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BACKGROUND:
I am migrating away from our Mitel SX-200 EL/ML PBX. I want to make the transition to the new PBX as smooth and quiet as possible.

CURRENT:
PSTN--T1-OLD MITEL

THE PLAN:
PSTN--PRI--NEW PBX--T1--OLD MITEL

-Then migrate off 1 extension at a time. (Setup forwarding on the old Mitel to the new PBX as I convert.)
------
ISSUE:
I'm getting different answers to the same question and wanted some guru opinions.
(My Mitel guys are angry because I'm moving away from their product... They wont answer any of my questions - and they wonder why I'm leaving.)

INFO:
My current connection to my PBX is a T1.
VIA THE CARD:
T1 DS1 Trunk Card 9109-021-001-NA

LINK 01 (2x T1 DS1)
Bay 02 Slot 06
Bay 03 Slot 06
Line Coding = B8ZS
Framing = D4
(Can be changed to ESF)

LINK 02 (1x T1 DS1)
Bay 04 Slot 06
Line Coding = AMI&ZCS
(Can be AMI or B8ZS)
Framing = D4
(Can be changed to ESF)

QUESTIONS:
What is my ISDN based signaling protocol? Or maybe a menu where I can find it? (I.E. National ISDN 2, Nortel DMS100, Q.SIG?)

What kind of 'connection' are attached to these 'T1 lines from XO'? Are they fiber up to the Demarc and then turned into copper? Or are they fiber all the way up to the card. (9109-021-001-NA)

How can I find out how many outbound circuits we use on average?

Why do you think we are using different framing and coding?

Does the system support ‘auto detect’ for ISDN based signaling protocol?

Will need a crossover cable to attach my new PBX? (Or a special cable/pin outs?)

Thanks everyone in advance!

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At first, I do understand Mitel guys. In my opinion, Mitel is one of the best PBXs on the market. It is menu-driven, and you can program almost everything without looking in the book. If you compare it to NEC 2000 or Toshiba DK424, Mitel is piece of cake.

My feeling is that if you feel the need to convince your client to get rid of it, then something is wrong. Is it possible that you might not know what you are doing with the system? This is not my business, but I felt that I should ask.

(Original text continues from here):

Yes whatever comes to PRI ISDN Mitel is a tough one. But ICP rel.3.0 solves all conversions for DIDs.

Now I will try answer your questions.
Your PRI ISDN protokol is a ESF, B8ZS,N2or any other like, DMS100, 5ESS or 4ESS depends on provider.

To find out how many outbound channels your customer using in average I will suggest to have CAS.

Smart jack is considered as a last Mile Loop an as much as I,ve seen always is a copper.

Different Framing and coding used because PRI can not be channelise, no E&M or TIE connection and no Wink,Delay or Immediate start. It is always CCS. And the most convinien future is a DID instead of ANY or DNIS.

If I would do migration I will try, if its possible, to install new system next to my old MDF,punch down all my 25 pairs to new MDF and then after ours will do cutover. Befor that I will connect all T1s back to back and test them for proper operation.

What is "Auto Detect"?

Last question. What kind a new PBX are you talking about?

[Edited first paragraph for spelling/grammar that is visible in preview pane.]

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Thank for your quick response.

I'm glad Mitel has treated you well in regards to quality and service. I had best come back and say Mitel hasn't been the worst PBX or key system I've worked with. I understand they are a business and helping me with my goals to move away from them isn't always in their best interest. (I probley shouldn't expect them to help.)

(Moving on)

I'm not really interested in any other PBX - unless it is comparable to a Trixbox/Asterisk/Video system does for ~$2,000. (~$200 for IP Phones.)
Here is how Asterisk has spoiled me:
-Record All calls (mp3, ogg, wav, gsm)
-MeetMe (Secritary GUI to forward or tear down calls.)
-Bluetooth Proximity Call Forward activation/deactivation.
-SugarCRM (Task Management GUI for Help Desk calls.)
-Basic Telephony features Caller ID, Call Waiting, Conferencing, etc.
https://www.asterisk.org/support/features
-Audio Dictation of Email/weather/sports/news. https://nerdvittles.com/index.php?p=135
-Free/Cheap Soft phone/Video Enabled. (H.264 Video). https://www.xten.com/index.php?menu=download
- -Background music and overhead paging system
https://var.abptech.com/s.nl/it.A/id.1842/.f?sc=2&category=2098

Does anyone know how I can enable CAS in Mitel SX-200 EL/ML? Perhaps someone can direct me to a manual? Or a Firmware Version I can give?

