Select the provider to be displayed/configured.

The domain (also known as a "realm", "SIP domain" or "SIP realm") is required for the VoIP address. The structure of a user's VoIP address is similar to an e-mail address. It is made up of the VoIP phone number and the domain, separated by the @ character: <user>@domain.

There are both country-specific and provider-specific reasons why name resolution is not always successful via the default gateway. In such cases, another way should be chosen here.

There are several accounts at this domain and the name resolution for another account at this domain has already been configured.

The name resolution takes place through the IAD selected as used Internet access. The DNS server provided by the provider is used.

The DNS servers that are entered below are used via the selected IAD as Internet access.

IPv4 or IPv6 address. In case of an IPv6 address, the IAD chosen as used Internet access must also have an IPv6 address.

Select the Internet access for this provider. A default IAD with the gateway of the first network interface is automatically created. If you only want to use the first network interface, you can select the default IAD. If using both network interfaces, if necessary, the IAD to be selected must be configured first.

REGISTER requests are sent to the specified registrar (also known as a SIP registrar, registry, SIP server, SIP registry server).

Registrar's IP address or URL.

Port of the registrar. The standard value is 5060 (recommended).

The registration time specifies how many minutes the PBX waits before reregistering itself with the provider.

The SIP protocol SIPconnect 1.1 is used by the PBX.

If a SIP Forbidden (403) is received on an INVITE, the involved provider is re-registered.

If a SIP Timeout (408, 503) is received on an INVITE, the involved provider is re-registered.

If NAT traversal is switched on and a query is sent from a local IP address to the public network, the sending IP address is swapped with the public IP address. This function is performed in the reverse direction for the reply.

NAT traversal is not performed by the PBX, but the local address is used. The router which connects the local area network with the Internet should be a properly functioning SIP-aware router which performs NAT traversal.

NAT traversal is performed by the PBX. The STUN server entered for the provider is used. Only in connection with IPv4.

NAT traversal is performed by the PBX. To achieve this, the registrar for the provider must be switched on. The SIP request additionally contains the part rport, in which the IP addresses used (local, public) are transported.

NAT traversal is performed by the PBX via a static public IP address.

The public IP address is used in the VIA Header.

These parameters are used in the VIA header and prevent the PBX from losing the registration (activation recommended).

The public IP address is used in the Contact Header.

This parameter is used in the VIA header and prevents the PBX from losing the registration (activation recommended).

NAT Keep Alive data packets are sent according to the selected method and the set interval.

Select the data packets sent to maintain the NAT mapping.

Sending NAT Keep Alive data packets

Sending SIP OPTIONS data packets

After the time specified here has passed, NAT Keep Alive data packets are sent to the Firewall to maintain the NAT mapping.

The PBX does not use outbound proxy.

The PBX automatically determines an outbound proxy.

The PBX uses the outbound proxy configured for the provider.

The SIP port on the PBX is used both as an incoming port and an outgoing port (i.e. for communications with the provider). The SIP port must be different for each provider and cannot be the same as the SIP port of the internal VoIP registrar or the SIPS port of the internal VoIP registrar.

At the specified intervals, the SIP session timer checks whether the connection is still present. (The PBX cannot detect whether a call connection is disrupted due to problems on the provider side.)

The negotiation of SIP option „100rel“ is disabled. This might be necessary if the provider supports „SIP-Forking“.

A connection is not set up to transmit the digits dialled here to the provider until the delay time after the last digit is dialled has passed, all the digits are then sent as a block. Time measurement restarts after the entry of each digit. After the selected time has passed, the PBX recognises the most recent digit as the final digit and will not allow any more numbers to be entered. Select as short a time as possible, because the PBX does not forward the entire phone number to the central office until the specified time has passed. Approximately three seconds is the default value for manual inputs.

This provider encrypts external connections to prevent listening in. The processes used to set up and shut down a connection, and signalling, are all encrypted by SIPS.

Encryption of SIPS via TLS.

Encryption of SIP via TLS. SRTP is used to encrypt call data.

No verification whether the certificate belongs to the domain/IP.

Displays information about the stored certificate or opens the certificates management.

Certificate for one-way SSL authentication.

Client certificate (.pem format) for the mutual SSL authentication.

Client key (.pem format) for the mutual SSL authentication.

Passphrase that may belong to the private key.

Transport protocol used.

IPv4 operating mode is used for communications with the provider.

IPv6 operating mode is used for communications with the provider.

The PBX first checks whether the provider supports IPv6. If so, operating mode IPv6 is used for communications with the provider. Otherwise, operating mode IPv4 is used.