The way our SIP net works is that phone B sends out a
BYE with an ALSO header in it. The ALSO header will
have the SIP address of phone C.
B --------------> Proxy
BYE with ALSO
Proxy ---------------> C
INVITE, from A
A is on hold
C ----------------> Proxy
180
Proxy -------------> A
180
A ----------------> Proxy
200
Proxy ---------------> C
200
A <====================> C
For more info check out:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios121/121newft/121t/121t3/dtsipgv2.htm#xtocid1219618
--- Hrishikesh Saria <[EMAIL PROTECTED]>
wrote:
>
>
> How do I support call transfer when a session has
> already been started.
>
> The scenario is like this
>
> A ----------------> B
> invite
>
> A <--------------- B
> 180
>
> A <--------------- B
> 200
>
> _________________________
>
> Media has started
>
> Now B has to to transfer the call to C so that the
> media between A and B is terminated and the media is
> started between A and C.
> The point is B has to make the transfer.
>
> How is this possible using the SIP protocol.
>
>
>
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