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

[Steve Totaro]

On Wed, Jun 11, 2008 at 2:30 PM, Brent Davidson
<brent at texascountrytitle.com> wrote:
> Steve Totaro wrote:
>
> On Wed, Jun 11, 2008 at 1:47 PM, Steve Totaro
> <stotaro at totarotechnologies.com> wrote:
>
>
> On Wed, Jun 11, 2008 at 11:53 AM, Raj Jain <rj2807 at gmail.com> wrote:
>
>
> 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
>
>
>
> 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