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Joined: Feb 2010
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Had a customer call me to say that no room phones ring when called. Every thing else appears to be working fine according to the GM.

I have no spare 200 Analog parts to take with me but I was told there might be a fuse or fuses on the interconnect board that controls the ring generator.

Any thoughts? I am trying to hunt down my 200 Analog manuals but in the mean time any assistance is appreciated.


Jon


"It is better to keep your mouth shut and appear stupid than open it and remove all doubt"

Mark Twain
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There is, I think it's F1, but the rear door power supply has no fuses on the system ring generator, just on the nite bells ring generator. It fails every now and then.

Carl


This model is end of life
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Sounds like you lost the ring generator on you back door power supply. I know EXACTLY where you can get a replacement!!!!


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But bring a strong back, darn things are 90+ pounds! I've never had the R/G fail, it's always been 1 of the 'little' voltages. I'm guessing that door has been repaired before.
The fuse on the interconnect board only affects the printer port and console.


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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Does anyone have a list of the voltages and test points since I still can't find my manuals?

I might mention that the backplane has been failing on this unit or at least no cards work in the right half of the top shelf anymore.


Jon


"It is better to keep your mouth shut and appear stupid than open it and remove all doubt"

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On the top left, and bottom left corner of each shelf, the screw terminals/test points are marked with the proper voltages. All of the voltages are DC, except the 90V, which is AC. I always use the -48 terminal as my neutral, when testing the AC ringing voltage.


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"no cards work in the right half of the top shelf anymore."
------------------------------------------------------------------------------------------------------
Probably not the problem, but the earlier cabinets had 2 fuses on the backplane of each shelf for the 48 VDC feed. One for slots 1-6 and one for slots 7-12. They were there to comply with some early NEC rule, which as it turned out, didn't apply! Open the back door and look along the bottom edge of the PCB on the back of the card cage. If there are fuses, they can be bypassed, or replaced. Yes, it was a MITEL approved field modification. Later production just had jumpers where the fuse clips used to be. I just soldered an 18 ga. wire to the fuse clips. Be quick because if you melt the solder where the clip is soldered into the board, you'll do more damage then good.


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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Customer wants to "bandaid" the unit until they have the money to upgrade (going on 3+ years now) so I will be changing out the R/G for them. All other voltages are correct and I found the faulty fuse to fix the shelf issue.

Thanks everyone for their assistance.


Jon


"It is better to keep your mouth shut and appear stupid than open it and remove all doubt"

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"I love it when a plan comes together." Better yet, when someone comes back with + results and appreciation. Of course, Jon is a tech, and we techs are nice guys! Right,Jon? smile


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Hello,

I have an issue with Mitel 3300 and Juniper SSG20
Plssssssssssss help.


There are two sites (Site A and B) that are connected by SSG 20 on each side via site-site VPN. We have SIP links coming into each end from the SIP provider which is NAT'ed to the Mitel controller. I can call extensions inernally across the sites and vice versa. However I'm having some issues:

1. Call comes in on site A, gets call attendant, request extension on the site B.
If someone answers there is no audio either direction.
If no one answers, caller just has dead air until the call dies.

2. A call comes in to the site A and is answered locally, then needs to be transferred to site B.
If done via conference the call is connected and communications works until the site A user hangs up, then both other users get dropped.
If done via transfer, the incoming caller gets dead air and the remote caller does get the ring but no audio

I have tried enabling SIp, etc. Pls help me to fix this issue.

Thanks
Sharu

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