Vad60: The migration plan you suggested sounds great, but I'm converting Digital Mitel phones over to VoIP. I will be using X-lite for most installs which means the PCs Network card will be used to carry my new traffic. In my case I don't think I can use that migration plan.

-For my migration I plan on using Asterisk/Trixbox as a man in the middle.
-The Dial-plan will forward all incoming and outgoing calls appropriately.
-I will provide clocking services to my Mitel as well as the proper coding and framing. Which I believe I can do successfully with the data above.
(Warn me if I'm missing something :-)
-My current plan: When I'm ready with a computer or Hard phone I will forward the Extension off my Mitel and direct it to the Asterisk. (There it will be forwarded to the network). I will change the dial plan on the Asterisk to forward calls for that extension to go the SIP device. (Or pass it on the PSTN).
(I'm open to suggestions if anyone can advise a better way.)

Auto Detect is a feature in Asterisk. It can detect which 'ISDN Protocol' and sync appropriately. I don't think my Mitel support this kind of feature. And I don't know if I trust 'auto'. For now I’ve manually specified my config. (span=1,1,0,D4,b8zs & Q.Sig)

So what does everyone think? I am open to suggestions.

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Unless your Mitel has a PRI card and Q-sig option enabled and link license, the only way to tie the 2 systems together would be T1 E&M Tie trunks. The card you listed does CAS. It appears that you have 3 T-1 circuits if fully populated would mean 72 trunks. Don't know if 1 PRI will be enough. Yes you will need a T-1 crossover.

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(More info: The Mitel does not have a PRI card. Only the T1 card mentioned. Thanks for pointing it out.)

Please forgive my arrogance: Can you tell me more about the differences between a PRI and a T-1 Circuit? I'm confused because it sounds like they are different yet they both use copper so maybe… They are the same but use different signaling? Or different demark equipment? (Google gave me a lot of results that didn’t have to do with Telco info).

If they are in-fact the same thing, just different signaling, can I unplug the connection from my Mitel and plug it into my Asterisk server (provided I have the same framing and coding)?
PSTN----2 Port PRI card---Asterisk---2 Port PRI card----Mitel
(https://www.voipsupply.com/product_info.php?products_id=1078)

If they aren’t the same thing what is the difference and what PCI card would I have to buy to accomplish a ‘man-in-the-middle’ Ear and Mouth Trunk?
PSTN----4 Port PRI card---Asterisk---4 Port ??? card----Mitel

(I already have a Sangoma A104D card.)

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Johnp meant that T1 has to be set in form 15 with E & M emulation. To set up PRI in Mitel ML/EL is a pain in the neck. You have to asign each and every PRI card as a separate ISDN Node.
I tryied to explane you what is the difference between T1 and T1 PRI. Righ now almost nobody uses T1 D4 except maybe if this is PTP or back to Back connection, which you probably would have to do to connect to your Asterisk.
Yes you are right there is no Autodetect on PBXs.
So you are trying to move all your traffic to VOIP and Asterisk is your gatway between Mitel and Host. Good luck. If this is privet IP Network thats OK otherwise you might bring lots of problems to yourself.

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Brief semi-hijack:

Quote
Originally posted by WalkerBoh:
Can you tell me more about the differences between a PRI and a T-1 Circuit? I'm confused because it sounds like they are different yet they both use copper so maybe… They are the same but use different signaling? Or different demark equipment?
WalkerBoh,
In keeping my response in simple terms, here's my explanation of the difference between a "T1" and a "PRI":

A PRI [Primary Rate Interface] is simply a switched service ISDN protocol delivered over a T1 circuit.

