* 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
* config->exists returns true or false but no action taken and variable
is unused
* remove unused variable db_type
* remove unused variable db_name
* remove unused variable db_username
* remove unused variable db_password
* remove unused variable db_secure
* remove unused variable db_cert_authority
* remove unused variable db_host
* remove unused variable db_path
* remove unused variable db_port
* remove unused variable db. The upgrade method never uses the db variable
and instead uses a new database connection each time.
* remove uninitialized parameters variable
* domain_count never used
* variable context is never used in the method or any app_defaults
* variable $row is clobbered by inner foreach loop
* variable domain_name is never used in method scope
* variable domain_array seems to be uninitialized in this scope so set an
empty string value so function lower_case is not receiving null.
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>
* Save the email response
* use the response variable from the email class
* Rename email_debug to email_response
* Update app_languages.php
* Show the email response
* Save the email response
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
* Add new static method to created newly connected database object
* Document database class and clean up and document some of the methods.
This removes the methods that should not be in each instance and places
them in the single instance class as to occupy less resources and be
able to create database objects more efficiently.
* More docs & removed the ability to set any value within the object.
Co-authored-by: Tim Fry <tim@voipstratus.com>
* allow setting hanging group mode on a per line basis for grandstream dp bases.
* Keep grandstream_hanging_group_mode for better backwards compatibility.
30,000 Microseconds = 0.03 Seconds. Longer timeout reduces the CPU. If the timeout is too long then the Status -> SIP STATUS page will take longer to load.
* 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.
The expansion module can use a different background image
The wallpaper_upload.url setting in yealink can be used multiple times to download multiple images.
setup for 2 more variable:
$yealink_t46u_wallpaper_expansion
and
$yealink_t46u_wallpaper_expansion_filename
This is only a template update. I did not push any additional varaibles up into the default settings.
The reason I think this should be removed is because setting the label on this model of phone removes the ability to see status of the line button.
For example (without the label), if I have a call on hold on my first line button, it will show the caller's number and the state of that call. For example: "4165551234 holding"
When you put the label, it removes that great feature.
* Create directory.xml
* More Yealink fixes
-Spelling
-Wrong var
-Updated power_saving schema to be uniform across all devices to best written variable logic. - Good job whoever built it. IF YOU HAVE SET THIS UP IN THE PAST IN THE OLD FORMAT IT MAY BREAK. $yealink_powersave_* vars have been depreciated in favor of $yealink_ps_* since logic has been built to allow more with less total vars.
-Added some documentation
* Add Polycom provisioning default settings, and update provisioning templates to support for additional polling options for supported versions (4.x, 5.x, 6.x)
* semicolons
* Add grandstream_dnssrv_transport setting to GRP and GXP21xx templates
Add grandstream_dnssrv_transport setting to enable setting of transport when dns/srv is selected. Added on GRP and GXP 21xx templates.
* Update app_config.php
Co-authored-by: FusionPBX <markjcrane@gmail.com>
* 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.
* 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.
nway_conference == true was evaluating as true regardless if nway_conference was set to true or false, just as long as it had a value of any kind.
This also fixes it so that you can set nway_conference to false in the default settings and it will disable the network conference feature, previously once it was enabled on yealink phones it could not be disabled without manually changing it in the phone or a factory default.
* Support for Disabling Non-configured ports on GXW42XX gateways
When patching in an ATA at facilities like residential facilities, we don't want dial tone on un-configured ports. This makes it so that the device will disable ports that have no user id configured so they don't provide dial tone. The behavior can be reversed by setting grandstream_fxs_enabled to 1 in the settings.
I fixed the voicemail access number (this was hard coded to *98)
Removed unsupported P values and added many hidden values that cannot be configured on the app but are present in the default app config file
TLS support (you can now easily toggle TLS using default settings)
Added grandstream_config_server_path, http_auth_username/password (useful if you want to point to another PBX)
* Add Grandstream Headset/Speaker Ring
Add variable to choose whether the speaker rings with the headset.
* Add Speaker Only option
Add note that speaker only is supported with a value of 2.
FusionPBX
fusionate
frytimo
Anthony
jonathanblack1000
Krushali Shah
Finn
Mendel
Norman King
Antonio Fernandez
David Jacob
Ben
agree
levishores
zrd740
Alex
Vincent
demonspork
garryavx
hershyheilpern
Marvin Skeete
Dan
Joyce Babu
yois615
Harry Foster
emak
Jesse Gruver
nacircoronado
Luis Daniel Lucio Quiroz
ggflow
nyash
chansizzle
shing6326
KRooney
TNSSF
Timothy Willey
smeyernyc
Andy Binder
Jason Kinner
AlexC
Vincent
Ricardo Cooper