On Wed, Jun 11, 2008 at 2:30 PM, Brent Davidson <[EMAIL PROTECTED]> wrote: > Steve Totaro wrote: > > 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. > > On the subject of CallerID and ringing, I'm not sure if it's like this > everywhere in the US, but where I live in Texas, our caller ID signal is > sent between the first and second rings. If the phone is answered in the > middle of the first ring then CID signal is never received. This might not > be an issue in the scenario being discussed, because it sounds more like > you're asking for Asterisk to connect the ringing Zap channel to a sip line > before issuing an "answer" in the dialplan. Correct me if I'm wrong. I'm > more used to using Asterisk in a PBX context with an automated attendant > that answers every call before ringing any of the extensions. The direct Zap > to Sip without without a menu is more of a switch context correct? > > -Brent
Not SIP necessarily, just progress into the dialplan, that includes IVR or anything that can be put in a dialplan. Your system using IVR still delays the call delivery into the dialplan. If you remove the callerid=yes statement, do a before and after and will see that the IVR picks up faster without having to wait for caller ID. Thanks, Steve Totaro _______________________________________________ -- 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