If someone were to say “I want to order a PRI to location X.”, what is being requested is a connection to a trunk side module capable of ISDN protocol to a Telco switch delivered to location X via a DS1 rate (T1) circuit.

Now if that same someone were to say “I want to order a T1 to that SAME location X.” That wouldn’t be enough info. A T1 from where to where? A T1 for what? By just asking for a T1, nothing is understood or implied about where or what the circuit would be used for.

1. A T1 can be used to truck data at the rate of 1.544 Mbps from one location to another.

2. In using channel bank systems and appropriate DS0-level cards (dual-sp-cu, fxs, fxo, 2T0, 4T0, etc.), a T1 can be used to truck up to 24 separate and distinct DS0 signals (analog data, analog voice, or digital data service) from one location to another. When voice channels are applied in this format with channel banks, this is commonly referred to as a "Channelized T1."

3. A T1 can be used to connect one location to an ISP Internet Edge device to connect a customer to the World Wide Web.

4. A "point-to-point" T1 can be used use for voice and/or data depending on the equipment you have at each end. Your provider is simply providing the "pipe" and it is up to you what it is used for.

In other words:

A T1 is simply a multi-use pipe, and the features..or lack thereof..directly depend on where the circuit is going and how it's being terminated at either end of it's path.

A PRI is a protocol that uses the 24th channel [D-Channel] for signalling and control of what the other 23 channels [B-Channels] are used for. And because this protocol is being delivered via a T1 circuit, it is referred to as a T1-PRI, or simply "PRI".

There are a lot more technical details dealing with ISDN Lines and Trunks (BRIs/PRIs respectively) National ISDN vs Custom ISDN, FAS vs NFAS, etc., but you've got the basic idea.

I hope that my explanation was able to assist your understanding the difference between the terms T1 and PRI. If not, please feel free to open a new topic of discussion over in our T1s, CSUs, and DSUs forum.

[My apoligies for the "semi-hijack", please continue the ongoing discussion..]

welcome

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5Etek-mike,
Thanks you very much for clearing that up. You confirmed what I already thought, I've been confused a lot by my telco. I'll make sure to find another tech for confirmation.

(Moving on)

Here are some of my notes. Feel free to double check and add suggestions:

MIGRATION:

STEP 1)
Contact Telco and move over one of the 'last mile' smart jacks from Mitel to Asterisk PRI.
-Configured Asterisk interface
span=1,1,0,esf,b8zs
span=2,1,0,esf,b8zs
bchan=1-23
dchan=24
bchan=25-47
dchan=48
switchtype=national
-Coordinate/Troubleshoot connection with Telco.
-Work out Red Alarms
-Telco supports Yellow Alarms

STEP 2) Make a T-1 Crossover Cable
1 — 4
2 — 5
4 — 1
5 — 2
(Someone verify the pin-outs for me)

STEP 3)
Enable E&M Emulation on one of my Mitel PBX trunks/T1.
-Connect Crossover to Asterisk.
-Make sure Asterisk is passing proper Coding and Framing.
-Do I need to providing time? Or is that only with PRI?
Clock mode = Master
span=3,1,0,sf,b8zs
span=4,1,0,sf,ami
switchtype=qsig (Might not be needed)
-Troubleshoot line as needed with Asterisk/Sangoma IRC channel and you guys. :-)

STEP 4)
As lines work Repeat 1-3 for another trunk.
-Two PRI interfaces from Telco to Asterisk.
-Two Mitel Crossover connections.
-Checking all the while for excessive busy signals.

