* 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.
FusionPBX
frytimo
Joyce Babu
Norman King
yois615
Harry Foster
Mendel
emak
garryavx
Antonio Fernandez
Jesse Gruver
Dan
nacircoronado
Ben
Anthony
AlexC
Luis Daniel Lucio Quiroz
ggflow
Marvin Skeete
nyash
chansizzle
agree
shing6326
fusionate
KRooney
TNSSF
Timothy Willey
smeyernyc
jonathanblack1000
Andy Binder
Vincent
zrd740
Jason Kinner
AlexC
Vincent