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

[Brian J. Murrell]

On Wed, 2008-06-11 at 13:30 -0500, Brent Davidson wrote:
> 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.

It's like that here in Canada too.

> 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.

Yeah.  I had never thought it through that fully but indeed, that would
be what I'm talking about.  I guess it never occurred to me that way
because I don't usually Answer() in my dialplan contexts anyway.  I
don't think I've every really understood why I need to given that it all
seems to work without doing that.

b.

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