Setting the SIP Channels ​
This page shows how to add and configure SIP channels. This can be done either manually or based on a template.
Overview ​
A channel is a connection to a telephony provider, which is able to give you access to the telephony network. Operating the system without using channels is possible, but you should know that you will be restricted to internal calls. For extended information on SIP Channels, read the area in the VoipNow User Guide
Interface Access ​
The fastest and easiest way to define your first SIP channel is by uploading a configuration file specific to the provider you want to use, if there is one. Whether your file is downloaded from the 4PSA website or obtained from your provider, the steps below are mandatory:
- Access the Unified Communications → SIP Provider Templates page and click the Add Provider Template icon.
- Upload the provider template. Use the available controls to upload the provider template. If the file you are trying to upload does not match the default provider template structure, it cannot be saved to the database. A warning message is displayed: Failed to upload the new template.
- Go to the Channels management page and click the Add SIP Channel icon.
SIP Channel Configuration ​
Use the controls available to personalize the channel that will use the settings provided in the uploaded configuration file. You may choose between two types of configuration:
Template Configuration ​
The SIP Channel will use the settings provided in the configuration file.
Click here for recommendations on template configuration.
- As this is your first channel, you cannot replicate any settings from other channels; you may leave the Clone settings from channel field as is;
- The Channel groups option allows you to organize channels; you can view channels by their assigned group;
- The I want to get service in list displays the countries the defined provider is available in, allowing you to target the SIP channel for certain destinations;
- Use the Recommended providers drop-down list to select the provider for which you have uploaded the configuration file; you must change the default selection, "-"!
- The Provider website field is auto-populated with the provider's web address, if specified in the configuration file.
Modifying any of the parameters available in the Channel Preferences section is not required.
Manual Configuration ​
If you are an experienced user and you want to manually configure your first SIP channel, you may use the entire customization options offered by VoipNow. To do so, simply navigate to the Channels management page and click the Add SIP Channel icon.
Click here for explanations on each setting.
| Option | Details |
|---|---|
| Accept calls from IPs | The IP addresses from which the system will accept calls. |
| Username | The login used to connect to the provider server. |
| Do not register | Enable if your provider authenticates your server by IP and not by username. If that is the case, the Username and the Password are no longer required. |
| Authorization username | Some SIP providers require an authorization username in order to connect to their channels. To verify whether the channel requires this parameter or not, you should check with the SIP provider. Specify the type of the channel:
|
| DTMF mode | It is advisable to select either auto or rfc2833. |
| Behind NAT | Enable if the channel is located behind a NAT (Network Address Translation) router. |
| Caller-ID for outgoing calls | This value will be displayed on the screen of the called party. Usually, the Caller-ID is the same as the public phone number. |
| From user/From domain | Some SIP providers require these two parameters to identify their users. To know for certain whether the channel requires this parameter or not, you may want to check with the SIP provider. |
| Authorization extension | This refers to the username part of the Contact SIP header used when creating a registration binding with the service provider's network (you can read this parameter as we are located at). Check with your provider if required. |
| Get DID from custom header | If a valid DID number is not present in the request line of the SIP packet, then it will be taken from a custom header. For example, if you fill in Test in the Get DID from custom header field and the DID number is 12345, then the custom header will be Test: 12345. |
| Qualify value | The server checks the remote party for presence regularly. If it does not answer in a time below the Qualify value setting, the device is considered offline. This feature can be disabled if you select None. |
| Prefix all calls with | This setting refers to the prefix that will be added to all the destination numbers of the calls routed through this channel. |
| Trusted channel | When the Trusted channel option is selected, VoipNow will consider all communications through this channel safe and will not authenticate incoming calls. It is required in order to receive incoming calls from certain providers. |
| Session timers | The user agents send periodic re-INVITE or UPDATE requests (session refresh requests) to keep the session alive. The interval for the session refresh requests is determined through a negotiation mechanism. If a session refresh request is not received before the interval passes, the session is considered terminated. From the drop-down list select the way the channel will handle the Session timers. |
| Session refresh interval | The value in this field is the maximum amount of time, in seconds, between session refresh requests in a dialog before the session is considered timed out. This time interval is included in the SIP Session-Expires header field. The user agent server (UAS) obtains this value form the Session-Expires header field in a 2xx response to a session request that it sends. The user agent client (UAC) determine this value from the Session-Expires header field in a 2xx response to a session request that it receives. Value: 10 to 84,000 seconds. Default: 1,800 seconds. |
| Minimum session refresh interval | The input in the field is the minimum value, in seconds, that will be accepted by the channel for the session interval. This value is included in the SIP Min-SE header field. Value: 90 to 18,000 seconds. Default: 90 seconds, representing little more than twice the duration that a SIP transaction can take in the event of a timeout. This value allows sufficient time for a user agent to attempt a refresh at the half-point of the session interval, and for that transaction to complete normally before the session expires. |
| Session refresh source | Here you can choose who will handle the SIP headers: the user agent client, UAC, or the user agent server, UAS. With the Use MD5 function selected, VoipNow can encrypt the passwords used in the authentication process. |