| 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) |
| Prefer Primary SIP Server: |
No
Will register to Primary Server if Failover registration expires
Will register to Primary Server if Primary Server responds, need to enable SIP OPTIONS/NOTIFY Keep Alive
|
| Outbound Proxy: |
(e.g., proxy.myprovider.com, or IP address, if any) |
| Backup Outbound Proxy: |
(e.g., proxy.myprovider.com, or IP address, if any) |
| Prefer Primary Outbound Proxy: |
No
Yes
( yes - will reregister via Primary Outbound Proxy if registration expires)
|
| From Domain: |
(Optional, actual domain name, will override the from header) |
| Allow DHCP Option 120 (override SIP server): |
No
Yes
|
| SIP Transport: |
UDP
TCP
TLS (default is UDP) |
| SIP URI Scheme When Using TLS: |
sip
sips |
| Use Actual Ephemeral Port in Contact with TCP/TLS: |
No
Yes |
| NAT Traversal: |
No
Keep-Alive
STUN
UPnP
Auto
VPN
|
| 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
Use Configured IP |
| DNS SRV Failover Mode: |
|
| Failback Timer: |
(in minutes. default 60 minutes, max 45 days) |
| Register Before DNS SRV Failover: |
No
Yes |
| Primary IP: |
|
| Backup IP1: |
|
| Backup IP2: |
|
| Tel URI: |
|
| SIP Registration: |
No
Yes |
| Unregister On Reboot: |
No
All
Instance |
| 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.) |
| Port Voltage Off upon no SIP Registration or SIP Registration Failure: |
(in minutes. Between 0-60, default is 0. 0 means port voltage is never turned off) |
| Delay Time of Port Voltage Off Timer Since Boot: |
(in minutes. Between 0-60, default is 0.) |
| Enable SIP OPTIONS/NOTIFY Keep Alive: |
No
OPTIONS
NOTIFY |
| SIP OPTIONS/NOTIFY Keep Alive Interval: |
(in seconds. Between 1-64800, default is 30) |
| SIP OPTIONS/NOTIFY Keep Alive Max Lost: | (Number of max lost packets for SIP OPTIONS/NOTIFY 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; 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 |
| Random RTP Port Range: |
Minimum:
Maximum:
(between 1024-65535.) |
|
Please check Minimum and/or Maximum RTP Port value.
|
| Enable RTCP: |
No
Yes |
| Hold Target Before Refer: |
No
Yes |
| Refer-To Use Target Contact: |
No
Yes |
| Transfer on Conference Hangup: |
No
Yes |
| Remove OBP from Route Header: |
No
Yes |
| Support SIP Instance ID: |
No
Yes |
| Support outbound: |
No
Yes |
| Support GRUU: |
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 |
| Trusted domain name list: |
(Support setting specific domain names and wildcard
domain names, wildcard domain names such as "*.grandstream.com", use ","
to separate domain names) |
| Authenticate server certificate chain: |
No
Yes |
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 |
| Use P-Access-Network-Info Header: |
No
Yes |
| Use P-Emergency-Info Header: |
No
Yes |
| Use P-Asserted-Identity Header: |
No
Yes |
| Use P-Early-Media Header: |
No
Yes |
| SIP REGISTER Contact Header Uses: |
LAN Address
WAN Address |
| Caller ID Fetch Order: |
Auto
Disabled
From Header |
| Allow SIP Factory Reset: |
No
Yes |
| Maximum Number of SIP Request Retries: |
(between 1 and 10, default is 2.)
|
| 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:
|
| Force DTMF to be sent via SIP INFO simultaneously: |
No
Yes
|
| Inband DTMF Duration: |
In 40-2000 milliseconds range, 0 means pass-through mode, duration:
inter-duration:
|
| Inband DTMF Tx Gain: |
(-12-12 db, default is 0) |
| DSP DTMF Detector Duration Threshold: |
In 20-200 milliseconds range, duration:
inter-duration:
|
| Disable DTMF Negotiation: |
No (negotiate with peer)
Yes (use above DTMF order without negotiation) |
| 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.)
