Manuale Utente (Inglese)

On the 'Telephone line settings' page, specify the connection settings as provided by your IP telephony service provider:

SIP account section

Enable SIP registration — SIP registration must be enabled to connect to most IP telephony operators. Disable SIP registration if you want to configure the line to connect to an operator without registration, with authentication by IP address. In addition, disabling registration allows you to set up direct calls between two Keenetic devices without using an IP telephony provider (hereinafter "p2p calls").

Provider — select an IP telephony operator from the list to automatically configure a line to connect to that provider. If your ISP is not on the list, select 'Other' and configure the phone line manually. You can contact Keenetic technical support to request to add a provider to the list.

Line name — the name of the line can be anything. It appears in the system call log and message log entries created for incoming and outgoing external calls.

By default, some of the settings configured automatically when an operator is selected are hidden. Click 'Show settings' if you wish to configure them manually.

SIP registrar — IP address or domain name of the SIP registration server. If the server uses a non-standard port (other than 5060), specify it to the right, separated by a colon. When SIP registration is disabled, this field is not displayed.

Registration timeout — the period of validity of SIP registration on the IP telephony operator's server, after the expiry of which registration should be resumed. The server can change this parameter during the registration process. Registration is required in order to receive incoming calls. When SIP registration is disabled, this field is not displayed.

SIP proxy — the proxy server of the IP telephony operator through which SIP signalling messages must be routed. If the server uses a non-standard port (other than 5060), specify it to the right with a colon. For p2p calls, you need to specify the IP address and port of the other Keenetic device here.

SIP domain — is the name of the SIP domain where the user is registered (right side of the SIP URI after the '@' symbol). When configuring p2p calls, the same domain (any domain) must be configured on both Keenetic devices.

SIP transport — the transport protocol (onwards: transport) used to transmit SIP signalling messages. Select the transport protocol that your IP telephony operator supports:

UDP, TCP and TLS transports use the IPv4 network layer protocol for data transfer. When using IPv4 to communicate with an IP telephony operator's server, you should select these transports.

UDP6, TCP6 and TLS6 are UDP, TCP, and TLS transports that use the IPv6 network layer protocol for data transfer. You should select these specific transports to communicate with an IP telephony operator's server using IPv6.

SIP security mode — this setting is only available when the TLS transport protocol is selected. Select the security type when using TLS:

Voice transmission protocol:

DTMF signal transmission method — during an established telephone connection, it is sometimes necessary to additionally dial the caller's extension number, voicemail control codes, etc. When the keypad buttons are pressed, the corresponding DTMF signals (0, 1...9, # and *) are transmitted to the remote party over the VoIP connection using one of three methods:

Select the DTMF transmission method that your IP telephony operator supports.

Display name — the name that is displayed on the called party's phone when an outgoing call is made.

SIP User ID — the SIP User ID (the left side of the SIP URI before the '@' symbol).

SIP Auth ID — the name used when authenticating to the IP telephony operator's servers. This field is not displayed when SIP registration is disabled.

Password — the password used when authenticating to the IP telephony operator's servers. This field is not displayed when SIP registration is disabled.

NAT traversal section

Use STUN — this option should be used when the Keenetic device is connected to the IP telephony operator's public SIP server via a NAT router and has a local IP address on the WAN interface. It allows you to receive the external IP address and NAT UDP ports associated with the local SIP and RTP ports of Keenetic Phone Station from the STUN server. The received data is placed in the Via, Contact, Connection Address and Media Port headers of SIP signalling messages. It should be noted that STUN technology does not work with symmetric type NAT.

Get to know own public IP address from SIP server — with this option, Keenetic Phone Station receives the IP address (or NAT IP address) from the SIP registration server and overwrites the relevant fields in the Via, Contact and SIP/SDP headers with it. This ensures two-way audibility and successful exchange of SIP signalling messages. Activate this option when a secondary channel such as a VPN tunnel is used to communicate with the server or when there is a symmetrical NAT between the Keenetic router and the public IP telephony server with which STUN technology does not work.

Keep-Alive interval — Keenetic periodically sends Keep-Alive messages to the SIP proxy signal port at a set interval to keep an open connection to the server via NAT.

This is necessary to ensure that incoming calls are received from the server.

Audio codecs section

When used with the Keenetic Linear adapter, Keenetic Phone Station supports the following codecs:

Codec priority corresponds to its position in the list, i.e. the codec in the top position has the highest priority. Codec priorities are taken into account for outgoing calls. The called party selects the codec with the highest priority that it supports. You can change the codec priority by dragging the line to the desired position. To drag and drop, use the special marker located at the beginning of each line.

The codecs can be turned on and off. Switched-off codecs are not used for voice communication.

Dial rules and priority of lines section

Line priority — the telephone line priority number. The higher the number, the higher the priority. Priorities are taken into account when selecting a line for an outgoing call. The telephone exchange selects the line with the highest priority among those that are allowed for this telephone and have appropriate dialling rules.

Line selection code — line selection code #0...#9 allows you to select the desired line for an outgoing call. To select a line, you need to dial its code, then the subscriber's number. When you select a line with the code, the dial rules are ignored, and you can call a number that does not comply with the dial rules of that line. Only lines that are allowed for the phone can be dialled with the code. To prohibit line selection using a code, select 'no' from the drop-down list of codes.

Dial rule — the dial rule describes the numbers for which outgoing calls are allowed through this line. In the absence of a dial rule, any numbers are allowed to be called.

When an outgoing call comes in, the Phone Station selects the line with the highest priority among those that have dial rules to which the dialled number corresponds. If the number does not match the dial rules, the line without dial rules with the highest priority is selected. A line without dial rules always has a lower priority than any line with dial rules. Only a line in which outgoing calls are allowed through this port can be selected for a call.

Dial rule syntax:

The '|' character separates two or more rules in a string.

Examples of dial rules:

Prefix substitute rule for incoming calls — allows you to replace or delete individual digits or groups of digits in a caller number that appears on the telephone display when a call comes in. The symbol '>' is used for replacement. To the left of the '>' must be a sequence of digits to be replaced by the sequence to the right of this symbol. The replacement expression must be enclosed in parentheses. There can be more than one replacement expression in a prefix replacement rule. Otherwise, the prefix substitution rule has the same syntax as the corresponding set rule discussed above.

An example of a prefix substitution rule:

Incoming call forwarding — this function allows you to forward incoming calls to certain numbers under certain conditions.

Activate the option corresponding to the required forwarding condition; on the right, indicate the telephone number to which you want to forward incoming calls.

Call forwarding conditions:

When a call is forwarded, Keenetic Phone Station sends a special SIP message to the calling party with the SIP URI and the forwarding number. All further actions on subscriber connection to this number are performed by the subscriber's equipment and/or the IP telephony operator's server.

Do not disturb (DND) — enable this option if you do not want to be disturbed by incoming calls. When DND is activated, calls are not received on the phones, callers are notified that you are busy, and information about missed calls is recorded in the call history and system log.

Delete line — press this button if you wish to delete the line. Once you have confirmed, the line will be permanently deleted, and you will no longer be able to use the IP telephony service of this operator.