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

[Steve Totaro]

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