Grandstream Networks GHP6 Series Instruction Manual

Grandstream Networks, Inc.
GHP6XX Series
GHP6XX –Administration Guide

GHP6XX/W – Administration Guide
The GHP series of hotel phones feature the GHP610(W)/GHP620(W) and the GHP611(W)/GHP621(W), easy-to-use IP phones
for any hotel room that can be programmed and customized based on the needs of the hotel and its guests. These devices
are equipped with integrated dual-band Wi-Fi (GHP6XX/W).
The features of the GHP series include an HD speaker, 2 SIP accounts/lines, 6 programmable keys (3 programmable keys for
GHP61X), 10-speed dial keys, 3-way voice conferencing, full band Opus voice codec, an advanced jitter-resilience algorithm
that tolerates up to 30% packet loss without impacting voice quality.
The GHP series is supported by the Grandstream Device Management System (GDMS), which provides a centralized interface
to configure, provision, manage and monitor the deployment of Grandstream endpoints.
The GHP series IP phones can be installed on a desktop or wall-mounted and can be powered by PoE
(GHP610/GHP611/GHP620/GHP621), power adapter (GHP610W/GHP611W/GHP620W/GHP621W) and USB Type-C
charger(GHP620/GHP621/GHP620W/GHP621W).
Its elegant and compact design makes it perfect for not only hotels but also hospitals, apartments, dormitories, and much
more.
PRODUCT OVERVIEW
Feature Highlights
GHP61X/W
GHP62X/W
○TrimStyle phone, small footprint and fashion design.
○2 SIP lines with wideband Opus codec.
○6 programmable keys (3 programmable keys for GHP61X) and 10-speed dial
keys to customize different services.
○One 100Mbps network port with PoE (PoE available on
GHP610/GHP611/GHP620/GHP621).
○Integrated dual-band 2.4Ghz & 5Ghz 802.11 a/b/g/n/ac Wi-Fi.
○Desk/Wall mounted.
○Tolerate up to 30% packet loss without impacting voice quality.
○Supports provision and management via GDMS.
○Customizable faceplate and speed dial label (Speed Dial label for GHP61X
only).
○New voice message and mute LED indicator.
○HAC/VCH (ADA) compliant handset volume boost.
○Magnetic Hook Switch feature.

Technical Specification
GHP61X
GHP62X
Protocols/Standards SIP: SIP RFC3261, TCP/IP/UDP, RTP/RTCP, RTCP-XR, Simple, TLS, SRTP.
Network Interface One Auto-Negotiation 10/100Mbps ethernet port, integrated PoE (Only GHP61X)
Wi-Fi GHP610W/GHP611W integrated dual-band 2.4Ghz & 5Ghz 802.11 a/b/g/n/ac.
Keypad
23 keys, including:
- 3 Programmable keys.
- 12 standard phone digit keys (0-9: speed dial keys configuration on web page, *,#).
- 5 function keys (Flash, Redial, Hands-free, Voice Mail, Hold).
- 3 volume control keys, up/down/mute (with red color LED).
HD Audio One HD handset, support for wide band audio.
Voice Conference 2 SIP accounts and lines, up to 3-way conference.
Voice Codecs Support for the G.729A/B, G.711µ/a-law, G726, G.722(wide band), G723, iLBC, full band OPUS.
Voice Capabilities In-band and out-of-band DTMF (in audio, RFC2833, SIP INFO), VAD, AEC, CNG, PLC, AGC, AJB.
Telephony Features Dial, Answer, Redial, Flash, Hands-free, Voice mail (with the server) Hold/Unhold, Mute/Unmute,
flexible dial plan, Speed dial server redundancy, fail-over Call out.
Advanced Features Support for multicast paging and E911 service.
Support for advanced jitter resilience algorithm.
Base Stand 1 wall-mountable base stand, can be put on a desk.
QoS Layer 2 QoS (802.1Q, 802.1p) and Layer 3 (ToS, DiffServ, MPLS) QoS
Security Secure boot, random default, unique security certificate per device, administrator level passwords, 256-
bit AES encrypted configuration file, SRTP, TLS, 802.1x media access control.
Protocols/Standards SIP: SIP RFC3261, TCP/IP/UDP, RTP/RTCP, RTCP-XR, Simple, TLS, SRTP.
Network Interface One Auto-Negotiation 10/100Mbps ethernet port, integrated PoE (Only GHP62X)
Wi-Fi GHP620W/GHP621W integrated dual-band 2.4Ghz & 5Ghz 802.11 a/b/g/n/ac.
Keypad
26 keys, including:
- 6 Programmable keys.
- 12 standard phone digit keys (0-9: speed dial keys configuration on web page, *,#).
- 5 function keys (Flash, Redial, Hands-free, Voice Mail, Hold).
- 3 volume control keys, up/down/mute (with red color LED).
HD Audio
One HD handset, support for wide band audio.

