On 01/04/2011 01:55 PM, A J Stiles wrote:
On Tuesday 04 Jan 2011, Gilles wrote:
Thanks Sebastian for the tip. The goal is to 1) have clients call the
usual landline number instead of asking them to try a cellphone in
case no one's home, 2) get Asterisk to handle the call, 3) have the
cellphone ring with the CID of the original caller instead of
Asterisk's.

The problem with doing no. 3 is, if you are routing the call over the PSTN at
any rate, your telephone company will  (silently)  *drop* the caller ID if
the number you are presenting does not actually "belong" to you.  This is
*good* most of the time, because it means you can trust other people's caller
ID to be accurate  (and untrustworthy caller ID makes caller ID pointless).

I agree with your point. That is why routing the divert part of the call through an (effectively) internal SIP extension - which is the case if you call your laptop or Android phone through SIP as an internal extension to your Asterisk server (through OpenVPN as well, optionally) has the advantage that you can transmit/present whatever Caller ID you want.

Sebastian


We first met this when we ordered our second E1 line and batch of presentation
numbers.  As a result of a mistake on somebody's part, the two lines appeared
(according to BT's records)  to belong to different companies.  As a result,
approximately half our calls were going out anonymously; because if we were
trying to go out on span 2 but using a number that was only allowed on span
1, or vice versa, then the ident would get stripped somewhere along the way.

Diagnosing this obscure fault rather stretched the definition of "fun"  :/





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