I wrote the following script to kick out logged in users to Asterisk. (FreePBX deviceanduser mode)
kickout script for Asterisk Hot desking
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Grandstream IP Phone auto-provisioning Template
Here is a php file that will read the list of IP phones from a CSV file including MAC address, host IP address, Extension and Secret. Then it will generate a binary for each cfgMAC.temp file.
Download the file here
Set Date/Time in Linux
Here are the steps to set date and time in linux
IP Phone provisioning option 66 and 43
Different IP Phones require different methods of provisioning. Below I describe how to do this on Grandstream phones (tested on GXP285) with firmware: 1.2.5.3
Create a startup service in CentOS
Here is a sample on running a service to startup in centos:
A2Billing configuration on FreePBX
Here is the step by step instruction on how to configure a2billing with FreePBX 2.4:
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Hotline calls (PLAR) with Asterisk
Hotline/Warmline, also known as Private Line Automatic Ringdown (PLAR), is a line used for priority telephone service. When using with a SIP phone you must make sure the phone supports this feature. As of this writing I know that Polycom SIP phones support sending a signal to Asterisk upon the offhook event.
Cisco 7912 with Asterisk
Asterisk phone lock/unlock feature
Below is a simple and intuitive dialplan by which you can add the phone lock feature to your Asterisk system. Take note that the provided dialplan only locks the phone from calling out to PSTN.
Cisco ATA 186 + Asterisk
The Cisco ATA 18X series can run either SCCP, H323 or SIP voice protocols making it very flexible.
Another note of importance is that they do not support two ports running the G.729 codec simultaneously. If one port is using G.729, the other will use G.711. In this manual we are using a SIP firmware (ata_03_02_04_sccp_090202_a). If you want to update to other firmwares or perhaps protocols, I suggest you read the last 2 links provided in the resources section.