Grandstream Networks GSM gateway - Manual (Page 25)

Februaray-2006 Page 25 / 42
MasterVoIP
VoIP to GSM Gateway
Dial Plan Prefix
This value contains the dial plan prefix string (typically an ASCII numeric
string). If it is not blank, then this string will be used as a prefix to the target
URI string in the “To” header field of an INVITE message.
No Key Entry
Timeout Default is 4 seconds.
Use # as
Send Key
This parameter allows the user to configure the “#” key to be used as the
“Send”(or “Dial”) key. Once set to “Yes”, pressing this key will immediately
trigger the sending of the
dialed string collected so far. In this case, this key is essentially equivalent to the “(Re)Dial”
key. If set to “No”, this # key will then
be included as part of the dial string to be sent out.
Local SIP port This parameter defines the local SIP port the IP phone will listen and transmit
on. The default value is 5060.
Local RTP port
This parameter defines the local RTP-RTCP port pair the IP phone will listen and transmit
on. It is the base RTP port for channel 0. When configured,
channel 0 will use this port value for RTP and the port_value+1 for its RTCP;
channel 1 will use port_value+2 for RTP and port_value+3 for its RTCP. The
default value is 5004.
Use Random Port
This parameter, when set to Yes, will force random generation of both the local
SIP and RTP ports. This is usually necessary when multiple IP phones are
behind the same NAT.
keep-alive interval
The VoIP Client ATA sends a UDP package to the SIP server periodically in
order to keep the port open on the router. This parameter defines the interval
time that HT286 send the UDP package. The default setting is 20 second.
Use NAT IP NAT IP address used in SIP/SDP message. Default is blank.
Proxy-Require SIP Extension to notify SIP server that the unit is behind the NAT/Firewall.
NAT Traversal
This parameter defines whether the phone NAT traversal mechanism will be activated or
not. If activated (by choosing “Yes”) and a STUN server is also specified, then the phone
will behave according to the STUN client specification. Under this mode, the embedded
STUN client inside the phone will attempt to detect if and what type of firewall/NAT it is
behind through communication with the specified STUN server. If the detected NAT is a
Full Cone, Restricted Cone, or a Port-Restricted Cone, the phone will attempt to use
its mapped public IP address and port in all the SIP and SDP messages it sends out.
If this field is set to “Yes” with no specified STUN server, then the phone will
periodically (every 20 seconds by default) send a blank UDP packet (with no
payload data) to the SIP server to keep the “hole” on the NAT open.
Firmware Upgrade This radio button will enable VoIP Client ATA to download firmware or configuration file
through either TFTP or HTTP.