GETTING STARTED
This chapter provides basic installation instructions including the list of the packaging contents and also information for
obtaining the best performance with the GHP6XX/W.
Equipment Packaging
GHP61X
GHP61X Package Content
Voice Conference 2 SIP accounts and lines, up to 3-way conference.
Voice Codecs Support for the G.729A/B, G.711µ/a-law, G726, G.722(wide band), G723, iLBC, full band OPUS.
Voice Capabilities In-band and out-of-band DTMF (in audio, RFC2833, SIP INFO), VAD, AEC, CNG, PLC, AGC, AJB.
Telephony Features Dial, Answer, Redial, Flash, Hands-free, Voice mail (with the server) Hold/Unhold, Mute/Unmute,
flexible dial plan, Speed dial server redundancy, fail-over Call out.
Advanced Features Support for multicast paging and E911 service.
Support for advanced jitter resilience algorithm.
Base Stand 1 wall-mountable base stand, can be put on a desk.
QoS Layer 2 QoS (802.1Q, 802.1p) and Layer 3 (ToS, DiffServ, MPLS) QoS
Security Secure boot, random default, unique security certificate per device, administrator level passwords, 256-
bit AES encrypted configuration file, SRTP, TLS, 802.1x media access control.
GHP61X/W
●1x Base
●1x Handset
●1x RJ11 (6P6C) Phone Cord
●1x Quick Installation Guide
●1x 12V Power Adapter (Only with GHP610W and GHP611W)
●1x MPK Label (On the back cover of the QIG)

GHP62X
GHP62X Package Content
GHP6XX/W Phone Setup
The GHP6XX/W can be installed on the desk or can be mounted on the wall. To mount the phone on the wall please follow
the instructions below.
GHP61X/W
GHP61X Wall Mount
Mark two dots horizontally with a 100 mm distance in between where to wall mount the GHP610/W & GHP611/W.
Using a drill, make a hole on each marked dot. Put a plastic expansion bolt and screw (not provided) on each hole. Leave
enough space to mount the GHP610/W & GHP611/W.
Mount the GHP610/W & GHP611/W on the screws using the Wall Mount Slots.
GHP62X/W
GHP62X/W
●1 x GHP62X Phone
●1 x Handset
●1 x Base Stand
●1 x Faceplate
●1 x 12V Power Adapter (GHP620W and GHP621W)
●1 x RJ9(4P4C) Phone Cord
●1 x Quick Installation Guide
Note
Please check the package before installation. If you find any one of the components missing, please contact your system
administrator.

GHP61X Wall Mount
1. On the wall where the GHP62X/W will be mounted, mark two dots with 100 mm distance in-between vertically.
2. Using a drill, make a hole on each marked dot. Put a plastic expansion bolt and screw (not provided) into each hole. Leave
enough space on the screws to mount the GHP62X.
3. Attach the wall mount spacers to the slot for the wall mount spacer on the back of the phones.
4. Mount the GHP62X on the screws using its “Wall Mount Slots”.
Connecting the GHP6XX/W
To set up the phone, follow the steps below:
1. Connect the phone cord with the handset (RJ-11 port in GHP61X/W, RJ-9 port in GHP62X/W) and the base.
2. Connect the LAN port of the phone to the RJ-45 socket of a hub/switch on a router ( LAN side of the router) using the
Ethernet cable. This step can be skipped if using GHP6XXW with Wi-Fi.
3. Powering the phone:
For GHP6XXW: Connect the 12V DC output plug into the power jack if the phone; plug the power adapter into the
electrical outlet.
For GHP6XX: A PoE switch must be used in step 2.
Phone Configuration Via Web Browser
1. Ensure your phone is powered up and connected to the internet.
2. Locate the MAC address on the MAC tag of the unit which is on the underside of the device, or on the package.
3. From a computer connected to the same network as the GHP6XX, type in the following address using GHP6XX’s MAC
address on your browser: https://<mac>.local
Example: https://c074adffffff.local
Note: The user can dial **47# and the IP address will be read by the IVR.
4. Enter the admin’s username and password to access the configuration menu. (The factory default username is “admin”
while the default random password can be found on the sticker at the back of the unit).
Phone Description
GHP61X
Note
For easy deployment, the GHP6xxW is out-of-the-box pre-configured to connect to default SSID named wp_master with a
password (WPA/WPA2 PSK) equal to wp!987@dmin, users can adapt these settings from the web UI as well to make it easier
for deployment on a customer’s site.

