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

[Steve Totaro]

On Wed, Jun 11, 2008 at 2:44 PM, Brian J. Murrell <brian at interlinx.bc.ca> wrote:
> 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.
>

If you ever have problems with a call dropping after 30 seconds,
Answer() is usually the cause.

Thanks,
Steve T