[Asterisk-Users] Hypothetical IAX2 situation

[Adam Goryachev]

Actually, it depends on how you have configured the system.

If I understand, things should be configured like this:
PSTN -> a -> (IAX2) -> b -> (SIP) -> phone1

At this stage, both a and b are in the media path, same if PSTN is in
fact SIP, however, if PSTN was actually IAX2, then a would bail, and the
call should go direct to b.

phone transfers the call to another phone connected to a

PSTN -> a -> (IAX2) -> b -> (SIP) -> phone1
        a <- (IAX2) <- b
        a -> (SIP)  -> phone2
After a few seconds:
PSTN -> a -> (SIP)  -> phone2

However, if it was done this way:
PSTN -> a -> (IAX2) -> b -> (SIP) -> phone1
        a <- (SIP) <- b
        a -> (SIP)  -> phone2

Then, again, depending on your config, a might drop the last two legs,
and let b talk directly to phone2 ....

In other words, if you are ping-pong'ing between two servers, AND they
all legs are running with the SAME protocol (ie, all SIP or all IAX)
then they can drop the extra legs *IF* you have configured it that way
(see notransfer or transfer keywords in sip.cfg and iax.cfg)

Hope this helps explains things...

As always, if I got it wrong, please someone else jump in....

Regards,
Adam

On Fri, 2004-12-03 at 07:48, Kris Boutilier wrote:
> This is the purpose of the 'native transfer' feature controlled on a
> per-peer basis by the 'notransfer' directive in /etc/asterisk/iax.conf.
> 
> Asterisk box '*a' will, some time shortly after a 'source -> *b -> *a -> *b
> -> destination' call path has been set up, quietly step out of the way and
> connect the endpoints directly to each other. In this case, it means the
> call will be taken off of the network and bridged internally to *b.
> 
> Hope that helps. 
> 
> Kris Boutilier
> Information Systems Coordinator
> Sunshine Coast Regional District
> 
> > -----Original Message-----
> > From:	Sean Kennedy [SMTP:skennedy at tpno.org]
> > Sent:	Wednesday, December 01, 2004 3:00 PM
> > To:	Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject:	[Asterisk-Users] Hypothetical IAX2 situation
> > 
> > Two * servers:  *a and *b.
> > 
> > Outside call comes in *b, and is automatically routed to *a.  Someone on 
> > a sip phone connected to *a then decides to transfer the call to someone 
> > on a sip phone connected to *b.  The transfer works.
> > 
> > At this point, is *a still in the converstation?  Or is * smart enough 
> > to see where the data stream is going/coming from?
> 	{clip}
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users