There are four applications within Asterisk that can be used to control the relationship between a queue and a member. They are AddQueueMember(), PauseQueueMember(), UnPauseQueueMember(), and RemoveQueueMember(). In this lesson we will look at adding and removing a user from a queue dynamically. In the previous lessons we added a static member to the techteam […]
To view detailed information about the Queue() application in the Asterisk CLI use the following command: core show application Queue Open the dialplan and add the following new extension within the [phones] context: exten => 300,1,NoOp(Tech Team) same => n, Queue(techteam,,,,120) #120 is application timeout same => n, Hangup
Move the original queues.conf file sudo mv /etc/asterisk/queues.conf /etc/asterisk/queues.conf.sample Create and edit the queues.conf file sudo nano /etc/asterisk/queues.conf Add the following information to your queues.conf file [general] persistentmembers = yes monitor-type = MixMonitor [techteam] member => SIP/matthew member => SIP/eva strategy=random timeout=9 Below is a list of optional strategies that can be used to direct […]
Assume we have a work schedule below Sunday: Off Work Monday: 13:00 – 13:50 Tuesday: 11:00 – 13:50 Wednesday: Off Work Thursday: 11:00 – 13:50 Friday: Off Work Saturday: Off Work We could then create conditions to send the phone system to the after hours voicemail when we are not at working using the following […]
If you are looking to control the routing of the phone system based on the time, then you will most likely want to use the GotoIfTime() application. To view detailed information about the application and it’s parameters while on an Asterisk server use the following command in from the Asterisk CLI: core show application GotoIfTime […]
If the CALLERID variable is not set, outside phone calls will most likely see “Private Caller” or “Unknown” on their phone screens. In this lesson we will set the phone number as the caller ID. The SIP provider may have specific documentation on what can be included with the CALLERID variable. Some allow names to […]
With many phone systems, a number such as “9” will need to be pressed in order to dial out and reach a phone number outside the Asterisk dialplan. In this lesson we will use the first digit 9 to indicate to Asterisk that the call will need to route through the SIP provider rather than […]
In this lesson we will redirect the incoming calls to an extension in the dialplan. In the previous lesson a SIP account was created for [LectureSnippets]. The context for this account was set to provider. We will need to create this context to put instructions in for incoming calls. By placing this account in a […]
A SIP account will need to be created both from the Provider and in the sip.conf file. When creating a SIP account on the providers web site you will need to enter in a public IP address. You can determine the public IP address of your location by visiting the page https://lecturesnippets.com/myip.php (Links to an external […]
In this lesson NAT and Public IP based Asterisk servers will be discussed. A majority of Asterisk servers setup for this course will be using a NAT connection to the Internet, so that will be the focus. NAT Connection The typical Internet connection for desktop computers is through a router that offers network address translation […]
