[OpenSIPS-Users] OpenSIPS & Asterisk Integration in a new way

[aamir chougule]

Hi Sammy,

Thanks a ton for saving the day. Yeah actually after I wrote this mail and searched on the internet I got this feature ready in front of my eyes. But I was searching this for a week on the internet how to do it but was out of luck. But your idea did changed the things going around in my mind.

Now my question is the call will be a new call and this new call will not me merged with the call that came into the OpenSIPS server for the first time. I wanted asterisk is between just for reading or fetching numbers from the customers and then control goes back to OpenSIPS server.

Thanks a lot again Sammy, appreciated your soonest reply. Any ideas on the new question.

 
Regards,

Aamir Chougule
Cell: 09167989111



________________________________
 From: SamyGo <govoiper at gmail.com>
To: aamir chougule <aamir_ryu at yahoo.com>; OpenSIPS users mailling list <users at lists.opensips.org> 
Sent: Monday, 2 July 2012 5:14 PM
Subject: Re: [OpenSIPS-Users] OpenSIPS & Asterisk Integration in a new way
 

First of this is not new. 
Just in your asterisk servers where you define the carriers replace all the carriers with just one opensips peer. Then in opensips.cfg on any incoming call detect that the call is coming from your asterisk server's - if that turns out to be yes use the LCR module to send out the call.
Obviously you need to define the carriers in your LCR db table in a new way ;)

Thanks
Sammy


On Mon, Jul 2, 2012 at 4:34 PM, aamir chougule <aamir_ryu at yahoo.com> wrote:

Hi Team,
>
>
>I have installed opensips and asterisk in this way as given below:
>
>
>Current Scenario:
>
>
>
>Calls comes in to OpenSIPS server ==> Authentication & Proxying part is done by OpenSIPS ==> Call is relayed to Asterisk Server ==> Asterisk Server ask the customer to dial the number that he/she wants ==> after fetching the number Asterisk relays the call to the carrier
>
>
>The Aim of the asterisk in the current scenario is to just fetch the numbers that the customer wants to dial and relays it to the carrier.
>
>
>
>Wanted Scenario:
>
>
>Calls comes in to OpenSIPS server ==> Authentication & Proxying part will be done by OpenSIPS ==> Call is relayed to Asterisk Server ==> Asterisk Server provides the IVR services to fetch the number from the customer ==> Asterisk passes on the fetched number to the OpenSIPS Server ==> OpenSIPS server relays the call to the carrier according to the LCR
>
>
>The Aim of the wanted scenario is to just rule out the possibility of the asterisk server to pass the call to the carrier, and OpenSIPS a more scalable in terms of a Proxier will be in between to deal with the LCR things.
>
>
>Team please help me out or point me to a direction so that I can achieve this goal.
>
>
>Thanking you in anticipation.
>
> 
>Thanks & Regards,
>
>Aamir
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>Users mailing list
>Users at lists.opensips.org
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>
>
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