GXP Business Phone Series

While upgrading/provisioning via TFTP, I’m not receiving a response from the TFTP server. Why?

GXP series phones can be upgraded and provisioned via TFTP. First, a local TFTP server needs to be installed on the PC. The computer running the TFTP server and the phone being upgraded/provisioned should be on the same LAN segment. Additionally, for Windows users, please try disabling the Windows Firewall.

For more steps on upgrading via TFTP, refer to the “Instructions for Local TFTP Upgrade” section in the GXP user manual.

Is the availability of certain features dependent on the connected PBX? Which ones?

The end user can choose to enable advanced calling features either locally on the phone or through the PBX (or server) to support them. To enable or disable this feature, use the GUI Account Page settings. Call features include call forwarding (all types), transferring (all types), call waiting, mute, DND, and conferencing.

Note: In the Web GUI -> Account Page, when “Enable Call Feature” is set to “No,” the phone’s LCD will no longer display the “Send All” softkey (for local feature code).

Does it display call duration or voicemail count?

During an active call, the call duration is displayed. Each line has a separate voicemail, and the voicemail count is shown for each line. Call duration and voicemail count are supported by all GXP models.

How many entries can be stored in the address book?

GXP21xx supports 2000 entries in the phone book. GXP14xx supports 500 entries in the phone book. For GXP20xx, it supports up to 500 entries in the phone book. XML synchronization depends on the file size and available memory allocation.

How many entries are allowed for call history, such as dialed, missed, etc.?

GXP21xx supports a total of 2000 entries in the call history. GXP1450 supports a total of 500 entries in the call history. GXP140x supports a total of 200 entries in the call history. For GXP20xx, the call log retains up to 50 entries per call type (Answered Calls | Dialed Calls | Missed Calls | Transferred Calls).

How can I create an XML configuration file for GXP21xx/14xx/110x provisioning?

Users can provision GXP21xx/14xx/110x by using the configuration tool, template, or by providing an XML file. Users need to create an XML file named cfgxxxxxxxxxxxx.xml (where xxxxxxxxxxxx is the phone’s MAC address) using the configuration template’s P values.

How can I reset GXP21xx/14xx/110x phones?

For GXP21xx/14xx, navigate to LCD -> MENU -> Configuration and select “Factory Reset” to reset the phone. For GXP110x, lift the handset, dial *** to access the IVR menu, enter 99 for the “Reset” option, then enter the MAC address of the GXP110x. The phone will restart upon correct entry; otherwise, it will return to the main menu.

How can I access voicemail using GXP21xx/14xx/110x?

When someone leaves a voicemail, the voicemail waiting indicator (LED on the top-right corner) will blink red. There are two methods to access voicemail:

  1. Dial the voicemail access code (supported by PBX).
  2. In Web GUI -> Account Page, enter the voicemail access code in the “Voicemail User ID” field. Then, press the Message key on the phone (for GXP140x, a voicemail screen key will appear for new voicemails).

Note: Some PBX systems may require setting “Enable Call Features” to “No” in Web GUI -> Account Page to access voicemail. Follow the system prompts for authentication and IVR options to listen, save, or delete voicemails.

How do I configure an “event list BLF” on GXP series for BLF functionality?

If the server supports it, configure an “event list BLF” URI on the server side (e.g., [email protected]). On the GXP, enter the “event list BLF URI” field under the account settings without the domain (e.g., BLF1006). In Basic Settings -> Multi-Purpose Key x, choose “event list BLF” for the key and select Account, Watched BLF Name, and Number.

How can I convert my .csv phone book to an acceptable XML format?

GXP series phones accept phone book files in XML format. To use a CSV phone book file, the end user needs to convert it to an acceptable XML format beforehand.

How can I disable weather, stock, and currency applications?

Go to Web GUI -> Basic Settings and modify the following options:

  • Enable Weather Update: No
  • Enable Stock Update: No
  • Enable Currency Update: No

To change this in the configuration file, use PValue like this:

  • Weather: 1402=0
  • Stock: 1403=0
  • Currency: 1404=0

Note: GXP140x/1450 only supports weather updates.

How do I use Distinctive Ring Tone with the ALERT INFO method?

The GXP series phone supports mapping three custom ringtone files to Alert-Info. For example, if you configure the Matching Caller ID to “priority,” and you receive an INVITE with the Alert-Info header as follows: Alert-Info:;info=priority, the corresponding ringtone will be used.

How can I use the call or intercom feature on the GXP series?

For GXP20xx, you need to set the following two fields (Account Page) to Yes on the client side:

Allow Auto Answer by Call-Info.
Turn Off Speaker at Remote Hangup.
For GXP21xx/GXP14xx, the following field (Account Page) should be set to Yes on the client side:

Allow Auto Answer by Call-Info.
Note: The PBX server needs to support this feature for it to work.

How can I utilize the PC port on GXP21xx/14xx for switching purposes?

