Grandstream Networks GXP-1200 User Manual page 35

SIP Registration Failure
Retry Wait Time
SIP T1 Timeout
SIP T2 Interval
SIP Transport
Use RFC3581
Symmetric Routing
NAT Traversal (STUN)
SUBSCRIBE for MWI:
SUBSCRIBE for
Registration Event
PUBLISH for Presence
Proxy-Require
Voice Mail UserID
Send DTMF
Early Dial
Dial Plan Prefix
BLF Call-pickup Prefix
Delayed Call Forward
Wait Time
Enable Call Features
Call Log
Grandstream Networks, Inc.
Retry registration if the process failed. Default is 20 seconds.
RFC 3261 SIP T1 timer. Default is 1 second.
RFC 3261 SIP T2 timer. Default is 0.5 seconds.
Choose SIP Transport between UDP and TCP. Default is UDP.
Default No. When selected the phone will follow the routing procedures specified
in RFC3581.
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.
Default is No. When set to "Yes" a SUBSCRIBE for Message Waiting Indication
will be sent periodically.
Default is No. This is mainly used for IMS purposes. When enabled, the terminals
should store the Service-Route header values after successfully registered, and
thereafter add a route header with the values stored in the Service-Route when
initiating a request excluding REGISTER.
Enable Presence feature.
SIP Extension to notify SIP server that the unit is behind the NAT/Firewall.
When configured, user can access messages by pressing "MSG" button. This ID
is usually the VM portal access number.
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.
Default is No. Use only if proxy supports 484 responses.
Sets the prefix added to each dialed number.
Default is '**". This prefix is prepended when answering call with BLF key.
Time waited before the call is forward to a number or VM.
Default is 20 seconds.
Default is Yes. If set to "No", Call transfer, Call Forwarding & Do-Not-Disturb are
supported locally provided ITSP support those features. In addition, "ForwardAll"
softkey will be hidden if call feature code is disabled for Account 1.
User can choose to disable Call Log and what kind of calls to log.
GXP User Manual
Firmware 1.2.3.5
Page 35 of 40
Last Updated: 7/2010