[asterisk-users] PRI failover to SIP trunk

[Steve Totaro]

On Fri, Jul 10, 2009 at 10:45 AM, Dave Fullerton <
dfullertasterisk at shorelinecontainer.com> wrote:

> 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
>
>
Good point.

I was unaware that "busy" back from a TDM circuit would progress in the
dialplan rather than going to the h exten.

What other cases are there like that?

-- 
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)
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