asterisk users - Jun 2005 - Call Routing based on number dialed (using S IP)

Is this even possible or am I better off getting a voip number for each of
the existing numbers I want to forward.

Thanks!
> -----Original Message-----
> From: Geoff Manning [mailto:gmanning@zoom.com]
> Sent: Friday, June 03, 2005 4:53 PM
> To: Asterisk Users (E-mail)
> Subject: [Asterisk-Users] Call Routing based on number dialed (using
> SIP)
> 
> 
> Is it possible to route calls based on the number called when 
> the inbound
> call is SIP based?
> 
> Here is what we are trying to do:
> 
> 1) Someone dials one of the companies 5 long standing, published phone
> numbers which have been forwarded to ONE Voip telephone 
> number by the telco.
> 
> 2) The SER server where that Voip number terminates is 
> passing it to our
> Asterisk server
> 
> 3) Is there a way to determine what the original number dialed was?
> 
> We want to avoid needing a Voip number for every forwarded number.
> 
> 
> Thanks in advance.
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Geoff Manning wrote:
>Is this even possible or am I better off getting a voip number for each of
>the existing numbers I want to forward.
>
>Thanks!
>
>  
>
>>-----Original Message-----
>>From: Geoff Manning [mailto:gmanning@zoom.com]
>>Sent: Friday, June 03, 2005 4:53 PM
>>To: Asterisk Users (E-mail)
>>Subject: [Asterisk-Users] Call Routing based on number dialed (using
>>SIP)
>>
>>
>>Is it possible to route calls based on the number called when 
>>the inbound
>>call is SIP based?
>>
>>Here is what we are trying to do:
>>
>>1) Someone dials one of the companies 5 long standing, published phone
>>numbers which have been forwarded to ONE Voip telephone 
>>number by the telco.
>>
>>2) The SER server where that Voip number terminates is 
>>passing it to our
>>Asterisk server
>>
>>3) Is there a way to determine what the original number dialed was?
>>
>>We want to avoid needing a Voip number for every forwarded number.
>>
>>
>>Thanks in advance.
>>_______________________________________________
>>Asterisk-Users mailing list
>>Asterisk-Users@lists.digium.com
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>>    
>>
>_______________________________________________
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>Asterisk-Users@lists.digium.com
>http://lists.digium.com/mailman/listinfo/asterisk-users
>To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>  
>
You can use the sipgetheader() application. If you pass a call to 
asterisk, the field "To" in the SIP header stay as originally dialed. 
So, with

sipgetheader(or_To=To)
Cut(or_To,or_To,:,2)
Cut(or_To,or_To,@,1)

in your dialplan, you can get the original dialed number.

with the cut function you can cut the "sip:" and the
"@domain.asd"
substrings.

 Mirko

> 
> sipgetheader(or_To=To)
> Cut(or_To,or_To,:,2)
> Cut(or_To,or_To,@,1)

That works! Thanks!

Correction to the cut command below, replaced , with = :
> sipgetheader(or_To=To)
> Cut(or_To=or_To,:,2)
> Cut(or_To=or_To,@,1)