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Jeremy,<br>
<br>
This is the best thing i was able to come up with. <br>
<br>
All incoming pots lines go to the "zapchans" context<br>
<br>
[zapchans]<br>
exten => 3,1,Dial(ZAP/1-1) ;ZAP3<br>
exten => 3,2,Hangup()<br>
exten => 4,1,Dial(ZAP/2-1) ;ZAP4<br>
exten => 4,2,Hangup()<br>
<br>
exten => s,1,Answer()<br>
exten => s,2,Goto(${CHANNEL:4:1},1)<br>
exten => s,3,Hangup()<br>
<br>
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).<br>
<br>
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.<br>
<br>
<br>
Jeremy Mann wrote:
<blockquote
cite="mid:5ED7AD82D22AE34E9000DDEAE5C999510C2563D4E4@exchsvr.txhmg.com"
type="cite">
<pre wrap="">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: <a class="moz-txt-link-abbreviated" href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a> [<a class="moz-txt-link-freetext" href="mailto:asterisk-users-bounces@lists.digium.com">mailto:asterisk-users-bounces@lists.digium.com</a>] 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:
</pre>
<blockquote type="cite">
<pre wrap="">
Rob Schall wrote:
</pre>
<blockquote type="cite">
<pre wrap="">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?
</pre>
</blockquote>
<pre wrap="">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: <a class="moz-txt-link-freetext" href="http://www.voip-info.org/wiki-Asterisk+tips+DID">http://www.voip-info.org/wiki-Asterisk+tips+DID</a>
(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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