[asterisk-users] sip trunking and call transfer

[nik600]

On Mon, Nov 24, 2008 at 12:14 AM, Raj Jain <rj2807 at gmail.com> wrote:
>
> Yes, by using the SIP REFER method. Caller 2 will send a SIP REFER to Caller
> 1 asking it to talk to Caller 3. This will cause Caller 1 to drop it's
> session with Caller 2 and send a new INVITE to Caller 3. So, this is how you
> do it from a SIP protocol perspective. I'm not sure to what extent Asterisk
> supports this capability.
> --
> Raj Jain

ok, thanks for your reply!

I'll search about Asterisk SIP referer implementation.

-- 
/*************/
nik600
http://www.kumbe.it