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Support

SIP Phone

Sipphone

KX-TGP500/550/551
  • *Installation
    *Configuration
    *Configuration file
    *Troubleshooting
    • Additional features do not work after updating the firmware of base unit to ver.22.68 or later. [ 2012/11/09 ]
      • Additional features may not work by difference of version between base unit and handset.
        Please refer to this document for more detail.
      I cannot hear Music On Hold (MoH). [ 2012/09/04 ]
      • As default, phone cannot play Music On Hold if Music On Hold is sent from server after server sent "inactive" SDP after hold had been placed by other party.
        We can change behavior by parameter.
        Please set the parameter
        HOLD_SOUND_PATH_n="1" by configuration file.
        The default value is "0" as phone play it's own hold tone.
        Example :
        HOLD_SOUND_PATH_1="1"
      The status LED goes umber from green after a while. A phone cannot be used. [ 2012/09/04 ]
      • It is happens some cases if you connect SIP server with multi steps like Phone - SIP Server1 - SIP Server2 behind SIP Server1.
        If the SIP servers are connected by two or more steps like this, please reconfirm these settings and connections.
      I have registered SIP phone to SIP server successfully. I can make a call, but I can not receive a call. [ 2012/09/04 ]
      1. Please check you've turned on some parametes for security related settings as follows:
          SIP_RCV_DET_HEADER_n="Y"
          SIP_DETECT_SSAF_n="Y"
        By setting these parameters, phone may reject the call as security reason. If you turned on these paramters, please try following procedure:
        -Solution:
         Set up appropriate security function of TGP for each usage environment.
         (case1)
        When the user part of the To header in reception INVITE is different from your telephone number, change setting as follows.
          SIP_RCV_DET_HEADER_n="Y" ->"N"
         (case2)
        When the To header in reception INVITE is correct but you cannot receive INVITE, change setting as follows.
          SIP_DETECT_SSAF_n="Y"->"N"
      2. Please check if Phone is installed under NAT environment or not. If yes, please try following procedure.
         Some router may function to close the port which is used for NAT. So, we need to access to router periodically before NAT is closed and information is deleted.
        For accessing to router periodically, please change following setting.
          PORT_PUNCH_INTVL_n="0" -> "20"(example)
      Initial setting value will not be retrieved even by "Reset Web Settings". [ 2012/09/04 ]
      Voice cuts off while talking on the phone. [ 2012/09/04 ]
      • Voice may cut off while downloading configulation file. Please update your firmware ver.12.10/22.10 or later.
        Please refter to No.04-003 of technical contents.
      When login to WEB setting menu, "403 Forbidden" message is displayed and cannot continue. [ 2012/09/04 ]
      Noise is heard, the sound is interrupted, the sound is delayed. [ 2012/09/04 ]
      1. It may be caused by someone using a computer connected to the same network as the unit.
      2. Confirm the speed of Internet connection. The unit requires 100 kbps for each upstream and downstream.
      Noise is heard, sound cuts in and outs. [ 2012/09/04 ]
      1. You are using the handset or base unit in an area with high electrical interference. Re-position the base unit and use the handset away from sources of interference.
      2. Move closer to the base unit.
      The unit does not work. [ 2012/09/04 ]
      1. Make sure the batteries are installed correctly
      2. Fully charge the batteries.
      3. Check the connections.
      4. Unplug the base unit's AC adaptor to reset the unit. Reconnect the adaptor and try again.
      5. The handset has not been registered to the base unit.
    *Operation/Features/Indication
    • I would like to disable the Key Click Tone of the phone when press the keys. [ 2012/09/04 ]
      • (TGP)From handset, please press [MENU] and [1][6][5], then press [0] to turn off key tone.
        Note : cannot disable key tones of Base Unit.
        (UT1xx/UT248)From handset:
        [Setting]->[Key Click Tone]->[Off] or [Automatic].
        If it is set as Automatic, the key tone rely on the setting from WEB or configuration file.
        WEB setting :
        [Telephone Settings]->[Telephone Settings]->[Key Click Tone]->[No]
        Configuration file:
        KEY_PAD_TONE="N" by configuration file.
        The default value is "Y" to active the key tone.
      Voice Message Waiting Indicator does not turn on. [ 2012/09/04 ]
      • We can choose the type of SIP message which phone can accept for Voice Message.
        As default phone cannot recognize the Voice Message if there is no "Voice-Message" line included.
        As default : phone can accept following notification message for Voice Message.
        ----------------------------------
        Messages-Waiting: yes
        voice-message: 1/0 (0/0)
        ----------------------------------
        Phone cannot accept following, or different structure of "Voice-Message" line.
        ----------------------------------
        Messages-Waiting: yes
        ----------------------------------
        Please set the parameter if voice message notification does not have "Voice-Message"
        VOICE_MESSAGE_AVAILABLE="N" by configuration file.
      How many digits of Caller ID will be displayed when receiving a call? [ 2012/09/04 ]
      How many digits will be displayed on LCD in each feature? [ 2012/09/04 ]
      How long does it take to turn off LCD backlit after operating the unit? [ 2012/09/04 ]
      • (Base unit)-Approx. 60sec after pressing a valid key
        -Approx. 5sec after disconnecting a call (Handset)-Approx. 10sec after pressing any key Please refter to No.05-003 of technical contents.
      Can I use intercom feature even if the SIP service is down? [ 2012/09/04 ]
      • Yes, you can use Intercom feature even when SIP server is down.
      How can I turn on "Do not disturb" and which unit will manage this feature? [ 2012/09/04 ]
      • "Do not disturb" feature can be set by WEB user interface or SIP terminal and it can set for each line.
        So even if one line set this feature, the other lines can receive a call.
      Caller information is not displayed. [ 2012/09/04 ]
      • Availability of this service depends on your phone system. Contact your administrator.
      I cannot barge in a call. [ 2012/09/04 ]
      How can I turn on Call Block feature and which unit will manage this feature? [ 2012/09/04 ]
      • Call block can be set by SIP terminal and you can register the list of number you'd like to block up to 30.
