Grandstream Device Configuration
STATUS BASIC SETTINGS ADVANCED SETTINGS FXS PORT FXO PORT
Account Active:   No      Yes
Primary SIP Server:   (e.g., sip.mycompany.com, or IP address)
Failover SIP Server:   (Optional, used when primary server no response)
Outbound Proxy:   (e.g., proxy.myprovider.com, or IP address, if any)
SIP Transport:   UDP       TCP       TLS   (default is UDP)
NAT Traversal (STUN):   No      No, but send keep-alive     Yes
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)
 
DNS Mode:   A Record      SRV      NAPTR/SRV
User ID is phone number:    No       Yes
SIP Registration:    No       Yes
Unregister On Reboot:    No       Yes
Outgoing Call without Registration:    No       Yes
Register Expiration:   (in minutes. default 1 hour, max 45 days)
SIP Registration Failure Retry Wait Time:   (in seconds. Between 1-3600, default is 20)
Local SIP port:   (default 5062)
Local RTP port:   (1024-65535, default 5012)
Use Random Port:   No      Yes
Remove OBP from Route Header:   No      Yes
Support SIP Instance ID:   No      Yes
Validate Incoming SIP Message:   No      Yes
Check SIP User ID for incoming INVITE:   No      Yes (no direct IP calling if Yes)
Allow Incoming SIP Messages
from SIP Proxy Only:
  No      Yes (no direct IP calling if Yes)
SIP T1 Timeout:  
SIP T2 Interval:  
 
DTMF Payload Type:  
Preferred DTMF method:
(in listed order)
  Priority 1:  
  Priority 2:  
  Priority 3:  
Proxy-Require:  
Use NAT IP:   (used in SIP/SDP message if specified)
 
Ring Timeout:   (10-300, default is 60 seconds)
Early Dial:   No       Yes   (use "Yes" only if proxy supports 484 response)
Dial Plan Prefix:   (this prefix string is added to each dialed number)
Use # as Dial Key:   No       Yes   (if set to Yes, "#" will function as the "Dial" key)
Dial Plan:  
SUBSCRIBE for MWI:   No, do not send SUBSCRIBE for Message Waiting Indication
  Yes, send periodical SUBSCRIBE for Message Waiting Indication
Anonymous Call Rejection:   No       Yes  
Special Feature:  
Session Expiration:   (in seconds. default 180 seconds)
Min-SE:   (in seconds. default and minimum 90 seconds)
Caller Request Timer:   No     Yes (Request for timer when making outbound calls)
Callee Request Timer:   No     Yes (When caller supports timer but did not request one)
Force Timer:   No     Yes (Use timer even when remote party does not support)
UAC Specify Refresher:   UAC   UAS     Omit (Recommended)
UAS Specify Refresher:   UAC   UAS (When UAC did not specify refresher tag)
Force INVITE:   No     Yes (Always refresh with INVITE instead of UPDATE)
INVITE Ring-No-Answer Timeout (sec):      (5-300 seconds. Default 40 seconds)
 
Preferred Vocoder:
(in listed order)
  choice 1:  
  choice 2:  
  choice 3:  
  choice 4:  
  choice 5:  
  choice 6:  
  choice 7:  
  choice 8:  
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)
AAL2-G726-16 Payload Type:   (between 96 and 127, default is 100)
AAL2-G726-24 Payload Type:   (between 96 and 127, default is 99)
AAL2-G726-32 payload type:   (between 96 and 127, default is 104)
AAL2-G726-40 Payload Type:   (between 96 and 127, default is 103)
G729E Payload Type:   (between 96 and 127, default is 102)
 
VAD:   No       Yes
Symmetric RTP:   No       Yes
Fax Mode:   T.38 (Auto Detect)   Pass-Through
Fax Tone Detection Mode:   Caller   Callee   Caller or Callee
Jitter Buffer Type:   Fixed   Adaptive
Jitter Buffer Length:   Low   Medium   High
SRTP Mode:   Disabled     Enabled but not forced   Enabled and forced
 
Caller ID Scheme:  
FSK Caller ID Minimum RX Level (dB):      (-96 - 0dB. Default -40dB)
FSK Caller ID Seizure Bits:      (0 - 800 bits. Default 70)
FSK Caller ID Mark Bits:      (1 - 800 bits. Default 40)
Caller ID Transport Type:  
Send Hook Flash To PSTN:   No      Yes   (If Yes, hook flash will be sent to PSTN upon receiving flash event from RFC2833 or SIP INFO)
Hook Flash Duration (ms):      (300 - 1500 milliseconds. Default 600)
Gain:  TX   RX
 
  FXO Termination
Enable Current Disconnect:   No       Yes    (Default Yes.  If set to yes, enter threshold below)
Current Disconnect Threshold (ms):     (50-800 milliseconds. Default 100 milliseconds)
Enable PSTN Disconnect Tone Detection:   No       Yes    (Default No)
 (If set to yes, the following tone is used as the disconnect signal)
PSTN Disconnect Tone:  
 (Syntax: f1=freq@vol, f2=freq@vol, c=on1/off1-on2/off2-on3/off3; [...])
 (Allowed Range: freq = 0 to 4000Hz; vol = -40 to -24dBm)
 (Default: Busy Tone: f1=480@-32,f2=620@-32,c=500/500;)
 
AC Termination Model   Country-based       Impedance-based    (Default Country-based )
Country-based  
Impedance-based  
 
Number of Rings:     (1-50. Default 4)
 (Number of rings for a PSTN incoming call before FXO port answers to accept VoIP number)
PSTN Ring Thru FXS:   No       Yes    (Default Yes)
 (If set to yes, all incoming PSTN calls will ring the FXS port after the Ring Thru Delay)
PSTN Ring Thru Delay (sec):     (1-10 seconds. Default 4 seconds)
 
  Channel Dialing
DTMF Digit Length (ms):    (40-127 milliseconds, Default 100 milliseconds)
DTMF Dial Pause (ms):    (40-127 milliseconds, Default 100 milliseconds)
First Digit Timeout (sec):   (1-20 seconds. Default 10 seconds)
Inter-Digit Timeout (sec):   (1-15 seconds. Default 4 seconds)
Wait for Dial-Tone:   No       Yes    (Default Yes - dial upon dial-tone)
Stage Method (1/2):    (Default 2 - 2 stage dialing)
 
     
All Rights Reserved Grandstream Networks, Inc. 2006-2008