Asterisk GUI. The first steps
Almost every time when it comes to choosing a web-interface to asterisk, the recommendations of "heavy artillery" are used: trixbox, elastix, freepbx. Asterisk GUI remains a kind of “dark horse”: it seems that you heard about it, like screenshots of simpatish ones, but really working systems - with a nose. This is partly due to the novelty, the non-rolling product. Partially - the lack of documentation. Partially - the inertia of the community. We will try to get things off the ground ...
The article provides step-by-step instructions on how to install and configure Asterisk with the Asterisk GUI.
Let's get started ... Downloading the source code from svn and compiling. You can copy the following commands in the console and lean back in the chair while waiting.
After the build process is complete, we will edit two configuration files:
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/etc/asterisk/manager.conf should look something like this:
And /etc/asterisk/http.conf - like this:
Run asterisk:
This completes the installation process.
Consider the simple case. We have one user in the system (one VoIP-phone). There is a SIP trunk to the my.sip.trunk.com provider. The provider provided us with one trial number - 7692450. Our task is to give our only user the opportunity to make and receive calls.
On the working machine, we launch the browser and type in the address bar xxxx: 8088, where xxxx is the IP of the machine where the Asterisk and Asterisk GUI are installed. Enter the username and password, and press the login button.
After short messages about updating the configuration files (the system will try to determine if we have any digium hardware installed), we will go to the main page.
First of all, let's configure the SIP trunk to the provider. Go to "Trunks", "VoIP Trunks" and click "New SIP / IAX Trunk". Fill in the fields in the window by analogy with the fact that the picture.
Configure the rule for outgoing calls: all outgoing calls - send to the newly created trunk. Go to "Outgoing Calling Rules". Click "New Calling Rule" and fill in the fields in the window.
Configure the call plan (Dialplan). Dialplan is nothing more than a group of rules for outgoing calls. In our case, we have only one rule. So, go to "Dial Plans", click "New DialPlan". In the box, tick the rule created in the previous step.
We create the user. Go to the "Users". Click "Create New User". We define our user name (CallerID Name), number for outgoing calls (CallerID Number) and a plan of calls.
It remains to determine the rule for incoming calls. This topic is quite interesting in itself and draws on a separate article. The fact is that an incoming call can be pushed into the interactive menu ("to contact technical support - press 1, to contact sales department - press 2, to send with an operator - press 0"), send to several phones at once (while someone do not pick up the phone), send to the answering machine, if no one answers for n seconds, redirect to the mobile ... and a lot more. We modestly direct incoming calls to our only user. So, go to "Incomming Calling Rules", click "New Incomming Rule" and fill in the fields in the window by analogy with what is in the picture.
Well that's all. Click the "Apply Changes" button in the upper right corner.
I hope someone this article will be useful.
The article provides step-by-step instructions on how to install and configure Asterisk with the Asterisk GUI.
Benefits:
- Developed by digium, the same company that develops himself asterisk
- Does not require Apache, MySQL, PHP. Uses asterisk embedded web server. Written in html and javascript
- It does not depend on the availability of third-party programs / libraries. Does not require installation on any specific distribution. All you need to install is a working asterisk
Installing Asterisk and Asterisk GUI
Let's get started ... Downloading the source code from svn and compiling. You can copy the following commands in the console and lean back in the chair while waiting.
svn co svn.digium.com/svn/asterisk/branches/1.6.0 asterisk
cd asterisk
./configure && make && make install && make samples && make config
cd ..
svn co svn.digium.com/svn/asterisk-gui/branches/2.0 asterisk-gui
cd asterisk-gui
./configure && make && make install
cd ..
After the build process is complete, we will edit two configuration files:
')
/etc/asterisk/manager.conf should look something like this:
[general]
enabled = yes
webenabled = yes
port = 5038
bindaddr = 127.0.0.1
[admin]
secret = 123456
read = system,call,log,verbose,agent,user,config,dtmf,reporting,cdr,dialplan
write = system,call,agent,user,config,command,reporting,originate
And /etc/asterisk/http.conf - like this:
[general]
enabled = yes
enablestatic = yes
bindaddr = 0.0.0.0
redirect = / /static/config/cfgbasic.html
[post_mappings]
backups = /var/lib/asterisk/gui_backups
Run asterisk:
/etc/init.d/asterisk start
This completes the installation process.
Configuring Asterisk with Asterisk GUI
Consider the simple case. We have one user in the system (one VoIP-phone). There is a SIP trunk to the my.sip.trunk.com provider. The provider provided us with one trial number - 7692450. Our task is to give our only user the opportunity to make and receive calls.
On the working machine, we launch the browser and type in the address bar xxxx: 8088, where xxxx is the IP of the machine where the Asterisk and Asterisk GUI are installed. Enter the username and password, and press the login button.
After short messages about updating the configuration files (the system will try to determine if we have any digium hardware installed), we will go to the main page.
First of all, let's configure the SIP trunk to the provider. Go to "Trunks", "VoIP Trunks" and click "New SIP / IAX Trunk". Fill in the fields in the window by analogy with the fact that the picture.
Configure the rule for outgoing calls: all outgoing calls - send to the newly created trunk. Go to "Outgoing Calling Rules". Click "New Calling Rule" and fill in the fields in the window.
Configure the call plan (Dialplan). Dialplan is nothing more than a group of rules for outgoing calls. In our case, we have only one rule. So, go to "Dial Plans", click "New DialPlan". In the box, tick the rule created in the previous step.
We create the user. Go to the "Users". Click "Create New User". We define our user name (CallerID Name), number for outgoing calls (CallerID Number) and a plan of calls.
It remains to determine the rule for incoming calls. This topic is quite interesting in itself and draws on a separate article. The fact is that an incoming call can be pushed into the interactive menu ("to contact technical support - press 1, to contact sales department - press 2, to send with an operator - press 0"), send to several phones at once (while someone do not pick up the phone), send to the answering machine, if no one answers for n seconds, redirect to the mobile ... and a lot more. We modestly direct incoming calls to our only user. So, go to "Incomming Calling Rules", click "New Incomming Rule" and fill in the fields in the window by analogy with what is in the picture.
Well that's all. Click the "Apply Changes" button in the upper right corner.
I hope someone this article will be useful.
Source: https://habr.com/ru/post/54816/
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