user manual

Grandstream Networks, Inc. GXV3175 USER MANUAL Page 49 of 119
FIRMWARE VERSION 1.0.0.32 Updated : 12/2010
service provider.
SIP User ID User account information provided by the VoIP service provider; normally
similar to a telephone number or an actual telephone number.
Authenticate ID The authenticate ID for the SIP user. It can differ or be the same as the SIP
user ID.
Authenticate password The password that the GXV uses to authenticate with the ITSP (SIP) server.
After it is saved, this will appear as blank for security reasons. The
maximum length is 25 characters.
Voice Mail User ID When this is configured, the user can dial to the voicemail server using the
MESSAGE button. This ID is normally the feature code for Voice Mail.
Name The Caller ID that will be displayed for the account.
Tel URI The default setting is “Disable”. If the Video phone has an assigned PSTN
number, this field should be set to “Enable”. If “User=Phone” is set, a
“User=Phone” parameter will be attached to the “From header” in the SIP
request to indicate the E.164 number.
Account/Network Settings
Outbound Proxy IP address or Domain name of the Outbound Proxy, or Media Gateway, or
Session Border Controller. Used by the GXV3175 for firewall or NAT
penetration in different network environments. If a symmetric NAT is
detected, STUN will not work and ONLY an Outbound Proxy will work.
DNS Mode The default is set to A Record. If the user wishes to locate the server by DNS
SRV, the user may select SRV or NATPTR/SRV.
NAT Traversal This setting decides whether the NAT traversal mechanism is activated. If it
is set to “STUN” and STUN server is configured, the GXV3175 will route
according the STUN server. In this mode, the STUN client embedded in the
phone will communicate with the appointed STUN server to examine which
type of Firewall/NAT setting is employed. If the type of NAT detected is Full
Cone, Restricted Cone or Port-Restricted cone, the phone will try to use
public IP addresses and port in all the SIP and SDP messages.
If the “NAT Traversal” is configured to be “Keep-alive”, the phone will send
an empty SDP packet (without payload data) to the SIP server once in 20
seconds to keep the NAT port open.
If VPN is used, users should select “VPN” for NAT Traversal.
Users could also set “Auto” or “UPnP” according to the network
configuration.
If an outbound proxy server is used, please configure this to be “NAT NO”.