asterisk users - Nov 2007 - Call Forward on SIP unreachable (network failure)

Hi,

I am trying to implement call forwarding on the event
that my ATA was not 
reachable to Asterisk, whether due to registration
timeout, network 
interruptions between the ATA and Asterisk, or simply
because the network on 
which the ATA resides in unreachable for any reason.

I there a way of implementing such a feature in
Asterisk?

I have implemented CF unconditional, and CF on busy,
CF on unavailable (ring 
but no answer) not nothing about CF on (SIP)
unreachable.

Thank you and best regards,

Antoine Megalla. 




     
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${DIALSTATUS} will be one of:


   - *CHANUNAVAIL* : Channel unavailable (for example in sip.conf, when
   using qualify=, the SIP chan is unavailable)
   - *BUSY* : Returned busy
   - *NOANSWER* : No Answer (i.e SIP 480 or 604 response)
   - *ANSWER* : Call was answered
   - *CANCEL* : Call attempt cancelled (i.e user hung up before the call
   connected)
   - *DONTCALL* : Privacy manager don't call
   - *TORTURE* : Privacy manager torture menu
   - *CONGESTION* : Means Congestion, or anything else (some other error
   setting up the call)

Did you test CHANUNAVAIL or CONGESTION ?
Debug DIALSTATUS var for this case using Noop application in dialplan.


On Nov 13, 2007 8:51 PM, Antoine Megalla <aatef at rocketmail.com> wrote:
> Hi,
>
> I am trying to implement call forwarding on the event
> that my ATA was not
> reachable to Asterisk, whether due to registration
> timeout, network
> interruptions between the ATA and Asterisk, or simply
> because the network on
> which the ATA resides in unreachable for any reason.
>
> I there a way of implementing such a feature in
> Asterisk?
>
> I have implemented CF unconditional, and CF on busy,
> CF on unavailable (ring
> but no answer) not nothing about CF on (SIP)
> unreachable.
>
> Thank you and best regards,
>
> Antoine Megalla.
>
>
>
>
>
> 
____________________________________________________________________________________
> Never miss a thing.  Make Yahoo your home page.
> http://www.yahoo.com/r/hs
>
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14 nov 2007 kl. 01.14 skrev Marco Mouta:
> ${DIALSTATUS} will be one of:
You can also use the SIPPEER() dialplan function to check the status  
of the peer,
provided you have a qualify= setting for that peer to enable health  
and latency
checking. If the peer is unavailable, you can do your call forward.

/O