        This feature and information is managed by SIP terminal. So all lines share same call block information.
      When I got a call, the phone rings but there is no one (no voice) , just silence when I pick up the handset to answer. [ 2012/09/04 ]
      The STATUS indicator lights in amber although the Ethernet cable is connected properly. [ 2012/09/04 ]
      • The unit's IP address may conflict with the IP addresses of other devices on your local network.
        Check the unit's static IP address. Then check the IP addresses of the other devices on your local network. If necessary, change the unit's static IP address.
      How can I de-register all handset at once? [ 2012/09/04 ]
      • Press and hold the Locator key [ALL] on base unit about 30 seconds.
      Caller information is displayed late. [ 2012/09/04 ]
      • Move closer to the base unit.
      I cannot make a call. [ 2012/09/04 ]
      1. Check if the STATUS indicator or indicator is lit in green.(TGP)
      2. You cannot make a call while the base unit is downloading the firmware update. Wait for the update to finish, and then try making a call.(TGP/UT1xx/UT670/UT248)
      3. The handset is too far from the base unit. Move closer and try again.(TGP)
      4. Check the VoIP status in the Web user interface and confirm that each line is registered properly.(TGP/UT1xx/UT670/UT248)
      5. Check that the SIP server address, URLs of the configuration files, encryption key, and other required settings are correct.(TGP/UT1xx/UT670/UT248)
      6. Check the firewall and port forwarding settings on the router.(TGP/UT1xx/UT670/UT248)
      7. Check [Multi Number Settings] in the [Telephone] tab in the Web user interface.(TGP)
      8. If 3 calls are being handled by the base unit and/orhandsets simultaneously, a new call cannot be made from the unit.(TGP)
      The STATUS indicator continues flashing in amber. [ 2012/09/04 ]
      1. An IP address may not have been acquired or the static IP address is not appropriate. Check the unit's IP address.It is recommended to perform the following.
        – If necessary, change the unit's static IP address.
        – If an IP address was not acquired, check your network devices' (router, modem, etc.) connections. If the connections are made properly but the problem persists, check your network devices' (router, modem,etc.) settings.
      2. Many installation issues can be resolved by resetting all the equipment. First, shut down your modem, router, hub,base unit, and PC. Then turn the devices back on, one at a time, in this order: modem, router, hub, base unit, PC.
      3. If you cannot access Internet Web pages using your PC, check to see if your phone system is having connection issues in your area.
      4. Check the VoIP status in the Web user interface and confirm that each line is registered properly.Even when one line fails to register, the indicator will flash in amber.
      5. Check that the SIP server address, URLs of the configuration files, encryption key, and other required settings are correct.
      6. Check the firewall and port forwarding settings on the router.
      How can I transfer a call? [ 2012/09/04 ]
      How can I mute key tone? [ 2012/09/04 ]
      • You can mute key tone by operating handset as follows.
        Press [Menu] -->Select [Initial settings]-->Select[Keytones]
      What is the rule of Tone Timing for Tone settings? [ 2012/09/04 ]
      • The meaning of four parameters for tone setting is described as.
        "on 1, off 1, on 2, off 2" ( on=ringing term, off=silent term).
        When it does not contain "0" as a parameter, the setting means as follows.
        (Example: "100, 200, 300, 400")
          --> "100msec ring, 200ms silent, 300ms ring, 400ms silent" (and this pattern will be repeated)
        A parameter "0" means "infinite time". Therefore, if "0" is included in the parameters, the unit will keep on ringing or being silent from some timing (depends on where "0" is put in these four parameters.)
        (Example1 :"100, 100, 0, 0")
          -->"100ms ring, 100ms silent, keep ringing"
        (Example2 :"100, 0, 100, 0")
          -->"100ms ring, keep silent" (because the first "0" is effective infinitely)
      The name stored in the phonebook is not displayed fully while an outside call is being received. [ 2012/09/04 ]
      • Edit the phonebook entry name to fit in 1 line of text.
      The displayed caller information is not correct. [ 2012/09/04 ]
      • Caller information may not be correct depending on your phone system.<
      The STATUS indicator is off. [ 2012/09/04 ]
      1. The base unit power is off.
      2. The Ethernet cable is not connected properly. Connect it.
      3. Your network devices (hub, router, etc.) are turned off. Check the LEDs for the link status of the devices.
      4. The base unit power is booting up.
      The SIP terminal does not ring. [ 2012/09/04 ]
      1. The ringer volume is turned off. Adjust the ringer volume.(TGP/UT1xx/UT670/UT248)
      2. Check the VoIP status in the Web user interface and confirm that each line is registered properly.(TGP/UT1xx/UT670/UT248)
      3. Check that the SIP server address, URLs of the configuration files, encryption key, and other required settings are correct.(TGP/UT1xx/UT670/UT248)
      4. Check the firewall and port forwarding settings on the router.(TGP/UT1xx/UT670/UT248)
      5. Check [Multi Number Settings] in the [Telephone] tab inthe Web user interface.(TGP)
      6. Check [Call Control] for each line in the [Telephone] tab in the Web user interface.(TGP/UT1xx/UT670/UT248)
        – If [Do Not Disturb] is set to [Yes], the base unit or handset does not receive calls.
        – If [Unconditional (Enable Call Forward)] is set to [Yes], the base unit or handset does not receive calls.
        – If [Block Anonymous Call] is set to [Yes], SIP terminal does not receive anonymous calls.
      7. Check that [Do Not Disturb], [Enable Call Forward], and [Block Anonymous Call] are not controlled by your phone system.(TGP/UT1xx/UT670/UT248)
      8. If 3 calls are being handled by the base unit and/or handsets simultaneously, a new call cannot be received and the unit will not ring.(TGP)
      The STATUS indicator flashes in amber rapidly although the IP address was acquired. [ 2012/09/04 ]
      • Unplug the unit's AC adaptor to reset the unit, then reconnect the AC adaptor. If the STATUS indicator is still flashing in amber rapidly, there may be a problem with the base unit hardware.
