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
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