Are you having an audio issues in your Asterisk?
Well it’s a common issue with PBX to have audio issues like one way audio or no audio. Sometimes only caller can hear remote party or remote party only can hear the caller. You must be wondering what causes this issue? This problem in audio is mainly because of the NAT issues. We recommend to use NAT with enabling 10000-20000 UDP ports on firewalls and also to enable natting on trunks
Learn how you can limit number of simultaneous calls on Asterisk based SIP Trunk. If you want to limit the number calls for your SIP peer or friend in Asterisk use call-limit in your trunk configuration.
This is a very common requirement that route the calls to Voice-mail after office hours. Or you can transfer the calls to your cell phone after certain time say 6:00pm. In Asterisk you can also control the call location based on time and date.
Asterisk can be used to originate calls from a web page. Asterisk Manager Interface (AMI) allows you to manage call origination. AMI also allows external programs to control Asterisk.
We have put together a list of dialplan functions that you can use to count calls from Asterisk Dialplan.
Find most frequently used asterisk commands. This is simple cheat sheet to view what’s happening inside your asterisk server. You can see how many calls, how many users who is communicating with your system.