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CUCM itself is a virtualised/software platform and doesn't have ISDN as such so the Cisco side of the PRI is likely terminating on a Cisco ISR or similar device. Are you able to run an ISDN trace on this device? It's entirely possible that there are number translations occurring there...

These sets do the same thing on Aura too and it's quite frustrating: https://www.tek-tips.com/threads/j179-display-question.1828747/post-7565401 I never found a fix for it but would love to know if the behaviour can be changed.

You will need a license for an Avaya IP Endpoint to use this phone on your IPO.

UM8000 has nothing to do with InMail.

I wrote a script that uses 'sox' which is preinstalled on the Server Edition/Application Server.

The J179 will run just fine on earlier versions in H323 mode and will identify themselves as a 9611. It will have the older interface and won't support the wifi/bluetooth module but otherwise it's a good intermediary step before upgrading.

Why don't you just buy 3x MOH players, load the audio file onto them, and plug them into the MOH port on your 3x IPOs?

Not to mention the microwaves. If you have a customer site that has voice/wifi issues around lunchtime every day that's a likely culprit!

I concur. I do this at home for some J179 sets I can't easily get cabling to but wouldn't roll them out in a commercial environment unless they were occasional phones (kitchen/lobby/etc). Even though I have a commercial grade Cisco WiFi network at home they're still lacking and will...

The 18D is a Partner set. You might have more luck posting in the legacy SMB forum which includes Partner.

Not to mention new handset firmware now requiring a PLDS login (something I always appreciated with Avaya was them making set firmware freely available). The mention of license files to unlock "enhanced features" in the latest J100 release notes is also a bit concerning. I wouldn't be...

They used to have the H200 series but I think they were discontinued recently.

You could just get a generic ATA and third party SIP licenses. I use Grandstream and Cisco ATAs on my IP Office.

It's definitely not a supported configuration but I can confirm that if you extract the older TTS RPM files using rpm2cpio and put the files on the file system (they live under /opt/Loquendo/LTTS7) it works just fine.

MADN came out in 10.1 so you won't be able to use that as you upgrade but this feature would work. On 9.1 your options are a bit more limited. A group will flash on both the appearance and group button in any case unless the user is logged out in which case only the group button will flash...

I've been doing the same for a couple of years. It works just fine. I have the IPO, printers, etc. all relaying through Google that way. Microsoft have a similar feature on Office365 that I used before switching over to Google Workspace...

The configuration steps in the documentation are not in order. Make sure you've configured your WebLM server before you start trying to add config.

Same as any other extension - using your incoming call routes. Send the DID to the extension associated with your phone port.

Make sure your fax transport support is set correctly on your SIP line under VoIP (I use T38 Fallback). Some SIP trunking providers don't support T38 so you might need to confirm with your provider. You'll also need an analogue phone port in the 500V2, and that should be set to be a fax machine:

The label on the back of the phone should say what variant it is - eg: 1603DxxA. The SIP firmware only works on 1603D03A.