Grandstream Networks GHP6 Series Administration Manual page 19

Allow Unsolicited REFER
Accept Incoming SIP from
Proxy Only
Check SIP User ID for
Incoming INVITE
Allow SIP Reset
Authenticate Incoming INVITE
On Hold Reminder Tone
Music On Hold URI
Special Feature
Phone Settings Page Definition
Basic Settings
Local RTP Port
Local RTP Port Range
Use Random Port
Keep-Alive Interval
STUN Server
Use NAT IP
Delay Registration
Configures whether to dial the number carried by Refer-to header after receiving out-of-
dialog SIP REFER request actively.
If set to "Disabled", the phone will send error warning and stop dialing.
If set to "Enabled/Force Auth", the phone will dial the number after sending authentication.
If the authentication fails, it will stop dialing.
If set to "Enabled", the phone will dial all numbers carried by SIP REFER.
When set to "Yes", the SIP address of the Request URL in the incoming SIP message will be
checked. If it does not match the SIP server address of the account, the call will be
rejected. The default setting is "No".
If set to "Yes", SIP User ID will be checked in the Request URI of the incoming INVITE. If it
does not match the phone's SIP User ID, the call will be rejected. The default setting is "No".
Allow SIP Notification message to perform factory reset.
The default setting is "No".
If set to "Yes", the phone will challenge the incoming INVITE for authentication with SIP 401
Unauthorized response.
The default setting is "No".
MOH ( Music on Hold )
Configures to play reminder tone when the call is on hold.
Music On Hold URI to call when a call is on hold if server supports it.
Specifies the server type for special requirements.
Configures the local RTP port used to listen and transmit. The valid range is 1024 to 65400
and it must be even.
Configures the range of local RTP port. Valid value is from 24 to 10000.
If set to "Yes", the parameter will force random generation of both the local SIP and RTP
ports. This is usually necessary when multiple phones are behind the same full cone NAT.
This parameter must be set to "No" for incoming direct IP calls (outgoing IP calls are not
affected).
Specifies how often the phone sends a blank UDP packet to the SIP server in order to keep
"ping hole" on the NAT router to open.
This option sets IP address or Domain name of the STUN server. Only non-symmetric NAT
routers work with STUN.
Configures the NAT IP address used in SIP/SDP messages. It should ONLY be used if
required by your ITSP.
Configures the specific time that the account will be registered after booting up.