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...
View ArticleOriginating 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...
View ArticleConfigure 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...
View ArticleConfigure 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 =>...
View ArticleHangup 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...
View ArticleMATH 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...
View ArticleChanging 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,...
View ArticleCalculate Difference between two Time Values from Asterisk Dialplan
For calculating the difference between two times, the time values should be converted to an epoch value first. This can be done using the asterisk function STRFTIME. Use the below dialplan to convert...
View ArticleFreepbx 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, login to your didforsale account, go to...
View ArticleSip Trunking Basics
Don’t know much about SIP Trunking? No worries! SIP Trunking is popular medium to provision calls for many who use VoIP. Along with flexibility and scalability SIP Trunking gives huge cost benefits....
View ArticleCount Calls From Asterisk Dialplan
We have put together a list of dialplan functions that you can use to count calls from Asterisk Dialplan. The post Count Calls From Asterisk Dialplan appeared first on DIDforSale.
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