[asterisk-users] FXS + Pots Extensions Help

[Rob Schall]

Jeremy,

This is the best thing i was able to come up with.

All incoming pots lines go to the "zapchans" context

[zapchans]
exten => 3,1,Dial(ZAP/1-1)    ;ZAP3
exten => 3,2,Hangup()
exten => 4,1,Dial(ZAP/2-1)    ;ZAP4
exten => 4,2,Hangup()

exten => s,1,Answer()
exten => s,2,Goto(${CHANNEL:4:1},1)
exten => s,3,Hangup()

Each one could have its own context, but I wanted to keep it all in one
place and make it easy for our phone guys to handle (they aren't linux
or asterisk saavy).

The only problem I've found so far, is when you dial in from a pots
line, the call connects fine, but they won't hang each other up. The
console shows the "hangup" command running, but neither side of the call
will hangup when the opposite side hangs their line up. Not sure if I
just missed a setting or what.


Jeremy Mann wrote:
> Here's a silly question, if these are standard POTS you obviously know which number corresponds to which line, being the case couldn't you tell that ZAP/1 is POTS 555-1234, ZAP/2 is POTS 555-1235, etc etc?
>
> I'm assuming you're trying to identify the inbound number from the telco that was dialed.  Unless you have the lines in a hunt group at the telco, but then you're implying you don't care which number was dialed, you just want failover at the telco.
>
>
>
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Rob Schall
> Sent: Wednesday, May 23, 2007 8:19 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] FXS + Pots Extensions Help
>
> We wanted a cheap last resort fail-over. A few really cheap pots lines
> are easy to run buy, as we can get them for a really low cost. My
> understanding with DIDs (and its limited), is they have to belong to a
> PRI. The only way that is cheaper than a few pots lines is if you needed
> 8 or more pots lines. Then the line fees balance out.
>
> I was hoping for a solution more along the lines of.... Use this "x"
> variable that contains what ZAP channel it came in on, then I can
> program that one to point to a particular person.
>
> Thanks,
> Rob
>
> Sean M. Pappalardo wrote:
>   
>> Rob Schall wrote:
>>     
>>> Normally I just use pri's with our asterisk systems, but a request came
>>> in to add some normal pots lines to the setup. We have 3 lines, and they
>>> run into the fxs ports. They hit the dialplan just fine, and they always
>>> dial the "s" extension. However, my question would be... Is there a way
>>> to determine what number was dialed and have it forward to a specific
>>> phone?
>>>       
>> Sure, it's called a DID trunk. It's basically just a regular analog
>> phone line but the CO switch sends down the digits dialed in one of a
>> few ways: Dial pulse (DP), Multi-freq (MF), Dual-tone multi-frequency
>> (DTMF). They are usually inbound-only, but some CO's can add outbound
>> service too if needed. Call your phone service provider's business
>> office and ask about analog DID lines/trunks. They should be around
>> $30/mo for the line and $1-2/mo for each number. Ask them what type of
>> signaling they use then you'll need to configure your zapata.conf to
>> match. After that, you can then start routing in the dialplan based on
>> the number called. For extra fun, have the CO set them up in a hunt
>> group to avoid busy signals.
>>
>> Take a look at: http://www.voip-info.org/wiki-Asterisk+tips+DID
>>
>> (BTW, Why are you adding analog lines if you're already big enough for
>> a PRI? Isn't it less expensive to just add a couple more DID numbers
>> to the PRI?)
>>
>> Sean
>>
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