STEP 5)
Setup a Basic forwarding dial plan(s)
-Asterisk forwards incoming PSTN calls to Mitel PBX. For every DID we have.
-Asterisk forwards all outgoing PSTN calls to Telco. For every DID we have.
-Add, strip, or forward 9-to-dial-out
-Troubleshoot Mitel/Asterisk forwarding as necessary.
(Note: All phones still on Mitel)

STEP 6)
VoIP Dail Plan.
-Extension 1234: Converts to VoIP.
-Redirect Asterisk incoming DID PSTN requests for that number to network phone.
-Redirect Mitel calls of 1234 to forward to pseudo 9-4321 (Which is directed outbound to Asterisk. I’ve done this for Cellphone forwarding before.)
-Asterisk picks up 4321 calls (received from Mitel) and directs it to Network Phone.
-Asterisk already programmed to forward 1234’s calls to Mitel via last script
-Asterisk already passes 1234 outbound if 9 is added.
-Notify users that they will receive new Voice Mail box.


More Questions Regarding this Migration:
-Do I need to provide timing to the Mitel?
-How do I setup E&M emulation? (Form 15 doesn’t appear to be the right screen.)
-Do I really need to setup E&M emulation if I’m just going to pretend to be the PSTN (as far as my Mitel is concerned).

Below is a list of my Mitel menu options:
CDE FORMS
01 = SYSTEM CONFIGURATION
02 = FEATURE ACCESS CODES
03 = COS DEFINE
04 = SYSTEM OPTIONS/SYSTEM TIMERS
05 = TENANT INTERCONNECTION TABLE
06 = TENANT NIGHT SWITCHING CONTROL
07 = CONSOLE ASSIGNMENTS
08 = ATTENDANT LDN ASSIGNMENTS
09 = DESKTOP DEVICE ASSIGNMENTS
10 = PICKUP GROUPS
11 = DATA CIRCUIT DESCRIPTOR
12 = DATA ASSIGNMENT
13 = TRUNK CIRCUIT DESCRIPTORS
14 = NON-DIAL-IN TRUNKS
15 = DIAL-IN TRUNKS
16 = TRUNK GROUPS
17 = HUNT GROUPS
18 = MISCELLANEOUS SYSTEM PORTS
19 = CALL REROUTING TABLE
20 = ARS: COR GROUP DEFINITION
21 = ARS: DAY ZONE DEFINITION
22 = ARS: MODIFIED DIGIT TABLE
23 = ARS: ROUTE DEFINITION
24 = ARS: ROUTE LISTS
25 = ARS: ROUTE PLANS
26 = ARS: DIGIT STRINGS
27 = ARS: MAXIMUM DIALED DIGITS
28 = FORM ACCESS RESTRICTION DEF
21 = ARS: DAY ZONE DEFINITION
22 = ARS: MODIFIED DIGIT TABLE
23 = ARS: ROUTE DEFINITION
24 = ARS: ROUTE LISTS
25 = ARS: ROUTE PLANS
26 = ARS: DIGIT STRINGS
27 = ARS: MAXIMUM DIALED DIGITS
28 = FORM ACCESS RESTRICTION DEF
29 = DTE PROFILE
30 = DEVICE INTERCONNECTION TABLE
31 = SYSTEM ABBREVIATED DIAL ENTRY
32 = CDE DATA PRINT
33 = ACCOUNT CODE ENTRY
34 = DIRECTED IO
35 = GLOBAL FIND ACCESS CODE
36 = MODEM ASSIGNMENTS
37 = GUEST RM SUPERSET KEYS TEMPLATE
38 = ACD KEYS TEMPLATE
39 = ACD AGENT GROUPS
40 = ACD SUPERVISORS
41 = ACD PATHS
42 = T1 LINK DESCRIPTORS
43 = T1 LINK ASSIGNMENTS
44 = NETWORK SYNCHRONIZATION
45 = BRI TRUNKS ASSIGNMENTS
46 = KEY SYSTEM TOLL CONTROL
47 = IP AND DHCP PARAMETERS
48 = IP NODES

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Form 15 is a Dial - in - Trunk. E$M emulation supposed to match numbering plan meaning that provider must send you number of digits that the same as your extensions numbers or at least last one has to match last digit in your extensions numbering plan. In Form 13 set up Discriptor for T1 E$M and do apropriate set up.

