GXW42XX Series - User Manual

Ethernet port

Connect to the internal LAN network or router

RESET

Factory Reset button. Press and hold for a while to reset factory default settings.

DC IN

Power adapter connection

Analog port

Connect to analog phones / fax machines with an RJ21 to RJ11 cable

FXS ports

FXS port to be connected to analog phones / fax machines.

Act LED

Remains ON if plug the network cable.

LINK LED

Indicates Ethernet port activity.

FXS LED

Indicate status of the respective FXS Ports on the back panel

Busy – ON (Solid Green)

Available – OFF

Slow blinking FXS LEDs indicates Voice Mail for that port.

All FXS LEDs slow blinking indicates provisioning.

Pin

Signal

Pin

Signal

1

RING1

26

TIP1

2

RING2

27

TIP2

3

RING3

28

TIP3

4

RING4

29

TIP4

5

RING5

30

TIP5

6

RING6

31

TIP6

7

RING7

32

TIP7

8

RING8

33

TIP8

9

RING9

34

TIP9

10

RING10

35

TIP10

11

RING11

36

TIP11

12

RING12

37

TIP12

13

RING13

38

TIP13

14

RING14

39

TIP14

15

RING15

40

TIP15

16

RING16

41

TIP16

17

RING17

42

TIP17

18

RING18

43

TIP18

19

RING19

44

TIP19

20

RING20

45

TIP20

21

RING21

46

TIP21

22

RING22

47

TIP22

23

RING23

48

TIP23

24

RING24

49

TIP24

GXW4216

GXW4224

GXW4232

GXW4248

Telephone Interfaces

16 RJ11 Ports/

1 RJ21 Port

24 RJ11 Ports/

1 RJ21 Port

32 RJ11 Ports/

2 RJ21 Ports

2 RJ21 Ports only

FXS LEDs

16

24

32

48

Network Interface

LAN, Single 10/100/1000 BASE-TX, RJ45

Power Input

Input:100-240VAC, 50/60Hz

Output: 12V DC, 5.0A (4216/24/32)

24V DC, 6.25A (4248 only)

LCD Screen

128x32 pixel

Telco Connector

1 RJ21 (50 pins)

1 RJ21 (50 pins)

2 RJ21 (50 pins)

2 RJ21 (50 pins)

RJ-11 Connectors

Yes

Yes

Yes

No

Function Buttons

1 button for Reset/Factory Reset

Environmental

Operation: 0°C to 45°C

Storage:     -20°C to 60°C

Humidity:  10％ to 90% Non-condensing

Mounting

Desktop and Rack mount

On-hook Voltage

Fixed, 48V

Ring Voltage

50Vrms (balanced ringing)

Ring Frequency

20-50 Hz

Short Haul Loop

2REN: Up to 2km on 24 AWG wire

2REN: Up to 2km on 24 AWG wire

2REN: Up to 2km on 24 AWG wire

2REN: Up to 2km on 24 AWG wire

Outdoor Protection

Over-voltage Protection and surge immunity

Signaling

FXS Loop-start

EMC

EN55022/EN55024 and FCC part15 Class B

Safety

UL

Compliance

FCC, CE, C-Tick

Menu

Voice Prompt

User’s Options

Main Menu

“Enter a Menu Option”

Enter “*” for the next menu option

Enter “#” to return to the main menu

Enter “001 – 005,007, 010, 013 – 017,047, 086 or 099” Menu option

Enter “9” for confirming an option

001

“DHCP Mode”,

“PPPoE Mode” or “Static IP Mode”

Enter ‘9’ to toggle the selection

If user selects “Static IP Mode”, user need configure all the IP address information through menu 002 to 005. If user selects “Dynamic IP Mode”, the device will retrieve all IP address information from DHCP server automatically when user reboots the device.

002

“IP Address “ + IP address

The current WAN IP address is announced

Enter 12-digit new IP address if in Static IP Mode.

003

“Subnet “ + IP address

Same as Menu option 002

004

“Gateway “ + IP address

Same as Menu option 002

005

“DNS Server “ + IP address

Same as Menu option 002

007

Preferred Vocoder

Enter “9” to go to the next selection in the list:

PCMU

PCMA

iLBC

G-726

G-723

G-722

G-729

010

“MAC Address”

Announces the Mac address of the unit.

013

Firmware Server IP Address

Announces current Firmware Server IP address. Enter 12-digit new IP address.

014

Configuration Server IP Address

Announces current Config Server Path IP address. Enter 12-digit new IP address.

015

Upgrade Protocol

Upgrade protocol for firmware and configuration update. Enter “9” to toggle between TFTP and HTTP

016

Firmware Version

Firmware version information.

