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It looks like everything is correct on the ds1 form. Is there something special I should be looking for in a DS1 cable? When I look online there are a multitude of options and I don't know which one I should be using.Thank you!

I have CM 5.2, the AdTran is connected directly to the demarc and was used to split channels off for ISDN voice before we had the gateway. Where do you see the CSU on the form. The only field I have that comes close to this is Near-end CSU type. It is currently set to other but when I go to edit...

My current G450 gateway has the PRI line connecting to it via an adtran device. It was left over from when there was an IP Office at this location and never was removed when the gateway was installed. I am trying to remove this out of the configuration now but I can't seem to get the PRI to come...

I am in Grand Rapids, MI. I tried pasting the screen shots but it won't allow it. I'm not sure how to make the traces visible to you?

I have run traces on the extensions and the trunks and all it says is recovery w/timer expiry. I can paste the screen shots of the tac trace if that would help.

The 3rd post from the top gives more detail on the situation and how this is working/not working. The issue appears to be when these 3 800 #'s try to route to MM after coming in on my LD trunk. The other 800 #'s all route over my local PRI which is where all of my lines are supposed to point...

4799 dialed internally goes to my receptionist because it isn't defined in the PBX. Dialed externally it just rings/no answer. Why would this only affect these 3 numbers if it needs to be defined in the system?

No it isn't. This extension doesn't actually exist in my PBX so it must be a routing thing that they did when they installed MM. 4700 is the number to reach VM but 457-4799 is defined in the VM hunt group. So a call routes to the hunt group which dials the group extension of 4700 and then it...

I would be hesitant to make this change because everything else is working fine. I don't know enough about SIP to understand what the disconnect is for these 3 numbers.

That is apparently what is happening. I tested this from a working number and a non-working number and they send the same digits.

The number that actually is called is 306-457-4799 although the SIP trace doesn't reflect that a simple extension trace shows all the number being used. The 306 drops and then the 457-4799 is routed through aar which strips off the first 3 digits and sends 4799, the number of the MM mailbox.

Yes 4799 is the extension for MM. No, that IP address is the server for the MAS.

Thank you for responding. We are at MM 5.2 SP12. It is a direct connection to CM, no SES involved. I have 2 other 800 numbers that also come in on this trunk but they route through a call center and I don't know if they ever reach a vm box. I do know the callers to those numbers would be...

Alrighty then...finally figured out where and how to get this information. I am attaching the trace for one of the extensions that is having a problem. Hope this can tell you what is going wrong because I couldn't see anything. Of course I am not surprised by that as this is beyond my...

Okay...I got that going but now where will I find the data?

I am trying to figure out how to make this happen but so far I'm still looking. I understand what you are getting at but this messaging system is still so new to me I haven't had much of a chance to learn how it works. As soon as I can make this happen I will post the results. Looking at your...

Below is the trace I ran on the mailbox. I tried to keep it small so I don't overwhelm this post. This is what it does every time someone tries to leave a message. When Object Event Type Call-Id Message Application Server Information Port Mailbox 10/22/2012 23:59 "TUI" 21217...

These lines have worked fine for over 6 years but now callers to these 3 800 numbers are actually routed to the voicemail system but the message starts playing and then they get a busy signal and disconnected. Only one of the numbers is routed to a DID and you can call that number directly from...

We changed from a MM 5.2 TDM connection to a SIP connection this summer and I don't know if this issue is related or not but it seems to have started around the same time. I have 9 800 #'s of which 6 work properly. The 3 that don't work do get routed to MM but after 4 seconds of hearing the...