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If the T1 is currently connected to your Mitel SX200 Digital, connecting it to your Asterisk will not make it a PRI.
Why would you configure the input to your Asterisk as a PRI?


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Man, I got a HEADACHE from this thread, gotta quit checking it!
3 questions;
1 How often do you reboot your Mitel system?
2 How often do you reboot your computer?
3 I assume ALL your certifications are computer hardware/software related. Never been to any phone system factory schools, have you?
How many of you phone techs have had to clean up the low $ techs' mess and then caught grief when you invoiced the customer, esp when the company insists on all certified computer techs and even HAS ONE ON THE PAYROLL?
But they can get by without their computer network for several days, but can't have their phones down for more than a couple of hours!
Who me, bitter? You betcha! John C.


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OOPs, I think I may have made WalkerBoh mad! Or maybe it as fifty5footer. Or maybe, just maybe, WalkerBoh figured out that he was going to either drop the whole idea, or pay the Mitel guys. OR his boss figured out that a LOT of $ was about to go down a rathole! John C.


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[Linked Image from e.deviantart.com]LOL--John--you're too much.....

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Thank you very much! (With an Elvis Presley accent) smile John C.


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Vad60 thanks for pointing that out, it took me a few days to get around to it but I carefully checked Form 15, and 13.

UPDATE:
Bay 04 Slot 06 has AMI&ZCS for coding because it connects to another remote Mitel PBX offsite. (This answers one of my earlier questions.) Additionally, it is currently setup to T1 E&M.

Bay 02 Slot 06 and Bay 03 Slot 06 are both bonded under a T1 DID/LOOP-TIE only 6 channels allocated on the first T1, and 24 on the other. (30 total) (I assume the 6 channels T1 was added to support more calls.)

I apologize in advance for being a newbie but could you please explain why I need to change the T1 DID to E&M emulation? Do you think it’s possible for me to simply 'pretend' to be the Telco on my Asterisk server - and leave the settings alone?

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Hi,
I am not saying that you have to change configuration of the T1 trunks. I was trying to tell you that in order to reach an extension from outside, your numbering plane has to match digits coming from Telco to the swith if you are T1 E&M or has to have at least two last numbers in sending digits string to match your last two digits extension's number to be able to use T1/DID discriptor with proper conversion set up.

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:sleep:

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Shame on you WT! John C.


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And, after 1 1/2 weeks, WalkerBoh has probably figured out that he's not really doing a smart thing. John C.


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Thanks Vad,

I'll take extra care planning out my dial-plan.

Any other helpful tips?

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Guess not! John C.


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Thanks everyone for your help.

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you guys are a awesome! Thanks for giving me a good laugh. I'm just curious how the project turned out?

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Gotta admit, you've gotten my curiosity up. How about it WalkerBoh? John C. (Not Garand)


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WalkerBoh,

I know this is an old thread, but I'm very interested in knowing how this project turned out and if you could post some of the things you learned?

I'm facing the same process myself and these phone system types are a real 'ole boys club', especially with regards to dealing with computer/networking types.

Fact is that these days we're forced to do lots of things outside our job description in the name of cost savings for the company, and keeping our jobs.

Anyway, I'm very interested in this project and it's outcome.

Presently I'm working on putting an Asterisk box along with an Astribank in between all our B1 lines. It will silently intercept all long distance calls and route them VoIP while sending the local calls out the PSTN. We hope to use some of the savings from this project down the road to migrate the whole system to Asterisk.

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KLGIT,

Thanks for stopping by to insult us. I hope your migration goes well.

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KLGIT, Welcome to the forum.

" Man, I got a HEADACHE from this thread, gotta quit checking it!
3 questions;
1 How often do you reboot your Mitel system?
2 How often do you reboot your computer?
3 I assume ALL your certifications are computer hardware/software related. Never been to any phone system factory schools, have you?"

Sorry, but the questions are just as valid today as they were almost TWO years ago.

