This page describes the SIP Channel and contains instructions on how to add one to the VoipNow system.
SIP channels allow the VoipNow PBX to communicate with other SIP exchanges and other VoIP networks.
The Session Initiation Protocol (SIP) is a signaling protocol widely used for controlling multimedia communication sessions such as voice and video calls over IP. The protocol can be used to create, modify and terminate two-party (unicast) or multi-party (multicast) sessions consisting of one or several media streams.
SIP channels can be configured using predefined provider templates, with the help of which you can instantly set up the channels for certain providers, without the need to manually fill in all the specific parameters. This feature simplifies the configuration process and reduces the time required to define an up and running SIP channel.
To set up a SIP Channel, first go to the System menu on the left and click on Channels. In the Channel Management page, click the Add SIP Channel icon under the Trunks area.
Enter channel details
- Clone settings from channel: Helps you create channels with a similar or identical structure more rapidly. You can replicate the settings of an existing SIP channel using the drop-down list.
- Name: Fill in a descriptive name to identify the channel.
- Channel groups: Click the channel name in the All groups list to assign it to one of the channel groups currently in the system. It will get automatically moved into the Assigned groups pool. The Assigned groups list contains the groups the channel is assigned to.
To remove the channel from a group, click the desired group and the channel will get automatically be moved back into the All groups list. For more information on how to create channel groups, please see the Channel Groups section.
- Channel notes: Add any channel details you want.
- I want to configure the channel: Select how you would like to set up the channel:
- Using a known provider - Set up the channel using the already defined settings included in the XML file specific to the desired provider (default selection). If you go with this option, you will have to fill in the Select Your Provider section detailed below.
- Manually - Manually fill in all the details required to set up the SIP channel (recommended for experienced users).
Customize channel preferences
If you set up the channel manually, all available options will be displayed and can be customized. If you configure the channel with a known provider, this section will only display the options for which the visibility level was set to 1 in the selected provider template XML file. In this case, the fieldset will be renamed Channel Preferences - Reveal All Fields and will become foldable; all the other options will be displayed with a click. When extended, the fieldset will be renamed Channel Preferences - Hide Fields; all the fields not included in the XML file will be hidden with a click.
|Send calls to <IP>:<port>||Hostname/IP address and port of SIP provider server. The port is not required unless the hostname cannot be resolved using |
|Accept calls from IPs/network||IP address(es) from which the system should accept calls. You may add/remove multiple IPs using the +/- buttons.|
The SIP provider's registrar server deals with the setup of all SIP calls in the SIP provider's network. Fill in the address of the record that is going to be created or updated on the registrar. The specified value will compose the username for the To and From SIP headers. If no Authorization username is specified, this value will be used for the Username SIP header as well.
Used for authorization with the SIP provider's
|Confirm password||Confirm password used to connect to the SIP provider's |
|Do not register||When enabled, you will not need to register before connecting.|
|Authorization username||Overrides the username used for authorization with the SIP provider's |
You can read this parameter as "we authenticate as."
|Concurrent calls (outgoing+incoming)||Applies if your SIP provider limits the number of concurrent calls through the channel. |
The channel will no longer be selected by the routing engine when the limit is reached and your provider will no longer accept other calls on that channel.
Specify the channel type:
Select the channel functionality:
|DTMF mode||Choose the DTMF (Dual-tone multi-frequency signaling) mode from the drop-down list. The default option is rfc2833. If you choose the auto option, Asterisk will automatically detect whether the channel supports rfc2833 or not.|
If it does not support it, Asterisk will use the inband mode.
|Behind NAT||Check this option if the channel is located behind a NAT (Network Address Translation) router.|
|CallerID for outgoing calls||Specify the number that will be displayed by phones with an active CallerID feature (e.g. John Doe <555890123>). |
Usually the CallerID is the same as the public phone number.
|From user||Some SIP providers require this parameter to identify the user. |
To verify if the channel requires this parameter, please check with the SIP provider.