      I cannot hear a dial tone. [ 2012/09/04 ]
      1. Confirm that the Ethernet cable is properly connected.
      2. Network settings may not be correct.
      3. Many installation issues can be resolved by resetting all the equipment. First, shut down your modem, router, hub, base unit, and PC. Then turn the devices back on, one at a time, in this order: modem, router, hub, base unit, PC.
      4. If you cannot access Internet Web pages using your PC, check to see if your phone system is having connection issues in your area.
      5. Check the VoIP status in the Web user interface and confirm that each line is registered properly.
      6. Check that the SIP server address, URLs of theconfiguration files, encryption key, and other required settings are correct.
      7. Check the firewall and port forwarding settings on the router.
      What is the rule and format about SYSLOG? [ 2012/09/04 ]
    *SIP signaling
KX-UT113/123/133/136
  • *Installation
    *Configuration
    *Configuration file
    *Troubleshooting
    • Acoustic gain settings specified by ADJDATA_GAIN cannot work. [ 2012/12/10 ]
      • (UT1xx/248)
        There are misdescriptions about ADJDATA_GAIN in Administrator Guide.
        Please refer to the correction document.
      I can not make/receive a call after setting BLF to all (24) Flexible keys. [ 2012/09/04 ]
      • At least 2 DN buttons must be assigned to each line.
        If DN buttons are not assigned, calls cannot be made or answered.
      I cannot hear Music On Hold (MoH). [ 2012/09/04 ]
      • As default, phone cannot play Music On Hold if Music On Hold is sent from server after server sent "inactive" SDP after hold had been placed by other party.
        We can change behavior by parameter.
        Please set the parameter
        HOLD_SOUND_PATH_n="1" by configuration file.
        The default value is "0" as phone play it's own hold tone.
        Example :
        HOLD_SOUND_PATH_1="1"
      I have registered SIP phone to SIP server successfully. I can make a call, but I can not receive a call. [ 2012/09/04 ]
      1. Please check you've turned on some parametes for security related settings as follows:
          SIP_RCV_DET_HEADER_n="Y"
          SIP_DETECT_SSAF_n="Y"
        By setting these parameters, phone may reject the call as security reason. If you turned on these paramters, please try following procedure:
        -Solution:
         Set up appropriate security function of TGP for each usage environment.
         (case1)
        When the user part of the To header in reception INVITE is different from your telephone number, change setting as follows.
          SIP_RCV_DET_HEADER_n="Y" ->"N"
         (case2)
        When the To header in reception INVITE is correct but you cannot receive INVITE, change setting as follows.
          SIP_DETECT_SSAF_n="Y"->"N"
      2. Please check if Phone is installed under NAT environment or not. If yes, please try following procedure.
         Some router may function to close the port which is used for NAT. So, we need to access to router periodically before NAT is closed and information is deleted.
        For accessing to router periodically, please change following setting.
          PORT_PUNCH_INTVL_n="0" -> "20"(example)
      Initial setting value will not be retrieved even by "Reset Web Settings". [ 2012/09/04 ]
      When login to WEB setting menu, "403 Forbidden" message is displayed and cannot continue. [ 2012/09/04 ]
    *Operation/Features/Indication
    • Can I set BLF to all (24) Flexible keys? [ 2012/09/04 ]
      • At least 2 DN buttons must be assigned to each line.
        If DN buttons are not assigned, calls cannot be made or answered.
      I would like to disable the Key Click Tone of the phone when press the keys. [ 2012/09/04 ]
      • (TGP)From handset, please press [MENU] and [1][6][5], then press [0] to turn off key tone.
        Note : cannot disable key tones of Base Unit.
        (UT1xx/UT248)From handset:
        [Setting]->[Key Click Tone]->[Off] or [Automatic].
        If it is set as Automatic, the key tone rely on the setting from WEB or configuration file.
        WEB setting :
        [Telephone Settings]->[Telephone Settings]->[Key Click Tone]->[No]
        Configuration file:
        KEY_PAD_TONE="N" by configuration file.
        The default value is "Y" to active the key tone.
      I would like to change the ringer tone. [ 2012/09/04 ]
      • (UT1xx/UT248)
        Setting from handset:
        [Setting]->[Ringer]->[Ringer Tone]
        If you have multiple lines, please select the line at first.
        If you'd like to change ringer tone by SIP Message for distinguish ring tone,
        please use following.
        Alert-Info:(x is the id)
        Example
        Alert-Info:
      Voice Message Waiting Indicator does not turn on. [ 2012/09/04 ]
      • We can choose the type of SIP message which phone can accept for Voice Message.
        As default phone cannot recognize the Voice Message if there is no "Voice-Message" line included.
        As default : phone can accept following notification message for Voice Message.
        ----------------------------------
        Messages-Waiting: yes
        voice-message: 1/0 (0/0)
        ----------------------------------
        Phone cannot accept following, or different structure of "Voice-Message" line.
        ----------------------------------
        Messages-Waiting: yes
        ----------------------------------
        Please set the parameter if voice message notification does not have "Voice-Message"
        VOICE_MESSAGE_AVAILABLE="N" by configuration file.
      Current Call is disconnected when I answer the incoming call during calling from someone. Why? [ 2012/09/04 ]
      • We can choose whether disconnect the existing call or not
        when answering the second call by configuration parameter.
        Please set the parameter
        AUTO_CALL_HOLD="Y" by configuration file.
        The default value is "N" as disabled.
      How many digits of Caller ID will be displayed when receiving a call? [ 2012/09/04 ]
      How can I turn on "Do not disturb" and which unit will manage this feature? [ 2012/09/04 ]
      • "Do not disturb" feature can be set by WEB user interface or SIP terminal and it can set for each line.
        So even if one line set this feature, the other lines can receive a call.
      Caller information is not displayed. [ 2012/09/04 ]
      • Availability of this service depends on your phone system. Contact your administrator.
