[OpenSIPS-Users] Use Opensips for wholesale sip trunking

Gabriel Bermudez elgabo81 at gmail.com
Fri Apr 17 17:21:22 CEST 2009


Hi Gary,

I'm also a newbie on opensips (I've been playing around with for only a 
few months), if got something wrong below fell free to correct me.

Gary Chen escribió:
> I am new to Opensips. I'd like to setup Opensips as wholesale sip 
> trunking switch. Here is what I want to do:
> 1) For incoming calls from carrier. Check called number and determine 
> which IP (customer PBX IP) to forward to. Which module should I use to 
> do this? (carrier routing, droute or something else?). I need to have 
> every customer DID in the database to determine which IP to send call 
> to. Can I do this with Opensips? Example will be great.
For this you can use the userloc module, each IP PBX should register to 
opensips and you will use the module to send SIP messages to the correct 
IP PBX

> 2) For outgoing calls from customer PBX. Check the customer PBX IP. 
> If  it is in our database, forward the call our carrier gateway. If 
> the IP is not in our database, reject it. Which module should I use 
> for this ?
The module permissions should help you on a IP based authorization. But 
you can also base you authorizations on a username, password with the 
auth module

> 3) log all the CDR for billing purpose. Do I need to have RTP going 
> through our to do the accurate accounting? I'd really like to not have 
> RTP going through our server. This will save a lot of bandwidth.
When you don't have RTP traffic information you have to relay on the 
INVITE and BYE messages to get an accurate accounting, if you can 
guaranty the BYE reception I don't see why you should need to handle RTP 
traffic.  Although you should handle RTP traffic when a UAC is behind a 
NAT (use a NAT traversal solution)

>
> If you  can show me the real configuration file that will be great.
>
> Thanks advance.
>
> Gary
>
>  
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Hope this helps.

Regards,



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