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Grandstream Networks, Inc. GXP User Manual Page 33 of 38
Firmware 1.1.6.44 Last Updated: 12/2008
Local SIP Port
This parameter defines the local SIP port used to listen and transmit. The default
value for Account 1 is 5060. It is 5062, 5064, 5066 for Account 2, Account 3 and
Account 4 respectively.
SIP Registration Failure
Retry Wait Time
Retry registration if the process failed. Default is 20 seconds.
SIP T1 Timeout
RFC 3261 SIP T1 timer. Default is 1 second.
SIP T2 Interval
RFC 3261 SIP T2 timer. Default is 0.5 seconds.
SIP Transport
Choose SIP Transport between UDP and TCP. Default is UDP.
Use RFC3581
Symmetric Routing
Default No. When selected the phone will follow the routing procedures specified
in RFC3581.
NAT Traversal (STUN)
This parameter activates the NAT traversal mechanism. If activated (by choosing
“Yes”) and a STUN server is also specified, the phone performs according to the
STUN client specification. Using this mode, the embedded STUN client detects if
and what type of NAT/Firewall configuration is used. If the detected NAT is a Full
Cone, Restricted Cone, or a Port-Restricted Cone, the phone will use its mapped
public IP address and port in all of its SIP and SDP messages. If the NAT
Traversal field is set to “Yes” with no specified STUN server, the GXP will
periodically (every 20 seconds or so) send a blank UDP packet (with no payload
data) to the SIP server to keep the “hole” on the NAT open.
Subscribe for MWI:
Default is No. When set to “Yes” a SUBSCRIBE for Message Waiting Indication
will be sent periodically.
PUBLISH for Presence
Enable Presence feature.
Proxy-Require
SIP Extension to notify SIP server that the unit is behind the NAT/Firewall.
Voice Mail UserID
When configured, user can access messages by pressing “MSG” button. This ID
is usually the VM portal access number.
Send DTMF
This parameter specifies the mechanism to transmit DTMF digit. There are 3
supported modes: in audio which means DTMF is combined in audio signal (not
very reliable with low-bit-rate codec), via RTP (RFC2833), or via SIP INFO.
Early Dial
Default is No. Use only if proxy supports 484 responses.
Dial Plan Prefix Sets the prefix added to each dialed number.
Delayed Call Forward
Wait Time
Time waited before the call is forward to a number or VM.
Default is 20 seconds.
Enable Call Features
Default is No. If set to “Yes”, Call transfer, Call Forwarding & Do-Not-Disturb are
supported locally provided ITSP support those features.
Call Log
User can choose to disable Call Log and what kind of calls to log.