[asterisk-users] Que on A2Billing

[Nitesh Divecha]

Thanks everyone for the input...

In real world we can not ask the customers to dial 9, if they want to 
call another SIP user... and trust me its confusing for a customer 
also... meaning when to dial 9 and when to not...

We have a custom proprietary system which does this part very well... 
Before it sends the call on a Trunk it will check the DID, if it exists 
within the local system. If it does then it will just use IP to IP call, 
else send the call to Trunk...

I think its possible to do this by creating some basic dial plans... 
Same like creating local extensions.

Cheers,
Nitesh




John Novack wrote:
> Given that Asterisk is modeled on, in the telephone industry, an 
> obsolete PBX design, without many of the modern day hybrid features, and 
> only recently has any effort been made to provide buttons and lights for 
> "lines" ( Is that yet working in 1.4??) one would have to do some very 
> careful number parsing to not use a trunk digit.
>
> If every phone in the system had buttons and lights representing 
> external connections and internal connections on other button(s) ( 
> intercom ) this wouldn't be an issue.
> Most "legacy" systems have been able to do this for the last 20 years or so.
>
> John Novack
>
>
> Nitesh Divecha wrote:
>   
>> Thanks man,
>>
>> Is there any other way without dialing 9... it will be kinda pain for a 
>> customer to dial 9 every time and plus they need to know also...
>>
>> Is there any intelligent way to identify? if its a local SIP then don't 
>> route to Trunk else route to Trunk.
>>
>> Cheers,
>> Nitesh
>>
>>
>> Guillermo Salas M. wrote:
>>   
>>     
>>> On Tue, 2007-06-19 at 09:36 -0400, Nitesh Divecha wrote:
>>>   
>>>     
>>>       
>>>> Thanks man...
>>>>
>>>> So far everything worked as expected...
>>>>
>>>> How can I make internal calls stay within the PBX. For example, when
>>>> one 
>>>> SIP-Friend tries to call another SIP-Friend without sending the call
>>>> out 
>>>> on Trunk and receive it back. Same like dialing from one extension 
>>>> number to another extension.
>>>>
>>>> My SIP-Friends are using US DID numbers and I would like to keep the 
>>>> local calls within the network.
>>>>
>>>> Right now when I try to call other SIP-Friend, I get a message saying 
>>>> "The number you have dialer is currently not available"... while the 
>>>> SIP-Friend is registered.
>>>>
>>>>     
>>>>       
>>>>         
>>> Try dialing the number 9 before the sip/iax2 friend number.
>>>
>>> Regards,
>>>
>>>
>>>   
>>>     
>>>       
>>>> Cheers,
>>>> Nitesh 
>>>>     
>>>>       
>>>>         
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