On 2009-06-03, at 22:30, Dale Worley wrote:

> This message is to describe an attack on the problem of grouping
> transferred calls in regard to CDR.  This proposal is to use a
> "References" header in the INVITE that establishes the latter,
> transferred dialog to show that it was derived from the first,  
> original
> dialog.
>

Just the post I needed :)

For the past two weeks I am all about SIP transfers.

a) hacking CSE XML builder to support additional info to allow future  
call resolvers and/or other programs to trace transfers.
b) fixing "external proxy" type transfer scenarios in FreeSwitch which  
are not existent at the moment.

So I see that in case of CSEs we need to export References header to  
be able to trace all transfer scenarios. :)
As for FreeSwitch fixing I see that my yesterdays patch 
(http://jira.freeswitch.org/secure/attachment/11573/urtho_nightmare_fix.txt 
)
does allow att transfers but incorrectly handles Refer-to headers.

There are multiple problems with SIP transfers in FreeSwitch and  
external proxy scenario and my first hack attempts to solve a scenario  
where FS is used as a PSTN gateway in front of sipXecs.
With FS 1.0.3 bundled in sipXecs 4.0 all att transfers fail. With FS  
1.0.4 they are totally not deterministic and with my latest patch they  
(seem) work all the time but apparently with Refer-To headers in the  
wrong place in the resulting INTIVE. This patch also fixes scenario  
where an assistant calls a conference room, then calls her boss and  
transfers the boss to the conference - IMHO this one is not in sipXecs  
tracker yet.

I will focus now on fixing other att transfer scenarios to AA and  
conference bridge which are present in multiple open issues on the  
tracker.

But apparently I need some RFC guru to explain me what is the correct  
behaviour so that future FS versions will be compatible with sipXecs  
4.2.

Can You show me two transfer examples - one where my UA (freeswitch)  
is hit with a REFER message (with all the invites, notifies and other  
that should follow) and the second one where my UA is hit with INVITE  
with REPLACES. I need an overdose of URI params and headers etc :)

Also a failure scenarios would be grate.

I am not good at synthesizing all the SIP knowledge/examples scattered  
across all of RFCs. Each one I find is oversimplified.
I need some hard core SIP transfer examples ;)

Pawel,


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