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 > __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com _______________________________________________ Users mailing list [email protected] http://openser.org/cgi-bin/mailman/listinfo/users
