All configuration functions rely on a profile. You must create a profile before you use any other configuration function except 'set working directory'. A profile is just a name you use to associate your configuration files with. You may have multiple profiles at one time. Technically, a profile is simply a subdirectory under the working directory where the PBX .conf files are stored. As a result, profile names adhere to the same rules as directory names on the native system.
2.1 Creating Profiles
You must first create a profile, before you can edit it. Once your profile has been created you may modify it if you like, however, in most cases the defaults created should be adequate for most users. The default profile produced by the create profile link has the following properties ...
Calls coming into the PBX from outside sources are permitted to any public extensions you might have previously set up (see create/edit SIP extensions for more information on setting up SIP extensions). If the call is for an unknown extension then the caller gets a message played which states the number is currently not in service. It then hangs up on them.
Calls originating from within the PBX are matched against both public and private extensions (again see create/edit SIP extensions for more information on this) and if no match is found then a message is played indicating an invalid extension has been reached. It prompts the user to try again, at which time the caller can attempt to dial another extension. If the new extension dialed is also invalid then the above is repeated until either the internal caller hangs up or times out (appx. 10 seconds).
The main voice mail application can be reached by both internal and external callers at extension 8500.
An extension at 8600 is created which the user can call (available only to internal extensions) and record a message which can later be used as a voice prompt by the system. This extension is initially disabled.
No SIP extensions are created by the create profile function. SIP extensions are added using the Create/Edit SIP Extensions function.
2.2 Changing a Profile
To change a profile select Edit Profile from the configuration menu. You will be prompted for the profile name and if the profile is found, then a screen will be displayed which shows the profile name near the top of the screen. The default inbound and outbound call flows, and the default voicemail, recording and parked calls extensions are also shown. Each of these parameters is explained below ...
2.2.1 Inbound Call Flow
The inbound call flow is what happens when a SIP call is received from an unknown extension. Inbound calls are also referred to as outside (or externally) originated calls. The pull down menu shows all the currently available call flows to choose from. If you would prefer a different call flow than what is offered by the system you can use the call flow editor to create new call flows. A call flow is defined as what happens if an extension match does not occur. In other words if a call comes in and the provided extension is found then the call is routed to that extension and that exension's definition defines the call behavior from that point on. If, however, the provided extension does not match one of the defined extensions (public extensions if the call originated from a client outside the server and public and private extensions if the call originated from within the system) then the call flow is what is used to determine what to do with the call. Currently the following call flows are provided by the system ...
play a congestion tone and hang up
play a message indicating the number is not in service and hang up
play a message indicating an invalid extension has been reached and hang up
play a message indicating an invalid extension and prompt the user to try again.
do not allow calls on this channel
Note that if you select the 'do not allow calls on this channel' option, even calls with valid (or known) extensions will not be routed. The system simply hangs up on any call received from a channel with the 'do not allow' option set.
2.2.2 Outbound Call Flow
The outbound call flow is the same as the inbound call flow except that it applies to calls that are received from devices which are known to the system. Remember that calls originating from known extensions are matched against both public and private extensions before the call flow is executed.
2.2.3 Voicemail Extension
The voice mail extension is the extension you dial to reach the voice mail system. The voice mail system allows known extensions to retrieve their voice mail messages and set various voice mail parameters like their greeting messages. By default the create profile function creates the voicemail extension at extension 8500 as a public extension. You can modify the this to anything you like, or you can completely disable the voice mail system by unselecting the 'enable' checkbox.
2.2.4 Recording Extension
The recording extension is the extension you dial to reach an application that will record what you say into the phone. The recorded message will be saved in your working directory as a GSM file which you can later use as a prompt in your PBX. You can then use this recorded message, for example, to play a menu that callers will hear when they reach your PBX. Be careful as the system uses the file name 'message.gsm' over and over again, so once you get a recording that you like you will need to copy it to the asterisk install sound directory so asterisk can use it.
