Grandstream GXP1160 IP Phone Firmware 1.0.8.4 Beta Download

Posted at November 14, 2024 by PCI-DB Team

Grandstream GXP1160 IP Phone Firmware 1.0.8.4 Beta 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 re-plugging 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

NOTE:

- Once upgraded to 1.0.7.8 and above, downgrading to any previous firmware version is not supported

  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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