GHP61X Handset
GHP61X Back Panel of the Handset and the Base
GHP62X
GHP62X Front Panel

GHP62X Back Panel
CONFIGURATION GUIDE
Status Page Definitions
Status → Account Status
Account Status This interface indicates the information of the accounts which are configured on the IP phone.
Status → Network Status
Ethernet
MAC Address This field indicates the MAC address of the device.
IPv4 Address Type This field indicated whether the obtained IP address was configured by DHCP or set statically.
IPv4 Address This field indicates the IP address allocated to the IP phone.
Gateway This field indicates the default gateway allocated.
IPv4 NAT Type This field indicates the type of NAT of the network.
IPv6 Address This field indicates the IPv6 IP address allocated.
Global Unicast Address This field indicates the global unicast address.
Link-Local Address This field indicates the IPv6 unicast address.
IPv6 Static Gateway This field indicates the IPv6 static gateway address.
IPv6 DUID This field indicates the DHCP unique identifier.
IPv6 NAT Type This field indicates the IPv6 NAT type.
Status → System Info
Information

Account Page Definitions
Basic Settings
Product model This field indicates the product model.
Part number This field indicates the part number.
Software version This field indicates the version of the firmware installed.
Language This field indicates the language used for displaying the web user interface.
Recommended Time Zone This field indicates the recommended time zone.
System Up Time This field indicates how long the devices has been turned on.
System Time This field indicates the time and date set on the system.
System Time Zone This field indicates the system time zone configured on the device.
Download System
Information This option allows downloading all the the system information.
Account Register
Account Active Indicates whether the account is active.
The default setting is “No”.
Account Name The name associated with each account to be displayed on the LCD. (e.g., MyCompany)
SIP Server The URL or IP address, and port of the SIP server. This is provided by your VoIP service provider
(e.g., sip.mycompany.com, or IP address)
Secondary SIP Server The URL or IP address, and port of the SIP server. This will be used when the primary SIP server fails
Outbound Proxy
IP address or Domain name of the Primary Outbound Proxy, Media Gateway, or Session
Border Controller.
If a symmetric NAT is detected, STUN will not work and ONLY an Outbound Proxy can
provide a solution
Secondary Outbound Proxy Defines secondary outbound proxy that will be used when the primary proxy cannot be connected.
SIP User ID User account information, provided by your VoIP service provider.
SIP Authentication ID SIP service subscriber’s Authenticate ID used for authentication. It can be identical to or different from
the SIP User ID.
SIP Authentication Password
The account password required for the phone to authenticate with the SIP server before the account
can be registered.
After it is saved, this will appear as hidden for security purpose.
Name The SIP server subscriber’s name (optional) that will be used for Caller ID display (e.g., John Doe).

SIP Settings
TEL URI
If the phone has an assigned PSTN telephone number, this field should be set to “user=phone”. A
“user=phone” parameter will be attached to the Request-URI and “To” header in the SIP request to
indicate the E.164 number. If set to “Enable”, “tel:” will be used instead of “sip:” in the SIP request.
Voice Mail Access Number Allows users to access voice messages by pressing the MESSAGE button on the phone. This value is
usually the VM portal access number.
Failback Expiration (m) Specifies the duration (in minutes) since failover to the current SIP server or Outbound Proxy before
making failback attempts to the primary SIP server or Outbound Proxy.
Network Settings
DNS Mode
This parameter controls how the Search Appliance looks up IP addresses for hostnames. If “Use
Configured IP” is selected, please fill in Primary IP, Backup IP 1 and Backup IP 2.
●A Record
●SRV
●NAPTR/SRV
●Use Configured IP
NAT Traversal
Set NAT traversal to activate the NAT penetration mechanism.
options to be chosen are No, STUN, or Keep-Alive.
●When set to "STUN" and the STUN server address is specified, detection will be based on the
STUN server.
●When set to "Keep-alive", messages can be configured to be sent at regular intervals, such as every
few minutes, and can be customized with specific packet content and timing parameters to
optimize performance in different network environments.
The Default value is set to "No".
Support rport (RFC3581)
Configures to use symmetric response routing. If it is used, the "rport" field will be added to the Via
header field in the SIP Request, and the information will be extracted from the SIP 200OK Response
for SIP Register to rewrite the SIP Contact information and apply it in subsequent SIP Requests.
Proxy-Require A SIP Extension to notify the SIP server that the phone is behind a NAT/Firewall.
Basic Settings
SIP Registration Selects whether the phone will send SIP Register messages to the proxy/server. The default setting is
“Enabled”.
UNREGISTER on Reboot
●If set to “No”, the phone will not unregister the SIP user’s registration information before new
registration.
●If set to “All”, the SIP Contact header will use “*” to clear all SIP user’s registration information.
●If set to “Instance”, the phone only needs to clear the current SIP user’s info.
REGISTER Expiration
Specifies the frequency (in minutes) in which the phone refreshes its registration with the specified
registrar.
The maximum value is 64800 minutes (about 45 days). The default value is 60 minutes.
SUBSCRIBE Expiration
Specifies the frequency (in minutes) in which the phone refreshes its subscription with the specified
registrar.
The maximum value is 64800 minutes (about 45 days). The default value is 60 minutes.