017

Firmware Upgrade

Firmware upgrade mode. Enter “9” to rotate among the following three options:

1. always check

2. check when pre/suffix changes

3. never upgrade

047

“Direct IP Calling”

Enter the target IP address to make a direct IP call, after dial tone. (See “Make a Direct IP Call”.)

086

Voice Mail

Number of voice mails

099

“RESET”

Enter “9” to reboot the device; or

Enter MAC address to restore factory default setting

(See Restore Factory Default Setting section)

700-748

Phone calls between different ports of the same GW42xx

GXW42XX support inter-port calling from voice menu for easy test/verification in factory

“700” Ring all ports and connect to first port pick up the call

“701-732” Call individual port

“Invalid Entry”

Automatically returns to Main Menu

Key

Call Features

*02

Forcing a Codec (per call): *027110 (PCMU), *027111 (PCMA), *02723 (G723), *02729 (G729), *0272616 (G726-r16), *0272624 (G724-r24), *0272632 (G726-r32), *0272640 (G726-r40), *027201 (iLBC)

*03

Disable LEC (pe call) Dial “*03” +” number”. No dial tone is played in the middle.

*16

Enable SRTP

*17

Disable SRTP

*30

Block CallerID (for all-config change)

*31

Send CallerID (for all-config change)

*67

Block CallerID (per call)

*82

Send CallerID (per call)

*47

Direct IP Calling. Dial “*47” + “IP address”. No dial tone will be played in the middle. Detail see Direct IP Calling section on page 12.

*50

Disable Call Waiting (for all-config change)

*51

Enable Call Waiting (for all-config change)

*69

Call Return Service: Dial *69 and the phone will dial the last incoming phone number received.

*70

Disable Call Waiting (Per Call)

*71

Enable Call Waiting (Per Call)

*72

Unconditional Call Forward: Dial “*72” and then the forwarding number followed by “#”. Wait for dial tone and hang up. (dial tone indicates successful forward)

*73

Cancel Unconditional Call Forward: Dial “*73” and wait for dial tone, then hang up.

*74

Enable Paging Call: Dial “*74” and then the destination phone number you want to activate in Paging mode.

*78

Enable Do Not Disturb (DND): When enabled all incoming calls will be rejected.

*79

Disable Do Not Disturb (DND): When disabled, incoming calls will be accepted.

*87

Blind Transfer

*90

Busy Call Forward: Dial “*90” and then the forwarding number followed by “#”. Wait for dial tone then hang up.

*91

Cancel Busy Call Forward: dial “*91”. Wait for dial tone. Hang up.

*92

Delayed Call Forward: Dial “*92” and then the forwarding number followed by “#”. Wait for dial tone then hang up.

*93

Cancel Delayed Call Forward: Dial “*93” for a dial tone, then hang up.

*98

Play registration ID: Dial “*98”, the registration ID will be announced.

*99

Provision start: Dial “*99”, the provisioning process will be triggered.

Flash/Hook

If user hears call waiting beep, flash/hook will switch to the new incoming call. Also used to switch to a new channel for a new call.

#

Pressing pound sign will serve as Re-Dial key.

User Level

Username

Password

Web Pages Allowed

End User Level

user

123

Only Status and Basic Settings

Administrator Level

admin

Browse all pages

Viewer Level

viewer

viewer

View all pages. Not allowed to modify the content.

Network Settings for Service Interface

IP Address Mode for Service Interface

Choose how the IP address obtained on the gateway for Service Interface

Preferred DNS Server

Enter the preferred DNS server that should be used for DHCP and PPPoE.

DHCP Settings

Host name (Option 12)

Specifies the name of the client. This field is optional but may be required by Internet Service Providers.

DHCP Domain

Specifies the DHCP Domain. This value is optional but may be required by Internet Service Providers.

Vendor Class ID (Option 60)

Used by clients and servers to exchange vendor class ID.

Default is "GXW4200"

Client Circuit ID (Option 82)

Used to Encode an Agent-Local identifier of the circuit, Circuit ID sub-option is supposed to include information specific to which circuit the request came in on.

Remote Agent ID (Option 82)

The relay agent may use this field in addition to or instead of Agent Circuit ID field.  The remote ID sub-option carries information relating to the remote host end of the circuit

PPPoE Settings

PPPoE Account ID

Configures PPPoE Account ID.

PPPoE Password

Configures PPPoE Password.

PPPoE Service Name

Specifies a name for PPPoE.

Static IP Settings

IP Address

Configures static IP Address

Subnet Mask

Configures Subnet Mask

Gateway

Configures Gateway address.

DNS Server 1

Configures primary DNS Server.

DNS Server 2

Configures secondary DNS Server.

