[OpenSIPS-Users] OpenSIPS Load Balancer of an Asterisk Cluster - REFER Issue

[Paris Stamatopoulos]

Hello Bogdan,

This seems rather interesting. I will try to setup an development instance of OpenSIPs to give it a try.

Regards,
Paris

-----Original Message-----
From: users-bounces at lists.opensips.org [mailto:users-bounces at lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu
Sent: Wednesday, May 05, 2010 6:52 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] OpenSIPS Load Balancer of an Asterisk Cluster - REFER Issue

Hi Stan,

Have you seen - 
http://lists.opensips.org/pipermail/users/2010-April/012414.html - this 
it about attended transfer when doing balancing for Asterisk.

Regards,
Bogdan

Stanisław Pitucha wrote:
> On 05.05.2010 11:50, Paris Stamatopoulos wrote:
>   
>> Has anyone experience this before? Do you have any working solutions?
>>     
>
> Yes and yes. Short version: It's impossible without hacking asterisk.
>
> Long version:
> First we tried routing all related calls to the same destination using
> the dialog module. It would work, but only in the most trivial cases. It
> will work only if you can always match the destination or the source
> number to an existing call. If you use call forking / huntgroups /
> queues / redirection on your asterisk - forget about easy transfers -
> you will get calls which don't match on any field, but are still going
> to the same phone.
>
> Now - theoretically transfers should work between asterisks based on the
> information in refer-to - it already includes the host which handles the
> call and in the transfer context you can extract that address and force
> a Dial() to that destination. I never discovered why it doesn't quite
> work. Half of the calls failed and sometimes I got deadlocks (still
> using 1.4 - maybe 1.6 is better here).
> AFAIK there's also no real possibility to set the callerid in that case
> to figure out who initiated the transfer.
>
> But we've got a working solution now:
> - The first part is a custom res_cluster module which will publish for
> each call (hooked into the Dial() command) the callid, ftag, ttag and
> the ip of the local asterisk into a shared mysql database.
> - The second part is activated on refer. It does some magic in
> handle_refer() (or something like that) and then magicks the call to
> magically know the correct callerid, customer id, checks whether the
> transfer is allowed and dials the destination asterisk with correct
> refer-to headers. (destination comes from the database)
>
> It could be much simpler if you don't have to handle different customers
> and bill them separately.
>
> Maybe I missed some other option... but that's the solution we got to
> after ~3 months of testing different methods, annoying asterisk mailing
> list and their IRC. And it works just fine so far.
>
> I'm not sure I can release all the needed patches, but if you want them,
> let me know.
>
> Stan
>
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>
>   


-- 
Bogdan-Andrei Iancu
www.voice-system.ro


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