[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