Integrating Asterisk And OpenSIPS In Realtime : Tutorial

Integrating Asterisk And OpenSIPS

We have been working with Asterisk and OpenSIPS for a long time (Even when OpenSIPS was OpenSER!). But we had to hammer out solutions as they fit and there were no methodology to it. Various engineers had their notes and were collected at one place only to be pulled out when the need arises to bring these two Open Source Telephony servers together.
SIP server like OpenSIPS usually provides registration and call handling, either by routing, forwarding or direct connections. The Asterisk on the other hand is a full blown telephony server, providing IVR, Voicemail, conferencing and announcements in addition to call handling, So by bringing these two servers (services) together, one will be able to provide full blown telephony solution integrating SIP, and Asterisk services.
So without saying that our method(s) are messy, I am going to point you to a very nicely written (Also very nicely thought out) article on how to integrated Asterisk and OpenSIPS.
Bogdan Andrei Iancu, a core OpenSIPS developer has written the tutorial. Go read it, Print it and save it! It is very good!
PS The tutorial currently addresses Asterisk 1.4 or 1.6 and OpenSIPS 1.5x. Once the OpenSIPS 1.6 is released the documentation will be updated appropriately.
Realtime OpenSIPS - Asterisk Integration

FreePBX 2.6 Beta 2 Released

FreePBX 2.6 Beta 2
Philippe from FreePBX has posted about the new beta release of FreePBX 2.6 (beta 2). We already have been working with the Beta 1 release and pretty happy with the performance. We are looking forward to install the FreePBX 2.6 Beta 2 as it has some new modules (We are really interested in Asterisk SIP and IAX settings as well as Bulk DID management);

• Asterisk SIP Settings & Asterisk IAX Settings
• Provides the ability to manage common global SIP and IAX settings that have often been confusing and error prone to do in configuration files, and includes a handy auto-configuration helper to determine your sip_nat.conf related settings (and further eliminates the need for that file).
• Outbound Route Messages
• Allows the ability to override the default “All Circuits Busy” message encountered when all trunks fail, and allows for explicit messages to be played for Emergency Routes.
• Weak Password Checks
• Provides an auditing function for all your SIP and IAX passwords to help harden your system and protect it from the many automated scripts on the Internet seeking out and exploiting such systems. Because of the security implications, this module was also introduced earlier into version 2.5.
• Bulk Extensions & Bulk DIDs
• Allows you to manage or provision large quantities of users and DIDs from a spreadsheet format.
• Custom Context & Route Permissions
• Provide some advanced functionality to limit routes or other features and capabilities to different extensions.

There are many other features and enhancements that Philippe has mentioned in his post like FOP 0.30 Updated to work with Asterisk 1.6, Virtual Extensions, many Queue updates.
You can download FreePBX 2.6 Beta 2 and other releases like FreePBX 3.0 from the usual download page.

Tags: FreePBX 3.0, Asterisk SIP, Asterisk IAX, FreePBX 2.6 Beta 2, Bulk DID management