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The only way I've ever been able to do this is via the Contact Center Client which is part of MiCC Business. It's super clunky, and the UI is still using an Office 2007 theme.

That's going to depend on a few factors, like if you have MiCC with IVR licensing; or if you're trying to set up the IVR in voice mail, that will depend on which voice mail platform you have.

I think that's the 3300 trying to reach out to OpsManager for some reason. OpsManager is an old network & user management system that's been discontinued for quite some time, but may still be around on some older installations.

Pretty much any PBX should be able to do this. On the NEC, if you have a spare analog station and trunk port, you could route the ringdown to the analog station port, then loop that right back to an analog trunk port to get around your answer limitation.

Make sure your MiCollab Feature Profile includes the Mobile SIP Softphone, make sure the mobile softphone extension is added to the MDUG, and sync with the PBX from the Synchronization tab in MiCollab Client Service

Do the devices you're trying to hand off between each have their own unique extension in the MDUG? A soft phone extension can only be active on one device at a time, so if you want to be able to hand off between different MiCollab clients, each device will need their own unique soft phone...

In YSE, for the Queue settings, in the "Recorded Announcements" tab, what's set for "Embedded Music Source" for each RAD? Should be 0 to use the system MOH, or another selection for a different embedded MOH file. Also check the Zone ID of the Queue, but you might have to get into the MiVB ESM...

You have trunks out of service. You need to either restore the trunks or deprogram them to clear the alarm.

Sounds like either a codec issue or something is disrupting the RTP stream. What does a packet capture tell you?

You sure they're logged out? You can issue Maintenance Command "busy reset [dn]" against the hot desk extension, which will log the extension out.

Might need to grab a packet capture and see what's going on then.

In the SIP peer profile, try changing the field "AVP Only Peer" to "Yes". This will force the connection to use RTP instead of SRTP for the audio stream, which can cause issues.

What system and software release?

Maybe run an SMDR log search against 2222, that may help track down the source.

locate number" should come back with something if the number exists. "dgt trace [number]" may also tell you something.

Not possible. All the features you listed are licensed per user so you'd have to buy at least UCC Entry licenses, I believe.

The pullstring is there for a reason. Use the cutter/stripper to get to the pullstring, then use that to strip back the jacket, and use electrician's scissors to cut the jacket.

I'm not aware of any actual standards, but nearly every building I've ever seen that still has old analog/digital voice cabling and data cabling usually uses white/beige cabling for voice, and blue for data, with putty jacks (or whatever color matches the faceplate) for voice and orange for...

Good luck. Are you doing this yourself or do you have help? Can't offer much advice on identifying anything without knowing what the existing endpoints are connected to, but hopefully you can call something from each ext that logs the call so you can identify the extension. Maybe not in real...

Sounds like a codec mismatch