[Spce-user] inbound rewriting rule

[Blagoja]

On 2/28/2013 6:11 PM, Kevin Masse wrote:
> That should still be ok.  Sipwise will see that the alias_number or
> E.164 number assigned to the Subscriber will match and stay local and
> not forward off to an Asterisk box.
> Kevin
>
>
> -----Original Message-----
> From: Blagoja [mailto:blagojaanusev at anutel.ca]
> Sent: Thursday, February 28, 2013 6:08 PM
> To: Kevin Masse
> Cc: <spce-user at lists.sipwise.com>
> Subject: [Spce-user] inbound rewriting rule
>
> Thanks for reply
> There is no asterisk involved i'm trying to route the call to local user
> and the number in my case is 7 digits eg. 510 1199
>
>
> Sent from my iPhone
>
> On 2013-02-28, at 5:35 PM, "Kevin Masse" <kmasse at questblue.com> wrote:
>
>> Good evening:
>>
>> Try to think of it this way:
>>
>> 15551231234 ->Your PSTN Carrier--> inbound call SIPWISE Outbound
> Rewrite
>> Rules for Callee ^(1|)([1-9][0-9][0-9]+)$               \2
>> -->  SIP Subscriber Asterisk Box where the call will go:
>> This will take a call inbound for your carrier like this 15551231234
> and
>> send it to your asterisk box (Subscriber) as 5551231234  The pattern
>> that your Asterisk Box is looking for is 5551231234 then send the call
>> to a destination.  Notice it dropped the 1 for the destination.
>>
>>
>> Let me know if this helps any.
>>
>> Kevin
>>
>>
>> -----Original Message-----
>> From: spce-user-bounces at lists.sipwise.com
>> [mailto:spce-user-bounces at lists.sipwise.com] On Behalf Of Blagoja
>> Sent: Thursday, February 28, 2013 5:28 PM
>> To: spce-user at lists.sipwise.com
>> Subject: [Spce-user] inbound rewriting rule
>>
>> Hello,
>> Thanks for your help I start to understand but i'm stuck again
>>
>> My peer authenticate to remote server so when i call the number call
> is
>> entering my server and disconect. I did rewrite rulle but in
>> kamailio.proxy.log file saying No matching rewrite rules for 'ngcp-lb'
>> found
>>
>> This is the line where i can see the call entering the server.
>> New request - M=ACK R=sip:127.0.0.1:5080
>> F=sip:025101165 at X.X.X.X:5060;user=phone;cpc=ordinary
>> T=sip:5101199 at X.X.X.X:5060;user=phone IP=X.X.X.X:5060 (127.0.0.1:5060)
>> ID=418ebe9cl43yuf9oa at 172.16.17.10
>>   My rulle is : ^([1-9][0-9]+)$ and i believe is muching the number
>> 5101199 I tried instead of the above rulle with the number but no
> luck.
>> Where and how the server is geting incoming did i cant findout
>>
>> Regards.
>>
>>
>> _______________________________________________
>> Spce-user mailing list
>> Spce-user at lists.sipwise.com
>> http://lists.sipwise.com/listinfo/spce-user
>>
>
         I think that SCPE is not reading dialed incomung  number 
properly. The number from my provider is 5101199 and the rulle I set for 
that is ^([1-9][0-9]+)$
I had this problem in asterisk when provider is not sending did number 
in sip header . I understand that asterisk is diferent from spce but if 
I can tell somehow to spce how to get  the did like i did in asterisk. 
This is the link for asterisk to help understend what is my problem.
http://www.freepbx.org/support/documentation/howtos/how-to-get-the-did-of-a-sip-trunk-when-the-provider-doesnt-send-it-and-

Blagoja