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

[Steven]

On Tue, 2006-08-08 at 09:05 -0600, Douglas Garstang wrote:
> > -----Original Message-----
> > From: Andrew Kohlsmith [mailto:akohlsmith-asterisk at benshaw.com]
> > Sent: Tuesday, August 08, 2006 7:31 AM
> > To: asterisk-dev at lists.digium.com
> > Subject: Re: [asterisk-dev] 'IAX2 call variable passing 
> > between servers
> > '
> > 
> > 
> > On Monday 07 August 2006 16:02, Douglas Garstang wrote:
> > > UA-A ---> Asterisk-1 ---> IAX2 ---> Asterisk-2 ---> UA-B
> > 
> > > The call from Asterisk-2 to UA-B has to be a SIP call, 
> > because UA-B only
> > > talks SIP. It doesn't know anything about IAX2. I 
> > ABSOLUTELY do care about
> > > the other side, because when Asterisk-2 calls my AGI script 
> > that dials
> > > UA-B, it sets agi_type to IAX2, not SIP. If that UA-B 
> > transfers or forwards
> > > a call, and sends a 'Moved Temporarily' SIP message back to 
> > Asterisk-2,
> > > Asterisk-2 re-enters the dial plan again with agi_type set 
> > to IAX2 again,
> > > not SIP. Because the type is IAX2, we've lost our rdnis 
> > value which is
> > > essential for forwarding calls, and we've lost dnid which 
> > is essential for
> > > transferring calls.
> > 
> > From Asterisk-2's point of view, the call is both IAX2 and 
> > SIP; unless you're 
> > leaving something out (I did see your other message), the 
> > call is BOTH IAX2 
> > *AND* SIP, and "what technology" the call is depends on how 
> > you look at it.  
> > This appears to be an issue outside of Asterisk because as 
> > Kevin said, 
> > Asterisk doesn't "flag" calls as being of one technology or 
> > another.  Indeed 
> > it makes no sense to do that from Asterisk's point of view, 
> > as it can be 
> > bridging between two or even more channel technologies.
> > 
> > I am guessing Asterisk is setting the call_type to IAX2 
> > because the source of 
> > the call *is* IAX2, but that is only a guess.
> 
> Still can't understand how a 'Moved Temporarily' SIP message from the
> phone, which causes Asterisk to re-enter the dialplan, can be an IAX2
> call.

Because you are having a problem accepting that the portion of the call
you are dealing with at this moment is simply

IAX2 -> Asterisk 2

When the SIP call is "moved" you re-entered the dialplan with ONLY a
IAX2 link to deal with.
 
> > It really does appear that your own scripting needs to be 
> > smarter, perhaps 
> > even so much so that you need to specifically set some 
> > dialplan variables 
> > before the AGI is called, and maybe even include some smarts 
> > to prevent it 
> > from looping calls when you receive the temporary failure messages.
> 
> I don't think it's an issue of the scripting. My script can only work
> with the variables it receives when Asterisk runs it. If vital info
> such as rdnis and dnid are not set, no amount of logic in the AGI is
> going to help. I don't know what you mean by 'looping calls'... phone
> sends a moved temporarily message back to Asterisk which causes it to
> re-enter the dialplan and try to dial the number moved to. (specified
> by the SIP 'Redirect' header I think).

Well, first we need you to get smarter, then you can work on making the
changes necessary to get your script to work right.

> > > Well, that seems to be the problem. IAX, from what I understood was
> > > supposed to be exactly designed for this kind of thing. As soon as I
> > > changed to SIP between the Asterisk boxes, these problems went away.
> > 
> > I really do fail to see this as an IAX2 issue... Asterisk 
> > will have the same 
> > trouble routing a call using any two different technologies.  
> > You can't have 
> > Asterisk drop out of the loop if it has to bridge two different 
> > technologies...  That's like saying you want to drive from 
> > New York to 
> > London, but you refuse to accept that the ocean liner you 
> > drive your car on 
> > to must be there for the trip across the ocean...  or something.  :-)
> 
> So, in summary, what we are essentially saying here is that IAX2
> cannot be used to trunk calls between Asterisk systems, when both
> end-points at SIP based devices unless you want to lose SIP
> functionality. That's really what we are saying here. We've since
> gotten this to work with SIP trunking between the Asterisk systems,
> and didn't have any of the problems associated when using IAX2.

Seems that MANY people use IAX2 to trunk calls from one side to another.
You are experiencing PEBKAC errors and suffering from poor choices. You
chose to originally try to use multiple transports which walled off
certain features.

> It just seems bizarre, because this was exactly the type of situation
> IAX2 was created for... I thought.

Where you privvy to the design documentation or meetings? You once again
prove your troll status by throwing out a comment that shows your wish
to stir up trouble. 
-- 
Steven <critch at basesys.com>