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.
For configuration you can refer to the following 2 links
In order to add Persian language we need to do some modifications on the Asterisk source code. For that, download the same Asterisk version as your current one (in case you’re using FreePBX/Trixbox). The main thing to change is the Asterisk say digits. Here is what must be changed:
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.
Here are the steps on how to push the Cisco phone 7942G to reboot.
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!)
Over time Asterisk log can become so heavy and space consuming, not to mention the time you have to spend reading through tons of lines of log files. The way to organise the log files is to rotate them using logrotate facility.