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Administrator setup: Configuring your Asterisk server to work with Flexor Manager


Server Configuration

Flexor Manager communicates with an Asterisk server using the Asterisk Manager Interface (AMI). In order to use Flexor Manager with Asterisk, you must enable the AMI on the server. This is done by ensuring the following lines are present in the manager.conf file of your server configuration:

[general]
enabled = yes
port = 5038

manager.conf entry to enable AMI

5038 is the standard port for AMI, but you can use any other available port, as this can be configured in Flexor Manager.

To allow your users to connect to the AMI server, you must define a user login in manager.conf. Although it is possible to have almost complete control of the server using the AMI, Flexor Manager requires access only to certain commands. As your users will need to be given the AMI username and password, it is recommended that the AMI user you create does not grant permission to access any more commands than is necessary.

Here is an example which defines the user 'amiuser' with password 'mysecret':

[amiuser]
secret = mysecret
read = system,call,agent
write = system,call

AMI user definition in manager.conf for Asterisk 1.4 (AMI 1.0)

[amiuser]
secret = mysecret
read = all,system,call,agent
write = system,call,originate

AMI user definition in manager.conf for Asterisk 1.6 (AMI 1.1)

The reason for the difference between the two versions is that the AMI version 1.1 provided in Asterisk 1.6 introduces an new permission 'originate' which is required if the user is to be allowed to originate calls. This is necessary for the Flexor Manager Asterisk driver to be able to perform click-to-dial requests from applications.

If your organisation does not use call queues, you can omit the 'agent' read permission from the user definition. Flexor Manager only requires information about agents in order to handle calls received from queues properly.

Finally, in order that events indicating that calls have been put on hold or taken off hold can be received, you need to add the following line to sip.conf if it is not already present:

callevents=yes

Entry in sip.conf to enable hold events

After you have finished editing the configuration files, you will need to restart Asterisk for the changes to take effect.

Auto-answer configuration

Normally, when a call from one extension to another is originated from the Asterisk server, the users of both phones have to accept the call. In the context of Flexor Manager, this means that when a user clicks on a button in a CRM application to dial a number, the first thing that will happen is that their phone will start to ring. Only when they answer the ring will the call be placed with the person they want to talk to.

Flexor Connect for Asterisk provides a facility to avoid this when using SIP devices that support auto-answer, by allowing users to configure an auto-answer extension. This is an extension which forwards to their real extension in such a way that the phone automatically answers the call. To enable this, auto-answer extensions need to be configured on the server. One way of achieving this is to add an entry like this to the extensions.conf file:

exten => _autoXXXX,1,SIPAddHeader(Call-Info: sip:10.0.0.26\;answer-after=0)
exten => _autoXXXX,n,Dial(SIP/${EXTEN:4})

Example entry in extensions.conf to provide auto-answer extensions (Replace 10.0.0.26 with the address of your Asterisk server)

This defines an auto-answer extension "autoABCD" corresponding to any given four-digit extension "ABCD". If a call is initiated from autoABCD, it has the effect of being a call from ABCD, but with the device registered to ABCD auto-answering. To make this available to a Flexor Manager user,all they need to do is to configure their device with the auto-answer extension corresponding to their 'real' extension.

Auto-answer of IAX devices is not currently supported in Flexor Manager.

This is only an example of how to provide auto-answer extensions on an Asterisk server, and is not intended to be a definitive solution. Some SIP clients may not support the answer-after header.

Not all SIP devices provide an auto-answer facility. You should check this in your SIP device's documentation

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