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This page lists all options related to provisioning the phone system. |
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- Administrator username: Fill in a username for logging in to the device's browser based configuration interface. Not all the devices support configuration via a web browser. For additional information, check the documentation of the equipment you want to provision.
- Password generationAdministrator password: Choose the method for providing the password:
- None: No password will be required to connect to the device's browser-based configuration interface.
- Auto Automatically generated: VoipNow will randomly generate a password for you.
- ManualManually set: If you like, you can manually set and confirm the Administrator password.
- Phone update interval: Optionally, you can fill in the text box with the number of minutes the device waits before checking for updates on the provisioning server. Between: 1 and 99,999 minutes/seconds, depending on the device's settings. Default: 10 minutes.
- Update protocol: The list is automatically populated with the protocols used by the selected device to access the configuration file on the provisioning server (e.g.
TFTP, HTTP). - Provisioning template: The most important step when adding a device is to select the appropriate Provisioning template. The drop-down list displays only those templates with configuration files defined for the selected equipment, plus the Server default template that includes the standard configuration files of all the devices supported by VoipNow. These configuration files contain the settings (default or custom) required by the device to become fully functional in the VoipNow system. It is advisable to double-check the template you are going to use and search for any possible error that could cause the device to malfunction. As a system administrator, you can only choose from your own templates. The provisioning templates defined by other users are not available.Regenerate provisioning files: Select this checkbox if you want the provisioning files available for the current extension to be regenerated with the new settings.
- Phone time zone: From the drop-down list, select a time zone that will be used by the phone device. You can either select the time zone of the extension(s) that this equipment will be assigned to or any other time zone, depending on your requirements.
SIP preferences
The users that log in to the VoipNow interface using a service provider, organization or user account have complete access to the Provisioning and SIP Preference's field SIP Devices area field sets only if the Phone extension SIP management permission is enabled.
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| title | Click here to read more on SIP preferences. |
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| Option | Details | | Media encryption | This option allows media (voice or video calls) to be encrypted. VoipNow supports the following crypto standards: SDES and DTLS-SRTP. SDES is a crypto standard we use for voice and video calls over mobile networks. DTLS-SRTP is a crypto standard we use for voice and video calls through WebRTC. From the drop-down list, you can choose among the following options: SDES, DTLS-SRTP, SDES/DTLS-SRTP (and/or). By default, this option is set to None. 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. | | DTMF | 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. | | A PBX Is Connected connected to this extension | 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. | | Allow re-invites from this extension | By enabling this option, your extension will be allowed to send re-INVITES. | | Extension publishes its own state | Enable 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 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. The drop-down list is disabled until an IP address is filled in.
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; | | SIP Signaling Transport | | | 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. | | Equipment description | Briefly describe your device. |
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Extension virtualization
Allow virtualization on this extension: By default, this option is disabled. If you enable it, any other member of the organization can use the phone device where the extension is provisioned.
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