I wrote the following script to kick out logged in users to Asterisk. (FreePBX deviceanduser mode)
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
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.
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.
Here is the Asterisk AGI to play Farsi numbers. The script is meant to play numbers as big as 1000, so for bigger numbers you need to do some simple modifications.
Recently we had a pack of cisco 7942G phones that we were required to get them up running with Asterisk. The good thing about 79XX series is that they all support SIP besides SCCP. Whereas, the bad thing is that they are by default running on SCCP and you have to upgrade them to SIP first. I spent a great deal of time trying to figure this out, as I were going to use SIP Version 8 and there is no to-the-point documentation (at least not that I could find!)
Here is the simple postfix configuration to make it work with freepbx.
Based on a customer request, we needed to have some virtual extensions (Voice Mail enabled) but the problem was that they wanted the incoming calls to extensions to ring 4 times before falling back to voice mail. Since by default FreePBX will get the extension status as UNAVAILABLE and I couldn’t get virtual extension running with this version of freepbx, i did this trick: