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

[Brian J. Murrell]

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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