By Default, Snom D717 has the smartlabel display mode set to "short". Snom D717 has an issue where for short labels it doesn't take the label that we configure in devices. Instead, it tries to look for the value in the system and if it doesn't find a contact or extension associated, it will just show the number itself.
* Fix sort on call broadcast display
* Grandstream Template Updates
For grp/gxp, added settings for intercom auto answer and multicast paging
For gxw42xx - Added off-hook auto-dial settings to all lines
For wp820 - Fixed incorrect P-codes and added setings for wifi roaming
Contributed new template for gxv3480
The template for codec enable settings was taking a value as 1 if it was set in the settings. It was not taking the actual value inside it. So I have edited the templates to check the value if it's set to true.
* Provisioning template changes
Grandstream
GAC2500 - Added account sip port to grandstream template after account server name
GRP2612 - Added Variable for grandstream dial timeout
GRP2612 - Added Variable for grandstream lock volume
GRP2612w - Added Variable for grandstream dial timeout
GRP2612w - Added Variable for grandstream lock volume
GRP2613 - Added Variable for grandstream lock volume
GRP2614 - Added Variable for grandstream dial timeout
GRP2614 - Added Variable for grandstream lock volume
GRP2614 - Added Variable for granstream call popup enabled
GRP2615 - Added Variable for grandstream dial timeout
GRP2615 - Added Variable for grandstream lock volume
GRP2616 - Added Variable for grandstream lock volume
GRP26xx - Added Variable for grandstream lock volume
GXP2160 - Changed template to default enable options keepalive
GXP2160 - Added option for multicast listen address with "if isset" for variable on P1569
GXP2160 - Added Variable for grandstream lock volume
GXP2160 - added variable grandstream_weather_enable to enable/disable weather function in phones
GXP2170 - Added Variable for grandstream dial timeout
GXP2170 - Added Variable for grandstream lock volume
GXP2170 - Added Variable for grandstream stun server
GXP2170 - Added Variable for granstream call popup enabled
GXP2170 - Added option for multicast listen address with "if isset" for variable on P1569
GXP2170 - added variable grandstream_weather_enable to enable/disable weather function in phones
GXP3240 - Added account sip port to grandstream template after account server name
HTek
UC903 - Added account sip port to htek template after account server name
UC903 - Added variable to specify account 1 ringtone
UC903 - Added ** as pickup code - Not defined
UC903 - Added variable for screen timeout
UC903 - Enabled call waiting tone P8850 - its off by default for some reason
UC903 - fixed variable P2 to set the admin password on the phone from default settings, was not defined
UC903 - Added variable to change the pc port mode if expansion module needed P231
UC903 - Fixed expansion module key types - not defined
UC923 - Added variable to specify account 1 ringtone
UC923 - Added variable to change the pc port mode if expansion module needed P231
UC923 - Added ** as pickup code - Not defined
UC923 - Added variable for screen timeout
UC923 - Enabled call waiting tone P8850 - its off by default for some reason
UC923 - fixed variable P2 to set the admin password on the phone from default settings, was not defined
UC923 - Fixed expansion module key types - not defined
UC924 - Added variable to specify account 1 ringtone
UC924 - Added variable to change the pc port mode if expansion module needed P231
UC924 - Added ** as pickup code - Not defined
UC924 - Added variable for screen timeout
UC924 - Enabled call waiting tone P8850 - its off by default for some reason
UC924 - fixed variable P2 to set the admin password on the phone from default settings, was not defined
UC924 - Fixed expansion module key types - not defined
UC926 - Added variable to specify account 1 ringtone
UC926 - Added variable to change the pc port mode if expansion module needed P231
UC926 - Added ** as pickup code - Not defined
UC926 - Added variable for screen timeout
UC926 - Enabled call waiting tone P8850 - its off by default for some reason
UC926 - fixed variable P2 to set the admin password on the phone from default settings, was not defined
UC926 - Fixed expansion module key types - not defined
Snom
D735 - Added code given to use by snom to light the BLF green when avail and red when in use
Fanvil
X210 - Added default setting for fanvil blf pickup code
X210 - Added if exist in template for fanvil x210 the pickup code var
* grandstream_stun_server variable already exists
---------
Co-authored-by: FusionPBX <markjcrane@gmail.com>
I was having an issue where my T41P phones were getting provisioned with IPv4 & IPv6 - even though I specified in default settings yealink_ip_address_mode=0 for IPv4 only. Checked the y000000000036.cfg, which was setting it correctly. The issue was that {$mac}.cfg was overriding the correct setting. There were several instances of "network.ip_address_mode = 2" in this file, which specifies ipv4&ipv6. I changed all 5 of those entries in this file to pull from the default setting {$yealink_ip_address_mode} rather than just setting a static value of 2. I don't know why it has this listed in this file 5 times. But, this change resolved my issue and will force the T41P provisioning to follow FusionPBX default settings.
* Fix the registration failed after provision Flyingvoice phone.
* Fix the parameter configuration of Flyingvoice not displayed in the Default Settings
* Fix: after configuring SIP line 1 of Flyingvoice phone, other SIP lines are disable and the configuration parameters of Flyingvoice are not displayed by default settings.
