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You can actualy log in to webpro and downgrade to the version you have although I wouldn't bother

Try basilisk browser

Bit rusty on these old things and has been a verrrrryyyyy long time but how I read it in the manual is -> 3112-01: KEYRG on all 3 (this refers to DAY overflow) 3112-04: 303 on all 3 3112-05: Yes on all 3 (this refers to NIGHT overflow) 3112-06: 700 on all 3 theres a timer somewhere too but...

They won't be compatible.

Why not replace the door phone for an IP intercom from fanvil or 2n and then you can use those ethernet surge protectors. Presumably you want to protect the dph2 card, back in the day we used those krone strip surge protectors. I've had the same instance of bad cards due to surge current.

Dont think thats how FXO works. Each SIP TRUNK channel on the SV is licensed and your converting SIP to analog so would not the FXO output connect to an Analog line card?

on the cpu card of the TDE is an analog input for MOH/BGM. IIRC the TDE has SVM so if I am correct you can upload a wav file to the OGM

Have you got automatic line seizure enabled on your extension?

Yes, but if you have access to the web programming you can change the operators extension (temporarily) and use the *782 code 2 System -> 2 Operator and BGM. *790 clears features set BTW like call forwarding

Actually its * 7 9 0 im used to NEC programming these days. If you are trying to dial out from these handsets and they transfer you to the operator then its likely the extension has been locked. You can unlock this from the operator phone by dialling SPEAKER * 7 8 2 EXTENSION NUMBER SPEAKER...

Hmm. Try dialling 790 from those problematic extensions and then try dialling them internally once done.

2 separate sip accounts you'll have a hard time sending company b ddi to the registered sip account as call will always come through on the lowest available line which in the above scenario belongs to company a.

Perfect thanks

Ns700

So you would forward the source RTP ports from your itsp and the destination would be 10020-10277 on the sl IPLB or rather the DSP address you have assigned in 10-12-09. If you have 8 sip trunks you don't reduce the RTP ports because the voice packets could fall anywhere within that port range...

Source 10000-60000 to Destination 10020-10277 UDP 10000 - 60000 UDP 10020 - 10277 If you reduce the port range you will have issues with audio.

Have you tried a different browser, I know Chrome and Firefox sometimes has issues, although certainly not having issues with my system

Quick question, can a NT55x IP Handset use wireless XDP to a SIP extension using the *48 feature code? Idea is to install an algo strobe and when the NT phone rings, the algo will flash.

Are you able to upgrade the FW to the latest and see if Incapture then appears?

What software version is your system on?