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I have an old A800 with the newer sip firmware that I've been using for
quite a while. They lose a bit being put into SIP (most noticeably the
ability to send dtmf out of band) but I had problems with chan_h323 a
while ago and got tired of all the hoops for oh_323 so was happy to
switch.<br>
<br>
I use the builtin dialplan as little as possible. As soon as someone
picks up the phone it goes straight to asterisk. This also means that
I can just use one hunt number for all outgoing pstn calls, even port
to port it goes through the * server.<br>
<br>
<br>
For the sip settings, I just use the proxy, not the registrar, as I
said all calls go straight to * anyway. <br>
<br>
Here's one pstn trunk group:<br>
Name = 27946-7<br>
Pass Through = no(0)<br>
Provide Call Progress Tone = no(0)<br>
Busyout = no(0)<br>
Hunt Algorithm = ascending(0)<br>
Modem Bypass = no(0)<br>
Direction = both(2)<br>
DN Used = public<br>
Forced IP Routing # = 1000<br>
Forced IP Routing # Type = public<br>
IP Extension = yes(1)<br>
channel ip-addr dnis rmt-line chan<br>
Maximum LAM Calls Allowed = 8<br>
LAM: Index Pattern Replacement NumberType<br>
1 < 9> < > 0<br>
<br>
That's only the stuff I changed or think is really that important. The
forced ip routing means it answers the phone immediately on pstn and
dials ext 1000 on the asterisk machine. The big thing is the lam
pattern stuff. You have to put in a pattern for the quintum to match,
otherwise it will give sip errors, as it doesn't understand where to
send any incoming calls. It could be anything, but I just started with
a 9, the only thing is it can't start with the same thing as any of the
extensions that're on the pbx side.<br>
<br>
On the pbx side:<br>
Name = 711<br>
Pass Through = no(0)<br>
Hunt Algorithm = ascending(0)<br>
Direction = both(2)<br>
DN Used = public<br>
Forced IP Routing # = 1000<br>
Forced IP Routing # Type = public<br>
IP Extension = yes(1)<br>
channel ip-addr dnis rmt-line chan<br>
Public Number of Digits = 3<br>
Public Hunt Ldn's:<br>
1: 711<br>
<br>
Pretty much the same. You'll see the pub hunt ldn which is the
extension that I dial from asterisk (see the extensions.conf below).
This also goes straight to * which means there's a bit of a delay when
you pick up the phone - not enough to notice but if you pick up the
phone and dial straight away it might not catch the first digit. The
caller id gets set to "Quintum" <name> which is the name of the
pbxtg, which is why it's set to the extension. <br>
<br>
Anyway on the asterisk side the sip.conf is pretty basic, but make sure
you have dtmf=inband.<br>
<br>
Some parts of my extensions.conf:<br>
<br>
exten => _71X,1,Macro(stdexten,${EXTEN},SIP/${EXTEN}@tenor800,30)<br>
<br>
Dials the pbx side... tenor800 is the name from the sip.conf<br>
<br>
exten => _9XXXXX,1,Dial(SIP/${EXTEN}@tenor800,30|mH)<br>
<br>
Dials out on the pstn... As you can tell I start extensions going out
on the local pstn with a 9 in asterisk as well... If you just dial them
straight you'll have to add the 9 before the exten variable.<br>
<br>
Hope that helps... I don't imagine that it does any kind of
authentication on calls coming into it but since mines natted behind
two firewalls on a lan with the * machine I've never really checked. <br>
<br>
Pulu<br>
<br>
<br>
<pre class="moz-signature" cols="72">--
Pulu 'Anau
27946 x 711
878-7856
</pre>
<br>
<br>
Jessie V. Mabanglo wrote:
<blockquote cite="mid006101c517c0$7995b060$e9be45ca@JDMJAMBOO"
type="cite">
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<div><font face="Arial" size="2">My fellows,</font></div>
<div> </div>
<div><font face="Arial" size="2">We have <a
href="mailto:Asterisk@home">Asterisk@home</a> installed and we want to
interconnect it with our existing quintum gateways.. any idea how to
config that?</font></div>
<div> </div>
<div><font face="Arial" size="2">Your time is very much appreciated..</font></div>
<div> </div>
<div><font face="Arial" size="2">Cheers,</font></div>
<div> </div>
<div><font face="Arial" size="2">Jessie</font></div>
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