[asterisk-users] PRI failover to SIP trunk

[Steve Totaro]

On Fri, Jul 10, 2009 at 11:40 AM, Dave Fullerton <
dfullertasterisk at shorelinecontainer.com> wrote:

> Steve Totaro wrote:
> > 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?
>
> It is my understanding (through trial and error, reading, etc) that any
> Dial command that does not result in an answered state will continue in
> the dialplan after a timeout (if specified) or some sort of progress is
> received. If the called channel results in an answer then dialplan
> processing stops as soon as one party hangs up (unless the g option is
> specified).
>
> This works on any channels that can pass progress (SIP/IAX and Zap/DAHDI
> PRI and BRI). Zap/DAHDI Analog channels are considered answered as soon
> as the dial is complete so you won't be able to use this trick under
> normal circumstances.
>
> -Dave
>
>
True I guess except that if the call fails as the OP posted, because the PRI
is down, it should work then right?

Another thing.  For outbound calls, I do not have a timeout.  So the user
hangs up when they are ready, or when the other side hangs up or gets
congestion, which amounts to the h exten, or am I not correct.

Why have a timeout on outbound dialing (unless you are a dialer app?)  It is
not like voicemail where you want it to ring for so many seconds and then
roll to VM.

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