What is the Extensions Module used for?
The Extensions Module is used to set-up each extension on your system. In the Extensions module, you will set-up the extension number, the name of the extension, the password, voicemail settings for the extension, and other options.
Normally, each physical phone will be assigned to one extension. If you have a phone that has more than one "line" button, you would normally make each line button register to the same extension number, and then use the line buttons to manage multiple calls to and from the same line. However, you could also create two or more extensions and assign each extension to a different line button.
The Extensions Module works together with any module that can route a phone call, including the Inbound Routes Module, the Ring Groups Module, the Queues Module, and the Paging Module. The Extensions Module also works together with the Follow Me Module, because each extension can have its own Follow Me options.
The Extensions Module is also related to the Advanced Settings Module. In the Device Settings section of the Advanced Settings Module, you can change a number of the default settings that will apply when you create a new extension.
In addition, the Advanced Settings Module can be used to enable Device and User Mode. When Device and User Mode is enabled, the Extensions Module will disappear and be replaced with two separate modules called "Devices" and "Users."
This guide walks you through information related to SIP extensions.
Options will vary based on installed modules. Configuration of these settings is covered by their respective user guides
Adding a SIP Extension
- From the Extensions landing click on the Device select box.
- Choose Generic CHAN SIP Device
- Click Submit
You will be brought to the extension page
This will be the extension number associated with this user and can not be changed once saved. In our example we have set this to “5000.” We recommend using 3 or 4 digit extension numbers.
This is the name associated with this extension and can be edited any time. This will become the Caller ID Name
CID Num Alias
The CID Number to use for internal calls, if different from the extension number. This is used to masquerade as a different user. A common example is a team of support people who would like their internal CallerID to display the general support number (a ringgroup or queue). There will be no effect on external calls.
If you want to support direct sip dialling of users internally or through anonymous sip calls, you can supply a friendly name that can be used in addition to the users extension to call them.
Here is where you set the caller ID for this extension. If you leave it blank the system will use the route or trunk caller ID. Please note this will only work on trunks that let you set the outbound caller ID and if your outbound routes and trunks are setup to allow this. Please see the Outbound Caller ID PDF for more information on caller ID logic.
How long the phone should ring, in seconds, before going to the no answer destination. The default value can be set in the General Settings module.
Call Forward Ring Time
If you setup a call forward for this extension, how long do you want to call before going to the “No Answer” destination of this extension. If you set it to “Always,” it will ring the call forward forever and never go to the “No Answer” destination.
Outbound Concurrency Limit
How many concurrent outbound calls this extension can make at one time. We usually recommend setting this to 3-4. You can pick “No Limit” to let it make unlimited concurrent outbound calls.
Choose if call waiting should be enabled. If not enabled and the extension is on a call, the second incoming call will be sent to the busy destination of this extension.
Internal Auto Answer
If set to answer, any time another extension calls this extension
Call screening prompts the caller to say their name before ringing the extension. You can pick from disabled and call screening with or without memory. With memory, the system will prompt the caller for this name then store the name recording to belong to the caller ID of the caller. Anytime someone calls from that caller ID, it will not prompt them to say their name again, but it will play the name from the saved recording.
If you have pinsets set on any outbound routes that are enabled, this extension will not be prompted to enter the pin. You can also enable pinless dialing on a per route basis.
If you have an outbound route tagged as emergency calling and this user makes a call, you can set the caller ID that should be sent for that call here. This is handy if the phone is at a remote location and you want to set a different caller ID for emergency calls, such as the remote locations phone number so the call gets routed to the proper 911 authority.
Queue State Detection
If this extension is part of a queue then the queue will attempt to use the user's extension state or device state information when determining if this queue member should be called. In some uncommon situations, such as a follow-me with no physical device or some virtual extension scenarios, the state information will indicate that this member is not available when they are. Setting this to “Ignore State” will make the queue ignore all state information, thus always trying to contact this member. Certain side effects can occur when this route is taken due to the nature of how queues handle local channels. For example, subsequent transfers will continue to show the member as busy until the original call is terminated. In most cases, this SHOULD BE set to “Use State.”
The Device Options Section is a mix of misc settings as it relates to devices and most of these settings should never be changed. We will only discuss the ones that you would need to change.
This is the password that your SIP phone will use to connect to register with this extension. It should be a random and hard to guess password that’s at least 6 characters long with a minimum of 2 letters and 2 numbers. When creating a SIP extension in newer versions of FreePBX a reasonably secure password will be auto-populated
The DTMF Signalling mode used by this device. In most cases this will be RFC 2833
SIP NAT Mode
NAT setting. Yes usually works for both internal and external devices. Set to No if the device will always be internal.
Always ignore info and assume NAT
No - (no)
Use NAT mode only according to RFC3581 (;rport)
Force rport - (force_rport)
Force rport to always be on.
Use rport if the remote side says to use it and perform comedia RTP handling.
Automatic Force rport - (auto_force_rport)
Force rport if Asterisk detects that an incoming SIP request crossed a NAT after being sent by the remote endpoint.
Use comedia if Asterisk detects that an incoming SIP request crossed a NAT after being sent by the remote endpoint.
never - (no)
Never attempt NAT mode or RFC3581 support
route - (force_rport)
Assume NAT, don't send rport
If voicemail is enables typically calls would go there if the extension was unavailable. Optionally you can set an alternative destination for these calls.
Where to send the callers if the call is not answered
Where to send the caller if the extension is busy or the call is rejected.
Where to send the call if the extension is unreachable, for example off line.
What to prefix the Caller ID with prior to sending the call to it's destination.