      How can I turn on Call Block feature and which unit will manage this feature? [ 2012/09/04 ]
      • Call block can be set by SIP terminal and you can register the list of number you'd like to block up to 30.
        This feature and information is managed by SIP terminal. So all lines share same call block information.
      When I got a call, the phone rings but there is no one (no voice) , just silence when I pick up the handset to answer. [ 2012/09/04 ]
      I cannot make a call. [ 2012/09/04 ]
      1. Check if the STATUS indicator or indicator is lit in green.(TGP)
      2. You cannot make a call while the base unit is downloading the firmware update. Wait for the update to finish, and then try making a call.(TGP/UT1xx/UT670/UT248)
      3. The handset is too far from the base unit. Move closer and try again.(TGP)
      4. Check the VoIP status in the Web user interface and confirm that each line is registered properly.(TGP/UT1xx/UT670/UT248)
      5. Check that the SIP server address, URLs of the configuration files, encryption key, and other required settings are correct.(TGP/UT1xx/UT670/UT248)
      6. Check the firewall and port forwarding settings on the router.(TGP/UT1xx/UT670/UT248)
      7. Check [Multi Number Settings] in the [Telephone] tab in the Web user interface.(TGP)
      8. If 3 calls are being handled by the base unit and/orhandsets simultaneously, a new call cannot be made from the unit.(TGP)
      How can I transfer a call? [ 2012/09/04 ]
      What is the rule of Tone Timing for Tone settings? [ 2012/09/04 ]
      • The meaning of four parameters for tone setting is described as.
        "on 1, off 1, on 2, off 2" ( on=ringing term, off=silent term).
        When it does not contain "0" as a parameter, the setting means as follows.
        (Example: "100, 200, 300, 400")
          --> "100msec ring, 200ms silent, 300ms ring, 400ms silent" (and this pattern will be repeated)
        A parameter "0" means "infinite time". Therefore, if "0" is included in the parameters, the unit will keep on ringing or being silent from some timing (depends on where "0" is put in these four parameters.)
        (Example1 :"100, 100, 0, 0")
          -->"100ms ring, 100ms silent, keep ringing"
        (Example2 :"100, 0, 100, 0")
          -->"100ms ring, keep silent" (because the first "0" is effective infinitely)
      The displayed caller information is not correct. [ 2012/09/04 ]
      • Caller information may not be correct depending on your phone system.
      The SIP terminal does not ring. [ 2012/09/04 ]
      1. The ringer volume is turned off. Adjust the ringer volume.(TGP/UT1xx/UT670/UT248)
      2. Check the VoIP status in the Web user interface and confirm that each line is registered properly.(TGP/UT1xx/UT670/UT248)
      3. Check that the SIP server address, URLs of the configuration files, encryption key, and other required settings are correct.(TGP/UT1xx/UT670/UT248)
      4. Check the firewall and port forwarding settings on the router.(TGP/UT1xx/UT670/UT248)
      5. Check [Multi Number Settings] in the [Telephone] tab inthe Web user interface.(TGP)
      6. Check [Call Control] for each line in the [Telephone] tab in the Web user interface.(TGP/UT1xx/UT670/UT248)
        – If [Do Not Disturb] is set to [Yes], the base unit or handset does not receive calls.
        – If [Unconditional (Enable Call Forward)] is set to [Yes], the base unit or handset does not receive calls.
        – If [Block Anonymous Call] is set to [Yes], SIP terminal does not receive anonymous calls.
      7. Check that [Do Not Disturb], [Enable Call Forward], and [Block Anonymous Call] are not controlled by your phone system.(TGP/UT1xx/UT670/UT248)
      8. If 3 calls are being handled by the base unit and/or handsets simultaneously, a new call cannot be received and the unit will not ring.(TGP)
      <
      I cannot hear a dial tone. [ 2012/09/04 ]
      1. Confirm that the Ethernet cable is properly connected.
      2. Network settings may not be correct.
      3. Many installation issues can be resolved by resetting all the equipment. First, shut down your modem, router, hub, base unit, and PC. Then turn the devices back on, one at a time, in this order: modem, router, hub, base unit, PC.
      4. If you cannot access Internet Web pages using your PC, check to see if your phone system is having connection issues in your area.
      5. Check the VoIP status in the Web user interface and confirm that each line is registered properly.
      6. Check that the SIP server address, URLs of theconfiguration files, encryption key, and other required settings are correct.
      7. Check the firewall and port forwarding settings on the router.
      What is the rule and format about SYSLOG? [ 2012/09/04 ]
    *SIP signaling
    *Application
KX-UT670
  • *Installation
    *Configuration
    *Configuration file
    *Troubleshooting
    • I can not make/receive a call after setting BLF to all (24) Flexible keys. [ 2012/09/04 ]
      • At least 2 DN buttons must be assigned to each line.
        If DN buttons are not assigned, calls cannot be made or answered.
      I have registered SIP phone to SIP server successfully. I can make a call, but I can not receive a call. [ 2012/09/04 ]
      1. Please check you've turned on some parametes for security related settings as follows:
          SIP_RCV_DET_HEADER_n="Y"
          SIP_DETECT_SSAF_n="Y"
        By setting these parameters, phone may reject the call as security reason. If you turned on these paramters, please try following procedure:
        -Solution:
         Set up appropriate security function of TGP for each usage environment.
         (case1)
        When the user part of the To header in reception INVITE is different from your telephone number, change setting as follows.
          SIP_RCV_DET_HEADER_n="Y" ->"N"
         (case2)
        When the To header in reception INVITE is correct but you cannot receive INVITE, change setting as follows.
          SIP_DETECT_SSAF_n="Y"->"N"
      2. Please check if Phone is installed under NAT environment or not. If yes, please try following procedure.
         Some router may function to close the port which is used for NAT. So, we need to access to router periodically before NAT is closed and information is deleted.
        For accessing to router periodically, please change following setting.
