[asterisk-dev] 'IAX2 call variable passing between servers '

[Steven Critchfield]

On Mon, 2006-08-07 at 13:33 -0600, Douglas Garstang wrote:
> > -----Original Message-----
> > From: Kevin P. Fleming [mailto:kpfleming at digium.com]
> > Sent: Monday, August 07, 2006 12:45 PM
> > To: Asterisk Developers Mailing List
> > Subject: Re: [asterisk-dev] 'IAX2 call variable passing 
> > between servers
> > '
> > 
> > 
> > ----- Douglas Garstang <dgarstang at oneeighty.com> wrote:
> > > The endpoint, UA-B, is a sip phone. How can an IAX2 call be 
> > placed to
> > > the new destination?
> > 
> > Please ask questions that make a little more sense... if the 
> > new destination is (for example) extension 3001 in the SIP 
> > peer's dialing context, and that extension ends up pointing 
> > to an IAX2 endpoint, then of course the outbound call that 
> > will be placed will be an IAX2 call.
> 
> I'm having a little trouble understanding how a Polycom 601 phone that
> has no knowledge of IAX can receive an IAX call. It may make perfect
> sense to you, but it doesn't to me. 

In your example, this is very easy. 

Sip-1 ---> SIP call --> Asterisk 1 --> IAX2 --> Asterisk 2

At asterisk 2, you are IAX2. You don't know about the other side, and
you shouldn't care about the otherside. Any call from asterisk 2 out
won't know that the link on the otherside of asterisk1 is and shouldn't
care.

I'm assuming that some where you are hoping to get a transfer done that
will get your asterisk machines out of the loop as well as your other
stated problems. Without using the same transport from end to end, it
won't happen.


> > However, you keep referring to these calls being 'flagged as 
> > IAX2 calls when they are SIP calls', which sounds like 
> > something that your AGI script (or something else) is doing, 
> > not Asterisk. Asterisk doesn't 'flag' calls as being any 
> > particular type, so unless you can provide an example of 
> > exactly what Asterisk is supposedly doing wrong, I don't 
> > think you have actually demonstrated any problem in Asterisk at all.
> 
> I am referring to the agi_type variable that Asterisk passes to an AGI
> script when it calls it. It is being set to IAX2, not SIP.

Hopefully this will be clear from the above comments.
-- 
Steven Critchfield <critch at basesys.com>