',')
> 

While this will work, be aware that there are circumstances where you 
may end up calling the number twice, once through each provider. One 
example is if the number you dial is busy, that progress will be passed 
via the PRI to asterisk and the dialplan will continue to the next 
priority. In this case, dialing the number again through the SIP 
provider. To avoid this you will need to use some dialplan logic and 
check the result of the DIALSTATUS variable. See this page for examples:

http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS

-Dave