Re-Register before Expiration Specifies the time frequency (in seconds) that the phone sends re-registration request before the
Register Expiration. The default value is 0.
Registration Retry Wait Time Specifies the interval to retry registration if the process is failed. The valid range is 1 to 3600. The
default value is 20 seconds.
Add Auth Header on Initial
REGISTER
If enabled, the phone will add Authorization header in initial REGISTER request.
Default is “Disabled”.
Enable OPTIONS Keep Alive
Configures whether to enable SIP OPTIONS to track account registration status. If enabled, the phone
will send periodic OPTIONS messages to server to track the connection status with the server.
Default is “Disabled”.
OPTIONS Keep Alive Interval Configures the time interval the phone sends OPTIONS message to the server. If set to 30 seconds, it
means the phone will send an OPTIONS message to the server every 30 seconds.
OPTIONS Keep Alive Max
Tries
Configures the maximum number of times the phone will try to send OPTIONS message consistently
to server without receiving a response. If set to “3”, the phone will send OPTIONS message 3 times. If
no response from the server, the phone will re-register.
SUBSCRIBE for MWI When set to “Yes”, a SUBSCRIBE for Message Waiting Indication will be sent periodically.
The default setting is “No”.
SUBSCRIBE for Registration When set to “Yes”, a SUBSCRIBE for Registration will be sent out periodically.
The default setting is “No”.
Use Privacy Header
Configures whether the “Privacy Header” is present in the SIP INVITE message.
●Default: the phone will add “Privacy Header” when special feature is not “Huawei IMS”.
●Yes: the phone will always add “Privacy Header”.
●No: the phone will not add “Privacy Header”.
The default setting is “default”.
Use P-Preferred- Identity
Header
Configures whether the “P-Preferred-Identity Header” is present in the SIP INVITE message.
●Default: the phone will add “P-Preferred-Identity header” when special feature is not “Huawei
IMS”.
●Yes: the phone will always add “P-Preferred-Identity header”.
●No: the phone will not add “P-Preferred-Identity header”.
Use X-Grandstream-PBX
Header
Configures to use X-Grandstream-PBX header in SIP request.
Default setting is “Yes”.
Use P-Access-Network-Info
Header
Configures to use P-Access-Network-Info header in SIP request.
Default setting is “Yes”.
Use P-Emergency-Info Header Configures to use P-Emergency-Info header in SIP request.
Default setting is “Yes”.
Use MAC Header
●If Register Only, all outgoing SIP message will include the MAC header.
●If Yes to all SIP, all outgoing SIP messages will include the MAC header.
●If No, the phone’s MAC header will not be included in any outgoing SIP messages.
The default setting is “No”.
Add MAC in User-Agent
●If Yes except REGISTER, all outgoing SIP messages will include the phone’s MAC address in the
User-Agent header, except for REGISTER and UNREGISTER.

●If Yes to All SIP, all outgoing SIP messages will include the phone’s MAC address in the User-
Agent header.
●If No, the phone’s MAC address will not be included in the User-Agent header in any outgoing SIP
messages.
The default setting is “No”.
SIP Transport Selects the network protocol used for the SIP transport.
The default setting is “UDP”.
Enable TCP Keep-alive Configures whether to enable TCP Keep-alive for the TCP connection between the terminal and the
SIP server.
SIP Listening Mode
Configures whether or not to listen to multiple SIP protocols.
●If set to “Dual“, phone will listen to TCP when UDP is selected.
●If set to “Dual (Secured)“, phone will listen to TLS/TCP when UDP is selected. If “TCP” or
“TLS/TCP” is selected, UDP will be listened too.
●If set to “Dual (BLF Enforced)“, phone will try to enforce BLF subscriptions to use TCP protocol
by adding ‘transport=tcp’ to the Contact header.
The default setting is “Transport Only”.
Local SIP Port Configures the local SIP port used to listen and transmit.
SIP URI Scheme when using
TLS
Specifies if “sip” or “sips” will be used when TLS/TCP is selected for SIP Transport. The default
setting is “sips”.
Use Actual Ephemeral Port in
Contact with TCP/TLS
Configures whether the actual ephemeral port in contact with TCP/TLS will be used when TLS/TCP is
selected for SIP Transport.
The default setting is “No”.
Support SIP Instance ID Configures whether SIP Instance ID is supported or not.
The default setting is “Yes”.
SIP T1 Timeout
SIP T1 Timeout is an estimate of the round-trip time of transactions between a client and server. If no
response is received the timeout is increased and request re-transmit retries would continue until a
maximum amount of time define by T2. The default setting is 0.5 seconds.
SIP T2 Timeout
SIP T2 Timeout is the maximum retransmit time of any SIP request messages (excluding the INVITE
message). The re-transmitting and doubling of T1 continues until it reaches the T2 value. Default is 4
seconds.
Outbound Proxy Mode
Configures whether to put the Outbound Proxy in the Route header, or if SIP messages should always
be sent to Outbound Proxy.
1. In route
2. Not in route
3. Always send to
Default is “in route”.
Enable 100rel
When enabled, the 100rel tag is appended to the value of the Supported header of the initial signaling
messages.
The default setting is “No”.
Use Route Set in Notify (Follow
RFC 6665)
Configures whether to use route set in NOTIFY (follow RFC 6665).
●If enabled, the Request URI of the refresh subscription will use the URI in the received NOTIFY
Contact (RFC 6665).
●If disabled, the URI in the previously subscribed 200 OK Contact will be used.
Session Timer