          PORT_PUNCH_INTVL_n="0" -> "20"(example)
      Initial setting value will not be retrieved even by "Reset Web Settings". [ 2012/09/04 ]
      When login to WEB setting menu, "403 Forbidden" message is displayed and cannot continue. [ 2012/09/04 ]
    *Operation/Features/Indication
    • Can I set BLF to all (24) Flexible keys? [ 2012/09/04 ]
      • At least 2 DN buttons must be assigned to each line.
        If DN buttons are not assigned, calls cannot be made or answered.
      Voice Message Waiting Indicator does not turn on. [ 2012/09/04 ]
      • We can choose the type of SIP message which phone can accept for Voice Message.
        As default phone cannot recognize the Voice Message if there is no "Voice-Message" line included.
        As default : phone can accept following notification message for Voice Message.
        ----------------------------------
        Messages-Waiting: yes
        voice-message: 1/0 (0/0)
        ----------------------------------
        Phone cannot accept following, or different structure of "Voice-Message" line.
        ----------------------------------
        Messages-Waiting: yes
        ----------------------------------
        Please set the parameter if voice message notification does not have "Voice-Message"
        VOICE_MESSAGE_AVAILABLE="N" by configuration file.
      Current Call is disconnected when I answer the incoming call during calling from someone. Why? [ 2012/09/04 ]
      • We can choose whether disconnect the existing call or not
        when answering the second call by configuration parameter.
        Please set the parameter
        AUTO_CALL_HOLD="Y" by configuration file.
        The default value is "N" as disabled
      How many digits of Caller ID will be displayed when receiving a call? [ 2012/09/04 ]
      How can I turn on "Do not disturb" and which unit will manage this feature? [ 2012/09/04 ]
      • "Do not disturb" feature can be set by WEB user interface or SIP terminal and it can set for each line.
        So even if one line set this feature, the other lines can receive a call.
      Caller information is not displayed. [ 2012/09/04 ]
      • Availability of this service depends on your phone system. Contact your administrator.
      How can I turn on Call Block feature and which unit will manage this feature? [ 2012/09/04 ]
      • Call block can be set by SIP terminal and you can register the list of number you'd like to block up to 30.
        This feature and information is managed by SIP terminal. So all lines share same call block information.
      When I got a call, the phone rings but there is no one (no voice) , just silence when I pick up the handset to answer. [ 2012/09/04 ]
      I cannot make a call. [ 2012/09/04 ]
      1. Check if the STATUS indicator or indicator is lit in green.(TGP)
      2. You cannot make a call while the base unit is downloading the firmware update. Wait for the update to finish, and then try making a call.(TGP/UT1xx/UT670/UT248)
      3. The handset is too far from the base unit. Move closer and try again.(TGP)
      4. Check the VoIP status in the Web user interface and confirm that each line is registered properly.(TGP/UT1xx/UT670/UT248)
      5. Check that the SIP server address, URLs of the configuration files, encryption key, and other required settings are correct.(TGP/UT1xx/UT670/UT248)
      6. Check the firewall and port forwarding settings on the router.(TGP/UT1xx/UT670/UT248)
      7. Check [Multi Number Settings] in the [Telephone] tab in the Web user interface.(TGP)
      8. If 3 calls are being handled by the base unit and/orhandsets simultaneously, a new call cannot be made from the unit.(TGP)
      How can I transfer a call? [ 2012/09/04 ]
      What is the rule of Tone Timing for Tone settings? [ 2012/09/04 ]
      • The meaning of four parameters for tone setting is described as.
        "on 1, off 1, on 2, off 2" ( on=ringing term, off=silent term).
        When it does not contain "0" as a parameter, the setting means as follows.
        (Example: "100, 200, 300, 400")
          --> "100msec ring, 200ms silent, 300ms ring, 400ms silent" (and this pattern will be repeated)
        A parameter "0" means "infinite time". Therefore, if "0" is included in the parameters, the unit will keep on ringing or being silent from some timing (depends on where "0" is put in these four parameters.)
        (Example1 :"100, 100, 0, 0")
          -->"100ms ring, 100ms silent, keep ringing"
        (Example2 :"100, 0, 100, 0")
          -->"100ms ring, keep silent" (because the first "0" is effective infinitely)
      The displayed caller information is not correct. [ 2012/09/04 ]
      • Caller information may not be correct depending on your phone system.
      The SIP terminal does not ring. [ 2012/09/04 ]
      1. The ringer volume is turned off. Adjust the ringer volume.(TGP/UT1xx/UT670/UT248)
      2. Check the VoIP status in the Web user interface and confirm that each line is registered properly.(TGP/UT1xx/UT670/UT248)
      3. Check that the SIP server address, URLs of the configuration files, encryption key, and other required settings are correct.(TGP/UT1xx/UT670/UT248)
      4. Check the firewall and port forwarding settings on the router.(TGP/UT1xx/UT670/UT248)
      5. Check [Multi Number Settings] in the [Telephone] tab inthe Web user interface.(TGP)
      6. Check [Call Control] for each line in the [Telephone] tab in the Web user interface.(TGP/UT1xx/UT670/UT248)
        – If [Do Not Disturb] is set to [Yes], the base unit or handset does not receive calls.
        – If [Unconditional (Enable Call Forward)] is set to [Yes], the base unit or handset does not receive calls.
        – If [Block Anonymous Call] is set to [Yes], SIP terminal does not receive anonymous calls.
      7. Check that [Do Not Disturb], [Enable Call Forward], and [Block Anonymous Call] are not controlled by your phone system.(TGP/UT1xx/UT670/UT248)
      8. If 3 calls are being handled by the base unit and/or handsets simultaneously, a new call cannot be received and the unit will not ring.(TGP)
      I cannot hear a dial tone. [ 2012/09/04 ]
      1. Confirm that the Ethernet cable is properly connected.
      2. Network settings may not be correct.
      3. Many installation issues can be resolved by resetting all the equipment. First, shut down your modem, router, hub, base unit, and PC. Then turn the devices back on, one at a time, in this order: modem, router, hub, base unit, PC.
