* 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
* 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>
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.
There was an illogical if/else statement that the "else" condition would always set a null value, so it didn't really matter. Re-wrote it to properly filter if the transport is set to dns srv and not affix the SIP Port to the server address.