[OpenSIPS-Users] Transfer issue

Jeff Kronlage jeff at data102.com
Sun Oct 25 03:50:56 CET 2009

Hi Bogdan,

Thanks for the fantastic news.

I don't suppose you have any samples of how to interpret a REFER and perform a transfer?

I've started pouring through the documentation for the B2BUA, but I'm still grinding through it :)



-----Original Message-----
From: users-bounces at lists.opensips.org [mailto:users-bounces at lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu
Sent: Saturday, October 24, 2009 11:18 AM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Transfer issue

Hi Iñaki,

Actually this is why the B2BUA module was designed in opensips - in most 
of the cases you need to control/change the dialog(s) but without any 
media dependencies/penalties (like you have now in most of the IP-PBXs).

So you actually can have a highly scalable signalling B2BUA - the 
opensips module could be used to locally (on opensips) interpret the 
REFER and do the call transfer, totally transparent to the other party.


Iñaki Baz Castillo wrote:
> El Sábado, 24 de Octubre de 2009, Jeff Kronlage escribió:
>>  Our setup has been initially
>>  engineered for thousands of concurrent calls, and we're hoping to avoid
>>  having a dozen Asterisk machines :)
> What you are looking for is the dream all want: a scalable SIP B2BUA (no media 
> handling), so a cluster of these B2BUA's would be located behind a proxy 
> (which does load balancing and failover). And it would be greater if the B2BUA 
> share information (about current dialogs and so) in some way (memcache? common 
> database?...).
> You could implement it with SipServlets (see Sailin SIP application server or 
> others), or FreeSwitch which allows calls without handling the media...
> Of course, Asterisk is not the most suitable solution: it involves media 
> handling ("canreinvite" is a hack), it has a very poor SIP stack... and 
> basically it's designed to be a single PBX box.

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