Network settings for Management Interface

Enable management interface

An interface for HTTP/HTTPS/SSH, the management interface will be a VLAN interface if management VLAN is configured and different to service VLAN, otherwise, a sub interface will be created on service interface.

Sub interface can only be configured with a static IP address without the default gateway if both management and service interface are using the same VLAN the management interface will automatically use the gateway of the service interface.

Management access

Allows to choose which interface can be reached for HTTP/HTTPS/SSH

Enable SNMP Through Management Interface

Whether to route the SNMP packets through the management interface

Enable Syslog Through Management Interface

Whether to route the Syslog packets through the management interface. (GXW42xx V2 only)

IP Address Mode for Management Interface

Allow to choose how the IP address will be obtained on the gateway for management interface.

Static IP Settings (Management Interface)

IP Address

Enter the IP address when static IP is used for the management interface.

The default setting is 192.168.10.10

Subnet Mask

Enter the Subnet Mask when static IP is used for the management interface..

The default setting is 255.255.255.0

Gateway

Enter the Gateway when static IP is used for the management interface.

The default setting is 192.168.10.1

DNS Server 1

Enter DNS Server 1 when static IP is used or the management interface.

The default setting is 192.168.10.1

DNS Server 2

Enter DNS Server 2 when static IP is used or the management interface.

The default setting is 192.168.10.1

LLDP

Enable LLDP

Configure to enable/disable the LLDP (Link Layer Discovery Protocol) service.

Layer 2 QoS Settings

Layer 2 QoS 802.1Q/VLAN Tag for Service Interface

Value used for layer 2 QoS 802.1Q/VLAN Tag for Service Interface.

Default setting is 0.

Layer 2 QoS 802.1Q/VLAN Tag for Management Interface

Value used for layer 2 QoS 802.1Q/VLAN Tag for Management Interface.

Default setting is 10.

Value used for layer 2 802.1p Priority Value for SIP signaling.

Default setting is 0.

Layer 2 QoS 802.1p Priority Value for RTP media

Value used for layer 2 802.1p Priority Value for RTP media.

Default setting is 0.

Layer 2 QoS 802.1p Priority Value for Management Interface

Assigns the priority value of the Layer 2 QoS packets for Management Interface. Valid range is 0 to 7.

STUN Settings

Use STUN

Enable STUN. Default is No.

STUN server

The IP address or Domain name of the STUN server. Only non-symmetric NAT routers work with STUN.

Number of STUN Response Misses Allowed

The Number of STUN response misses allowed before restarting DHCP. The minimum is 3 misses.

Keep-Alive Interval

Specifies in seconds how often the phone sends a blank UDP packet to the SIP server in order to keep the “ping hole” on the NAT router to open. The default is 20 seconds.

Firewall Setting

Blacklist for WAN Side Port

allows users to block access for specific IP address.

Reply to ICMP

This feature allows users to Enable/Disable ICMP response

Local DNS Setting

Local DNS

This feature allows user to configure the table to map an FQDN with its IP address.

Format is: FQDN0/IP0; FQDN1/IP1; FQDN2/IP2; FQDN3/IP3

Lock Keypad Update

If set to “Yes”, the configuration update via keypad is disabled.

Firmware Upgrade and Provisioning

Specifies how firmware upgrading and provisioning request to be sent.

There are three options to choose from: “Always Check for New Firmware”, “Check New Firmware only when F/W pre/suffix changes”, and “Always Skip the Firmware Check”.

XML Config File Password

The password used for encrypting the XML configuration file using Open SSL. This is required for the phone to decrypt the encrypted XML configuration file.

HTTP/HTTPS User Name

The user name needed to authenticate with the HTTP/HTTPS server.

HTTP/HTTPS Password

The password needed to authenticate with the HTTP/HTTPS server.

Always send HTTP Basic Authentication Information

Default is No. If set to Yes, device will send configured user name and password within HTTP request before server sends authentication challenge.

Upgrade via

Allows users to choose the firmware upgrade method via TFTP, HTTP or HTTPS.