Codec Settings
Enable Session Timer
Configures whether to enable session timer function. It enables SIP sessions to be periodically
“refreshed” via a SIP request (UPDATE, or re-INVITE). If there is no refresh via an UPDATE or re-
INVITE message, the session will be terminated once the session interval expires. If set to “Yes”, the
phone will use the related parameters when sending session timer according to “Session Expiration”.
If set to “No”, session timer will be disabled.
The default setting is “No”.
Session Expiration
Session Expiration is the time (in seconds) where the session is considered timed out, provided no
successful session refresh transaction occurs beforehand.
The default setting is 180. The valid range is from 90 to 64800.
Min-SE The minimum session expiration (in seconds). The default value is 90 seconds. The valid range is from
90 to 64800.
Caller Request Timer
If set to “Yes” and the remote party supports session timers, the phone will use a session timer when it
makes outbound calls.
The default setting is “No”.
Callee Request Timer
If set to “Yes” and the remote party supports session timers, the phone will use a session timer when it
receives inbound calls.
The default setting is “No”.
Force Timer
If set to “Yes”, the phone will use the Session Timer even if the remote party does not support this
feature. Otherwise, Session Timer is enabled only when the remote party supports it.
The default setting is “No”.
UAC Specify Refresher
As a caller, select UAC to use the phone as the refresher, or select UAS to use the callee or proxy
server as the refresher. When set to “Omit”, the refresh object is not specified.
The default setting is “UAC”.
UAS Specify Refresher
As a callee, select UAC to use caller or proxy server as the refresher, or select UAS to use the phone
as the refresher.
The default setting is “UAC”.
Force INVITE Select “Yes” to force using the INVITE method to refresh the session timer.
The default setting is “No”.
Audio
Preferred Vocoder
(Choice 1 – 8)
Multiple vocoder types are supported on the phone, the vocoders in the list is a higher preference.
Users can configure vocoders in a preference list that is included with the same preference order in
SDP message.
●Opus
●G722
●PCMU
●PCMA
●G.723.1
●G.729.1
●iLBC
●G.726-32
Codec Negotiation Priority
Configures the phone to use which codec sequence to negotiate as the callee. When set to “Caller”, the
phone negotiates by SDP codec sequence from received SIP Invite. When set to “Callee”, the phone
negotiates by audio codec sequence on the phone. The default setting is “Callee”.

Use First Matching Vocoder in
200OK SDP
When set to “Yes”, the device will use the first matching vocoder in the received 200OK SDP as the
codec. The default setting is “No”.
iLBC Frame Size Selects iLBC packet frame size. Users can choose from 20ms and 30ms. The default setting is “30ms”.
iLBC Payload Type Specifies iLBC payload type. Valid range is 96 to 127. Cannot be the same as Opus or DTMF payload
type. Valid range is 96 to 127. The default setting is “97”.
G.726-32 Packing Mode Selects “ITU” or “IETF” for G726-32 packing mode.
The default setting is “ITU”.
G.726-32 Dynamic Payload
Type Specifies G.726-32 payload type. Valid range is 96 to 127. Default is 127.
Opus Payload Type Specifies Opus payload type. Valid range is 96 to 127. It cannot be the same as iLBC or DTMF
Payload Type. Default value is 123.
Send DTMF
Specifies the mechanism to transmit DTMF digits. There are 3 supported modes:
1. In audio: DTMF is combined in the audio signal (not very reliable with low-bit-rate codecs).
2. RFC2833 sends DTMF with RTP packet. Users can check the RTP packet to see the DTMFs sent
as well as the number pressed.
3. SIP INFO uses SIP INFO to carry DTMF.
Default setting is “RFC2833”.
DTMF Payload Type Configures the payload type for DTMF using RFC2833. Cannot be the same as iLBC or OPUS
payload type.
Enable Audio RED with FEC If set to “Yes”, FEC will be enabled for audio call.
If set to “Yes”, FEC will be enabled for audio call.
Silence Suppression
If set to “Yes”, when silence is detected, a small quantity of VAD packets (instead of audio packets)
will be sent during the period of no talking. For codec G.723 and G.729 only.
Default setting is “No”.
Jitter Buffer Type Selects either Fixed or Adaptive for jitter buffer type, based on network conditions. The default setting
is “Adaptive”.
Jitter Buffer Length Selects jitter buffer length from 100ms to 800ms, based on network conditions. The default setting is
“300ms”.
Voice Frames Per TX
Configures the number of voice frames transmitted per packet. It is recommended that the IS limit
value of Ethernet packet is 1500 bytes or 120 kbps. When configuring this, it should be noted that the
“ptime” value for the SDP will change with different configurations here. This value is related to the
codec used in the codec table or negotiate the payload type during the actual call. For example, if set
to 2 and the first code is G.729, G.711 or G.726, the “ptime” value in the SDP datagram of the
INVITE request is 20 ms. If the “Voice Frame/TX” setting exceeds the maximum allowed value, the
phone will use and save the maximum allowed value for the selected first codec. It is recommended to
use the default setting provided, and incorrect setting may affect voice quality.
The default setting is 2.
G.723 Rate Selects encoding rate for G723 codec.
RTP Settings
SRTP Mode Enable SRTP mode based on your selection from the drop-down menu.
●No
●Enabled but Not forced

