Very kind of you to offer, thanks.
I managed to find the version you mention online after searching around with Google, and have updated my phone system to it. Glad to hear it’s the latest one available.
The company that supports our phone system told me that 8.3 “wasn’t available in the UK...
imran4212 - did you ever solve your UT670 screen flickering problem? From this thread: https://www.tek-tips.com/viewthread.cfm?qid=1798675
I can't reply to that thread.
I think I love you Mike :) Assigning a DN to one of the flexible buttons fixed it. Now when I pick up the handset (or press 1 on the onscreen keypad) I get a dialtone, can make calls and voice works both ways.
Thanks so much!
Ok I have made some progress.
I reset all of the flexible buttons on the phone to a default set. They now look like this:
If I pick up the handset I don't get a dial tone, but if I press one of those flexible buttons (01 to 08) then I get a dial tone. If I try and place a call it doesn't...
Ok I have made some progress, but the phone still isn't working.
It seems that uploading the config I took from the PBX (UT_MRG_HTTPS_01NS700.cfg) to the phone in installer mode (user: instoperatoruserid, pass: instpass) then I don't have to set the VOIP settings in the Embedded Web myself, as...
Thanks Mike.
Does the SIP source port HAVE to be 44822? I have two SIP phones on the same network, I thought the SIP source port had to be unique unless using SBC?
Assuming I can use 7010 as the source port, would the SIP Service Domain be: sip.domain.com:7010 or sip.domain.com:44822 ?
No, do I need to do that?
Here are my settings on the phone, pre-registration:
(I changed the SIP source port to 7010 because apparently this has to be unique, and I already have a working KX-HDV230 that was using the same port - but they are now using unique ports, but it has not made any...
Thanks for the continued help.
I have checked that document, and with the exception of the HTTP ports (I have set MRG to use HTTPS) they are all open and forwarded correctly.
The phone didn't work (no dialtone) when it was on the same LAN, connected to the same switch as the PBX. I have been...
What ports would stop a dial tone from being heard? Or a call being received on the extension (other extensions see it as busy)
I have a KX-HDV230 SIP phone with similar settings that works, but the UT670 doesn't.
I have factory reset it and registered it remotely, and I can see packets on...
Thanks Mike.
I had a look at your image - unfortunately I have all of those settings in place already. The phone registers successfully, and looks from the handset like it is working, but there is no dial tone on the handset or speakerphone.
I will try factory resetting the UT670 again...
Hi,
Yes, everything open between UT and PBX, they are on the same switch.
I can get the phone to register, but I have no dial tone and trying to call the phone from another handset (non-UT) in the office says its busy :(
I don't know if its relevant but after I have registered the phone I...
UT670 is running firmware 01.122 and I did factory reset it. I am able to manually register the phone (it says "Registration Complete" with a green tick) but it does not work after that.
Thanks.
I'm running version 004.22016 at the moment.
I can register a KX-UT670 phone on the same network as the NS700, but there is no dial tone or ability to make calls, and it shows "Fault" in the SIP-MLT section. The MAC address and Program Version is shown, but no IP address (but phone...
Hi,
Does anyone have the latest available firmware for a NS700? I am currently running 004.22016 according to WebMC.
I have downloaded a PFMPR_008_030_230 file - which looks to be 008.030230 (8.03?). I'm guessing that isn't the latest?
My PBX supplier, who is being quite unhelpful, is...
Hi,
I'm trying to add a KX-UT670 phone to our NS700 PBX. I have not purchased any additional licences for it, but it says that it has 4 "IP Proprietary Telephone/P-SIP Extension (ch)" ones pre-installed. We appear to be running firmware 004.22016.
I am able to manually register the phone...
I finally got this working remotely.
The problem was that the local SIP port on the phone was set to 5060. I presume since I am going through several routers when remote that one of the ones in the office (probably the Gamma one that I don't have access to) is running ALG on port 5060. When I...
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