|
| Callee Flash to 3WC: |
No
Yes
|
| Offhook Auto-Dial: |
(User ID/extension to dial automatically when offhook) |
| Offhook Auto-Dial Delay: |
(0-60 seconds, default is 0) |
| Offhook Auto-Dial DTMF: |
(DTMF digits to send after offhook auto-dial call gets connected) |
| Proxy-Require: |
|
| Use NAT IP: |
(used in SIP/SDP message if specified) |
| SIP User-Agent: |
|
| SIP User-Agent Postfix: |
|
| Add MAC in User-Agent: |
|
| Use MAC Header: |
|
| RFC2543 Hold: |
No
Yes |
| 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 Reminder Ring for DND: |
No
Yes |
| Disable Visual MWI: |
No
Yes |
| Visual MWI Type: |
FSK
NEON |
| MWI Tone: |
Default
Special Proceed Indication Tone |
| Ringback Tone at No Early Media: |
No
Yes |
| Do Not Escape '#' as %23 in SIP URI: |
No
Yes |
| Disable Multiple m line in SDP: |
No
Yes |
| Ring Timeout: |
(0-300, default is 60 seconds, 0 means no timeout) |
| 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)
|
| Disable # as Redial Key: |
No
Yes (if set to Yes, "#" will not function as ReDial key)
|
| Dial Plan: |
|
| SUBSCRIBE for MWI: |
No, do not send SUBSCRIBE for Message Waiting Indication
Yes, send periodical SUBSCRIBE for Message Waiting Indication
|
| Subscribe Retry Wait Time upon 403 Forbidden: |
(0-10080 in minutes. Default is 0, max 7 days) |
| Send Anonymous: |
No
Yes (caller ID will be blocked if set to Yes) |
| Anonymous Call Rejection: |
No
Yes |
| Special Feature: |
|
| Enable Session Timer: |
No
Yes |
| 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) |
| When To Restart Session After Re-INVITE received: |
Immediately
After replying 200OK |
| Enable 100rel: |
No
Yes |
| Add Auth Header On Initial REGISTER: |
No
Yes |
| Enable Call Waiting Alert-Info in 180 Ringing Response: |
No
Yes |
| Conference URI: |
|
| |
| 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:
choice 8:
|
| 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 |
| SRTP Key Length: |
|
| Crypto Life Time: |
Disabled
Enabled |
| |
| SLIC Setting: |
|
| Caller ID Scheme: |
|
| DTMF Caller ID: |
Start Tone
Stop Tone
|
| Disable Unknown Caller ID: |
No
Yes |
| Replace Beginning '+' in Caller ID with: |
|
| Number of Beginning Digits to Strip from Caller ID: |
(between 0 and 10, default is 0.)
|
| Polarity Reversal: |
No
Yes (reverse polarity upon call establishment and termination)
|
| Loop Current Disconnect: |
No
Yes (loop current disconnect upon call termination)
|
| Play busy/reorder tone before Loop Current Disconnect: |
No
Yes (play busy/reorder tone before loop current disconnect upon call fail)
|
| Loop Current Disconnect Duration: |
(100 - 10000 milliseconds. Default 200 milliseconds)
|
| Enable Pulse Dialing: |
No
Yes
|
| Pulse Dialing Standard: |
|
| 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) ) |
| Incoming Call Duration Limit: |
(0-180 minutes, default is 0 (No Limit) ) |
| Ring Frequency: |
|
| Enable High Ring Power: |
No
Yes |
| OnHook DC Feed Current: |
|
| RFC2833 Events Count: |
(between 2 and 10, default is 8, 0 means continuous RFC2833 events) |
| RFC2833 End Events Count: |
(between 2 and 10, default is 3) |
| RFC2833 Convert Mode: |
|
| |
| Distinctive Ring Tone: |
used if incoming caller ID is
used if incoming caller ID is
used if incoming caller ID is
|
| 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: |
|
| |
| Distinctive Call Waiting Tone: |
used if incoming caller ID is
used if incoming caller ID is
used if incoming caller ID is
|
| Call Waiting Tones |
Syntax: f1=val[,f2=val[,c=on1/off1[-on2/off2[-on3/off3]]]];
(Frequencies are in (10, 4000) Hz and cadence on and off are in (0, 64000) ms)
|
| Call Waiting Tone 1: |
|
| Call Waiting Tone 2: |
|
| Call Waiting Tone 3: |
|
| Call Waiting Tone 4: |
|
| Call Waiting Tone 5: |
|
| Call Waiting Tone 6: |
|
| Call Waiting Tone 7: |
|
| Call Waiting Tone 8: |
|
| Call Waiting Tone 9: |
|
| Call Waiting Tone 10: |
|
| |
| Call Features Settings |
| Enable Call Features: |
No
Yes
Enable All
(When enabled, Do No Disturb, Call Forward and other
call features can be used via the local feature codes on the phone.
