[asterisk-users] SIP call, updated with CID as it becomes available

[Raj Jain]

On Wed, Jun 11, 2008 at 9:17 AM, Brian J. Murrell <brian at interlinx.bc.ca> wrote:
> I'm wondering if the SIP lines can start ringing as soon as the zap line
> gets a call and when the zap line finally gets the CID, that is passed
> down to the already ringing SIP phones.

This is actually an interesting problem. The SIP protocol didn't
originally support this notion, but a recent extension to SIP adds
this capability to the protocol. This concept is known as
Connected-Identity in SIP and is defined in RFC 4916. The idea is to
be able to update remote party's identity in either direction after
the call has been answered or while it is ringing. I don't think
people were really aware of the scenario that you've described, but it
is an interesting one and I think RFC 4916 covers it.

The thing though is that even if somebody added this capability to
Asterisk, you'll need SIP phones that support this capability as well.
Right now, I don't think there is any SIP phone out there that
supports this.

--
Raj Jain