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Hi Pulu 'Anau,
thank you very much for the advice.. we already did
some part.. and still we are on simulation to prefect it...
thank you very much again..
Regards,
Jessie
----- Original Message -----
Sent: Wednesday, February 23, 2005 5:55
PM
Subject: Re: [Asterisk-Users] PLease
help: Asterisk to Quintum interconnection
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.
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.
For the sip settings, I just use
the proxy, not the registrar, as I said all calls go straight to *
anyway.
Here's one pstn trunk group:
Name = 27946-7
Pass
Through = no(0)
Provide Call Progress Tone = no(0)
Busyout =
no(0)
Hunt Algorithm = ascending(0)
Modem Bypass = no(0)
Direction =
both(2)
DN Used = public
Forced IP Routing # = 1000
Forced IP Routing
# Type = public
IP Extension =
yes(1)
channel
ip-addr
dnis rmt-line chan
Maximum LAM Calls Allowed =
8
LAM: Index Pattern Replacement
NumberType
1
< 9>
< >
0
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.
On the pbx side:
Name =
711
Pass Through = no(0)
Hunt Algorithm = ascending(0)
Direction =
both(2)
DN Used = public
Forced IP Routing # = 1000
Forced IP Routing
# Type = public
IP Extension =
yes(1)
channel
ip-addr
dnis rmt-line chan
Public Number of Digits =
3
Public Hunt Ldn's:
1:
711
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.
Anyway on the asterisk side the sip.conf
is pretty basic, but make sure you have dtmf=inband.
Some parts of my
extensions.conf:
exten =>
_71X,1,Macro(stdexten,${EXTEN},SIP/[EMAIL PROTECTED],30)
Dials the pbx
side... tenor800 is the name from the sip.conf
exten =>
_9XXXXX,1,Dial(SIP/[EMAIL PROTECTED],30|mH)
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.
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.
Pulu
--
Pulu 'Anau
27946 x 711
878-7856
Jessie V. Mabanglo wrote:
My fellows,
We have [EMAIL PROTECTED] installed and we want to
interconnect it with our existing quintum gateways.. any idea how to config
that?
Your time is very much
appreciated..
Cheers,
Jessie
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