Firmware Server Path

IP address or domain name of firmware server. That URL of the server that hosts the firmware release. The default server is: fm.grandstream.com/gs

Note: Valid input needs to match the following values ^$|^[a-zA-Z_.0-9-]{1,253}$|^[a-zA-Z_.0-9-]{1,253}/[a-zA-Z~%+_= ./0-9-]*$|^[a-zA-Z_.0-9-]{1,253}:[0-9]{1,5}$|^[a-zA-Z_.0-9-]{1,253}:[0-9]{1,5}/[a-zA-Z~%+_= ./0-9-]*$|^[[0-9a-f]{1,4}:[0-9a-f]{1,4}:[0-9a-f]{1,4}:[0-9a-f]{1,4}:[0-9a-f]{1,4}:[0-9a-f]{1,4}:[0-9a-f]{1,4}:[0-9a-f]{1,4}]$|^[[0-9a-f]{1,4}:[0-9a-f]{1,4}:[0-9a-f]{1,4}:[0-9a-f]{1,4}:[0-9a-f]{1,4}:[0-9a-f]{1,4}:[0-9a-f]{1,4}:[0-9a-f]{1,4}]/[a-zA-Z~%+_= ./0-9-]*$|^[[0-9a-f]{1,4}:[0-9a-f]{1,4}:[0-9a-f]{1,4}:[0-9a-f]{1,4}:[0-9a-f]{1,4}:[0-9a-f]{1,4}:[0-9a-f]{1,4}:[0-9a-f]{1,4}]:[0-9]{1,5}$|^[[0-9a-f]{1,4}:[0-9a-f]{1,4}:[0-9a-f]{1,4}:[0-9a-f]{1,4}:[0-9a-f]{1,4}:[0-9a-f]{1,4}:[0-9a-f]{1,4}:[0-9a-f]{1,4}]:[0-9]{1,5}/[a-zA-Z~%+_= ./0-9-]*$

Config Server Path

IP address or domain name of the configuration server. The server hosts a copy of the configuration file to be installed on the gateway. The default server is: fm.grandstream.com/gs

Note: Valid input needs to match the following values ^$|^[a-zA-Z_.0-9-]{1,253}$|^[a-zA-Z_.0-9-]{1,253}/[a-zA-Z~%+_= ./0-9-]*$|^[a-zA-Z_.0-9-]{1,253}:[0-9]{1,5}$|^[a-zA-Z_.0-9-]{1,253}:[0-9]{1,5}/[a-zA-Z~%+_= ./0-9-]*$|^[[0-9a-f]{1,4}:[0-9a-f]{1,4}:[0-9a-f]{1,4}:[0-9a-f]{1,4}:[0-9a-f]{1,4}:[0-9a-f]{1,4}:[0-9a-f]{1,4}:[0-9a-f]{1,4}]$|^[[0-9a-f]{1,4}:[0-9a-f]{1,4}:[0-9a-f]{1,4}:[0-9a-f]{1,4}:[0-9a-f]{1,4}:[0-9a-f]{1,4}:[0-9a-f]{1,4}:[0-9a-f]{1,4}]/[a-zA-Z~%+_= ./0-9-]*$|^[[0-9a-f]{1,4}:[0-9a-f]{1,4}:[0-9a-f]{1,4}:[0-9a-f]{1,4}:[0-9a-f]{1,4}:[0-9a-f]{1,4}:[0-9a-f]{1,4}:[0-9a-f]{1,4}]:[0-9]{1,5}$|^[[0-9a-f]{1,4}:[0-9a-f]{1,4}:[0-9a-f]{1,4}:[0-9a-f]{1,4}:[0-9a-f]{1,4}:[0-9a-f]{1,4}:[0-9a-f]{1,4}:[0-9a-f]{1,4}]:[0-9]{1,5}/[a-zA-Z~%+_= ./0-9-]*$

Firmware File Prefix

This field enables user to store different versions of firmware files in one single directory on the firmware server. If configured, only the firmware file with the matching prefix will be downloaded.

Firmware File Postfix

This field enables user to store different versions of firmware files in one single directory on the firmware server. If configured, only the firmware file with the matching postfix will be downloaded.

Config File Prefix

This field enables user to store different configuration files in one single directory on the configuration server. If configured, only the configuration file with the matching prefix will be downloaded.

Config File Postfix

This field enables user to store different configuration files in one single directory on the configuration server. If configured, only the configuration file with the matching postfix will be downloaded.

Allow DHCP Option 43 and Option 66 to Override Server

If set to “Yes”, configuration and upgrade server’s information can be obtained using DHCP option 66 from DHCP server. This option specifies the URL of the TFTP server.

Note: If DHCP Option 66 is enabled, the gateway will attempt downloading a configuration file from the server URL provided by DHCP, even though Config Server Path is left blank.

Enable Using tags in URL

Allows users to configure variables on the configuration server path to differentiate the directories on the server.

Example: When provisioning, a user can define the mac address and IP address when sending  the HTTP Send request link in the following form "192.168.5.96:8060/?mac=[MAC]&lan_ip=[IP]", the link will look like this example:http://192.168.5.99/mac=000b89a9064&lan_ip=192.168.5.99/cfg.xml

Default Value is "No".

