Recently we had a customer with this cisco conference phone (7937) which doesn’t support SIP protocol. To get it to work with Asterisk we need to either use chan_skinny or chan_sccp. As the latter is a lot more solid is brief you how it work here:
Recently I had a customer who wanted to push all VoiceMail voice files into a table in MSSQL database. After going through a lot of searching and trying I managed to get it running like this:
I wrote the following script to kick out logged in users to Asterisk. (FreePBX deviceanduser mode)
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.
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.
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:
To get Skype for SIP (Skype Connect) to work with FreePBX follow these steps:
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.
