[asterisk-users] Que on A2Billing

[Nitesh Divecha]

Strange...
Got it working now... I can receive incoming call...

Changed following parameters in additional_a2billing_sip.conf of the DID 
to: -

qualify=yes
canreinvite=no

Cheers,
Nitesh



Guillermo Salas M. wrote:
> On Fri, 2007-06-15 at 17:42 -0400, Nitesh Divecha wrote:
>   
>> When I call from my cell to the above DID, it hits on the Asterisk and
>> I 
>> see A2Billing trying to call SIP/2486543210, but it fails because 
>> Asterisk says "Unable to create channel of type 'SIP' (cause 3 - No 
>> route to destination) ". 
>>     
>
> I know it, but the error is saying that you don't have one 2486543210
> user registred.
>
> Show us the output of:
>
> sip show peers
>
> Regards,
>
>