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The phones cache the security code on the station form. That's the password they login with. If that's not working, maybe there's a mismatch on the main CM and the ESS/LSP. If you list survivable, do you see translations updated as recently as yesterday? Maybe you've got expired certificates...

https://www.startech.com/en-ca/faq/change-com-port-number-windows Maybe that helps?

that might be a bug depending what release you're at. SMGR certs have auto-renewed since 7.1 I think. Just needs a reboot to kick in. Maybe something about after renewing the CA made it do that

how close are they to expiring? they should autorenew. SMGR usually needs a reboot to pick it up but SM I've never really had to do.

How are your network regions setup and sip domains setup? Network map should have subnets pointing to a network region. CM procr should be region 250. Subnets in other regions should be directly connected to procr. The authoritative domain of the region should be the root sip domain as should...

Look at the subscribes and notifys in a trace and check the registration details for what feature packages are subscribed to. Get all endpoint configuration all in PPM gives you the button, but the subscription to the feature package is what gets the SIP notify from CM to the phone to tell it...

You can GET $MACADDR in the settings file and have something like SET FORCE_SIP_USERNAME and SET FORCE_SIP_PASSWORD in that named for the MAC address. Users swapping desks wouldn't need to move their phones and the change process would be just updating the MAC txt file. That presupposes...

Special Application 8096 does what you want. You should be able to enable it with "change system special". It adds fields in the ISDN trunk group form to overwrite name and/or number going out to the PSTN.

https://sourceforge.net/projects/avayacdr/ Maybe something like this can help? Otherwise, CDR solutions aren't that expensive

Thanks for the heads up!

Is that for Oceana only or snap-ins that depend on it, or are you talking about every Presence server too? Got a PSN?

I think your problem is that CM can only be authoritative to one domain. If one SIP user is calling another and they have different domains, the originating or terminating application sequence is going to fail for one of them. I don't have a good answer for you beyond changing all your domains...

I think you could try doing just plain G711 and not offer t38. AFAIK, the Viper phones do a V150.1 negotiation that is encapsulated within T38. I think you might be in a situation where t38 is offered by the carrier, and is transported end to end, but the media gateways in the PSTN between you...

Pretty sure that's a feature request. Otherwise, you can run a little script against the web APIs for IX and probably change it that way if you want the automation. Not fun.

Yeah. This part: Calls hit the group normally through a vdn/vector. Have a global variable 1-3 for 3 people on call If inside of business hours, do the normal thing if out of business hours, go to step 10 if global variable = 1, 20 if variable = 2, 30 if variable = 3 The treatment to look for...

EC500 is still a call on your station. A virtual station can't take a call, it can only use a coverage path. You can try a real x ported station to see if it's any different, but AFAIK, once the call hits coverage, it's not on the station anymore so you can't record the station.

So everybody needs to get calls on their extension and hit their EC500 after hours for personal calls but there needs to be a separate configuration so that calls to a hunt group or other department number would go to just the one person that's on call and you want a button on each phone to...

If you can record all a station's calls and you're always going out to the same number, I'd try setting up EC500 on a station to the destination number and try recording the station that way

It's on WebLM. Either standalone or in SMGR or in System Platform. Should be an option to export licenses from there to a zip file on the local file system of whatever is hosting WebLM.

I have no idea. I'd imagine you're just recording whatever codec the stream is in, which means if you're using H323 via DMCC that you're limited to the codecs supported by H.323 would be up to G.722. If you have an Avaya Media Server and not G430/450, you could try setting up opus only in a...