[asterisk-dev] Asterisk SIP Channels Bridge

[Mayank Mathur]

Dear Eric
I Got your reqm. but im not able to understood why ur using AGI scripting
to place a call or even to do conferencing in Asterisk using SIP.
R all your users accessing same Asterisk Server ??
We don;t req PRI if all your users r using Public IP and can access Server.
U can directly place a conference using Asterisk's Inbuilt features.


> Hi Mayank,
>
> Yes, I am trying to conference 2 users in through SIP. I am not using
> any Digium card and calls come in from the carrier via SIP.
>
> First caller would call in and be placed on hold. And I have the unique
> name of the channel saved in the database. And the subscriber will get a
> text message indicating that you have a call. And if the subscriber
> wants to talk to the original caller who is still on hold, he/she will
> call into the system and the system would bridge both calls together. I
> am getting sporadic results with the bridging. And while the original
> caller is on hold , music on hold will not play most of the time.
>
> I have read that this is a pretty simple feature to do if we use a PRI.
>
> thanks
>
> Eric Lee
>
>
> Mayank Mathur wrote:
>> hi
>> ru looking to do Conferencing b/w users thru SIP / just want 2
>> simultaneous users to get connected thru SIP ??
>> And what Prob ru facing ??
>> Let me know whether if i can help u out .
>>
>>
>>
>>> Hi there,
>>>
>>> I'm trying to bridge 2 SIP channels together via AGI script. The first
>>> caller would call in and be placed on hold and the second caller would
>>> call in and both the calls gets connected together.
>>>
>>> But I am having problem with the second caller finding the first
>>> channel.
>>>
>>> Can someone point me to the right direction?
>>>
>>> thanks
>>>
>>> Eric
>>>
>>> _______________________________________________
>>> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>>>
>>> asterisk-dev mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>    http://lists.digium.com/mailman/listinfo/asterisk-dev
>>>
>>>
>>
>>
>>
>
>
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-dev mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-dev
>


-- 

Regards,
Mayank Mathur