asterisk users - Jun 2008 - SIP call, updated with CID as it becomes available

Right now I have an Asterisk 1.4.18ish server and a Wildcard POTS
interface.  As it is now, when the zap line gets a call, Asterisk
answers it and waits for the analog CID to be presented, then rings the
SIP phones with the call and the CID.  There's a significant latency
involved in doing this.

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.

That way if a SIP phone user wants to wait for the CID, they can, but if
they just want to answer the phone without waiting for the CID, they can
do that too.

One might suggest that everyone wants to see the CID anyway, so why
bother?  Because in some situations, the phone is not at an arms reach
and the person only starts making their way towards it when they start
to hear the ringing, so if the ringing starts before the CID is
available it is likely that by the time they have gotten to the phone,
the CID is available and yet the latency between the availability of the
call on the zap line and it being picked up at a ringing phone has been
reduced a ring or two.

b.

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On Wed, Jun 11, 2008 at 9:17 AM, Brian J. Murrell <brian at
interlinx.bc.ca> wrote:
> Right now I have an Asterisk 1.4.18ish server and a Wildcard POTS
> interface.  As it is now, when the zap line gets a call, Asterisk
> answers it and waits for the analog CID to be presented, then rings the
> SIP phones with the call and the CID.  There's a significant latency
> involved in doing this.
>
> 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.
>
> That way if a SIP phone user wants to wait for the CID, they can, but if
> they just want to answer the phone without waiting for the CID, they can
> do that too.
>
> One might suggest that everyone wants to see the CID anyway, so why
> bother?  Because in some situations, the phone is not at an arms reach
> and the person only starts making their way towards it when they start
> to hear the ringing, so if the ringing starts before the CID is
> available it is likely that by the time they have gotten to the phone,
> the CID is available and yet the latency between the availability of the
> call on the zap line and it being picked up at a ringing phone has been
> reduced a ring or two.
>
> b.
>

Paul B just posted the same issue and suggested the same thing you did
in this thread "decrease the time it takes for asterisk (fxsks) to
answer"

This was my reply to his issue and a feature request I guess.

"That brings up a question though, on a regular landline with caller ID
the phone rings right away, it just doesn't display caller ID info
until a couple of rings.  Why not have that option in Asterisk?"

Thanks,
Steve T

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