Download and Process All Available Config Files

By default, the device will provision cfgMAC at first, then try to provision the first available config in the order of "DHCP option 67 boot file", cfgMAC.xml => cfgMAC => cfgMODEL.xml => cfg.xml. (corresponding to option 67 specific, device-specific, model specific and global configs).
If set to Yes, the device will download and apply (overwrite) all available configs in the order of cfgMAC.xml => cfgMAC => cfgMODEL.xml => cfg.xml => cfgMAC_override.xml, and "DHCP option 67 boot file".
Option Disabled by Default.

Additional Override DHCP Option

Allow users to configure additional DHCP Options to be used for the firmware server instead of the configured firmware server or the server from DHCP Options 43 and 66. It can be set to None or Option 150, it is set to None by default. 

3CX Auto Provision

Default is No. If set to Yes, phone will multicast SUBSCRIBE for provision if this feature is enabled.

Automatic Upgrade

Choose “Yes” to enable automatic upgrade and provisioning. When set to No, GXW42XX will only do upgrade once at boot up.

When “Check every day” or “Check every week” is checked, user can specify “Hour of the day (0-23)” or “Day of the week (0-6)”. Default time is Monday 1AM.

Randomized Automatic Upgrade

With this feature enabled, device will specify a random time to upgrade the device within the range of hours of the day or postpone the upgrade every X minute(s) by random 1 to X minute(s)

Hour of the Day (0-23)

Define the hour of the day to check HTTP/TFTP server for firmware upgrades or configuration files changes

Day of the Week (0-6)

Define the days of the week to check HTTP/TFTP server for firmware upgrades or configuration files changes

Authenticate Conf File

If set to Yes, configuration file is authenticated before being accepted. This protects the configuration from unauthorized modifications.

Firmware Key

For firmware encryption. It should be 32-digit in Hexadecimal Representation. End user should keep it blank.

Configuration File Types Allowed 

Allow users to choose Allowed Configuration File Types, All or XML Only, and Set by default to All.

Provision

Allows the user to manually provision the device based on the provided firmware path, without the need to reboot the unit.
Once  "Provision" is pressed the device will request the firmware file from the server.
If the device is under provisioning, if you attempt to click on "Provision" again, a pop mesage indicating that the device is currently provisioning will appear.

Upload Device Configuration

Upload Device Configuration

Upload the Config file to reload settings.

Restore from Backup Configuration

Restore the device configuration from the backup file.

Upload Device Firmware

Upload Device Firmware

Upload .bin file to the device to upgrade the firmware.

Download Device Configuration

Device Configuration

(Text format)

Click to download the device configuration file in .txt format.

Device Configuration

(XML format)

Click to download the device configuration file in .xml format.

Export Backup Configuration

(Encrypted XML)

Click to export all device configurations in encrypted .xml format

HTTP Web Port

By default, HTTP uses port 80. This field is for customizable web port.

HTTPS Web Port

By default, HTTP uses port 443. This field is for customizable web port.

Web Access Mode

Specify Web Access Protocol. HTTP or HTTPS. Default is HTTP.

Disable SSH

If set to Yes, SSH access will be disabled. Default is No

SSH Port

Defines the SSH Port Number, Set to 22 by default.

Security Controls for SSH Access

Allow users to configure security privileges. The security privilege options can be as follows : 

The default setting is: "Allow all SSH users to set any Pvalue" 

Web Session Timeout

Configure timer to logout web session during idle. Default is 10 min. Range is 2-60 min

Configure attempt limit before lockout. Default is 5. Range is 1-10

Lockout Time Interval

If login attempt failed 5 times, login would be locked out for the time length. (Default 15 mins. Range 1-15 min).

Disable User Level Web Access

If set to yes, User will not be able to access web UI

Disable Viewer Level Web Access

If set to yes, Viewer will not be able to access web UI

Viewer Password

New Password

Set new password for web GUI access as Viewer. This field is case sensitive with a maximum length of 30 characters.

User Password

New Password

Set new password for web GUI access as User. This field is case sensitive with a maximum length of 30 characters.

Admin Password

New Password

Set new password for web GUI access as Admin. 

Access Control Lists

White list for WAN side

White list for WAN side: only IP in this list can access web and SSH.

Multiple IPs are supported and need to be separated by “space”. Example: 192.168.5.222 192.168.5.223 192.168.7.0/24

Black list for WAN side

Blacklist for WAN side: IP in this list can’t access web and SSH.

Multiple IPs are supported and need to be separated by “space”. Example: 192.168.5.222 192.168.5.223 192.168.7.0/24

Enable SNMP

Enables/Disables the SNMP feature. Default settings is No.

Version

Version of SNMP Agent.