Account Call Settings
●Enabled and Forced
●Optional
The default setting is “No”.
SRTP Key Length
Allows users to specify the length of the SRTP calls. Available options are:
●AES 128&256 bit
●AES 128 bit
●AES 256 bit
Default setting is AES 128&256 bit
Crypto Life Time Enable or disable the crypto lifetime when using SRTP. If users set to disable this option, phone does
not add the crypto lifetime to SRTP header. The default setting is “Yes”.
RTCP Mode
Configure RTCP port negotiation rules.
●Default: Use the traditional RTCP port, which is "RTP port+1".
●Negotiate RTCP Port: Use attribute RTCP to negotiate.
●RTCP Mux: The caller actively negotiates the RTCP port and indicates RTCP Mux at the same
time.
●RTCP Mux Only: The caller forces RTCP Mux, generated by the local media port only apply for
RTP port.
RTCP Keep-Alive Method
Configures the RTCP channel keep-alive packet type.
●Receiver Report: The RTCP channel will sends "receiver report+source description+RTCP
extension" as keep-alive data.
●Sender Report: The RTCP channel will sends "Sender report+source description+ RTCP
extension" as keep-alive data.
RTP Keep-Alive Method
Configures the RTP channel keep-alive packet type.
●No: No data will be sent
●RTP Version 1: The wrong version infor "1" will be carried when sending RTP data packets.
Symmetric RTP Configures whether Symmetric RTP is used or not. Symmetric RTP means that the UA uses the same
socket/port for sending and receiving the RTP stream. The default setting is “No”.
RTP Timeout (s)
Configures the RTP timeout of the phone. If the phone does not receive the RTP packet within the
specified RTP time, the call will be automatically disconnected. The default range is 0 and 6-600. If
set to 0, the phone will not hang up the call automatically.
General
Key as Send Pressing the selected key will immediately dial out.
No Key Entry Timeout Configures the timeout (in seconds) for no key entry.
If no key has been pressed after the timeout, the collected digits will be sent out.
Send Anonymous If set to "Yes", the "From" header in outgoing INVITE messages will be set to anonymous.
Anonymous Call Rejection If set to "Yes", anonymous calls will be rejected.
Enable Call Waiting Enables the call waiting feature. If set to "No", new incoming calls will be rejected after the call is
established.

RFC2543 Hold If set to "Yes", c=0.0.0.0 will be used in INVITE SDP for hold.
Ring Timeout Configures the timeoput (in seconds) for the phone to ring when an incoming call is not answered.
Valid range is 30 to 3600.
Auto Answer
Auto Answer If set to “Yes”, the phone will automatically turn on the speaker phone to answer incoming calls after a
short reminding beep. Default setting is “No”.
Auto Answer Numbers Allows the user to configure specific numbers to auto answer. If not set, all numbers will be auto
answered If auto answer is enabled. Up to 10 numbers can be configured.
Intercom
Play warning tone for Auto-
Answer Intercom If enabled, phone will play warning tone when auto answering intercom.
Custom Alert-Info for Auto
Answer
Used exclusively to match the contents of the Alert-Info header for auto answer. The default auto
answer headers will not be matched if this is defined.
Allow Auto Answer by Call-
Info/Alert-Info
If set to “Yes”, the phone will automatically turn on the speaker phone to answer incoming calls after a
short reminding beep, based on the SIP Call-Info/Alert-Info header sent from the server/proxy.
Default is “Yes”.
Allow Barging by Call-
Info/Alert-Info
When enabled, the phone will automatically put the current call on hold and answer the incoming call
based on the SIP Call-Info/Alert-Info header sent from the server/proxy. However, if the current call
was answered based on the SIP Call-Info/Alert-Info header, then all other incoming calls with SIP
Call-Info/Alert-Info headers will be rejected automatically. Default setting is “No”.
Mute on Intercom Answer If enabled, the phone will mute the microphone after answering an intercom call via Call-Info/Alert-
Info.
Transfer
Transfer on Conference
Hangup Configures whether the call is transferred to the other party if the conference initiator hangs up.
Enable Local Call Features
When enabled, the phone will use the local call function, dial *87 plus number to use local blind
forwarding; When disable, the server-side call function code will be used.
Enabled by Default.
Enable Recovery on Blind
Transfer Enable Recovery to the call to the transferee on failing blind transfer to the target.
Blind Transfer Wait Timeout Configures the timeout (in seconds) when waiting for sipfrag response in blind transfer. Valid range is
30 to 300
Refer-To Use Target Contact If enabled, the "Refer-to" hear uses the transferred target's Contact header information for attended
transfer.
Dial plan
Dial Plan Prefix Configures a prefix added to all numbers when making outbound calls.
Bypass Dial Plan Bypass the dial plan when dialing from one of the available items:

