[asterisk-users] Que on A2Billing
Al Bochter
Al.Bochter at bochterservices.com
Tue Jun 19 16:03:04 CDT 2007
In a2billing just change the 9 to what you need it is right in the conf
file.
Best regards,
Al Bochter
Bochter Services
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Nitesh Divecha wrote:
>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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>>>
>>>
>>>
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>>
>
>
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