[OpenSIPS-Users] Forwarding REGISTER messages to Asterisk

Simon Shaw sshaw at interwise.com
Fri Sep 5 10:34:33 CEST 2008

Hi Darren,


I will also want use this setup where the remote PBX is Cisco, Avaya or Nortel.  I need this setup in my lab purely to make some of my testing and monitoring easier (long story, not really important here).







From: Darren Sessions [mailto:dmsessions at gmail.com] 
Sent: Thursday, September 04, 2008 6:28 PM
To: Simon Shaw; users at lists.opensips.org
Subject: Re: [OpenSIPS-Users] Forwarding REGISTER messages to Asterisk



I would highly highly highly recommend you *DO NOT* use Asterisk if you'll have any sort of volume of registrations (for the record, I am an Asterisk advocate).


I speak from experience and can say first hand that this is a **bad idea**.


Asterisk's sip stack and configuration setup really just don't allow for a high concentration of SIP registrations without a very significant penalty on the Asterisk host. I had a couple hundred registrations on a single Asterisk server that started to visibly effect the performance of the host server. When I hit 500 registrations, the Asterisk server was unusable. I have consistently been able to reproduce this effect to varying degrees on several Linux distros and on four different hardware vendor platforms.


Offloading the registrations to OpenSIPS is the smart way to do it. If you're wanting to send the call simultaneously to Asterisk and the end-point you can make a creative OpenSIPS config file using the LCR module or dbalias or something.


If you're wanting to do processing on the Asterisk server before the end-point is called, you could use the opensips config to route all the calls to the Asterisk server(s) and then have the Asterisk servers do their processing bit and send the call back to OpenSIPS which could detect the call as coming from your pool of Asterisk server(s) and do a location lookup at that point.


In any case, be creative, it's quite possible to do almost anything with the call flows you'd like having OpenSIPS at the front of your network. 


Good Luck!





Darren Sessions

dmsessions at gmail.com








On Sep 4, 2008, at 8:58 AM, Simon Shaw wrote:

I was hoping to solve this issue in a more generic manner.
Any idea how I change the IP address in the contact header?

-----Original Message-----
From: users-bounces at lists.opensips.org [mailto:users-bounces at lists.opensips.org] On Behalf Of I?aki Baz Castillo
Sent: Thursday, September 04, 2008 11:33 AM
To: users at lists.opensips.org
Subject: Re: [OpenSIPS-Users] Forwarding REGISTER messages to Asterisk

El Wednesday 03 September 2008 19:44:42 Simon Shaw escribió:

I would like to use OpenSIPS as a pure router between some of my SIP UAs

	and Asterisk such that when one of these UAs registers with OpenSIPS the

	REGISTER message will be redirected to Asterisk where it will be

	handled.   When a UA that is connected directly to the Asterisk box

	dials one of the OpenSIPS UAs I would like the SIP INVITE to be sent

	from the Asterisk box via OpenSIPS.


	I have managed to forward the REGISTER messages to Asterisk using

	rewritehost(<IPofAsterisk>), however when a UA that is registered

	directly with the Asterisk dials a UA that registered via OpenSIPS the

	INVITE message is sent directly from Asterisk to the UA.


	I guess this is happening because Contact header in the REGISTER message

	has the IP of the UA and not that of OpenSIPS.


	If this assumption is correct what is the correct method to change the

	IP in the contact header?  Could the NATHelper be useful here?

It's easy: if you forward the REGISTER from OpenSIPS to Asterisk then when 
Asterisk calls that user it will do it directly, since of course "Contact" in 
REGISTER has the UA IP as it's logical.

This would be possible with "Path" feature (see "path" module and related 
RFC), but of course, be sure that Asterisk doesn't implement it.

Iñaki Baz Castillo
ibc at in.ilimit.es

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