The “Providers” module is located in the “IP telephony” menu. This module is designed to configure incoming and outgoing calls to an external network. When you open the “Providers” module, you will see a list of providers and tunnels created in the ICS CUBE, as well as a search bar and function buttons (add, delete, disable, edit) that are activated when you select an object.
The ICS CUBE supports two types of channels - SIP and IAX, as well as two types of similar tunnels. The tunnels are designed to connect the telephony of two ICS CUBE. To set up a new channel, click the “Add” button and select the type of channel (trunk) you need.
The “SIP provider” object is intended for establishing a connection using the SIP protocol. To add an external channel, click “Add” - “IP-telephony SIP provider”. A dialog box will open in which you will be asked to enter the required parameters: name, server address and port.
The “Server address” and “Port” fields are intended for specifying the appropriate settings of the IP telephony provider to connect to.
Important: when selecting a port, you need to consider the value of the “Type of connection” field, since providers usually ensure connectivity for different protocols on different ports.
The “Number” field is intended for specifying an external number for making calls via the ICS CUBE telephony server.
The “Login” and “Password” fields are used to control authorization data when connecting the ICS CUBE to the provider's server.
The “Max number of connections” field is intended for specifying the maximum number of simultaneous connections via the provider (by default, 100).
The “Automatically create a rule using the prefix” flag activates the field for specifying the default external call prefix. This prefix is a number that the module uses to determine whether to send a call to an external network. For example, a call to 555-3333 with the specified prefix 9 will be dialed by the client as 9-555-3333.
“DTMF mode”, “Insecure”, “Canreinvite” options allow you to configure the tone dial modes, while the latter two are only available when you select the chan_sip driver in the Telephony settings. The “direct_media” option, on the contrary, is only available for the chan_pjsip driver and determines whether media data can be transmitted directly between endpoints. If no, all RTP streams go through Asterisk.
In the “Type of connection” field you can choose whether to encrypt sip packages and media data (RTP) for this number using the certificate selected on the general telephony settings tab; there are two possible connection options without encryption: via UDP (this is the default value) and via TCP. To enable encryption, select the “Use encryption (TLS and SDES sRTP)” value in the “Type of connection” field. This value activates simultaneous encryption of SIP signaling via TLS and sRTP media data.
Important: each type of connection requires the appropriate configuration of its transport means, which is specified in the IP telephony settings. In other words, you need to set the appropriate port for each connection type.
If the provider has specific settings, you should fully enter the registration line by setting the appropriate flag. Please note that the registration line is only available for the chan_sip driver. It is created automatically for the chan_pjsip driver.
Important: in some cases, the SIP telephony provider cannot recognize the caller that is making an external call. If incoming calls work when registering the provider, but outgoing calls do not work, then you need to specify the phone number or username in the “From User” field (depending on the features of the provider).
The “Keep connection” field is used to specify whether the remote device is available for making calls. Asterisk will periodically send a SIP message of the OPTIONS type to check availability. If this device does not respond within the specified period (or the default period of 2000 ms) in milliseconds, then Asterisk considers this device to be turned off and unavailable for making calls. This option is only used if telephony is behind NAT.
IAX2 (Inter-Asterisk eXchange protocol) is a protocol for exchanging VoIP data between IP-PBX Asterisk. It is most adapted to NAT network address broadcasting, unlike SIP and H.323 it uses only one port 4569 of the UDP protocol for signaling and media stream. Similar to the SIP provider, the IAX provider asks for the server and connection port, phone number, login and password (if necessary), external prefix, DTMF mode and the registry string (if necessary) as parameters.
The tunnels are designed to connect telephony servers of several ICS CUBE. When creating a tunnel, you need to enter the following required parameters: name and login. Configuration process of the fields for specifying the max number of connections, password, prefix, registry string, and the “From User” option is similar to configuration of the corresponding fields of the “SIP provider” object.
Option that differs from the SIP provider settings is “Type of connection”. When using a communication channel to connect to an external server of the provider, you need to use the “Client” option. If other clients connect to the ICS via an external channel, use the “Server” option. In other words, one of the ICS CUBE is selected by the server, and the second one is selected by the client.