By default the create profile function creates the recording extension at extension 8500 as a private extension. You can modify the this to anything you like, or you can completely disable the recording application by unselecting the 'enable' checkbox. Note the create profile function initially diasables this extension so you must use the edit profile function to enable it if you wish to use this functionality.
2.2.5 The Parked Call Extension
The parked call extension is currently unsupported.
2.3 Viewing an Existing Profile
You may view the .conf files produced by AM at any time using the 'Show Profile' option from the configuration menu. You will be prompted to provide a profile name and if the profile name is valid, a page which shows the sip.conf and extensions.conf files associated with the supplied profile will be displayed.
2.4 Creating and Modifying SIP Extensions
When configuring your SIP extensions the system handles the defaults for all advanced parameters for you. If you like, you may control all parameters by clicking the 'ADVANCED' link on the lower left side of the page. When you click the advanced link, you will be presented with a page which shows all the advanced parameters. If you don't know what a particular parameter does you should probably leave it alone as the default is probably the safest choice. If you don't even want to see these parameters just stay on the standard screen, or from the advanced screen simply click the 'STANDARD' link on the lower hand corner of the screen and you will be returned to the standard extension editing page.
The main screen allows you to ceate, edit, delete and find SIP extensions associated with a profile. To find an existing extension (for subsequent processing) simply ener the profile name and the extension then press the 'FIND' button and if it exists, the extension information will be displayed on the screen and available for other operations (like edit and delete). To create a SIP extension you must provide the profile name, extension and IP Address of the SIP device (or computer in the case of soft phones).
2.4.1 Standard SIP Extension Parameters
These are the parameters you see when you first select the 'Create/Edit SIP Extension' option.
This is the name of the profile you would like to associate this extension with. You must have previously created a profile in order to configure extensions.
This is the extension number of the extension you wish to configure. While you can use alpa extensions, it is recommended you use simply numeric extensions. Typically a 3-5 character extension is a good idea. It is also a good idea to keep all extenions the same length.
188.8.131.52 IP Address
This is the IP address of the SIP phone you are configuring. If you don't know the IP Address you can try to enter 'dynamic' here and if the SIP phone registers properly, the Asterisk server will automatically pick it up. If you can not get your SIP phone working using 'dynamic' then you will probably need to determine the IP address of the machine. To do this under Windows use ipconfig and under linux you can use ifconfig. Remember, this is the IP address of the machine which is running the SIP soft phone. For a Cisco ATA 186, you pick up the handset, press the red button on top and then dial 80# and the ATA will tell you its current IP address.
This is the mailbox number of the mailbox you create for this extension (if any). Mailboxes are optional. If you don't want to use a mailbox with this extension leave this field blank.
184.108.40.206 Call Flow
This pull-down lets you select the call flow assigned to the extension. The call flow defines how the extension behaves when a call is transferred to it. Two default call flows are provided. One supports voicemail on the extension, the other does not.
This checkbox indicates whether this extension is assigned to the public or private extensions list. By default, public extensions can be seen/reached by anyone including outside callers calling into the system. Private extensions on the other hand can only be seen within the system. In other words, only extensions (phones) within this PBX may see (and place calls to, transfer to, etc.) private extensions.
2.4.2 Advanced SIP Extension Parameters
You must click the 'ADVANCED' link on the lower left hand corner of the 'Create/Edit SIP Extension' page to have access to these parameters. When you do, an advanced screen which looks something like this will replace the normal data input screen ....
The user parameter is typically the same as the extension number. It is defaulted to the extension number but if you wish to override this you may do so by entering your user name in this field. If you leave this field blank then the system will default it to the extension number which is recommended.
220.127.116.11 DTMF Mode
The DTMF mode may be either 'rfc2833', 'inbound', or info. The default (or if you leave it blank) is rfc2833.
The type parameter may be either 'friend', 'peer', or 'user'. The default is friend.
This parameter is used when the PBX is located behind a firewall. The default is yes.