Some SIP providers require this parameter to identify its users.
|Authorization extension||The username part of the Contact SIP header used for creating a registration binding with the service provider's network.|
You can read this parameter as "we are located at."
|Get DID from custom header||If a valid DID number is not present in the request line of the SIP packet, it will be taken from a custom header. |
This is where you can define the name of that header.
For example, if you fill in Test and the DIDnumber is 12345, then the custom header will be Test: 12345.
|Qualify value||The server checks the remote party for presence regularly. Specify a time-frame using the drop-down list. If it does not answer within the time-frame, the device will be considered offline. Can be disabled if you select None.|
|Prefix all calls with||The prefix that will be added to all destination numbers of the calls routed through this channel.|
|Trusted channel||When enabled, VoipNow considers that all communication through this channel comes from a trusted source (based on IP) and does not authenticate incoming calls.|
|Trust Remote-Party-ID SIP header||When this parameter is set to 1 (enabled), VoipNow trusts and uses the Remote-Party-ID headers in the SIP messages received on the channel.|
|Send Remote-Party-ID SIP header||When this parameter is set to 1 (enabled), VoipNow includes Remote-Party-ID headers in the SIP messages sent through the channel.|
|ICE Support||Select this option if you want to enable Interactive Connectivity Establishment (ICE) support for NAT Traversal in multimedia sessions.|
The user agents send periodic re-INVITE or UPDATE requests (session refreshm 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 by the end of the interval, the session is considered terminated.
|Session refresh interval|
Maximum amount of time (in seconds) between session refresh requests in a dialog before the session will be considered timed out. This time-frame is included in the SIP Session-Expires header field.
|Minimum session refresh interval|
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.
This value allows enough 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. If the UAC's Session-Expires is lower than the Min-SE defined for another SIP device, then that device rejects the session refresh request with a 422 response. This response contains a Min-SE header identifying the minimum session interval it is willing to support. The UAC will try again, this time including the specified Min-SE header. This way, the minimum timer meets the constraint.
|Session refresh source|
Indicate who will handle the SIP headers:
|SIP Signaling Transport |
Select the network protocol used on channels: UDP or TCP. Default value: UDP.
|Channel codecs||Select the codecs for encoding–decoding the voice signal over this channel.|
|Use MD5||VoipNow encrypts the passwords used in the authentication process. |
The method used is an MD5 hash function.
Select your provider
This option is only displayed if you have previously selected to configure the channel with a known provider. Providers get displayed in alphabetical order, starting with the Gold certified providers, followed by the Silver ones and ending with those that are not certified at all. Recommended providers are displayed in the Name (Certification: <level>) format, e.g. Service Provider (Certification: Gold). If the provider is not certified, it will be displayed as Service Provider (Certification: None).
- I want to get service in: Lists all the countries the defined providers are available in. This allows you to target the SIP channel for certain destinations. If you just want to set up a generic channel, stick to the No preference option, which is the default value.
- Recommended providers: Lists all the providers you have previously defined for the country.
--is the default selection; for the No preference option, the list displays all the providers in the system, regardless of the country they are available in.
- Provider website: The homepage of the service provider. Click the link to open the site in a new window. The link is automatically updated, depending on the selected provider.
Set the SIP server
Set the SIP server that proxies all the SIP traffic for this channel. Use the SIP node assign option from the Calls Distribution section.
Configure public phone numbers and DIDs
Configure public phone numbers and DIDs as sent by the channel provider. Select the Set up the public phone numbers assigned to the channel checkbox and click OK. If you want to add the new channel without configuring the public phone numbers, deselect the Set up the public phone numbers assigned to the channel checkbox and click OK. For more info on how to add public phone numbers, check theManage Public Numbers section.
Set channel costs
Set up the channel costs charged by the channel provider, based on the call destination. Select the Setup channel costs checkbox. You won't be able to do this unless you have defined a Paid charging plan in the Channel setup page. For more info on how to add channel costs, check the section.
Click OK to add the new channel. If you want to add the new channel without configuring the public phone numbers, deselect the Set up the public phone numbers assigned to the channel checkbox and click OK to save the channel costs.
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