On Wed, Jun 11, 2008 at 1:47 PM, Steve Totaro <[EMAIL PROTECTED]> wrote: > On Wed, Jun 11, 2008 at 11:53 AM, Raj Jain <[EMAIL PROTECTED]> wrote: >> 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 >> > > If you search the archives, you will see this topic come up again and > again, and in reality it is an issue. If nobody answers a phone in > say five to ten seconds (including voicemail), I hangup. > > Ok, then build it in now. Make it work for DAHDI and when the phones > start implementing the capability, Asterisk will be ready. People > with channel banks or similar can benefit immediately. > > Thanks, > Steve Totaro >
Correction, seconds should read rings. _______________________________________________ -- 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
