On Wed, Jun 11, 2008 at 9:17 AM, Brian J. Murrell <[EMAIL PROTECTED]> 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 _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
