SIP
Proceed as follows:
•Separately for VoIP providers
•Using the Configuration Manager (Public switched tel. networks > Voice over IP (VoIP page) > Provider > Configure)
Note: You can obtain the data for the settings either directly from the VoIP provider or from suitable lists in the Internet.Requirements:
•For entering a URL: configured DNS server
•For using a STUN server: IP operating mode IPv4
PROVIDER
Select the provider to be displayed/configured.
Domain
The domain (also known as a "realm", "SIP domain" or "SIP realm") is required for the VoIP address. The structure of a subscriber'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: <subscriber>@domain.
Name 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 alreday 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 via the selected IAD under 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 under Used Internet access.
First DNS server/Second DNS server
IPv4: 4 blocks with 3 digits separated by a dot (.).
IPv6: 8 blocks with 4 digits or letters separated by a colon (:).
Note: If an IPv6 address is entered as DNS server, then the IAD selected under Used Internet access must also have an IPv6 address.Registrar | Port
REGISTER requests are sent to the specified registrar (also known as a SIP registrar, registry, SIP server, SIP registry server). Enter this data as an IP address or a URL.
Note: The page Overviews > Ports shows an overview of the PBX ports (incoming and outgoing).Note: You should only switch off the registrar if the VoIP provider requires you to do so. No NAT Keep-Alive is performed if the registrar is disabled. An internal network is usually protected from external manipulation 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 VoIP provider's configuration.
Caution: Each time a port is opened on the router, this creates a potential source of danger.•It is essential you put additional protective measures in place.
Time limit for registration (1 to 60 min.)
The registration time specifies how many minutes the PBX waits before reregistering itself with the VoIP provider.
Note: In the test phase, after configuring the VoIP 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 VoIP providers also reject calls if too many attempts are made to register.SIP registration in accordance with RFC 6140
The SIP protocol SIPconnect 1.1 is used by the PBX.
NAT traversal
NAT (Network Address Translation) enables you to convert a particular IP address that is used within a network (e.g. a local network) into a different IP address that can be used by a different network (e.g. the public switched phone network). 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.
Disabled (use local address
NAT traversal is not performed by the PBX. The router which connects the local area network with the Internet should be a properly functioning, SIP-aware router which performs NAT traversal.
Enabled with use of STUN
NAT traversal is performed by the PBX. To achieve this, a STUN server must also be specified for the VoIP provider.
Enabled
NAT traversal is performed by the PBX. To achieve this, the registrar for the VoIP provider must be switched on. The SIP request also contains an "rport" part in which the IP addresses used here (local, public) are transported.
Note: If Enabled is selected: If problems occur with single-sided call connections, use Enabled with use of STUN instead.Caution: Each time a port is opened on the NAT router, this creates a potential source of danger.•It is essential you put additional protective measures in place.
Use static public IP address
Requirements:
•Registered fixed IP address (see Preferences)
NAT traversal is performed by the PBX. The VoIP provider requires a connection via a static public IP address.
If the VoIP provider still requires this static IP address in a header, select the appropriate option:
•Use static public ID address in VIA Header
•Use static public IP address in Contact Header
Interval 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 VoIP providers block accounts if the Interval for NAT Keep-Alive is too short. As a rule, this is reported with error message 503 during SIP registration. If you experience this problem, it is advisable to increase the value (e.g. to 180).Default
Default setting
SIP options
Use this setting for non-registered connections between the VoIP provider and the PBX for re-registration.
Outbound proxy
The outbound proxy is an intermediate server that processes all VoIP requests and connections going to the provider (except for registration).
automatically
The PBX automatically determines an outbound proxy.
Manual Port
The outbound proxy is a fixed default setting. Enter this data as an IP address or a URL.
Note: The Overviews > Ports page shows an overview of the PBX ports (incoming and outgoing).SIP port (1 to 65535)
The SIP port on the PBX is used both as an incoming port and an outgoing port (i.e. for communications with the VoIP provider).
Note: A different SIP port must be used for each VoIP provider.Note: The SIP port for a VoIP provider cannot be the same as the SIP port of the internal VoIP registrar or the SIPS port of the internal VoIP registrar.Note: The Overviews > Ports page shows an overview of the PBX ports (incoming and outgoing).SIP session timer (5 to 60 min.)
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 may result in the call being interrupted more frequently after the specified interval, if a VoIP provider has not implemented session renewal properly. In this case, you should adjust the interval of the SIP session timer or switch it off.Deactivation of SIP Prack (RFC 3262)
The negotiation of SIP option „100rel“ is disabled. This might be necessary if the provider supports „SIP-Forking“.
Only en-bloc dialling | Delay time after last digit dialled (3 to 15 sec.)
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.
Note: En-bloc dialling is a good idea, for example, when dialling single digits causes problems in sub-system operation.Note: Select as short a time as possible, because the PBX does not forward the entire phone number to the public exchange until the specified time has passed.Note: Approximately three seconds is the default value for manual inputs.SIPS URL structure
Requirements:
•External VoIP channel that supports SIPS (optional, see Technical Data in the Instructions)
•A certificate for the VoIP provider stored in the PBX
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.
SIPS
Encryption of SIPS via TLS.
SIP/TLS
Encryption of SIP via TLS.
Note: SRTP is used to encrypt call data.Disable host name check
No verification whether the certificate belongs to the domain/IP.
Manage certificates
Opens the certificates management.
SIP transport
Selection of the transport protocol.
IP protocol
IPv4
IPv4 operating mode is used for communications with the VoIP provider.
IPv6
IPv6 operating mode is used for communications with the VoIP provider.
automatically
The PBX first checks whether the VoIP provider supports IPv6. If so, operating mode IPv6 is used for communications with the VoIP provider. Otherwise, operating mode IPv4 is used.
Note: If the actual IP operating mode is IPv6, STUN cannot be used as a NAT method either for SIP or RTP.Increased security
The provider provides better levels of security to protect the PBX against external attacks.