Hangup Active Calls from Asterisk CLI

SIP Trunks for Asterisk Try Now Compatible PBX & VoIP Phones check out Flexible SIP Trunk Pricing check out Hangup Active Calls from Asterisk CLI Asterisk CLI provides Hangup command to hangup live calls. For using the hangup command, you need to get the name of the...

Configure SIP Trunk on Grandstream PBX

For Configuring Grandstream PBX with didforsale, you need to create four sip trunks. Two trunks for incoming calls and two trunks for outgoing.   For creating trunks, go to PBX => VoIP Trunks => Create new SIP Trunk. Add the details as shown in below figure  ...

Configure SIP Trunk on goautodial

For configuring our DID number with goautodial, you will have to create two trunks in your system to allow calls from our server. You can do that by going to Admin section in your goautodial and choose carriers. Click on “Add a New Carrier” and add the...

Originating calls from a webpage using asterisk

Asterisk can be used to originate calls from a web page. Asterisk Manager Interface (AMI) is used for this purpose. AMI allows external programs to control asterisk.   For doing this , you should have A working asterisk server A SIP termination provider for sending...

Add Condition in Asterisk Dial-plan

While building a dial plan you will always run in scenario where you have to choose the action based on a if statement. In this example we can use a counter variable and based on the value of the variable we can make another decision.  Lets start with normal counter...

Asterisk Realtime conference

For asterisk 1.6 and above Create a new database and table in your mysql database. For adding the table use the below query CREATE TABLE meetme ( confno char(80) NOT NULL default ‘0’, starttime datetime NOT NULL default ‘0000-00-00 00:00:00’,...