Grandstream Device Configuration
Profile Active:   No      Yes
Primary SIP Server:   (e.g.,, or IP address)
Failover SIP Server:   (Optional, used when primary server no response)
Prefer Primary SIP Server:   No      Yes      ( yes - will register to Primary Server if Failover registration expires)
Outbound Proxy:   (e.g.,, or IP address, if any)
Allow DHCP Option 120( override SIP server ):   No      Yes    
SIP Transport:   UDP       TCP       TLS   (default is UDP)
NAT Traversal:   No      Keep-Alive     STUN     UPnP
DNS Mode:   A Record      SRV      NAPTR/SRV
DNS SRV use Registered IP:   No       Yes
Tel URI:  
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)
Reregister before Expiration:   (0-64800. Default 0 second)
SIP Registration Failure Retry Wait Time:   (in seconds. Between 1-3600, default is 20)
SIP Registration Failure Retry Wait Time upon 403 Forbidden:   (in seconds. Between 0-3600, default is 1200. 0 means stop retry registration upon 403 response.)
Enable SIP OPTIONS Keep Alive:    No       Yes
SIP OPTIONS Keep Alive Interval:   (in seconds. Between 1-64800, default is 30)
SIP OPTIONS Keep Alive Max Lost:   (Number of max lost packets for SIP OPTIONS Keep Alive before re-registration. Between 3-10, default is 3)
Layer 3 QoS:   SIP DSCP (Diff-Serv value in decimal, 0-63, default 26)
  RTP DSCP (Diff-Serv value in decimal, 0-63, default 46)
Local SIP Port:   (default is 5060 for UDP and TCP; 5061 for TLS)
Local RTP Port:   (even number between 1024-65535, default 5004)
Use Random SIP Port:   No      Yes
Use Random RTP Port:   No      Yes
Refer-To Use Target Contact:   No      Yes
Transfer on Conference Hangup:   No      Yes
Disable Bellcore Style 3-Way Conference:   No       Yes (Using star code *23 for 3-way conference)
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)
Authenticate incoming INVITE:   No      Yes
Authenticate server certificate domain:   No      Yes
Authenticate server certificate chain:   No      Yes
Trusted CA certificates:  
Allow Incoming SIP Messages
from SIP Proxy Only:
  No      Yes (no direct IP calling if Yes)
Use Privacy Header:   Default      No      Yes
Use P-Preferred-Identity Header:   Default      No      Yes
SIP REGISTER Contact Header Uses:   LAN Address      WAN Address
SIP T1 Timeout:  
SIP T2 Interval:  
SIP Timer D:     (0 - 64 seconds. Default 0)
DTMF Payload Type:  
Preferred DTMF method
(in listed order):
  Priority 1:  
  Priority 2:  
  Priority 3:  
Disable DTMF Negotiation:   No (negotiate with peer) Yes (use above DTMF order without negotiation)
Generate Continuous RFC2833 Events:   No      Yes (RFC2833 events are generated until key is released)
Send Hook Flash Event:   No      Yes   (Hook Flash will be sent as a DTMF event if set to Yes)
Flash Digit Control:   No      Yes   (Overrides the default settings for call control when both channels are in use.)
Enable Call Features:   No      Yes (if Yes, call features using star codes will be supported locally)
Offhook Auto-Dial Delay:   (0-60 seconds, default is 0)
Use NAT IP:   (used in SIP/SDP message if specified)
Use SIP User-Agent Header:  
Distinctive Ring Tone:     used if incoming caller ID is
    used if incoming caller ID is
    used if incoming caller ID is
Disable Call-Waiting:   No      Yes
Disable Call-Waiting Caller ID:   No      Yes
Disable Call-Waiting Tone:   No      Yes
Disable Connected Line ID:   No      Yes
Disable Receiver Offhook Tone:   No      Yes   (ROH tone will not be played after offhook for 60 seconds)
Disable Reminder Ring for On-Hold Call:   No      Yes
Disable Visual MWI:   No      Yes
Do Not Escape '#' as %23 in SIP URI:   No      Yes
Disable Multiple m line in SDP:   No      Yes
Ring Timeout:   (10-300, default is 60 seconds)
Delayed Call Forward Wait Time:   (Allowed range 1-120, in seconds.)
No Key Entry Timeout:   (1-15, default is 4 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 "(Re-)Dial" key)
Dial Plan:  
SUBSCRIBE for MWI:   No, do not send SUBSCRIBE for Message Waiting Indication
  Yes, send periodical SUBSCRIBE for Message Waiting Indication
Send Anonymous:   No       Yes   (caller ID will be blocked if set to Yes)
Anonymous Call Rejection:   No       Yes  
Special Feature:  
Session Expiration:   (90-64800. default 180 seconds)
Min-SE:   (90-64800. default 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)
Enable 100rel:   No     Yes
Add Auth Header On Initial REGISTER:   No     Yes
Use First Matching Vocoder in 200OK SDP:   No      Yes
Preferred Vocoder
(in listed order):
  choice 1:  
  choice 2:  
  choice 3:  
  choice 4:  
  choice 5:  
  choice 6:  
  choice 7:  
Voice Frames per TX:  
G723 Rate:   6.3kbps encoding rate       5.3kbps encoding rate
iLBC Frame Size:   20ms       30ms
Disable OPUS Stereo in SDP:   No     Yes (removes "/2" from offer)
iLBC Payload Type:   (between 96 and 127, default is 97)
OPUS Payload Type:   (between 96 and 127, default is 123)
VAD:   No       Yes
Symmetric RTP:   No       Yes
Fax Mode:   T.38   Pass-Through
Re-INVITE After Fax Tone Detected:   Enabled   Disabled
Jitter Buffer Type:   Fixed   Adaptive
Jitter Buffer Length:   Low     Medium   High
SRTP Mode:   Disabled     Enabled but not forced   Enabled and forced
Crypto Life Time:   Disabled     Enabled
SLIC Setting:  
Caller ID Scheme:  
DTMF Caller ID:   Start Tone    Stop Tone
Polarity Reversal:   No      Yes   (reverse polarity upon call establishment and termination)
Loop Current Disconnect:   No      Yes   (loop current disconnect upon call termination)
Loop Current Disconnect Duration:     (100 - 10000 milliseconds. Default 200 milliseconds)
Enable Hook Flash:   No      Yes  
Hook Flash Timing:   In 40-2000 milliseconds range, minimum:       maximum:
On Hook Timing:   (In 40-2000 milliseconds range, default is 400)
Gain:   TX   RX
Disable Line Echo Canceller (LEC):   No      Yes
Disable Network Echo Suppressor:   No      Yes
Outgoing Call Duration Limit:   (0-180 minutes, default is 0 (No Limit) )
Ring Frequency:   (15-60 Hz, default is 20 Hz )
Ring Tones (Syntax: c=on1/off1-on2/off2-on3/off3;)
Ring Tone 1:  
Ring Tone 2:  
Ring Tone 3:  
Ring Tone 4:  
Ring Tone 5:  
Ring Tone 6:  
Ring Tone 7:  
Ring Tone 8:  
Ring Tone 9:  
Ring Tone 10:  
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