Hmn. Very nice! It works!
On the matter of timing --
Asterisk appears to wait two full PSTN rings before it dials the SIP
extension. Is there any way we can tighten up this interval? Is that
done in the Zap configuration? The driver? The dialplan?
Asterisk is waiting for the CallerID, which is sent between the first
and second ring. If you disable CallerID you will shorten the
interval, but you will loose CallerID
hth
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