SIP
Provider
Select the provider to be displayed/configured.
Further help under External Internet TelephonyDomain
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.
Further help under External Internet TelephonyName resolution
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.
Name resolution via IP configuration
There are several accounts at this domain and the name resolution for another account at this domain has already been configured.
Note: It is possible to set the name resolution differently for the accounts of the same domain.via used Internet access
The name resolution takes place through the IAD selected as used Internet access. The DNS server provided by the provider is used.
Use own DNS (via IAD)
The DNS servers that are entered below are used via the selected IAD as Internet access.
Further help under External Internet TelephonyFirst DNS server / Second DNS server
IPv4 or IPv6 address. In case of an IPv6 address, the IAD chosen as used Internet access must also have an IPv6 address.
Further help under External Internet TelephonyUsed Internet access
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.
Further help under External Internet TelephonyRegistrar
REGISTER requests are sent to the specified registrar (also known as a SIP registrar, registry, SIP server, SIP registry server).
Note: The registrar should only be disabled if the provider requires this. No NAT Keep-Alive is performed if the registrar is disabled. An internal network is usually protected from actions from the outside by, for example, a firewall integrated in the router. Without NAT Keep-Alive, the firewall's security functions will defend you against attempted accesses from the Internet. For this reason, you may need to implement port forwarding for incoming SIP packets on the SIP port set up in the provider's configuration.Further help under External Internet TelephonyRegistrar|Address
Registrar's IP address or URL.
Further help under External Internet TelephonyRegistrar|Port
Port of the registrar. The standard value is 5060 (recommended).
Caution:Divergent entries may lead to malfunctions as soon as the provider makes changes. Only the standard value allows an SRV query.Caution:Each time a port is opened on the router, this creates a potential security risk. It is essential you put additional protective measures in place.Note: The page Administration > Network > Ports shows an overview of the PBX ports (incoming and outgoing).Further help under External Internet TelephonyRegistrar|Time lapse for the registration
The registration time specifies how many minutes the PBX waits before reregistering itself with the provider.
Note: In the test phase, after configuring the provider, we recommend you enter a low value (e.g. three minutes). After completing the test phase, you can enter a much higher value, since some providers also reject calls if too many attempts are made to register.Further help under External Internet TelephonyRegistrar|SIP registration in accordance with RFC 6140
The SIP protocol SIPconnect 1.1 is used by the PBX.
Further help under External Internet TelephonyNAT traversal
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.
deactivated
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.
activated (STUN)
NAT traversal is performed by the PBX. The STUN server entered for the provider is used. Only in connection with IPv4.
activated (rport)
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.
Note: If problems occur with single-sided call connections, use the option with use of STUN instead.Caution:Each time a port is opened on the NAT router, this creates a potential security risk. It is essential you put additional protective measures in place.Public IP
If the provider requires it: NAT traversal is performed by the PBX via a static public IP address.
use static public IP address in VIA Header
If the provider requires it: The static IP address is used in the VIA Header.
use static public IP address in Contact Header
If the provider requires it: The static IP address is used in the Contact Header.
Further help under External Internet TelephonyInterval for NAT Keep-Alive|active
After the time specified here has passed, NAT Keep-Alive packets are sent to the firewall to maintain the NAT mapping.
Note: Some providers lock accounts if the NAT keep alive interval is too short. As a rule, this is reported with error message 503 during SIP registration. If you encounter this problem, we recommend you set a higher value (e.g. 180).Further help under External Internet TelephonyOutbound Proxy/Session Border Controller
deactivated
The PBX does not use outbound proxy.
automatically
The PBX automatically determines an outbound proxy.
manually
The PBX uses the outbound proxy configured for the provider.
Note: The page Administration > Network > Ports shows an overview of the PBX ports (incoming and outgoing).Further help under External Internet TelephonySIP port
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.
Note: The page Administration > Network > Ports shows an overview of the PBX ports (incoming and outgoing).Further help under External Internet TelephonySIP session timer
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.)
Note: When the SIP session timer is switched on, this can result in increased breaks in conversations after the set interval, if a provider has not implemented the renewal of the session cleanly. In this case, you should adjust the interval of the SIP session timer or switch it off.Further help under External Internet TelephonyOnly en-bloc dialling
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.
Further help under External Internet TelephonySIPS
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.
Further help under External Internet TelephonySIPS|URL structure
SIPS
Encryption of SIPS via TLS.
SIP/TLS
Encryption of SIP via TLS. SRTP is used to encrypt call data.
Further help under External Internet TelephonySIPS|Disable host name check
No verification whether the certificate belongs to the domain/IP.
Further help under External Internet TelephonySIPS|Manage certificates
Displays information about the stored certificate or opens the certificates management.
Further help under CertificatesSIP transport
Transport protocol used.
Further help under External Internet TelephonyIP protocol
IPv4
IPv4 operating mode is used for communications with the provider.
IPv6
IPv6 operating mode is used for communications with the provider.
Automatically
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.
Note: If the actual IP operating mode is IPv6, STUN cannot be used as a NAT method for neither SIP nor RTP.Further help under External Internet Telephony