The Phone Features module for FreePBX implements feature key synchronization of the do not disturb (DND) and call forward buttons with FreePBX feature codes. The module requires Asterisk to be patched and the phone to have server feature event keys enabled. If you are using the Polycom Phones module the DND and call forward client and server options will appear next to each line.
The module is released under the GPLv2 license and distributed without any warranty. Support will only be provided to customers of Excalibur Partners, LLC.
- FreePBX 2.11, 12, 13
- Asterisk 1.8 (depreciated), 11 or 13
- Asterisk: Must be patched and built from source.
- Polycom Phones module 184.108.40.206 or higher
Asterisk Server Feature Events
Asterisk Jira issue 13145 has a patch that adds support for Cisco phone DND and call forward synchronization along with presence and other features. In order to limit the amount of code changes I have rewritten the patch to only include changes necessary for server feature events to work.
The patch adds two Asterisk CLI commands "sip donotdisturb on/off peer" and "sip callforward on/off peer number" along with the dialplan variable SIPPEER(peer, donotdisturb) and SIPPEER(peer, callforward).
I have also included hooks to execute dialplan code when a DND or call forward event is received from the phone or CLI. This allows the module to sync with FreePBX feature codes and users that have multiple devices. Currently the hooks are hard coded for DND events to call context sip-feature-events-dnd with extension on or off and for call forward events context sip-features-events-cf with the extension of the forwarding number or off.
The patch has been tested with Polycom phones but should work with other phones that support the server feature event synchronization like Cisco, Snom, and Yealink.
Asterisk Releases Tested
- Certified Asterisk 1.8.15-cert5
- Certified Asterisk 11.2-cert2
- Certified Asterisk 11.6-cert4
- Certified Asterisk 11.6-cert13
- Certified Asterisk 13.8-cert3
Polycom Phones and UC Software Tested
- SoundPoint IP 335 4.1.0 Rev I PolycomSoundPointIP-SPIP_335-UA/220.127.116.11959
- SoundPoint IP 550 4.1.0 Rev I PolycomSoundPointIP-SPIP_550-UA/18.104.22.168959
- VVX 300 4.1.7 PolycomVVX-VVX_300-UA/22.214.171.1240
- VVX 400 4.1.7 PolycomVVX-VVX_400-UA/126.96.36.1990
- VVX 300 5.5.0 PolycomVVX-VVX_300-UA/188.8.131.5256
- VVX 311 5.5.0 PolycomVVX-VVX_311-UA/184.108.40.20656
- VVX 400 5.5.0 PolycomVVX-VVX_400-UA/220.127.116.1156
- 18.104.22.168 - Initial release
It you would like to contribute bug fixes or additional functionality the source code is available on GitHub.