Grandstream IP Phone Configuration
MAC Address:   00.0B.82.11.C8.47
Software Version:   Program--1.0.4.43    Bootloader--1.0.0.10    HTML--1.0.0.10
    detected NAT type is Core NAT   ( Nat is supported)
   
Admin Password:   (password to configure this IP phone)
IP Address:( Both dynamical and static will work)

 
dynamically assigned via DHCP or PPPoE
  PPPoE account ID:
  PPPoE password:
statically configured as:
       IP Address:   . . .
       Subnet Mask:   . . .
       Default Router:   . . .
       DNS Server 1:   . . .
       DNS Server 2:   . . .
SIP Server:   (e.g., sip.mycompany.com, or IP address)
Outbound Proxy:   (e.g., proxy.myprovider.com, or IP address, if any)
SIP User ID:   (the user part of an SIP address)
Authenticate ID:   (can be identical to or different from SIP User ID)
Authenticate Password:  
Name:   (optional, e.g., John Doe)
 
Advanced Options:  
Preferred Vocoder:
(in listed order)		   choice 1:   
  choice 2:   
  choice 3:   
  choice 4:   
  choice 5:   
  choice 6:   
G723 rate:   6.3kbps encoding rate       5.3kbps encoding rate
Silence Suppression:   No      Yes
Voice Frames per TX:   (up to 10/20/32/64 frames for G711/G726/G723/other codecs respectively)
IP QoS:   (IP Diff-Serv or Precedence value for RTP)
VLAN Tag:   (802.1Q/802.1p classification for RTP)
SIP User ID is
phone number: 		  No      Yes
Dial Plan:   (dial plan prefix string)
SIP Registration:   Yes     No
Clear Registration On Reboot:   Yes     No
Register Expiration:    (in minutes. default 1 hour, max 45 days)
Early Dial:   No      Yes (use "Yes" only if proxy supports 484 response)
Use # as Dial Key:   No      Yes (if set to Yes, "#" will function as the "(Re-)Dial" key)
local SIP port:   (default 5060)
local RTP port:   (1024-65535, default 5004)
Use random port:   No      Yes
NAT Traversal:   No   
  Yes, STUN server is: (URI or IP:port)
keep-alive interval:   (in seconds, default 20 seconds)
TFTP Server:   . . . (for remote software upgrade and configuration)
Voice Mail UserID:   (User ID/extension for 3rd party voice mail system)
Offhook Auto-Dial:   (User ID/extension to dial automatically when offhook)
Send DTMF:   in-audio     via RTP (RFC2833)     via SIP INFO
DTMF Payload Type:  
Send Flash Event:   No      Yes   (Flash will be sent as a DTMF event if set to Yes)
NTP Server:   (URI or IP address)
Time Zone:  
Daylight Savings Time:   No      Yes   (if set to Yes, display time will be 1 hour ahead of normal time)
Send Anonymous:   No      Yes   (caller ID will be blocked if set to Yes)
     

 

1 There is bug in old firmware version.  Please update to the latest firmware version.

http://grandstream.com/y-firmware.htm

2   How do I configure two or more terminals behind one router in a network?

 

In order to allow two SIP terminals on a network to operate behind one router, you must make the following configuration changes.

Configuration of the IP devices

Reconfigure the DHCP mode and assign a static IP address to both devices.

Call the IP address of a device in the browser.

The devices are pre-configured on the following ports: