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…
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…
In Asterisk 1.6, FreePBX, and Exchange UM I mentioned how to setup an extension to transfer a caller to voicemail. Let’s take this one step further and add a soft key on the Polycom. Instead of having to dial ##407ext, you will be able to press the Xfer VM button and it will prompt for the extension. Read more…
They are two methods of setting up BLF on Polycom phones. The first gives you idle or inuse status only. A ringing phone will show up with a solid inuse light. The good thing about this method is users can create BLF for extensions themselves. If this is all you need start by enabling the presence feature in the phone provisioning file. Read more…
Setting up Asterisk 1.6.1 and FreePBX 2.5 or 2.6 to work with Exchange 2007 UM is easier than Asterisk 1.4 thanks to Asterisk 1.6 including support for SIP over TCP. However only Asterisk 1.6.1.4 and lower work without modification. Read more…