Port

SNMP Trap IP Address

SNMP Trap Port

Port of the SNMP trap receiver (Default 162).

SNMP Trap version

Version of SNMP Trap.

SNMP Trap Interval

The interval between each trap sent to the trap receiver. (Default 5)

SNMPv1/v2c Community

Name of SNMPv1/v2c community.

SNMPv1/v2c Trap Community

Name of SNMPv1/v2c trap community. It must match the community string of the trap receiver.

SNMPv3 User Name

User Name for SNMPv3.

SNMPv3 Security Level

noAuthUser: Users with security level noAuthnoPriv and context name as noAuth.

authUser: Users with security level authNoPriv and context name as auth.

privUser: Users with security level authPriv and context name as priv.

SNMPv3 Authentication Protocol

Select the Authentication Protocol: “None” or “MD5” or “SHA”.

SNMPv3 Privacy Protocol

Select the Privacy Protocol: “None” or “DES” or “AES”.

SNMPv3 Authentication Key

Enter the Authentication Key.

SNMPv3 Privacy Key

Enter the Privacy Key.

SNMPv3 Trap Username

User name for SNMPv3 Trap.

SNMPv3 Trap Security Level

noAuthUser: Users with security level noAuthnoPriv and context name as noAuth.

authUser: Users with security level authNoPriv and context name as auth.

privUser: Users with security level authPriv and context name as priv.

SNMPv3 Trap Authentication Protocol

Select the Authentication Protocol: “None” or “MD5” or “SHA”.

SNMPv3 Trap Privacy Protocol

Select the Privacy Protocol: “None” or “DES” or “AES”.

SNMPv3 Trap Authentication Key

Enter the Trap Authentication Key

SNMPv3 Trap Privacy Key

Enter the Trap Privacy Key.

Download MIB

Click on download to download the MIB file.

RADIUS for Web Access Authentication

Enable RADIUS Web Access Control

Default is No

Action upon RADIUS Auth Server Error

Choose action upon RADIUS server error. Default is Authenticate Locally

RADIUS Auth Protocol

Configure RADIUS authentication protocol, the available options are:

PAP (Password Authentication Protocol): PAP sends the username and password in plaintext, making it less secure than other methods.
 
CHAP (Challenge Handshake Authentication Protocol): CHAP uses a challenge-response mechanism for authentication, enhancing security.
 
MsCHAPv1 (Microsoft Challenge Handshake Authentication Protocol version 1): An older Microsoft-specific CHAP variant.
 
MsCHAPv2 (Microsoft Challenge Handshake Authentication Protocol version 2): A more secure and improved Microsoft-specific CHAP variant with mutual authentication support.
 
Default value is "PAP".

RADIUS Auth Server Address

Address of RADIUS Auth server

RADIUS Auth Server Port

Port of RADIUS Auth server (Default 1812)

RADIUS Shared Secret

Set RADIUS shared secret

RADIUS VSA Vendor ID

Configure RADIUS VSA Vendor IS to match RADIUS server’s configuration. Default is 42397 for Grandstream Networks Inc

RADIUS VSA Access Level Attribute

Configure RADIUS VSA Access Level Attribute to match RADIUS server’s configuration.

Note: Incorrect setting would cause RADIUS authenticate fail

RADIUS for call

Primary RADIUS Server

Set Primary RADIUS Server address.

Primary RADIUS Authentication Port

Default is 1812.

Primary RADIUS Account Port

Default is 1813.

Primary RADIUS Server Secret

Set Primary RADIUS Server Secret

Secondary RADIUS Server

Set Secondary RADIUS Server address.

Secondary RADIUS Authentication Port

Default is 1812.

Secondary RADIUS Account Port

Default is 1813.

Secondary RADIUS Sever Secret

Set Secondary RADIUS Server Secret

RADIUS Timeout

Default is 2.

RADIUS Retry

Default is 3.

Trusted CA Certificate

(A, B, C, D)

The certificate entered here will be accepted as valid CA for authenticating the server TLS certificate.

Load CA Certificates

This feature specifies which CA Certificates the gateway needs to trust. Following options are available:

Default is “Built-in Trusted Certificates”.

NTP Server

URI or IP address of the NTP (Network Time Protocol) server. Used by the phone to synchronize the date and time. An extensive list of public NTP servers can be found at http://www.ntp.org

NTP Update Interval

Defines the update interval (in minutes) to obtain the date and time from the server.

Allow DHCP Option 42 to NTP Server

Default is Yes. DHCP Option 42 is enabled, the NTP server can be changed. If set to No will disable the NTP server be changed.

Time Zone

Configures the date/time used on the phone according to the specified time zone.

