Call Control integration on OpenSIPS

I have finished integrating the codes for call control in the opensips.cfg. At initial testings call control was working fine. However, I had a hard time accounting using radius, the call control generated BYE when account balance reaches zero. Since the BYE which was generated by internally, should be accounted using the local_route. Unfortunately, I was not aware that this was the proper way to account the BYE. After reading some articles on the mailing list I was able to realize that my accounting script for an internally generated BYE should not be located at the main route. Please note that the main route and local_route are not the same (I guess I still need to read more documentation).

Here is the code for accounting internally generated BYE:
local_route {

if (is_method(“BYE”) ) {
acc_db_request(“Internally generated BYE”, “acc”);
acc_rad_request(“Internally generated BYE”); #// This is sending an Bye to the radius server to create the stop time.
}
}

OpenSIPS Webinar Schedule

Guys the AG-Projects Team is organizing another webinar!!!

Hi,

The next free webinar is scheduled for Wednesday 8th of July 2009. The
topic is "Routing in SIP"

This webinar will help you to understand SIP routing concepts. SIP
routing is very often a misunderstood concept. It is essential for
anyone trying to script OpenSIPS to fully understand these concepts.

Registration form : https://www2.gotomeeting.com/register/429289130

Also see http://www.opensips.org/Training/Webinars for future updates.

Best regards,
Bogdan

Hope you can set a time to listen to this webinar. This would be a great opportunity to listen to the experts and ask some questions.

OpenSIPS: Case Study Overview

Started:

January 2009

Operating System:
Fedora Core 10

Billing and Accounting:

CDRTool Version 6.8

FreeRadius 1.1.7

Presence:

OpenXcap

Chat

MSRPRelay

Status:

June 2009

Everything is working using a modified version of the original opensips.cfg configuration. Accounting and billing using CDRTool and FreeRadius is functional. After the difficulties in figuring out why each call has been classified as a Free Call, which can be pointed to the dial plan configuration in the opensips.cfg file.

A very simple NAT traversal support has been created in the routing logic using the nathelper module.

The next focus of the study is the dial plan and create an advance nat traversal routing logic which will determine NAT situation on register request,

Intro

At last I was able to finish installing all the necessary applications for VOIP project. Althought there still some problems. I was able to install opensips 1.5, OpenXcap, CDRTool, Mediaproxy, Radius Server, MSRPRelay.

I have created this blog to document all the problems that I will encounter and if possible post the solutions to the problem.

Currenlty, our project is encountering on the billing part of the CDRTool. The price for each call is not computed.