Randomely Ends Voicemail

[Tyl0r]

Hey Everyone, We have a few IP403 units on version 3.2 (53)and a voicemail PC running Voicemail Pro 3.2.

If you call from a cell phone or any incoming call, everything will work *most* of the time, but about 25% of the time you will get to a users voicemail and it will give their greeting, then you'll hear the *beep* to leave a message and you start to talk and then immediately it will send you (the caller) to the voicemail help options menu and cut you off from speaking.

Also rarely but it does happen, you can listen to our Auto Attendant greeting at the very beginning and it will cut you off into complete silence and nothing happens after like 5 seconds of listening to it. Any idea's why this is happening?

I have the monitor log from when it happened but I can't really see what the problem is. It's a lot of data too so I didn't want to post like 15 pages of it if someone has already heard of this problem before. Thanks!

 

[amriddle01]

You may well just be getting noise on the line which the system thinks is DTMF tones and it starts routing according to what it thinks it has detected. Do you ever hear noise on your lines at all?

ACS - IP Office Implement

"I'm just off to Hartlepool to buy some exploding trousers

 

[Tyl0r]

Actually sometimes we kinda do. I've heard people say on some occasions it sounds like a demonic exorcist voice has taken over their sound quality while on the phone LOL.

Do you have an idea what would cause this?! We have a MultiTech FaxFinder fax server, but it's plugged into it's own 4 seperate analog ports, 2 on an Avaya IP400 phone expansion unit and the other 2 on the IP403. but since you mention the DTMF thing... hmm

 

[amriddle01]

It'll more than likely be the incoming lines, do you have analogue or PRI lines? If PRI try and call in/out on each channel and see if you find a particular one at fault, if analogue try a POT handset plugged into the lines instead of the system and see if you hear noise coming from them.

ACS - IP Office Implement

"I'm just off to Hartlepool to buy some exploding trousers

 

[Tyl0r]

OK thx! So it's basically PRI there's a T1 line. We dedicate 15 channels of it for voice. What is the easiest way to test each channel?

Just curious, how do you check the Codec being used and the peak hours of calls? I'm curious to see the bandwidth that's required for the codec but I can't find what Codec we actually use in the Avaya Manager. Also when our peak call hours actually are or like some reporting data about peak usage.

Thanks! :]

 

[amriddle01]

For the channels you can look at the Line settings in Manager and each of the channels will have a line ID agianst it, you can then give yourself a line appearence key for that ID and when pressed will seize that particular channel, so go through each channel and see if you have a noisy one. As for peak hour calls etc there is nothing in the system that will tell you that I'm afraid.

ACS - IP Office Implement

"I'm just off to Hartlepool to buy some exploding trousers

 

[Tyl0r]

Okay awesome thanks! Do you happen to have any idea where you can find what codec our system is using?

 

[amriddle01]

If it's PRI you are using ISDN codecs which are set at 64 kbps so you can't change that, these are not to be confused with IP trunks and IP handsets that use H323 or SIP and on those you can change the codecs used

ACS - IP Office Implement

"I'm just off to Hartlepool to buy some exploding trousers