The Polycom Phones module for FreePBX 2.11 has been updated to include additional network based settings with codec priority / authentication options, phone override support, additional attendant options, and flexible line key assignment. Read more…
Cox Communications provides a SIP trunk service through an EdgeMarc gateway installed at your office location. This article covers the configuration of FreePBX to connect to the EdgeMarc gateway and send calls over the Cox SIP trunk. Read more…
The Polycom Phones module for FreePBX 2.11 has been updated to include additional phone options, alert info settings, contact directory editor, and AD / Exchange integration settings.
Refer to Polycom Phones 2.11.0.0 UC Provisioner for FreePBX 2.11 for additional screenshots and recommended DHCP and FreePBX settings. Read more…
The Polycom Phones module for FreePBX 2.11 allows for quick and easy provisioning of phones running the Polycom UC software.
Before creating this module I had looked at the OSS PBX Endpoint Manager and the Commercial Endpoint Manager. Development on the OSS module seems to have slowed and the commercial module doesn’t support FreePBX device and user mode.
Using the idea of the Digium Phones module I decided to create a module specifically for Polycom phones, which removes the need to edit XML configuration files. Read more…
This article assumes that you have an Exchange UM server already configured. The directions cover the use of a module I created for FreePBX 2.11 and assume FreePBX is in device and user mode. The module removes the need to override macro-vm, which allows you to use Exchange UM for some users and FreePBX voicemail for others.
NOTE: Support for the MWI (message waiting indicator) and play on phone requires patching and compiling Asterisk from source. The patches were created against Certified Asterisk 11.2-cert2. Read more…
After upgrading to Asterisk 1.8 from 1.6.1 I noticed the CALLERID() function was not updating the clid and src fields in the CDR. Previously on Asterisk 1.6.1 with FreePBX 2.7 the clid and src would be set to the outbound cid. With Asterisk 1.8 those fields were staying as the sip device id. Read more…
With a traditional PBX when dialing an extension the name is displayed on the phone. With Asterisk 1.4 and 1.6.x the name is not displayed even though phones, like the Polycom SoundPoint series, have support for it with the Remote-Party-ID SIP header. Read more…