* Fix after configuring SIP line 1 of Flyingvoice phone, other SIP lines are disable.
* Fix the configuration parameters of Flyingvoice are not displayed by default settings.
* Fix: when only one sip line's shared line is enable, the shared lines of all SIP lines will be enable.
* Fix: unable to generate configuration file of Flyingvoice due to syntax error.
* Add the P1X, P2X, P3X, P5X, i86Box, iMetalBox, audioKit Series for Flyingvoice in Devices.
* Update all model templates of Flyingvoice.
Added a line to set SRTP encryption. the default setting yealink_srtp_encryption was not setting properly to my T56a devices. I downloaded my config files and did not find an entry for srtp at all.
compared with the t54w template and it did have a srtp setting. So I pretty much stole the line from the T54w template and dumped it in here.
The T54w template is vastly superior to this one and this probably needs a rewrite to bring it in line.
Tested on my T56a and it at least gets the job done.
* Fix the registration failed after provision Flyingvoice phone.
* Fix the parameter configuration of Flyingvoice not displayed in the Default Settings
* Fix: after configuring SIP line 1 of Flyingvoice phone, other SIP lines are disable and the configuration parameters of Flyingvoice are not displayed by default settings.
* Fix after configuring SIP line 1 of Flyingvoice phone, other SIP lines are disable.
* Fix the configuration parameters of Flyingvoice are not displayed by default settings.
* Fix: when only one sip line's shared line is enable, the shared lines of all SIP lines will be enable.
* Fix: unable to generate configuration file of Flyingvoice due to syntax error.
* Add the P1X, P2X, P3X, P5X, i86Box, iMetalBox, audioKit Series for Flyingvoice in Devices.
Add grandstream_missed_call_log, grandstream_missed_call_notification, grandstream_missed_call_backlight, grandstream_firmware_upgrade_protocol, grandstream_onhook_dial_barging, grandstream_transfer_mode_via_vpk, grandstream_enable_call_features.
Add condition on grandstream_distinctive_ringtone_name_1, and grandstream_distinctive_ringtone_name_2.
Update Connection request user/pw to work better with GDMS.
On grandstream_wallpaper_url, use current setting name on GRP2613
* allow setting hanging group mode on a per line basis for grandstream dp bases.
* Keep grandstream_hanging_group_mode for better backwards compatibility.
* Added two new minor hardware revisions, the CP925 and T42U, and added the voice.tone.country = {} to all templates, previously missing on most
* Minor typo from a wrong window issue, and removing the commented examples for easier grepping
* Update deprecated sip profile params
Parameter names have changed. See here: https://freeswitch.org/confluence/display/FREESWITCH/Sofia+Configuration+Files
When starting a profile, the following warnings would appear:
[WARNING] sofia.c:5332 rtp-hold-timeout-sec deprecated use media_hold_timeout variable.
[WARNING] sofia.c:5325 rtp-timeout-sec deprecated use media_timeout variable.
Updating the parameters fixes the issue.
* Update external.xml.noload
Co-authored-by: FusionPBX <markjcrane@gmail.com>
* Update deprecated sip profile params
Parameter names have changed. See here: https://freeswitch.org/confluence/display/FREESWITCH/Sofia+Configuration+Files
When starting a profile, the following warnings would appear:
[WARNING] sofia.c:5332 rtp-hold-timeout-sec deprecated use media_hold_timeout variable.
[WARNING] sofia.c:5325 rtp-timeout-sec deprecated use media_timeout variable.
Updating the parameters fixes the issue.
* Update external-ipv6.xml.noload
Co-authored-by: FusionPBX <markjcrane@gmail.com>
* Update deprecated sip profile params
Parameter names have changed. See here: https://freeswitch.org/confluence/display/FREESWITCH/Sofia+Configuration+Files
When starting a profile, the following warnings would appear:
[WARNING] sofia.c:5332 rtp-hold-timeout-sec deprecated use media_hold_timeout variable.
[WARNING] sofia.c:5325 rtp-timeout-sec deprecated use media_timeout variable.
Updating the parameters fixes the issue.
* Update internal-ipv6.xml.noload
Co-authored-by: FusionPBX <markjcrane@gmail.com>
* Update deprecated sip profile params
Parameter names have changed. See here: https://freeswitch.org/confluence/display/FREESWITCH/Sofia+Configuration+Files
When starting a profile, the following warnings would appear:
[WARNING] sofia.c:5332 rtp-hold-timeout-sec deprecated use media_hold_timeout variable.
[WARNING] sofia.c:5325 rtp-timeout-sec deprecated use media_timeout variable.
Updating the parameters fixes the issue.
* Update internal.xml.noload
Co-authored-by: FusionPBX <markjcrane@gmail.com>
This should be paired with a default setting to match. it controls the behavior of the headset if you make a call while it is connected. by default (0), yealink phones will not connect the call via the headset. if you set it to 1, the phone will immediately start using the headset for the call, instead of requiring you to press the headset button after dialing.