What brand of system are you migrating from? How old is it? Do you have any complaints other than VOIP incompatibility? Have you calculated the TRUE cost of converting to VOIP for LD? (You might want to PM Kumba, or post in the VOIP forum.) And the Feds ARE going to figure out that Voip 'lines' are a NEW source of revenue (TAXES), and then the whole exercise will be a waste of time.

And you might want to investigate the VOIP capabilities of your present system, no matter the brand. The latest and greatest Mitel is Voip capable, with automatic routing of calls. Of course there are other brands that have the same capabilities.

And you might consider lightening up on the
"I'm facing the same process myself and these phone system types are a real 'ole boys club', especially with regards to dealing with computer/networking types." if you really want some assistance.
Later Gator, John C.


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No insult intended.
I was mostly referring to the telco's and hardware vendors. Tried getting info from your telco on their system? My telco outright told me they wouldn't help us at all and they wouldn't support us anymore either. How's that for service.

And you have to admit, there's a lot of "serves you IT guys right for trying to do a telco guys job" type comments on various Internet forums. It's generally pretty unfair since most of the IT guys bumbling through this telco stuff had these projects dumped on them by their company. "Hey phones have plugs and wires, so this must be your job"
What are we supposed to do in these troubled economic times, refuse?

Anyway, the comment was not intended for those here. I mistakenly assumed that was obvious since you wouldn't be here in a help forum if you were an unhelpful old boys club member. Right?

;')

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I think you will find that most of my colleges who contribute to this board will go out of their way in providing assistance. That being said, John C has some valid points. Why are you considering replacing a system that has been developed of many years, having a proven track record and available manufacturer support with one that doesn't provide any of these?

The SX200 is fairly easy to program and maintain. There are also programs that enable migration to the 3300.

What you are trying to do with using Asterisk as a gateway, is likely fairly straight forward. You could use the routing inside the 200 to choose to send the calls to the asterisk box or dial out a local attached trunk. Without knowing your current configuration, I can't say what my best approach would be. I do think you will need some type of T1 E&M conectivity between the 2 systems.

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Old-boys club? The are very knowledgeable and helpful girls (ladies) here also…

These techs here have been extremely helpful to WalkerBoh… John points out the obvious and asks questions that are important to the project. Knowing what you have and what you are trying to achieve before you plan an upgrade is also critical… 2 weeks into the post, quote from walker “UPDATE:Bay 04 Slot 06 has AMI&ZCS for coding because it connects to another remote Mitel PBX offsite.” If they want to replace their system, that’s their choice. As you can see he went to Mitel, who has always been extremely helpful to me for the last 21 years, and received no assistance, then he came here to get a lot of advice. I think everyone would like to know how the project turned out, hopefully well, but he never came back to tell us.


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Well, just to be clear. I sincerely apologize to anyone that was offended.
Months of lurking before joining has made me well aware of the amount and quality of help available here.

Thanks.

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OK, we STILL need to know the Make and model of system(s) you are trying to integrate, and why, if you think it may be of help. smile John C.


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Let me jump in here and mediate

KLGIT you have to understand that a lot of us old "phone fogeys" have spent countless hours(and I do mean so many you can't count them - approaching bailout bill status) being derided by IT types. We have untold hours being patient with "computer whizzes" at PMS (Property Management Systems) companies trying all manner of things they suggest because their egos will not let them listen to a lowly "phone tech" (who knew how to solve the problem in five minutes). Most of the phone guys on here have a knowledge of computers and networking that is well above average - most of it learned by the seat of their pants - asking questions, listening and observing - and then by trial and error. Trial and error because we have had to figure out ways to make things work when the computer whizzes would just sit and scratch their heads.

We have also watched as customers (especially hotels) have had priests come in to bless their computer systems that are encased in environmentally sterile glass boxes) and then still pay thousands of dollars for someone to fix it while we walk into phone rooms to see lamps, tables, and all other kinds of detritus piled so close to the phone system that there is no air circulation in a room that is hotter than Iraq in August with the customer leaning over our shoulder saying, "I can't believe you guys can't keep this thing running - it's just a phone system."

In other words even though most phone systems these days are very sophisticated us "phone techs" are treated like GWB by Hollywood and the media."

