asterisk users - Mar 2008 - How to capture destination number when receive call through ZAP

Hi all,

I am using Digium PCI board to receive PSTN call through regular phone
line.  It is no problem for me to receive calls, but I am not able to
capture the destination number through the ZAP channel


exten => s, n, Verbose(1|destination to ${EXTEN}  )


${EXTEN} returns 's' instead of the actual destination number.  Since I
have
multiple phone numbers, I want to be able to route different calls to
different places.

Is this possible to do with Asterisk?

Thanks,
Mark
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Without knowing the line type, card model, etc, I doubt anyone can help 
you.  FXO signaled ports do not support receiving the dialed number.

mark morreny wrote:
> Hi all,
> 
> I am using Digium PCI board to receive PSTN call through regular phone
> line.  It is no problem for me to receive calls, but I am not able to
> capture the destination number through the ZAP channel
> 
> 
> exten => s, n, Verbose(1|destination to ${EXTEN}  )
> 
> 
> ${EXTEN} returns 's' instead of the actual destination number. 
Since I have
> multiple phone numbers, I want to be able to route different calls to
> different places.
> 
> Is this possible to do with Asterisk?
> 
> Thanks,
> Mark
> 
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
> 
-- 
Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, 
T-1, PRI, Frame Relay, Linux, and network design.  Based near 
Birmingham, AL.  Now accepting clients worldwide.

The only method I'm familiar with for an analogue line to signal which 
number was called is a very old service that loops the line first and 
then dials the number.  The only way to capture this would be to handle 
the incoming line as a standard extension with a different context.  
I've only run in to one of these services myself - and that was attached 
to a legacy PABX.  I seriously doubt you'd be able to order these 
services any more.

If this is a standard PSTN service, the only way you know which number 
has been called is by matching it against the zap channel that the call 
has been received on.  The ${EXTEN} variable won't tell you this as 
you've already found out - you'll need to examine ${CHANNEL} and match 
the channel to the connected DID yourself.



mark morreny wrote:
> Hi all,
>
> I am using Digium PCI board to receive PSTN call through regular phone 
> line.  It is no problem for me to receive calls, but I am not able to 
> capture the destination number through the ZAP channel
>
>
> exten => s, n, Verbose(1|destination to ${EXTEN}  )
>
>
> ${EXTEN} returns 's' instead of the actual destination number. 
Since
> I have multiple phone numbers, I want to be able to route different 
> calls to different places. 
>
> Is this possible to do with Asterisk?
>
> Thanks,
> Mark
> ------------------------------------------------------------------------
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
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Distinctive Ringing might be available from your telecom provider.

mark morreny wrote:
> Hi all,
>
> I am using Digium PCI board to receive PSTN call through regular phone 
> line.  It is no problem for me to receive calls, but I am not able to 
> capture the destination number through the ZAP channel
>
>
> exten => s, n, Verbose(1|destination to ${EXTEN}  )
>
>
> ${EXTEN} returns 's' instead of the actual destination number. 
Since
> I have multiple phone numbers, I want to be able to route different 
> calls to different places. 
>
> Is this possible to do with Asterisk?
>
> Thanks,
> Mark
> ------------------------------------------------------------------------
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users