2N NetStar Admin manual - Manual (Page 57)

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3.6 SIP
SIP Gateway
The SIP Gateway virtual port is used for creating a trunk between two PBXs or
connecting a PBX to the public network via a VoIP provider.
Stack status
This field displays information on the stack and its current status.
SOCK_TCP_ERROR – the TCP socket has not been opened.
SOCK_UDP_ERROR – the UDP socket has not been opened.
CREDS_IN_ERROR – the authorisation server is unavailable.
CREDS_OUT_ERROR – the authorisation client is unavailable.
REALM_CONFLICT – the Realm collides with another port's Realm/Alias.
STUNNING – the public IP address is being obtained from the STUN server.
STUN_TIMEOUT – the STUN server is unavailable.
EXPIRED – the public IP address validity has expired.
SIP_REGISTERING – the gateway registration is in progress.
REG_TIMEOUT – the REGISTRAR server is unavailable.
REG_NOT_AUTH – the registration has not been authorised.
REG_REJECTED – the registration has been rejected with an error.
Common parameters
Port – here define the local gateway port to communicate with the other party.
Realm (Domain) – define the domain over which the gateway communicates.
The domain and ports specified here help route calls to the gateway. The
Realm(Domain) + port items are checked in the Request–URI field for incoming
INVITE messages. If the setting matches the gateway SIP, the packets are
routed to the gateway. The INVITE messages whose Request–URI items are
included in the Alias field are served too.
Via/Contact – define the contents of the
and headers. The Via Contact
following options are available:
IP address – fill in the CPU IP address.
FQDN – fill in the PBX Hostname as entered for the PBX IP interface.
NAT – fill in the public IP address and NAT port for the opponent's sending
of signalling messages. Packets are routed to the PBX according to the port
routing and router IP address settings.
STUN – enter the STUN server address and port for finding the current
address behind the NAT.
Authorisation required – use this option to enable the other party's
authorisation for incoming calls. User login data are used for this purpose. All
logins are always used.
Send congestion tone – enable transmission of the congestion tone from the
PBX or network in case the opposite subscriber hangs up.