This should be paired with a default setting to match. it controls the behavior of the headset if you make a call while it is connected. by default (0), yealink phones will not connect the call via the headset. if you set it to 1, the phone will immediately start using the headset for the call, instead of requiring you to press the headset button after dialing.
This should be paired with a default setting to match. it controls the behavior of the headset if you make a call while it is connected. by default (0), yealink phones will not connect the call via the headset. if you set it to 1, the phone will immediately start using the headset for the call, instead of requiring you to press the headset button after dialing.
This should be paired with a default setting to match. it controls the behavior of the headset if you make a call while it is connected. by default (0), yealink phones will not connect the call via the headset. if you set it to 1, the phone will immediately start using the headset for the call, instead of requiring you to press the headset button after dialing.
This should be paired with a default setting to match. it controls the behavior of the headset if you make a call while it is connected. by default (0), yealink phones will not connect the call via the headset. if you set it to 1, the phone will immediately start using the headset for the call, instead of requiring you to press the headset button after dialing.
This should be paired with a default setting to match. it controls the behavior of the headset if you make a call while it is connected. by default (0), yealink phones will not connect the call via the headset. if you set it to 1, the phone will immediately start using the headset for the call, instead of requiring you to press the headset button after dialing.
This should be paired with a default setting to match. it controls the behavior of the headset if you make a call while it is connected. by default (0), yealink phones will not connect the call via the headset. if you set it to 1, the phone will immediately start using the headset for the call, instead of requiring you to press the headset button after dialing.
This should be paired with a default setting to match. it controls the behavior of the headset if you make a call while it is connected. by default (0), yealink phones will not connect the call via the headset. if you set it to 1, the phone will immediately start using the headset for the call, instead of requiring you to press the headset button after dialing.
This should be paired with a default setting to match. it controls the behavior of the headset if you make a call while it is connected. by default (0), yealink phones will not connect the call via the headset. if you set it to 1, the phone will immediately start using the headset for the call, instead of requiring you to press the headset button after dialing.
This should be paired with a default setting to match. it controls the behavior of the headset if you make a call while it is connected. by default (0), yealink phones will not connect the call via the headset. if you set it to 1, the phone will immediately start using the headset for the call, instead of requiring you to press the headset button after dialing.
* Update y000000000150.cfg
This should be paired with a default setting to match. it controls the behavior of the headset if you make a call while it is connected. by default (0), yealink phones will not connect the call via the headset. if you set it to 1, the phone will immediately start using the headset for the call, instead of requiring you to press the headset button after dialing.
* Grandstream HT802 Added Offhook Auto-dial variables
* Grandstream HT802 Added Offhook Auto-dial variables
* Grandstream GRP261x Make Configuration Keypad Lock configurable, fix default
* Grandstream Add grandstream_dhcp_time_zone across all templates
* Yealink Added yealink_missed_calllog to control whether missed calls are logged or not
* Grandstream HT802 update variable name to not use dash (which was breaking the template)
* added default settings for grandstreams
added "grandstream_configuration_via_keypad" and
"grandstream_dhcp_time_zone" as settings
also changed "grandstream_long_label" to 1
* Add custom transcription providers to email queue, fix spelling error
* Correct missing $ in Grandstream template
Co-authored-by: Jesse Gruver (piajesse) <me@piajesse.com>
* Added varibles to grandstream configs for idle mute fuction
I have changed them all to the default of "0" and then added a
varible called "grandstream_idle_mute_function" that overwrites
the default option of 0
There seems to be older configs where 0 = no instead of dnd
so I've added grandstream_idle_mute_function_old for these configs
* Added curls for grandstream_idle_mute_function
* Grandstream HT802 Added Offhook Auto-dial variables
* Grandstream HT802 Added Offhook Auto-dial variables
* Grandstream GRP261x Make Configuration Keypad Lock configurable, fix default
* Grandstream Add grandstream_dhcp_time_zone across all templates
* Yealink Added yealink_missed_calllog to control whether missed calls are logged or not
* Grandstream HT802 update variable name to not use dash (which was breaking the template)
* Add grandstream_label_background varable
Allows the label background to be toggled
* Add ability to set static ipv4 for gxp2130_40_60_70_35
* Add ability to set static ipv4 for dp750
* Update {$mac}.xml
<P35> expects user_id not display_name
Changes to make the templates a bit more consistent while also fixing some typos and creating a new variable for the NAT Update setting.
By default, the account.x.nat.udp_update_enable is set to "3" which means to have the phone periodically send NOTIFY messages. This is not the case though on additional accounts. This makes for inconsistent configs that sometimes cause issues with phones ringing.
This fixes the issue while also giving the ability to use a variable to set it.
FusionPBX
Mihir Gajjar
Zachary Kitchen
Finn
jonathanblack1000
Krushali Shah
Finn
Mendel
Norman King
Antonio Fernandez
David Jacob
Ben
levishores
zrd740
Vincent
garryavx
hershyheilpern
Marvin Skeete
Dan
Alex
Joyce Babu
yois615
Harry Foster
emak
Jesse Gruver
Anthony
ggflow
chansizzle
agree
shing6326
KRooney