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...

Asterisk time based routing

This is a very common requirement that route the calls to Voice-mail after office hours. Orr transfer the calls to cell phone after 6:00. In Asterisk you can control the call location based on time and date. [AutoAttendant] exten => start,1,Verbose(2,Entering our...

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’,...