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KX-UT670 2
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Mr. Durzel
For V6.01 or later, V-UTEXT cards have to be disabled for using new features. This condition is applied for KX-NS1000 and KX-NS300/500/700 in KX-NS1000 one look network.
KX-UT SIP phone can work as ordinary SIP-SLT with V-SIPEXT card without UT specialized feature such as DSS, CO key etc. At same time you need 3rd party SIP device AK(KX-NSM7xx) to connect UT as SIP-SLT.
For V6.01 or later, V-UTEXT cards have to be disabled for using new features. This condition is applied for KX-NS1000 and KX-NS300/500/700 in KX-NS1000 one look network.
KX-UT SIP phone can work as ordinary SIP-SLT with V-SIPEXT card without UT specialized feature such as DSS, CO key etc. At same time you need 3rd party SIP device AK(KX-NSM7xx) to connect UT as SIP-SLT.
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 has assigned IP address from DHCP server on same network).
The Powerpoint you linked doesn't show anything about configuring the phone itself, in terms of ports and addresses.
Any help you can provide would be gratefully received
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 has assigned IP address from DHCP server on same network).
The Powerpoint you linked doesn't show anything about configuring the phone itself, in terms of ports and addresses.
Any help you can provide would be gratefully received
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 cannot turn on the "Embedded Web" on the UT, it just says "Failed to save setting". I have to unregister it from the PBX to be able to access the embedded web again.
Thanks for your help.
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 cannot turn on the "Embedded Web" on the UT, it just says "Failed to save setting". I have to unregister it from the PBX to be able to access the embedded web again.
Thanks for your help.
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 though.
edit: Factory resetting the UT670 made no difference unfortunately.
These are the settings on my UT670:
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 though.
edit: Factory resetting the UT670 made no difference unfortunately.
These are the settings on my UT670:
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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 the router relating to Media Relay Gateway, in particular for CWMP and "Data Transmission Protocol (HTTPS) Port No, for SIP-MLT". The phone connects to the PBX the same as it did on the LAN, but there is still no dial tone.
What are these settings in WebMC? Are they important?
Voice (RTP) UDP Port No. (Server)
Voice (RTP) UDP Port No. (IP-PT / SIP-MLT)
I have UDP ports 16000-16511 forwarded to the DSP IP.
I am not totally sure that this is a router/firewall problem because the phone is behaving the same both on the same LAN as the PBX, and at a remote location.
Thanks in advance.
Also - as I mentioned before - once the phone is registered to the PBX I am unable to enable the "Embedded Web". It just tells me "Failed to save setting". I don't know if that is a permissions thing or not, but the only way I can get back on it is by de-registering the phone.
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 the router relating to Media Relay Gateway, in particular for CWMP and "Data Transmission Protocol (HTTPS) Port No, for SIP-MLT". The phone connects to the PBX the same as it did on the LAN, but there is still no dial tone.
What are these settings in WebMC? Are they important?
Voice (RTP) UDP Port No. (Server)
Voice (RTP) UDP Port No. (IP-PT / SIP-MLT)
I have UDP ports 16000-16511 forwarded to the DSP IP.
I am not totally sure that this is a router/firewall problem because the phone is behaving the same both on the same LAN as the PBX, and at a remote location.
Thanks in advance.
Also - as I mentioned before - once the phone is registered to the PBX I am unable to enable the "Embedded Web". It just tells me "Failed to save setting". I don't know if that is a permissions thing or not, but the only way I can get back on it is by de-registering the phone.
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 able to register it both on the same LAN and remotely - so I think the ports are ok.
I would think at this point that the phone itself was faulty, or that the speaker/handset didn't work, but I can change the ringing tone on the menu and it makes a sound, so I think it's ok.
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 able to register it both on the same LAN and remotely - so I think the ports are ok.
I would think at this point that the phone itself was faulty, or that the speaker/handset didn't work, but I can change the ringing tone on the menu and it makes a sound, so I think it's ok.
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