Grandstream GXP1450 IP Phone Firmware 1.0.8.4 Download

Posted at December 22, 2024 by PCI-DB Team

Grandstream GXP1450 IP Phone Firmware 1.0.8.4 Description

BUG FIXES:

- Fixed phone crash issue associated with SIP OPTIONS
- Fixed phone sends MOH URI INVITE with incomplete From header
- Updated Hebrew language missing strings
- Fixed Secondary SIP Server was modified successfully after turning on provider lock
- Fixed SIP TLS Private Key and Certificate cannot be provisioned using XML file
- Fixed phone will use random port as REGISTER and INVITE port after unplugging and replugging network cable
- Fixed phone cannot receive incoming calls under certain conditions
- Fixed phone freezes on conference page after long time conference on PCMU
- Fixed phone behaves abnormally if entering audio loopback when enable HEADSET Key Mode
- Fixed Phonebook export/upload are accessible without authentication request
- Updated New Zealand time zone
- Updated Russian Time Zone settings
- Fixed phone does not stop the RTP after BYE and get mixed on subsequent calls
- Fixed changing configuration does not take effect under certain conditions
- Fixed rejecting a call on "Audio Loopback" page would result in sound lost
- Fixed volume setting is incorrect if adjusting it under headset mode
- Fixed phone fallbacks to PCMU upon re-invite during the session
- Fixed phone crashed after switching audio mode during a call
- Fixed DNS-SRV connection failed if first option is unavailable using TCP
- Fixed Phonebook containing Cyrillic characters is not sorted by alphabetical order
- Fixed phone cannot enter Audio Loopback when it is in headset mode
- Fixed audio channel in audio loopback is error after enabling Toggle Headset/Speaker mode
- Fixed there was no audio after 14min conversation when using SRTP enabled and forced
- Fixed phone crashes when configured with long dial plan after registering to SIP server
- Fixed phone drops calls on re-INVITE
- Fixed no dial tone and unable to make or receive calls after weekend
- Fixed SRTP with TLS SIP Transport - no voice speech path after secondary server is disconnected while having an active call on primary server
- Fixed TCP port 4 is open when logging in from phone UI
- Fixed no language cursor change when switching Auto/Downloaded to the language same as current one in LCD mode
- Fixed failed to change LCD language after factory reset
- Fixed pressing Line key when phone is downloading phonebook will lead abnormal display on LCD
- Fixed there is no audio channel icon and no sound on audio loopback page after entering and exiting audio loopback page continuously
- Fixed LCD display is abnormal when transfer a call
- Fixed phone plays busy tone in audio loopback mode after remote party ends the call
- Fixed failed to dial Voice Mail via MPK in dial page
- Fixed audio does not stop after sending number with * and no call feature is set

ENHANCEMENTS:

- Added option to ignore Alert-Info header when used for distinctive ringtone
- Added dual Outbound SIP Proxy support
- Added support for sending SIP Option messages to verify connectivity to the SIP server
- Added option to select Call Pickup mode
- Added support of OPTION 160
- Enabled the HTTP(S) ID/Password prompt
- Added option to disable the Call park subscription
- Added support to check the status of registration to Action URI

  It is highly recommended to always use the most recent driver version available.

Try to set a system restore point before installing a device driver. This will help if you installed an incorrect or mismatched driver. Problems can arise when your hardware device is too old or not supported any longer.

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