Self-Defined Time Zone

This parameter allows the users to define their own time zone. For syntax and examples, please refer to user manual.

Allow DHCP Option 2 to override time zone

Default is Yes. DHCP Option 2 is enabled, the time zone can be offset. If set to No, it will disable the time zone be offset.

Syslog Protocol

This feature allows users to configure the protocol used to send syslog. Protocol supported are: UDP, SSL and TLS. Default is: UDP.

Syslog Server

The IP address or URL of System log server. The server collects system log information from the device.

Syslog Level

Select the GXW42XX to report the log level. Default is NONE.

The level is one of NONE, DEBUG, INFO, WARNING, ERROR, or EXTRA DEBUG. Syslog messages are sent based on the following events:

1. product model/version on boot up (INFO level)

2. NAT related info (INFO level)

3. sent or received SIP message (DEBUG level)

4. SIP message summary (INFO level)

5. inbound and outbound calls (INFO level)

6. registration status change (INFO level)

7. negotiated codec (INFO level)

8. Ethernet link up (INFO level)

9. SLIC chip exception (WARNING and ERROR levels)

10. memory exception (ERROR level)

The Syslog uses USER facility. In addition to standard Syslog payload, it contains the following components:

GS_LOG: [device MAC address][error code] error message

Example: May 19 02:40:38 192.168.1.14 GS_LOG: [00:0b:82:00:a1:be][000] Ethernet link is up

Print SIP in Syslog

Enable or disable printing of full SIP messages in Syslog.

SIP Log Option

By default, the device will split the allowed memory for SIP file into 2 parts. Device will create the first SIP file which is half of the allowed size, when it is full, device will create the second file. When “SIP File Option” is set to Keep, device will keep the SIP files when both files are full, no more new record will be stored. When this feature is set to Override, device will clear the first SIP file and start storing again.

Save Syslog

Save Syslog to device. 3 rotate files save syslog. Each file could save at maximum 3 MB.

Note: Scroll down to “Syslog List” and “The list for SIP log files” to download the saved logs.

CDR record option

Enable or disable override cdr records.

By default, the device will split the allowed memory for CDR file into 2 parts. Device will create the first CDR file which is half of the allowed size, when it is full, device will create the second file.

When “CDR File Option” is set to Keep, device will keep the call records when both files are full, no more new record will be stored.

When this feature is set to Override, The device will clear the first CDR file and start storing again.

Port

Enter the port id for port recording, range: 1 - max port id

With secret Key information

Allow users to configure whether the packet capture file contains secret key information or
not. It is set to Yes by Default

Status

Start/stop Ethernet capture.

Warning: This feature can only be used for debugging purposes, please delete all files after finishing debugging.

Ring Tones

System Ring Cadence

Configuration option for all FXS ports rings cadence for all incoming calls.
(Syntax: c=on1/off1-on2/off2-on3/off3; […]) Default is set to c=2000/4000; (US standards)

Call Progress Tones

Using these settings, the user can configure tone frequencies according to user preference. By default, the tones are set to North American frequencies. Frequencies should be configured with known values to avoid uncomfortable high pitch sounds. ON is the period of ringing (ON time in ms) while OFF is the period of silence. In order to set a continuous ring, OFF should be zero. Otherwise, it will ring ON ms and a pause of OFF ms and then repeat the pattern.

The below tones can be configured with the following syntax :
f1=val, f2=val[, c=on1/ off1[- on2/ off2[- on3/ off3]]];
(Frequencies are in Hz and cadence on and off are in 1 ms units)

Please refer to the document below to determine your local call progress tones:
http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf

FXO Failover

Failover to FXO Gateway

Enable or disable the Failover FXO Gateway.

FXO Gateway IP

IP Address or URI of the FXO gateway.

System Features

Disable Direct IP Call

Disables the Direct IP Call function. Default is “No”. If set to “Yes” direct IP-to-IP calling will not be supported.

Inter-port Calling

With this feature enabled, when the FXS port is unregistered, users can use the same extensions of the FXS port for inter-port dialing instead of dialing ***7xx.

Disable SIP NOTIFY Authentication

Disables challenging SIP NOTIFY reboot and resync messages.

Disable Voice Prompt

Disables the voice prompt configuration. Default is “No”. If set to “Yes” accessing integrated voice menu will be impossible.

IVR Language

Choose English, Chinese, Russian or Spanish.

Upload/Delete Language Pack

Upload/delete language pack.

Display Language

Choose language for LCD display.

Prompt Dial Tone Code

Simulates an analog PBX where a code is required to dial an outside line.

Port Lock Code

Play Busy Tone When Account is unregistered

If set to "Yes", busy tone will be played when user goes offhook from an unregistered account.
Default value is "No".