●Contacts
●Call History Incoming Call
●Call History Outgoing Call
●Dialing Page
●MPK
●API
Dial Plan
Configures the dial plan rule. For syntax and examples, please refer to user manual for more details.
Dial Plan Rules:
1. Accepted Digits: 1,2,3,4,5,6,7,8,9,0, *, #, A,a,B,b,C,c,D,d;
2. Grammar: x – any digit from 0-9;
3. Grammar: X – any character from 0-9, a-z, A-Z.
4. xx+ – at least 2 digit numbers
5. xx – only 2 digit numbers
6. XX – two characters ( AA, Ab, 1C, f5, 68,…)
7. test : only string “test” will pass the dial plan check
8. ^ – exclude
9. [3-5] – any digit of 3, 4, or 5
10. [147] – any digit of 1, 4, or 7
11. <2=011> – replace digit 2 with 011 when dialing
12. | – the OR operand
●Example 1: {[369]11 | 1617xxxxxxx}
Allow 311, 611, and 911 or any 11 digit numbers with leading digits 1617;
●Example 2: {^1900x+ | <=1617>xxxxxxx}
Block any number of leading digits 1900 or add prefix 1617 for any dialed 7 digit numbers;
●Example 3: {1xxx[2-9]xxxxxx | <2=011>x+}
Allows any number with leading digit 1 followed by a 3-digit number, followed by any number
between 2 and 9, followed by any 7-digit number OR Allows any length of numbers with leading digit
2, replacing the 2 with 011 when dialed.
●Example of a simple dial plan used in a Home/Office in the US: { ^1900x. | <=1617>[2-9]xxxxxx |
1[2-9]xx[2-9]xxxxxx | 011[2-9]x. | [3469]11 }
Explanation of example rule (reading from left to right):
●^1900x. – prevents dialing any number started with 1900;
●<=1617>[2-9]xxxxxx – allows dialing to local area code (617) numbers by dialing7 numbers and
1617 area code will be added automatically;
●1[2-9]xx[2-9]xxxxxx |- allows dialing to any US/Canada Number with 11 digits length;
●011[2-9]x – allows international calls starting with 011;
●[3469]11 – allows dialing special and emergency numbers 311, 411, 611 and 911.
Note: In some cases, where the user wishes to dial strings such as *123 to activate voice mail or other
applications provided by their service provider, the * should be predefined inside the dial plan feature.
An example dial plan will be: { *x+ } which allows the user to dial * followed by any length of
numbers.
Max length of dial plan is up to 1024 characters.
Ringtone
Account Ring Tone
Allows users to configure the ringtone for the account. Users can choose from different ringtones from
the dropdown menu.
Note: User can also choose silent ring tone.
Ignore Alert-Info header Configures to play default ringtone by ignoring Alert-Info header.
The default setting is “No”.
Match Incoming Caller ID
Specifies matching rules with number, pattern, or Alert Info text (up to 10 matching rules). When the
incoming caller ID or Alert Info matches the rule, the phone will ring with selected distinctive

Account Advanced Settings
ringtone. Matching rules:
●Specific caller ID number. For example, 8321123.
●A defined pattern with certain length using x and + to specify, where x could be any digit from 0 to
9. Samples:
xx+ : at least 2-digit number.
xx : only 2-digit number.
[345]xx: 3-digit number with the leading digit of 3, 4 or 5.
[6-9]xx: 3-digit number with the leading digit from 6 to 9.
●Alert Info text
Users could configure the matching rule as certain text (e.g., priority) and select the custom ring tone
mapped to it. The custom ring tone will be used if the phone receives SIP INVITE with Alert-Info
header in the following format: Alert-Info: <http://127.0.0.1>; info=priority
Selects the distinctive ring tone for the matching rule. When the incoming caller ID or Alert Info
matches one of the 10 rules, the phone will ring with the associated ringtone.
Security Settings
Check Domain Certificates Configures whether the domain certificates will be checked when TLS/TCP is used for SIP Transport.
The default setting is “No”.
Validate Certificate Chain Validate certification chain when TCP/TLS is configured.
The default setting is “No”.
Validate Incoming SIP
Messages
Specifies if the phone will check the incoming SIP messages Caller ID and CSeq headers. If the
message does not include the headers, it will be rejected. The default setting is “No”.
Allow Unsolicited REFER
Configures whether to dial the number carried by Refer-to header after receiving out-of-dialog SIP
REFER request actively.
If set to “Disabled“, the phone will send error warning and stop dialing.
If set to “Enabled/Force Auth“, the phone will dial the number after sending authentication. If the
authentication fails, it will stop dialing.
If set to “Enabled“, the phone will dial all numbers carried by SIP REFER.
Accept Incoming SIP from
Proxy Only
When set to “Yes”, the SIP address of the Request URL in the incoming SIP message will be checked.
If it does not match the SIP server address of the account, the call will be rejected. The default setting
is “No”.
Check SIP User ID for
Incoming INVITE
If set to “Yes”, SIP User ID will be checked in the Request URI of the incoming INVITE. If it does
not match the phone’s SIP User ID, the call will be rejected. The default setting is “No”.
Allow SIP Reset Allow SIP Notification message to perform factory reset.
The default setting is “No”.
Authenticate Incoming INVITE
If set to “Yes”, the phone will challenge the incoming INVITE for authentication with SIP 401
Unauthorized response.
The default setting is “No”.
MOH ( Music on Hold )
On Hold Reminder Tone Configures to play reminder tone when the call is on hold.
Music On Hold URI Music On Hold URI to call when a call is on hold if server supports it.
Special Feature Specifies the server type for special requirements.

