Hi,
take a look on the network trace and see if the REFER is properly routed
also in the second case.
regards,
bogdan
Frogger wrote:
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 <[EMAIL PROTECTED]> 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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