[asterisk-users] Que on A2Billing

[Nitesh Divecha]

Hello All,

Stable release of A2Billing has solved most of my problems and so far 
everything is OK...

Right now the only problem I am facing with my SIP clients are: -
    - Three-way Calling
       Three-way calling works fine, but when SIP client hangs up the 
call, the other two channels are still active and talking.

    - Call Forwarding
       I created the context for *72, *73, *90, *91, *52, and *53. SIP 
client can enable and disable but it never works because "a2billing.php" 
will time out and     hang up the SIP channel.

    - Voice Mail
       I created the context for voice mail, but the calls will never go 
to voice mail because "a2billing.php" after 60 sec will hang up the 
channel.

No doubt A2Billing is a great software, but the above features are also 
essential for home SIP users...

Anyone can show or share their setup if they have implemented the above 
features with A2Billing Software.

Cheers,
Nitesh



Al Bochter wrote:
> In a2billing just change the 9 to what you need it is right in the 
> conf file.
> Best regards,
>
> Al Bochter
> Bochter Services
>
> ----------------------------------------------------------
> Need to call me use our web phone at the link below
> http://www.bochterservices.com/voip/iaxphone.php?cn=250
> ----------------------------------------------------------
> Can you WIN gold today? Click on the link and see.
> http://www.bochterservices.com/?t=USbill_email
> ----------------------------------------------------------
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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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>> Inbound (clean). Database: 000750-2, 06/19/2007 - 6/19/2007 4:22:13 PM
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>>   
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