Februaray-2006 Page 25 / 42 MasterVoIP VoIP to GSM Gateway
Dial Plan Prefix
This value contains the dia l plan prefix string (typically an ASCII nu meric
string). If it is not blank, then this string will be used as a prefix to the target
URI string in the “To” header f ield of an INVITE message.
No Key Ent ry
Timeout
Default is 4 seconds.
Use # as
Send Key
This parameter allows the us er to configure the “#” key to be use d as the
“Send”(or “Dial”) key. Once set to “Yes”, pressing this key will immedia tely
trigger the sending of the
dialed string collected so f ar. In this case, this key is essentially equ ivalent to the “(Re)Dial”
key. If set to “No”, this # key wil l 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 a nd transmit
on. The default value is 50 60.
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 f or channel 0. When configured,
channel 0 will use this port value for RTP and the port_value+1 for i ts RTCP;
channel 1 will use port_val ue+2 for RTP and port_value+3 for it s RTCP. The
default value is 5004.
Use Rand om 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-aliv e interval
The VoIP Client ATA sen ds a UDP package to the SIP server period ically in
order to keep the port op en on the router. This parameter defines th e interval
time that HT286 send the U DP package. The default setting is 20 second.
Use NAT I P
NAT IP address used in SI P/SDP message. Default is blank.
Proxy-Requ ire
SIP Extension to notify SIP se rver that the unit is behind the NAT /Firewall.
NAT Trav ersal
This parameter defines wh ether the phone NAT traversal mechani sm will be activated or
not. If activated (by choosin g “Yes”) and a STUN server is also s pecified, then the phone
will behave according to t he STUN client specification. Under this m ode, the embedded
STUN client inside the ph one will attempt to detect if and what typ e of firewall/NAT it is
behind through communi cation with the specified STUN server. If the detected NAT is a
Full Cone, Restricted Cone, or a Port-Restricted Cone, the phone w ill attempt to use
its mapped public IP addre ss and port in all the SIP and SDP messa ges it sends out.
If this field is set to “Yes” with no specified STUN server, then the phone will
periodically (every 20 secon ds by default) send a blank UDP pac ket (with no
payload data) to the SIP se rver to keep the “hole” on the NAT open .
Firmware Upgrade
This radio button will ena ble VoIP Client ATA to download firmwar e or configuration file
through either TFTP or HTTP .