[Users] REFER To: (how to handle in OpenSER)

Frogger froggerandbongo at yahoo.com
Fri Jul 14 07:53:11 CEST 2006


I have done a lot of investigation and debugging. 
Here is where I am so far.

I am able to get transfers working in the following
situation:

++This works++
==============
SIP-ua1 calls pstn phone through gw

SIP-ua1 transfers pstn phone to final-callee:  SIP-ua2
==============



**This does NOT work**
==============
PSTN phone calls SIP-ua1

SIP-ua1 transfers pstn phone to final-callee:  SIP-ua2
==============

Basically, if sip-ua1 is the primary caller, the
transfer works.  If pstn gw is the primary caller, the
transfer does not work.

Any thoughts?

Please, even the slightest comment can be helpful to
crack this case.

Thank you!  FR  


--- Juha Heinanen <jh at tutpro.com> wrote:

> Frogger writes:
> 
>  > I am concerned about the "@sip.refer.com".  I am
> not
>  > sure how the gateway is handling this.
> 
> my understanding is that cisco doesn't look host
> part at all.  as i
> said, debug your dial plan when refer comes in.
> 
> -- juha
> 


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