The PC port on GXP21xx/14xx allows the device to act as a switch. After setting up GXP21xx/14xx, connect a PC or another phone to the PC port of GXP21xx/14xx to access the Internet. This way, you can connect PCs and other devices to your network without using a router or switch.

When configured with “SIP Server” under Web GUI->Account settings, how does the “Secondary SIP Server” work?

The “Secondary SIP Server” field for GXP21xx/14xx/11xx includes the URL or IP address of a second SIP server. When configured, the phone will send Register Requests and Subscription messages (except for message waiting) to both “SIP Server” and “Secondary SIP Server” for the same account.
During a call, the phone will first use the registered primary “SIP Server.” If the primary is unavailable, the registered “Secondary SIP Server” will be used. If the primary is not registered but the “Secondary SIP Server” is, the phone will directly use the “Secondary SIP Server.”

Note: Please avoid configuring duplicate SIP Server addresses in both “SIP Server” and “Secondary SIP Server.”

Does call recording work? Does it require additional devices?

If the server or PBX supports call recording, we can support it. The GXP Series does not require additional add-on devices to support call recording. Call recording is supported by all models if the server or PBX supports this feature.

The keypad on my phone is locked with a “Lock” icon on the LCD. How can I unlock it?

On GXP21xx, when you press and hold the STAR * key for about 4 to 5 seconds, the keypad will be locked. To unlock, press and hold the STAR * key for 4 to 5 seconds again. A window will appear to enter a password, followed by the message “LEFT for BACK, MENU for OK.” Enter the password (default is blank), then press the MENU key (the round button in the middle of the navigation buttons). The keypad will now be unlocked.

If you wish to disable the STAR key Keypad Lock feature, log in to the phone’s web GUI, go to the Advanced Settings page, and configure the following two options:

Enable STAR Key Keypad Lock: No
Password for Locking/Unlocking: Leave blank
Save the settings from the Web GUI and reboot the phone.

The GXP Series supports multiple languages. Which languages are available?

GXP21xx/GXP14xx supports languages including English, Spanish, German, French, Polish, Italian, Arabic, Hebrew, Croatian, Hungarian, Japanese, Korean, Dutch, Polish, Portuguese, Russian, Slovenian, Simplified Chinese, Traditional Chinese, and more as requested. GXP20xx has a language pack supporting English, Spanish, German, French, Polish, Italian, and others as desired. The language pack can be found in the Resources section and is regularly updated.

Does GXP21xx support SRTP? How can we use it with Asterisk 1.8?

GXP21xx supports SRTP with server-side support, and Asterisk 1.8 comes with the SRTP feature. To use GXP21xx with Asterisk 1.8, you need to enable SRTP on the server side and apply a patch allowing the crypto lifetime in SDP to be ignored. Please refer to the following steps:

Register the Asterisk account to GXP21xx. Under the web GUI -> Account settings, select “Enable SRTP and Set as Mandatory.” Then choose the supported codec component.

Apply a patch on the Asterisk server where the Asterisk source code is compiled, in the sip.conf file, to allow the “ignorecryptolifetime=yes|no” option.

In the sip.conf file, set ignorecryptolifetime=yes for extensions on the GXP21xx for using SRTP.

For example:


You should now be able to make calls using SRTP.

I can be heard by the other party, but I can’t hear them. Why?

In this case, please ensure that the handset is securely connected. If not, reattach it. This issue could also result from the following:

The other party might have pressed the “Mute” button, which could silence the call. Ask the other party to check if their mute is on and to unmute the call.
If your phone is behind a router for Internet connectivity, it could be a NAT traversal issue. If GXP21xx is on a LAN and needs to register with a SIP server on a public IP, enable NAT traversal by selecting the appropriate NAT traversal method based on your network environment. If unsure, it’s recommended to select “Auto” to enable automatic NAT traversal configuration.
Which configuration changes in the web GUI require a reboot?

Basic and Account settings generally do not require a reboot (“Secondary SIP Server” being the exception). Changes under Advanced settings that would require a reboot include:

Administrator Password

STUN Server
Download Phone Book XML
Offhook Auto Dial
Call Progress Tones
Custom Ring Tone 1 – 3 if set for a caller ID
Intercom User ID
Disable Call-Waiting
Disable Call-Waiting Tone
Disable Direct IP Call
Use Quick IP Call Mode
Disable Conference
Enable DTMF Sending MPK
Disable Transfer
Auto Attended Transfer
Display Language

Why do I see a strange symbol, a down arrow inside a square, going to [_] in my web GUI?

This symbol indicates that the phone is writing the call history detail file. It occurs when the phone has been idle for about 5 minutes or when the call history reaches more than 100 entries.

Why can’t my GXP21xx device register with the SIP server?

Ensure the phone is connected to the network and has acquired an IP address. Secondly, check if the account is set to “Active” by configuring “Account Active” on the Account page of the web interface. Also, verify the login credentials and SIP server details. If the SIP server is incorrect, the phone can’t communicate with the server for registration