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You can also edit this parameter when a device is already provisioned for this account.

Use MAC based provisioning

When enabled, this option will have the provisioning link generated based on the device MAC address as long as the Allow MAC based provisioning on HTTP(S) option is enabled from the Unified Communications →  → Provisioning templates →  → Global preferences section and the Update protocol is either HTTP or HTTPS.

For Aastra devices, the provisioning link could be:

• If this check box is selected and the Update protocol is set to HTTP: http://192.168.4.170/pro/p/00085D000000

• If this check box is not selected and the Update protocol is set to HTTP:http://192.168.4.170/pro/p/0019/0020/001

For some manufacturers, such as Polycom, the provisioning links may look different: http://192.168.14.39/pro/p/0038/0039/001/0004f2111111.cfg

If a duplicate MAC address is found in the database, the provisioning link will not be generated and an error message will be displayed.

If you just want to define and assign the device without provisioning it, then you should select Disable and the provisioning file will not be generated; On the other hand, if you want to finalize the provisioning process, select Enable;

If the extension's Allow to provision devices permission level is set to Modify and a device is already provisioned for this account, then you are able to modify the selection.

Notes

Add supplementary information about the device.

This option allows media (voice or video calls) to be encrypted. VoipNow supports two crypto standards: SDES and DTLS.

If you want to use any of these crypto standards, you must first ensure that your client (IP phone or softphone) supports it. If the crypto standard you have selected is not supported by your client, calls will not work.

By default, this option is set to None.

Choose the DTMF mode. Default: rfc2833.
If you choose the auto option, Asterisk automatically detects whether the channel supports rfc2833 or not. If not, Asterisk uses the inband mode.

This option allows the system to direct an incoming call made to a public phone number to a particular extension on the PBX server connected to the extension for which the current setting is enabled.
When the call is sent to the PBX server, the public phone number that was called is saved and thus the call can be directed to the chosen extension on the PBX server.
This setting is not available unless your license supports SIP trunking (the Maximum number of SIP trunking channels must be bigger than 0).
If the license supports it, this option can only be enabled if the extension's Maximum public concurrent calls value has not been previously set to Unlimited; if the value is set to Unlimited, the line is still available, but you will not be allowed to select the checkbox and you'll see warning message displayed next to it:

To make it available, setup the Maximum public concurrent calls to a limited number (now it is unlimited).

Ping the extension to check its status

When enabled, the server sends ping SIP messages to the extension regularly. Usually, this option is used for extensions behind NAT.

Extension is on private network (<ip_address>/<network_mask>)

This option is available only if you choose the NAT (Network Address Translation) or Public/Private Networks Routing infrastructure types from Cloud Management → Infrastructure Properties.
Enable this option if your extension is located inside the private network of your SIP server, in order to prevent unnecessary NAT processing.

Extension publishes its own state

Enable this option if you do not want the server to send presence notifications to the phones watching this extension for presence.
If enabled, the server will no longer send any notification events unless this extension explicitly publishes its presence by sending PUBLISH messages to the SIP server.

Force enable of MWI

Enable this option if you want to receive Message Waiting Indicator notifications and your phone does not send explicit subscriptions for MWI. Most phones do not need this option.

Allowed codecs

Select the codecs supported by the phone device.
The list of possible codecs displayed in the Allowed codecs section can be modified from the Unified Communications →  → Zero Priority →  → General page.

Phone does not register, is located on IP <> Port <> and <has to/does not have to> authenticate

All incoming calls from this IP/Port require/do not require authentication.
More than one extension can use the same IP, as long as every single one authenticates. If the same IP is being used by another extension that does not have to authenticate, an error message will be displayed, telling you that this configuration cannot be saved.

The drop-down list is disabled until an IP address is filled in.

Select the network protocol used on channels: UDP or TCP. Available for extensions that do not register, located on a fixed IP. Default: UDP.

The drop-down list is disabled until an IP address is filled in the the Phone does not register, is located on IP field.

Allow extension SIP connection only from IP <IP_address> (maximum class C (/24)

Limit the extension usage to an IP or a network. Only the IP addresses specified here will be allowed to receive and make calls from this extension.
Registration on the phone is still needed in order to receive calls.
You can add several IP addresses by using the +/- buttons.