Phone Settings Page Definition
Basic Settings
Call Settings
Local RTP Port Configures the local RTP port used to listen and transmit. The valid range is 1024 to 65400 and it must
be even.
Local RTP Port Range Configures the range of local RTP port. Valid value is from 24 to 10000.
Use Random Port
If set to "Yes", the parameter will force random generation of both the local SIP and RTP ports. This is
usually necessary when multiple phones are behind the same full cone NAT. This parameter must be set
to "No" for incoming direct IP calls (outgoing IP calls are not affected).
Keep-Alive Interval Specifies how often the phone sends a blank UDP packet to the SIP server in order to keep "ping hole"
on the NAT router to open.
STUN Server This option sets IP address or Domain name of the STUN server. Only non-symmetric NAT routers
work with STUN.
Use NAT IP Configures the NAT IP address used in SIP/SDP messages. It should ONLY be used if required by your
ITSP.
Delay Registration Configures the specific time that the account will be registered after booting up.
Enable Outbound Notification Configures whether to enable outbound notifications such as Action URL.
General
Preferred Default Account
Select the preferred default account for on-hook or off-hook dialing. when the selected
account is unavailable, system will use the first available account to dial out.
Mute Key Functions While Idle If set to 'Idle Mute", clicking the MUTE key while idle will cause the phone to be muted
automatically when answering incoming calls.
Do Not Escape '#' as %23 in SIP URI Replaces # by %23 for some special situations.
User-Agent Prefix Configures the prefix in the User-Agent header
Outgoing
Enable Direct IP Call Enables Direct IP Call feature.
Onhook Dial Barging When the option is set to "Disabled", onhook dialing won't be interrupted by an incoming call.
Off-hook Auto Dial Configures the digits to be dialed via the first account when the phone is off-hook.
Off-hook Auto Dial Delay Configures the timeout (in seconds) for off-hook auo dial. Valid range is 0-10. If set to 0, it
will be dialed out immediately. Otherwise, it will be dialed out after the configured timeout.

Ringtone
This section is used to customize ringtones for each specific phone feature
Off-hook/On-hook Timeout (s)
If configured, when the phone is in the off-hook or on-hook, dialing state, it will go idle after
the timeout (in seconds). Valid range is 10 to 60.
Redial Expiration (min)
Sets a timeout in minutes for the redial key, after which the previous callout trace will be
cleared and the redial key will be invalid. If set to 0, the previous callout trace will not be
removed permanently unless the device is turned off.
Note: This configuration item also affects the call return soft key timeout in MPK.
The default value is set to 30mins.
Incoming
Allow Incoming Call before Ringing This allows incoming calls after dialed but before ringing. This can be used under custom user
configuration based on need.
Enable Call Waiting Enables the call waiting feature. If set to "No", new incoming calls will be rejected after the
call is established.
Enable Call Waiting Tone Enables Call Waiting alert tone when another incmong call is received while a call is in
progress.
Auto Answer Delay Configures the delay for automatically answering the incoming call. Valid range is 0 to 10
(seconds).
In Call
Enable Auto Unmute
If the option is enabled, automatically unmute phone when a user unholds the call or
establishes a new call.
Enable Busy Tone on Remote
Disconnect Playing busy tone when call is disconnected remotely.
Enable Mute Key In Call When set to "No", the mute key will not work while on call.
Transfer
Enable Transfer Enables Call Transfer feature.
Hold Call Before Completing Transfer When set to "No", the phone will not hold the current call or the transfer target for an Attended
Transfer.
Conference
Enable Conference Enables the Conference feature.
Call Progress Tones ON is the period of ringing ("On time" in "ms") While OFF is the period of silence. Up to three cadences are
supported.
System Ringtone
Dial Tone