[asterisk-ha-clustering] Asterisk clustering and transferring calls

Chris Mylonas chris at mrvoip.com.au
Tue Apr 21 19:51:40 CDT 2009


Feature Request.
Make an offer for a paid feature.



On Tue, Apr 21, 2009 at 12:41 AM, Brent Thomson <bthomson at getjive.com>wrote:

> Leif Madsen wrote:
> > I was not the OpenSER administrator (and do not have a ton of OpenSER
> > experience), but I think we ended up making all the requests and such
> > go to the OpenSER server, which I think had transaction support, and
> > used the outboundproxy option in sip.conf to make all the requests go
> > to the OpenSER server.
> >
> > I may be able to get access to the configuration that we used, but
> > that is a former client and I'll have to ask for permission to get it.
> > Not sure if any of that information is useful or not, because it was a
> > couple of years ago that we did it, and I'm just remembering bits and
> > pieces.
>
>
> Leif: I'd love to have a look at that if it's not too much trouble.
>
> Chris: Correct. SIP transfers using INVITE and REFER.
>
> Yehavi: I've tried sending the same handset to the same PBX every time,
> but this breaks down if a call is routed to the handset from elsewhere,
> since it could land on any PBX and then get forwarded on to the handset.
> The work-around for this is to either keep an entire logical PBX on a
> single physical PBX (so that all calls, in and out, go to the same piece
> of hardware), or route all calls *to* a particular handset through the
> same PBX where its registration landed. But that kind of defeats the
> whole load balancing and failover thing.
>
> I did some poking around and came across this:
>
>        http://tools.ietf.org/html/rfc5359#page-58
>
> The document is excellent. It full of examples relating to RFC 3261 (and
> some of its extensions). The example for attended transfer clearly shows
> Alice initiating a call with Bob, then Carol, then directing Bob to
> connect to Carol. Bob then sends and INVITE to Carol with a Replaces:
> directive to complete the transfer. Replace Alice with Phone, Bob with
> PBX-1, and Carol with PBX-2 and you'll see what I'm trying to do. It
> appears that Asterisk may not follow spec in this case. Sniffing the
> traffic on PBX-1 verifies that Asterisk doesn't ever attempt to contact
> PBX-2 when it receives a REFER referencing a Call-ID that it doesn't
> recognize. Does this warrant a bug report/feature request?
>
> -Brent
>
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-- 
Kind Regards,
Chris Mylonas
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