[asterisk-users] log incoming calls without answering

[kevin.larsen at pioneerballoon.com]

> I've already proposed your solution (is the most reasonable) but they
> have more than 60 analogs lines (no faxes) and some of them terminate in
> appliances like alarms, etc, so the solution must not touch in any way
> the connection between the line and his termination: doing a analog to
> digital conversion, passing it to asterisk and the convert it back to
> analog is prone to problems (what if asterisk crashes? or if a gateway
> fail?).
> I can split the existing lines (there are no complex things like adsl or
> digital signaling), convert the branches to digital and terminate then
> into an asterisk machine, so any failure will not affect the old
> circuit, but of course I've to configure asterisk to ONLY LOG calls and
> nothing more.
> 
> This is what they want:
> - line 1 ring
> - line 1 is splitted in two, the first branch (let's say the "analog"
> branch) go to an analog phone, that rings
> - the second branch go through a gateway and then to asterisk
> - asterisk log (with an AGI for example) "line 1 rings at .... from 
...."
> no more is required from asterisk, if someone answer the analog phone or
> not is not my business.
> 
Ok, so I would agree with them that a conversion to digital and back again 
would tend to break things like fax lines and alarm lines. My analog lines 
in my facilities are there because a lot of alarm systems just don't work 
with SIP at all. It's something the alarm companies are going to have to 
figure out in the next decade or so as the Telcos are moving away from 
copper and switched networks and towards fiber and packet based networks.

I honestly don't know if you can do what you want without some piece of 
equipment picking up the line. What I would do is get an analog line, an 
analog phone, an analog to sip device (there are many to choose from) and 
a basic asterisk instance. I would then make a small test setup where the 
analog line goes to a splitter. One side of the splitter goes to your 
analog phone. One side goes to your analog to SIP converter and then into 
your asterisk instance via your ethernet network. Use your cell phone to 
call the number of your analog line and see if it works. You would have to 
code a basic dialplan on the asterisk side and set up the trunk to your 
converter, which I am assuming you know how to do.

This would at least give you a fairly low cost way to test to see if you 
can trigger what you want on the Asterisk side without also triggering the 
line itself to be answered. I would also note that you would only be able 
to log incoming calls this way. I can't see a way you would be able to 
detect an outgoing call from the analog extension.

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