      4. If you cannot access Internet Web pages using your PC, check to see if your phone system is having connection issues in your area.
      5. Check the VoIP status in the Web user interface and confirm that each line is registered properly.
      6. Check that the SIP server address, URLs of theconfiguration files, encryption key, and other required settings are correct.
      7. Check the firewall and port forwarding settings on the router.
      What is the rule and format about SYSLOG? [ 2012/09/04 ]
    *SIP signaling
    *Application
KX-UT248
  • *Installation
    *Configuration
    *Configuration file
    *Troubleshooting
    • Acoustic gain settings specified by ADJDATA_GAIN cannot work. [ 2012/12/10 ]
      • (UT1xx/248)
        There are misdescriptions about ADJDATA_GAIN in Administrator Guide.
        Please refer to the correction document.
      I cannot hear Music On Hold (MoH). [ 2012/09/04 ]
      • As default, phone cannot play Music On Hold if Music On Hold is sent from server after server sent "inactive" SDP after hold had been placed by other party.
        We can change behavior by parameter.
        Please set the parameter
        HOLD_SOUND_PATH_n="1" by configuration file.
        The default value is "0" as phone play it's own hold tone.
        Example :
        HOLD_SOUND_PATH_1="1"
      I have registered SIP phone to SIP server successfully. I can make a call, but I can not receive a call. [ 2012/09/04 ]
      1. Please check you've turned on some parametes for security related settings as follows:
          SIP_RCV_DET_HEADER_n="Y"
          SIP_DETECT_SSAF_n="Y"
        By setting these parameters, phone may reject the call as security reason. If you turned on these paramters, please try following procedure:
        -Solution:
         Set up appropriate security function of TGP for each usage environment.
         (case1)
        When the user part of the To header in reception INVITE is different from your telephone number, change setting as follows.
          SIP_RCV_DET_HEADER_n="Y" ->"N"
         (case2)
        When the To header in reception INVITE is correct but you cannot receive INVITE, change setting as follows.
          SIP_DETECT_SSAF_n="Y"->"N"
      2. Please check if Phone is installed under NAT environment or not. If yes, please try following procedure.
         Some router may function to close the port which is used for NAT. So, we need to access to router periodically before NAT is closed and information is deleted.
        For accessing to router periodically, please change following setting.
          PORT_PUNCH_INTVL_n="0" -> "20"(example)
      Initial setting value will not be retrieved even by "Reset Web Settings". [ 2012/09/04 ]
      When login to WEB setting menu, "403 Forbidden" message is displayed and cannot continue. [ 2012/09/04 ]
    *Operation/Features/Indication
    • Can I set BLF to all (24) Flexible keys? [ 2012/09/04 ]
      • At least 2 DN buttons must be assigned to each line.
        If DN buttons are not assigned, calls cannot be made or answered.
      I would like to disable the Key Click Tone of the phone when press the keys. [ 2012/09/04 ]
      • (TGP)From handset, please press [MENU] and [1][6][5], then press [0] to turn off key tone.
        Note : cannot disable key tones of Base Unit.
        (UT1xx/UT248)From handset:
        [Setting]->[Key Click Tone]->[Off] or [Automatic].
        If it is set as Automatic, the key tone rely on the setting from WEB or configuration file.
        WEB setting :
        [Telephone Settings]->[Telephone Settings]->[Key Click Tone]->[No]
        Configuration file:
        KEY_PAD_TONE="N" by configuration file.
        The default value is "Y" to active the key tone.
      I would like to change the ringer tone. [ 2012/09/04 ]
      • Setting from handset:
        [Setting]->[Ringer]->[Ringer Tone]
        If you have multiple lines, please select the line at first.
        If you'd like to change ringer tone by SIP Message for distinguish ring tone,
        please use following.
        Alert-Info:(x is the id)
        Example
        Alert-Info:
        (UT1xx/UT248)
      Voice Message Waiting Indicator does not turn on. [ 2012/09/04 ]
      • We can choose the type of SIP message which phone can accept for Voice Message.
        As default phone cannot recognize the Voice Message if there is no "Voice-Message" line included.
        As default : phone can accept following notification message for Voice Message.
        ----------------------------------
        Messages-Waiting: yes
        voice-message: 1/0 (0/0)
        ----------------------------------
        Phone cannot accept following, or different structure of "Voice-Message" line.
        ----------------------------------
        Messages-Waiting: yes
        ----------------------------------
        Please set the parameter if voice message notification does not have "Voice-Message"
        VOICE_MESSAGE_AVAILABLE="N" by configuration file.
      How many digits of Caller ID will be displayed when receiving a call? [ 2012/09/04 ]
      How can I turn on "Do not disturb" and which unit will manage this feature? [ 2012/09/04 ]
      • "Do not disturb" feature can be set by WEB user interface or SIP terminal and it can set for each line.
        So even if one line set this feature, the other lines can receive a call.
      Caller information is not displayed. [ 2012/09/04 ]
      • Availability of this service depends on your phone system. Contact your administrator.
      How can I turn on Call Block feature and which unit will manage this feature? [ 2012/09/04 ]
      • Call block can be set by SIP terminal and you can register the list of number you'd like to block up to 30.
        This feature and information is managed by SIP terminal. So all lines share same call block information.
      When I got a call, the phone rings but there is no one (no voice) , just silence when I pick up the handset to answer. [ 2012/09/04 ]
      I cannot make a call. [ 2012/09/04 ]
      1. Check if the STATUS indicator or indicator is lit in green.(TGP)
      2. You cannot make a call while the base unit is downloading the firmware update. Wait for the update to finish, and then try making a call.(TGP/UT1xx/UT670/UT248)
      3. The handset is too far from the base unit. Move closer and try again.(TGP)
      4. Check the VoIP status in the Web user interface and confirm that each line is registered properly.(TGP/UT1xx/UT670/UT248)
      5. Check that the SIP server address, URLs of the configuration files, encryption key, and other required settings are correct.(TGP/UT1xx/UT670/UT248)
      6. Check the firewall and port forwarding settings on the router.(TGP/UT1xx/UT670/UT248)
      7. Check [Multi Number Settings] in the [Telephone] tab in the Web user interface.(TGP)
      8. If 3 calls are being handled by the base unit and/orhandsets simultaneously, a new call cannot be made from the unit.(TGP)
      How can I transfer a call? [ 2012/09/04 ]
      What is the rule of Tone Timing for Tone settings? [ 2012/09/04 ]
      • The meaning of four parameters for tone setting is described as.
