FreePBX SIP Trunk Configuration

For creating a sip trunk between didforsale and your FreePBX system, first create a sip account from your didforsale account. For creating the sip account, log in to your didforsale dashboard, go to Interconnection > Manage SIP Accounts and then click Add New SIP...

Changing Default SIP Port in Asterisk

Asterisk by default use 5060 as its sip signalling port. It is a good idea to change the default sip port as most of the SIP vulnerable attacks occurs on its default port 5060. To change the SIP port, open /etc/asterisk/sip.conf in your favorite text editor, look for...

MATH Dialplan Function in Asterisk

Asterisk provides the MATH function to do mathematical operations from dialplan. It allows to perform mathematical operations between two parameters. The syntax for math function is MATH(expression,type) The operators supported by math function are...

Hangup Active Calls from Asterisk CLI

Asterisk CLI provide Hangup command to hangup live calls. For using the hangup command, you need to get the name of the channel that you want to hangup. Use the below command to get all the active channels in your asterisk server. core show channels This command will...