There are many occasions where you need to have some sort of communication between your Asterisk system and an external system, be it another telephony platform or external application. The beauty of Asterisk is it’s superb flexibility in terms of collaboration or integration with external phone systems or applications. One common scenario is integration to Avaya system. Here in this post I try to highlight some of the tips and tricks in order to establish a successful communication with Avaya.
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: