I have a Mitel SX-200 Light PBX that has a single incoming T1. I would like to insert Asterisk between the telco and the Mitel to handle a majority of the calls. My current hangup is that I can't seem to pull the right info out of the Mitel box to know how to configure the Asterisk box. From what I understand I need to find info on framing and encoding. So my question is twofold:

1. What information do I need?
2. How can I retrieve it?

I am a complete noob when it comes to Mitel, and would greatly appreciate any help.

Thanks,

Try a search in this forum using "Asterisk" as the search word. A lot of information in those threads.

If you are, indeed, a "noob" with Mitel--then you might have to pay someone to help you (train you) in the Mitel programming. It is not something that you can learn overnite. Good Luck.

Two forms you have to look in: Form 13's subform and form 42's subform

I'm a real noob too. Here is what I did.

Dial in to your PBX. (PC with a Modem & HyperTerminal with the phone number of your PBX).

Enter the username and password for your Mitel Box. (You may need to contact a Mitel Rep for this.)

Enter menu 13 & 15 to count how many T1 channels are being used.
(Esc+9 to review)
Doing this allowed me to discover a fractionalized T1 so I could setup the Channels correctly.
I.E. Bay 03 Slot 06 has 24 channels using E&M/DID/Loopstart But Bay 02 Slot 03 had only 6 channels using E&M/DID/Loopstart. (Telco connections)
Another T1 line goes over to another deparment. (It was labeled the comments section)

Enter menu 42 (Esc 8 to review/edit T1)
Here is the current Line Coding your PBX is using.
Record the Framing.
Doing this I found the 30 channels were using D4 with B8ZS
My remote Department is using D4 AMI&ZCS

I also discovered 'Incoming/Outgoing Start Type' of Wink - which is useful later in the zapata config file.


Here is my config for a 4 port A104D card:
********************************
ZAPTEL:
span=1,1,0,d4,b8zs
span=2,2,0,d4,b8zs
span=3,0,0,d4,b8zs
span=4,0,0,d4,b8zs

e&m=1-24 #24 channels
e&m=25-30 #Fractionalized T1 6 Channels
clear=31-48 #Clear the rest

e&m=49-72 #24 channels
e&m=73-78 #6 Channels
clear=79-96 #Clear the rest

loadzone = us
defaultzone = us

********************************
ZAPATA

[trunkgroups]

[channels]
;-----------------------------------
context=from-pstn

signalling=em_w

prewink=500

; Based upon our Mitel config
wink=200 ;wink timer
debounce=40 ;debounce timer
flash=300 ;flash timer

group=1
channel => 1-24, 25-30

;----------------------------------------
; Now for the MITEL PBX

context=to-mitel

signalling=em_w

prewink=500

; Based upon our Mitel config
wink=200 ;wink timer
debounce=40 ;debounce timer
flash=300 ;flash timer

group=2
channel => 49-72, 73-78

#include zapata_additional.conf

********************************
Here are some links for your research:

Your Question answered by Diguim:
https://kb.digium.com/entry/52/93/

A Zaptel PRI Explanation:
https://www.asteriskguru.com/tutorials/pri_zaptel.html

Sample Zaptel File:
https://asterisk.drunkcoder.com/zaptel.conf

Both Zapata and Zaptel Sample:
https://www.asterisk.org/doxygen/1.2/Config_zap.html

Oh, Lord, here we go again! Will someone PULEEEZE tell me what the Asterisk/Digium can do that a late release Mitel can't do. I know, I know, $$$. Remember, the Mitel doesn't have to be rebooted regularly, or shut down to install/replace periphial control/interface cards. And it has proprietary phones that have information displays to tell you what's going on and what you can do so you don't have to remember 2-5 digit command strings. (Command string in computerese = access code in PBX-talk.) And you don't have to have a full-time tech on site. "OH, OH, (Hand waving in air) I know, teacher, JOB PROTECTION!" "Right you are Johnny. Every IT person's dream, job protection." John C.

Hey JohnC,
Forget about that Asterisk. Let them go for it. Its not worth it. I understud that guys determined to have IP. Let them try. All big guys who implemented VOIP admited that unfortunatelly IP never going to work the way real PBX is working because no metter how smart you are you can not supersid phisics law. Latency, priorities, Network congession, echo that is the summary of VOIP. But this is trend right now and nothing we can do about it. I think it is not going to be forever that Internet connection for Voice would be free of charge. Eventually it would cost more then regular lend line.

VAD60: Yeah, I know. But, surely these guys have something better to do with their time! Let 'em create another OS that will run every program written for Windows, without lawsuits, And without crashing, then start working on an Asterisk type system! They obviously have too much time on their hands now! Then, when they're successful with those assignments, take a shot at VOIP. (Note that I was charitable and said 'when' not 'if'.) John C.