[OpenSIPS-Users] Transfer issue
Anca Vamanu
anca at opensips.org
Fri Nov 6 17:49:25 CET 2009
Hi,
The REFER handing support has been added in B2BUA. Please update from
svn to use this feature.
To enable it you have to load a simple scenario document that describes
the behavior of the B2BUA when a REFER message is received and then call
b2b_init_request("refer") for the initial Invite message.
I have also updated the documentation page and you can find there also
the scenario document for this feature:
http://www.opensips.org/Resources/B2buaTutorial#toc15.
Regards,
Anca
Jeff Kronlage wrote:
> 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 :)
>
> Thanks,
>
> Jeff
>
> -----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.
>
> Regards,
> Bogdan
>
> 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.
>>
>>
>>
>>
>>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
More information about the Users
mailing list