[OpenSIPS-Users] Transfer issue
jeff at data102.com
Sun Oct 25 03:50:56 CET 2009
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 :)
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
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
> 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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