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

[Douglas Garstang]

Jared,

I sure can. He's a first, basic scenario. UA-A wants to reach UA-B. We first execute a ChanAvail() application command to determine the location of UA-B. We then attempt to reach UA-B via the supplied IAX path. There's nothing too unusual about this. Lots of people are doing this.

 +------+           +-------+                          +-------+           +------+
 |      |           |       | DUNDi Lookup of UA-B     |       |           |      |
 | UA-A | - SIP --> | PBX-1 | -------- IAX2 ---------> | PBX-2 | - SIP --> | UA-B |
 |      |           |       |                          |       |           |      |
 +------+           +-------+                          +-------+           +------+

Now, when UA-B forwards a call that came from UA-A to UA-C, it sends a 'Moved temporarily' message back to PBX-2. PBX-2 re-enters the dial plan at this  point, looking for a new match for the forwarded call. It executes my AGI script again. Normally, the rdnis would be set, and the type of the call would be SIP. However, when the call has been trunked from another Asterisk system, the call type is instead IAX2, eventhough it should be SIP. As a result, because IAX2 does not convery rdnis, it gets lost.

 +------+           +-------+                          +-------+           +------+           +-------+           +------+
 |      |           |       | DUNDi Lookup of UA-B     |       |           |      | Moved     |       |           |      |
 | UA-A | - SIP --> | PBX-1 | -------- IAX2 ---------> | PBX-2 | - SIP --> | UA-B | - SIP --> | PBX-2 | - SIP --> | UA-C |
 |      |           |       |                          |       |           |      |           |       |           |      |
 +------+           +-------+                          +-------+           +------+           +-------+           +------+
                                                                                              ^                          ^
                                                                                              |                          |
                                                                                              +- Appears as an IAX call--+

A similar thing happens in the case of a transferred call. However, the dnid is set to the UA-B, and the extension is set to UA-C. Once again, because the call comes in as an IAX2 call, the dnid is lost.

 +------+           +-------+                          +-------+           +------+           +-------+           +------+
 |      |           |       | DUNDi Lookup of UA-B     |       |           |      |           |       |           |      |
 | UA-A | - SIP --> | PBX-1 | -------- IAX2 ---------> | PBX-2 | - SIP --> | UA-B | - SIP --> | PBX-2 | - SIP --> | UA-C |
 |      |           |       |                          |       |           |      |           |       |           |      |
 +------+           +-------+                          +-------+           +------+           +-------+           +------+
                                                                                              ^                          ^
                                                                                              |                          |
                                                                                              +- Appears as an IAX call--+

I'm not sure if these diagrams help much.... but it's a start. Does this make any sense?

Doug.


> -----Original Message-----
> From: Jared Smith [mailto:jaredsmith at jaredsmith.net]
> Sent: Friday, August 04, 2006 8:46 AM
> To: Asterisk Developers Mailing List
> Subject: RE: [asterisk-dev] 'IAX2 call variable passing 
> between servers
> '
> 
> 
> On Fri, 2006-08-04 at 08:32 -0600, Douglas Garstang wrote:
> > A far more serious issue is that at the other end of that 
> trunk, when
> > Asterisk makes calls to SIP phones, and those phones transfer or
> > forward calls, all new calls generated from those are flagged as IAX
> > calls instead of SIP calls. 
> 
> I'm not sure we have enough information of your setup to be 
> able to help
> you out here.  Can you please explain (preferably on a web page, with
> pictures or diagrams) what you're trying to do, so we can understand
> what your complaint is?  I certainly can't offer any suggestions if I
> can't visualize what it is you're trying to accomplish.
> 
> -Jared
> 
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