Otherwise, the ITSP feature codes will be used. Enable
All will override all individual features enable setting.) |
| Reset Call Features: |
No
Yes |
| ------------------------ |
| SRTP Feature: |
No
Yes |
| Enable SRTP: |
|
| Disable SRTP: |
|
| ------------------------ |
| SRTP per call Feature: |
No
Yes |
| Enable SRTP per call: |
|
| Disable SRTP per call: |
|
| ------------------------ |
| CID Feature: |
No
Yes |
| Enable CID: |
|
| Disable CID: |
|
| ------------------------ |
| CID per call Feature: |
No
Yes |
| Enable CID per call: |
|
| Disable CID per call: |
|
| ------------------------ |
| Direct IP Calling Feature: |
No
Yes |
| Direct IP Calling: |
|
| ------------------------ |
| CW Feature: |
No
Yes |
| Enable CW: |
|
| Disable CW: |
|
| ------------------------ |
| CW per call Feature: |
No
Yes |
| Enable CW per call: |
|
| Disable CW per call: |
|
| ------------------------ |
| Call Return Feature: |
No
Yes |
| Call Return: |
|
| ------------------------ |
| Unconditional Forward Feature: |
No
Yes |
| Enable Unconditional Forward: |
|
| Disable Unconditional Forward: |
|
| ------------------------ |
| Busy Forward Feature: |
No
Yes |
| Enable Busy Forward: |
|
| Disable Busy Forward: |
|
| ------------------------ |
| Delayed Forward Feature: |
No
Yes |
| Enable Delayed Forward: |
|
| Disable Delayed Forward: |
|
| ------------------------ |
| Paging Feature: |
No
Yes |
| Paging: |
|
| ------------------------ |
| DND Feature: |
No
Yes |
| Enable DND: |
|
| Disable DND: |
|
| ------------------------ |
| Blind Transfer Feature: |
No
Yes |
| Enable Blind Transfer: |
|
| ------------------------ |
| Disable LEC per call Feature: |
No
Yes |
| Disable LEC per call: |
|
| ------------------------ |
| Disable Bellcore Style 3-Way Conference: |
No
Yes |
| Star Code 3WC Feature: |
No
Yes |
| Star Code 3WC: |
|
| ------------------------ |
| Play registration id Feature: |
No
Yes |
| Enable playing registration id: |
|
| ------------------------ |
| Set Offhook Auto-Dial Feature: |
No
Yes |
| Enable set Offhook Auto-Dial: |
|
| ------------------------ |
| Forced Codec Feature: |
No
Yes |
| Forced Codec: |
|
| PCMU Codec Feature: |
No
Yes |
| PCMU Codec: |
|
| PCMA Codec Feature: |
No
Yes |
| PCMA Codec: |
|
| G723 Codec Feature: |
No
Yes |
| G723 Codec: |
|
| G729 Codec Feature: |
No
Yes |
| G729 Codec: |
|
| iLBC Codec Feature: |
No
Yes |
| iLBC Codec: |
|
| G722 Codec Feature: |
No
Yes |
| G722 Codec: |
|
|
|