That being said the people here (probably just like you) think it's "cool' to figure things out and get them working - especially when the manufacturer tells you can't do it" and they like to share that knowledge. (I hope that you will do the same on the networking and computer side of this forum).

I have found people on the Sundance Forums to be incredibly patient with any questions I have had - even ones that turned out to be very basic. I have had people be incredibly generous with their time - even offering to take a system home with them so they could upload a file and print it out for me when I was in a bind. And whenever I have figured something out I have tried to make sure I come back and share that information with everyone else.

So FLGIT I am glad you are here and keep asking question if you have them, but cut us "good ole boys" some slack if we get a little testy sometimes. It might be that we are a little "backed up" by the Opossum stew and collard greens we had the night before and well be in better humor when we get "reg'lar" again.

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"OK, we STILL need to know the Make and model of system(s) you are trying to integrate, and why, if you think it may be of help. John C."

Sorry to be repititious but...... John C.


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Yeah - sorry I got sidetracked

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Hey, sometimes the 'side-track' is the only way to 'get there'. smile John C.


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Dane,

As a "phone tech" for the past 40 years, I can say you hit the nail on the head. I just came from a client that is a 250 room hotel with an IBM PMS running Opera and a Mitel SX-200 EL 6 bay system. The IBM system is in a really cool looking stainless steel and glass enclosure, located in one of the most climate controlled rooms I have ever seen with absolutely sterile conditions. They pay a computer whiz more than I make a year to "resolve" their problems with that system.

The lowly Mitel is located in the basement, directly adjacent to an old elevator control room where, occasionally, one can hear arcing and relay chatter. The "phone room" is also the Video on Demand room (the big blue box containing the porn), the cable TV amplifier and distribution room, the cable company's "head-end" with two racks of channel modulators, the whiz-bang "DSL" to every room internet solution, the graveyard for all things broken....AND....all this is contained in a 10' by 10' room. The heat in that room, even with an air conditioner running 24/7 is enough to cause heat stroke in a human.

The IBM is constantly "down." Waiting several days for a fix is "OK." Interfacing with the Mitel is a series of interface boxes and "devices" so the IBM can accept data from the Mitel.

The Mitel is on a UPS. The maintenance people like to use the UPS to run their saws, drills and sewer cleaning equipment. It is absolutely unacceptable for the Mitel to be down for more than a minute. Does ANYONE know what a circular saw does to the electrical circuits?

Between having to figure out how to keep the metal cabinets of the Mitel from melting and peeling layers of old metal lamps, bare electrical wire surplus and endless mop heads, broken broom handles, etc. from off the MDF, I probably spend 20 minutes a week on the Mitel. It just keeps running.

My point is that there is a "class distinction" between the pampered and revered IT people and telephone crafts people which is totally ridiculous. The fact is all modern telephone systems are single purpose computer systems. The people that complain that telco people are less than forthcoming with "information" should look to their own house. Have any of you ever tried to get a port opened up on a client's network from the IT person? I'd rather hang 1,000' of bridle wire and install 500 42a blocks that have to deal with these prima donnas. How about getting information on passwords on a network? Yet, they are "offended" when we hesitate to allow them installer access into the equipment that we must maintain and keep working "no matter what."

I get it that companies today are cutting back. The interesting thing is that the most critical equipment of ANY company is the phone system and most companies treat the phone equipment, which is probably the most reliable of the office equipment, worse than the garbage.

I propose a solution. Open a forum topic and moderate it with experienced network IT types and allow the telco people to ask all kinds of questions like the IT types ask of us. What do you think will happen?

Rcaman


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Rcaman clap

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Appluase, Applause!!!!! That said, your Great idea probably would flop. Us phone types are so used to having to dig it out for ourselves, and/or getting a ration of ---- from the IT folks, we'd ignore the forum topic. But, it's STILL a good idea! frown John C.


When I was young, I was Liberal. As I aged and wised up, I became Conservative. Now that I'm old, I have settled on Curmudgeon.
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