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Posts Tagged ‘VOIP’

Cox SIP Trunk for FreePBX Configuration Guide

November 25th, 2013

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…

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Polycom Phones 2.11.0.2 UC Provisioner for FreePBX 2.11

November 11th, 2013

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…

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Polycom Phones 2.11.0.0 UC Provisioner for FreePBX 2.11

October 30th, 2013

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…

Asterisk, VOIP , ,

Exchange 2010 UM with FreePBX 2.11 and Asterisk 11

October 14th, 2013

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…

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Asterisk 1.8 Caller ID Wrong in CDR

October 23rd, 2011

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…

Asterisk, Linux, VOIP , ,

Asterisk Called Party Name

July 2nd, 2010

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…

Asterisk, Linux, VOIP , , ,

Polycom Soft Keys and Exchange UM

March 6th, 2010

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…

Asterisk, Linux, VOIP , , ,