[OpenSIPS-Users] SIP Trunking

[Steven C. Blair]

I did this once before. I would suggest dividing the config file into two pieces. The first handled outbound to the PSTN, the second inbound from the PSTN. This allows you to rewrite header fields based on your requirements or those of your ITSP in a fairly straightforward way

-steve

-----Original Message-----
From: users-bounces at lists.opensips.org [mailto:users-bounces at lists.opensips.org] On Behalf Of Matthew S. Crocker
Sent: Thursday, August 20, 2009 1:54 PM
To: users at lists.opensips.org
Subject: [OpenSIPS-Users] SIP Trunking


Hello,

 I'm brand new to OpenSIPS, just going through the make process now.  

 I need to configure OpenSIPS to act like a SBC for some SIP trunks coming off a VoIP switch.  Where should I look for Documentation/Examples of a working config?

Here is my scenario:

OpenSIPS has two interfaces,  private & public.  
VoIP Gateway is on private LAN with no gateway configured (it can only talk to local machines, no routing)

End user has an Asterisk server on a private lan behind their firewall (NAT)

I need to configure OpenSIPS to listen for SIP messages on :5060 from the end user firewall.  It then need to rewrite the SIP message and send it to the Gateway.  The Gateway would see the messages coming from the internal IP of the OpenSIPS server.  Once all of the SIP messages get processed I then need the OpenSIPS server to proxy the RTP streams (plan on using mediaproxy) between the Asterisk server and VoIP Gateway.

Any helpful hints on where to look?

-Matt


-- 
Matthew S. Crocker
President
Crocker Communications, Inc.
PO BOX 710
Greenfield, MA 01302-0710
http://www.crocker.com
P: 413-746-2760


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