[OpenSIPS-Users] Transfer issue

[Jeff Pyle]

Yes, samples/examples would be fantastic.  My first adventure into the b2bua
module ended about as quickly as it began with mountains of auth problems.
For example, I'd love to see the differences between a "standard"
(non-b2bua) call routing config and one that accomplished much the same
thing but with the topology hiding scenario in use.  And while solving this
REFER situation sounds less of a priority to me than to Jeff K., I feel I
must walk with the tophiding before I can run with REFERs.


- Jeff



On 10/24/09 10:50 PM, "Jeff Kronlage" <jeff at data102.com> 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.
>> 
>> 
>> 
>>   
> 
> 
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