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 channelsis possible, but you should know that you will be restricted to internal calls.
For extended information on SIP Channels, read theChannels 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.
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.
Option
Details
Accept calls from IPs/network
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:
Paid - The channel charges a fee every time it is used; this option must be chosen when the provider charges you a fee for the channel.
Free - The channel can be used for free to call any destination.
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.
Video tutorial
To set up your SIP channel and learn useful tips on how to manage your routing channels, watch the video below or read the related transcripts.
Hi and welcome to our series of VoipNow server configuration tutorials. In this video, you will learn how to set up a SIP channel.
Before we start, let's cover some useful info; SIP, ENUM or IAX channels can be created from the Channel Management page.
PRI channels can also be added from here, but only if an E1/T1 Asterisk-supported card is installed on the server.
Costs can be added on any channel; on SIP, IAX and PRI channels you can also add public phone numbers and DIDs.
Don't forget that you can use more than one provider; this will give you service redundancy as well as allow you to route your calls using the best cost option, also known as least cost routing.
To create and configure a SIP channel, you will need a SIP account from a SIP provider; channels can be configured either manually or using settings for a provider approved by 4PSA.
In our example, we will set up a new channel manually; start by adding the IP address from your provider.
If the provider uses credentials-based authentication, then also fill in the username and password.
Authentication can also be done based on your server IP, in which case you will not need a username and password.
If your server IP is on your provider’s whitelist, then check the “do not register” checkbox.
Set the number of concurrent calls supported by the channel and the charging plan type; the charging plan you assign on the channel can be either paid or free.
If you decide for a paid channel, you must also assign costs on that channel.
Finally, select the codecs used by your Sip provider, usually g711 and ulaw or g729.
If the setup was done correctly, the SIP channel will show the “registered” status in the list of SIP channels; this means the channel can now be used for public calls.
The "registered" status will only be displayed on channels assigned to a provider. Channels with IP-based authentication will not be "registered" in the channel list.
If your provider has given you DID numbers, you can add them from the Add Phone Numbers section.
You can add DID numbers one by one or, if you want to load a batch of DIDs, you can load them from a file.
Then you can set your number, DID format, location, and monthly cost to provider.
In special cases, you can also allocate an incoming call cost (for example, sharing costs phone services or toll free services such as 0800).
If your SIP channel is paid, you need to load the costs based on the dialed area code.
If your SIP provider is an A to Z provider, which means that it offers termination all around the world, then there could be more than 20,000 area codes. So it’s better to load a .csv file.
Remember that you will not be able to place calls to destinations without configured costs. In this example, we’ll be adding three channel costs.
For Germany all, 49 code, 0.002 USD per 6 second.
Then add other area codes - 4915, 4916 and 4917 for Germany mobile.
Finally, add area code 402 for Romania landline and 407 for Romania mobile.
What you could also do is prepare a .csv file and load it into the system.
Once all this is set, in the Routing Rules area, you can manage how outgoing calls are routed outside the VoipNow system using the channels available.
To learn how to do this, watch the video tutorial in the Setting the Routing Rules page.
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