Grandstream Device Configuration
STATUS BASIC SETTINGS ADVANCED SETTINGS 1 ADVANCED SETTINGS 2
Admin Password:   (purposely not displayed for security protection)
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:   (purposely not displayed for security protection)
Name:   (optional, e.g., John Doe)
Home NPA:  
 
Advanced Options:  
Preferred Vocoder:
(in listed order)
  choice 1:  
  choice 2:  
  choice 3:  
  choice 4:  
  choice 5:  
  choice 6:  
  choice 7:  
G723 rate:   6.3kbps encoding rate       5.3kbps encoding rate
iLBC frame size:   20ms       30ms
iLBC payload type:   (between 96 and 127, default is 97)
Silence Suppression:   No       Yes
Voice Frames per TX:   (up to 10/20/32/64 for G711/G726/G723/other codecs respectively)
Fax Mode:   T.38 (Auto Detect)   Pass-Through
Layer 3 QoS:   (Diff-Serv or Precedence value)
Layer 2 QoS:   802.1Q/VLAN Tag     802.1p priority value (0-7)
Allow incoming SIP messages
from SIP proxy only:
  No      Yes
Use DNS SRV:   No      Yes
User ID is phone number:    No       Yes
SIP Registration:   Yes     No
Unregister On Reboot:   Yes     No
Register Expiration:   (in seconds. default 1 hour, max 45 days)
Early Dial:   No       Yes (use "Yes" only if proxy supports 484 response)
Allow outgoing call without Registration:   No       Yes
Dial Plan Prefix:   (this prefix string is added to each dialed number)
No Key Entry Timeout:   (in seconds, default is 4 seconds)
Use # as Dial Key:   No       Yes (if set to Yes, "#" will function as the Dial key)
local SIP port:   (default 5060)
local RTP port:   (1024-65535, default 5004)
Use random port:   No      Yes
SIP Registration Failure Retry Wait Time:   (in seconds. Between 1-3600, default is 20)
NAT Traversal:   No   
  Yes, STUN server is: (URI or IP:port)
keep-alive interval:   (in seconds, default 20 seconds)
Use NAT IP   (used in SIP/SDP message if specified)
Use STUN keep-alive to detect networks connectivity:   No   
  Yes, total STUN response misses (mininum=3) before restart
Proxy-Require:  
SUBSCRIBE for MWI:   No, do not send SUBSCRIBE for Message Waiting Indication
  Yes, send periodical SUBSCRIBE for Message Waiting Indication
Offhook Auto-Dial:   (User ID/extension to dial automatically when offhook)
Enable Call Features:   No      Yes
(if yes, call features using star codes will be supported locally)
Use Bell-style
3-way Conference:
  No      Yes (if Yes, *23 will be disabled)
Disable Call-Waiting:   No      Yes
Disable Call-Waiting Caller-ID:   No      Yes
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)
     
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