        "on 1, off 1, on 2, off 2" ( on=ringing term, off=silent term).
        When it does not contain "0" as a parameter, the setting means as follows.
        (Example: "100, 200, 300, 400")
          --> "100msec ring, 200ms silent, 300ms ring, 400ms silent" (and this pattern will be repeated)
        A parameter "0" means "infinite time". Therefore, if "0" is included in the parameters, the unit will keep on ringing or being silent from some timing (depends on where "0" is put in these four parameters.)
        (Example1 :"100, 100, 0, 0")
          -->"100ms ring, 100ms silent, keep ringing"
        (Example2 :"100, 0, 100, 0")
          -->"100ms ring, keep silent" (because the first "0" is effective infinitely)
      The displayed caller information is not correct. [ 2012/09/04 ]
      • Caller information may not be correct depending on your phone system.
      The SIP terminal does not ring. [ 2012/09/04 ]
      1. The ringer volume is turned off. Adjust the ringer volume.(TGP/UT1xx/UT670/UT248)
      2. Check the VoIP status in the Web user interface and confirm that each line is registered properly.(TGP/UT1xx/UT670/UT248)
      3. Check that the SIP server address, URLs of the configuration files, encryption key, and other required settings are correct.(TGP/UT1xx/UT670/UT248)
      4. Check the firewall and port forwarding settings on the router.(TGP/UT1xx/UT670/UT248)
      5. Check [Multi Number Settings] in the [Telephone] tab inthe Web user interface.(TGP)
      6. Check [Call Control] for each line in the [Telephone] tab in the Web user interface.(TGP/UT1xx/UT670/UT248)
        – If [Do Not Disturb] is set to [Yes], the base unit or handset does not receive calls.
        – If [Unconditional (Enable Call Forward)] is set to [Yes], the base unit or handset does not receive calls.
        – If [Block Anonymous Call] is set to [Yes], SIP terminal does not receive anonymous calls.
      7. Check that [Do Not Disturb], [Enable Call Forward], and [Block Anonymous Call] are not controlled by your phone system.(TGP/UT1xx/UT670/UT248)
      8. If 3 calls are being handled by the base unit and/or handsets simultaneously, a new call cannot be received and the unit will not ring.(TGP)
      I cannot hear a dial tone. [ 2012/09/04 ]
      1. Confirm that the Ethernet cable is properly connected.
      2. Network settings may not be correct.
      3. Many installation issues can be resolved by resetting all the equipment. First, shut down your modem, router, hub, base unit, and PC. Then turn the devices back on, one at a time, in this order: modem, router, hub, base unit, PC.
      4. If you cannot access Internet Web pages using your PC, check to see if your phone system is having connection issues in your area.
      5. Check the VoIP status in the Web user interface and confirm that each line is registered properly.
      6. Check that the SIP server address, URLs of theconfiguration files, encryption key, and other required settings are correct.
      7. Check the firewall and port forwarding settings on the router.
      What is the rule and format about SYSLOG? [ 2012/09/04 ]
    *SIP signaling
    *Application
KX-UDS124/UDT111/UDT121
  • *Installation
    • Is Tree Survey needed to replace previous multi DECT system? [ 2012/09/25 ]
      • When replacing Panasonic NCP0158 with UDS/UDT at the same position, Site Survey can be skipped. However, we recommend to do Tree Survey.
        In the case of replacement from other models, we strongly recommend that I carry out Site Survey and Tree Survey.
        Please refer the procedure to Installation Guide or document No.41-001 of technical contents.
      Basic procedure of Install for UDS/UDT series? [ 2012/09/07 ]
      Why is Tree Survey necessary? [ 2012/09/07 ]
      • "CS Registration" groups CSs into each air synchronization group and just confirms the transmissive area of reference clock signal from master CS.
        Therefore you need to decide the best and fail-over path of clock signal and share this information in all CSs. Additionally, 2nd Master CS is decided at this stage.
        Please refer to No.41-015 of technical contents.
      What is the difference in Tree Survey between Web user interface and CS Maintenance Tool? [ 2012/09/07 ]
      How many steps are the cascadable connection of CS? [ 2012/09/04 ]
      • It is possible to connect up to 8 steps.
      Is it possible to mix with other DECT systems? [ 2012/09/04 ]
      • We do not recommend.
      How can I supply power in the step of Site Survey? [ 2012/09/04 ]
      • Please supply power by AC adaptor(KX-A239), PoE or Battery(DC 9-12V).
      Can I assign IP address of CS by DHCP? [ 2012/09/04 ]
      • If the system consists of one air synchronization group, it is Yes.
        If the system consists of several air synchronization group, you must set static IP to the super master CS. And other CS's IP addresses can be assigned by DHCP.
      Though I input a PS name with web screen in the step of PS Registration, it is not registered. [ 2012/09/04 ]
      • Please set "All save" after inputing the PS name into the field of web screen.
      Can I register PS with pressing the "OK" button of several PSs at the same time for PS registration? [ 2012/09/04 ]
      • When registering multiple PSs, perform the registration procedure on each PS individually.
        Performing the registration procedure on multiple PSs at the same time may result in an error.
        In this case, reperform the registration procedure.
      How can I register PS to the system used by IPEI ? [ 2012/09/04 ]
      Where can Site Survey Tool or PC Maintenance Tool be downloaded? [ 2012/08/30 ]
      How to implement Pre-Provisioning by DHCP option 66? [ 2012/09/04 ]
      SIP phone doesn't complete provisioning after pre-provisioning. [ 2012/09/04 ]
      How can I configure provisioning setting by using Multiple configuration files like Standard, Master and Product? [ 2012/09/04 ]
    *Configuration
    *Configuration file
    *Troubleshooting
    • How to check firmware version? [ 2012/09/07 ]
      • PS firmware version with PS
           Turn on PS and then select menu according to following procedure;
           "MENU"->"Setting Handset"(OK)->"System Option"(OK)->"H/S Information"(OK)
        PS firmware version with Web UI
           Login to Super Master CS, and then select menu according to following procedure;
           "Status"->"PS Information"
        CS firmware version with Web UI
           Login to Super Master CS, and then select menu according to following procedure;
           "Status"->"CS Verion List"
      More than 33 PSs can't be registered to the system. [ 2012/09/07 ]
      • The number of PS per 1CS depends on "SIP Register Expire Time" of the server.
        Please refer to No.41-011 of technical contents.
        Even if you can't change the SIP register expire time setting, you can register PS to other CSs. In other way, you can register PS to same CS when registered PS does handover to other CS.
      I cannot hear Music On Hold (MoH). [ 2012/09/04 ]
      • As default, phone cannot play Music On Hold if Music On Hold is sent from server after server sent "inactive" SDP after hold had been placed by other party.
        We can change behavior by parameter.
        Please set the parameter
        HOLD_SOUND_PATH_n="1" by configuration file.
        The default value is "0" as phone play it's own hold tone.
        Example :
        HOLD_SOUND_PATH_1="1"
      I have registered SIP phone to SIP server successfully. I can make a call, but I can not receive a call. [ 2012/09/04 ]
      1. Please check you've turned on some parametes for security related settings as follows:
          SIP_RCV_DET_HEADER_n="Y"
          SIP_DETECT_SSAF_n="Y"
        By setting these parameters, phone may reject the call as security reason. If you turned on these paramters, please try following procedure:
        -Solution:
         Set up appropriate security function of TGP for each usage environment.
         (case1)
        When the user part of the To header in reception INVITE is different from your telephone number, change setting as follows.
          SIP_RCV_DET_HEADER_n="Y" ->"N"
         (case2)
        When the To header in reception INVITE is correct but you cannot receive INVITE, change setting as follows.
          SIP_DETECT_SSAF_n="Y"->"N"
      2. Please check if Phone is installed under NAT environment or not. If yes, please try following procedure.
         Some router may function to close the port which is used for NAT. So, we need to access to router periodically before NAT is closed and information is deleted.
        For accessing to router periodically, please change following setting.
          PORT_PUNCH_INTVL_n="0" -> "20"(example)
    *Operation/Features/Indication
    • Is it possible to update firmware during operation? [ 2012/09/07 ]
      • It is possible.
        CS reboots automatically to update new firmware, but it doesn't occur while CS communicates with PS.
        Please refer to No.42-001 of technical contents.
      Is it possible to update firmware during conversation? Can I make/receive a call during updating firmware? [ 2012/09/07 ]
      How to deregister all PSs at once? [ 2012/09/07 ]
      Voice Message Waiting Indicator does not turn on. [ 2012/09/04 ]
      • We can choose the type of SIP message which phone can accept for Voice Message.
        As default phone cannot recognize the Voice Message if there is no "Voice-Message" line included.
        As default : phone can accept following notification message for Voice Message.
        ----------------------------------
        Messages-Waiting: yes
        voice-message: 1/0 (0/0)
        ----------------------------------
        Phone cannot accept following, or different structure of "Voice-Message" line.
        ----------------------------------
        Messages-Waiting: yes
        ----------------------------------
        Please set the parameter if voice message notification does not have "Voice-Message"
        VOICE_MESSAGE_AVAILABLE="N" by configuration file.
      Current Call is disconnected when I answer the incoming call during calling from someone. Why? [ 2012/09/04 ]
      • We can choose whether disconnect the existing call or not
        when answering the second call by configuration parameter.
        Please set the parameter
        AUTO_CALL_HOLD="Y" by configuration file.
        The default value is "N" as disabled.
      What is the rule and format about SYSLOG? [ 2012/09/04 ]
    *SIP signaling
    • Is it possible to use TCP for SIP message? [ 2012/09/07 ]
      • Yes, it is possible.
      Can I register additional PS or change PS name after installing? [ 2012/09/04 ]
      • It is possible to name PS after registering to the system.
      I would like to have Automatic answer for intercom, click to dial. [ 2012/09/04 ]
      • We can use auto answer by including following options for making call from SIP server to phone as SIP INVITE message.
        -Auto answer (intercom or answer-after=0)
        example for intercom
         Alert-Info:intercom
        example for answer-after
         Call-Info:answer-after=0
      I would like to have distinguish ring tone. [ 2012/09/04 ]
      • We support following options for distinguish ring tone.
        Please use following into SIP INVITE message.
        Distinguish ring tone. (From Bellcore-dr1 to 5)
        Example:
        Alert-Info:
      I would like to use DTMF with DTMF Relay (SIP INFO). [ 2012/09/04 ]
      • You can use DTMF Relay (SIP INFO) by using following settings.
        Please set the parameter
        DTMF_RELAY_n="Y" by configuration file.
        The default value is "N" as disabled.
        Example :
        DTMF_RELAY_1="Y"
      There are some PBXs which cannot recognize SIP ACK message because Contact header is missing. Can we configure to have SIP message ACKNOWLEDGE? [ 2012/09/04 ]
      • Please set the parameter
        SIP_CONTACT_ON_ACK_n="Y" by configuration file.
        The default value is "N" as disabled.
        Example :
        SIP_CONTACT_ON_ACK_1="Y"
      How can I activate anonymous call feature? [ 2012/09/04 ]
      How to activate the Private Extension (RFC3325)? [ 2012/09/04 ]
      We'd like to monitor what kind of SIP message is going thru the phone. [ 2012/09/04 ]
      We'd like to see the statistical information of VoIP (RTP) packet information. [ 2012/09/04 ]
    *Application

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