Country Specific Deployment

This feature allows to customize the tones to match the country’s specifications. There are 5 supported options: None (default), USA, China, China ITSP, and Germany.

Automatic Reboot

Default is No. When “Yes, reboot every day” or “Yes, reboot every week” is checked, user can specify “Hour of the day (0-23)” or “Day of the week (0-6)”. Default time is Monday 1AM.

LED Pattern

This feature allows users to customize LED patterns. There are three patterns to choose from:

General Settings

Profile Active

When set to Yes, the SIP Profile is activated.

SIP Server

SIP Server’s IP address or Domain name provided by VoIP service provider.

Failover SIP Server

Failover SIP Server’s IP address or Domain name provided by VoIP Service provider. This server will be used if the Primary SIP server becomes unavailable.

Prefer Primary SIP Server

Control how the FXS Gateway handles SIP server selection, allowing you to influence the routing and distribution of VoIP traffic based on your network's requirements, The options are

The default value is set to "No".

Primary Outbound Proxy

IP address or Domain name of Outbound Proxy, or Media Gateway, or Session Border Controller. Used by GXW42XX for firewall or NAT penetration in different network environments. If symmetric NAT is detected, STUN will not work and ONLY outbound proxy can correct the problem.

Backup Outbound Proxy

Backup Outbound Proxy which will be used when the primary outbound proxy cannot be connected

Prefer Primary Outbound Proxy

If yes, the profile will re-register via the primary outbound proxy when registration expires.

Network Settings

Layer 3 QoS Settings

Configure the Diff-Serv value for SIP and RTP. Default is:SIP Diff-Serv 24RTP Diff-Serv 46

DNS Mode

One from the 3 modes available for “DNS Mode” configuration:A Record (for resolving IP Address of target according to domain name)SRV (DNS SRV resource records indicates how to find services for various protocols)NAPTR/SRV (Naming Authority Pointer according to RFC 2915)Use Configured IP (If selected, please fill in Primary IP, Backup IP 1 and Backup IP 2.)One mode can be chosen for the client to look up server.The default value is “A Record”

DNS SRV Use Registered IP

If yes, under DNS SRV mode, use Registered IP as INVITE destination instead of using high priority IP from resolved IP list.

Primary IP

Configures the primary IP address where the phone sends DNS query to when “Use Configured IP” is selected for DNS mode.

Backup IP 1

Configures the backup IP 1 address where the phone sends DNS query to when “Use Configured IP” is selected for DNS mode.

Backup IP 2

Configures the backup IP 2 address where the phone sends DNS query to when “Use Configured IP” is selected for DNS mode.

NAT Traversal

This parameter defines whether the GXW42XX NAT traversal mechanism is activated or not.

Default setting is No.

Use NAT IP

The NAT IP address used in SIP/SDP messages. It should ONLY be used if required by your ITSP.

Proxy-Require

A SIP Extension to notify the SIP server that the phone is behind a NAT/Firewall.

SIP Settings – Basic Settings

SIP transport

User can select UDP or TCP or TLS. Please make sure you’re SIP Server or network environment supports SIP over the selected transport method. Default is UDP.

SIP Registration

This parameter controls whether the GXW42XX needs to send REGISTER messages to the proxy server. The default setting is “Yes”.

Unregister on Reboot

Default is No. If set to “Yes”, the SIP user’s registration information is cleared on reboot.

Add Auth Header on Initial REGISTER

Default is No. If set to “Yes”, an Authentication Header with blank nonce will be added in the initial REGISTER.

Outgoing Calls Without Registration

If set to “Yes,” user can place outgoing calls even when not registered (if allowed by Internet Telephone Service Provider) but is unable to receive incoming calls.Any port, member of a Hunting Group that is not registered with a SIP account, will be able to place outbound calls using the SIP credentials of the primary Hunting Group port. Default is No.For example: Port 1, 3 and 5 are members of the same Hunting Group. Port 1 is registered with a SIP account. Ports 3 and 5 are not registered. Ports 3 and 5 will be able to place outbound calls using the SIP account of port 1, even if Outgoing Call without Registration is set to No.

Register Expiration

Allows the user to specify the time frequency (in minutes) for the GXW42XX to refresh its registration with the specified registrar. The default interval is 60 minutes (or 1 hour). The maximum interval is 65535 minutes (about 45 days).

SIP Registration Failure Retry Wait Time

Allows the user to specify the time frequency (in seconds) for the GXW42XX to re-register after registration failure. The default interval is 20 seconds. The maximum interval is 3600seconds (1 hour).

SIP Registration Failure Retry Wait Time upon 403 Forbidden

Default is 1200 sec. Specifies the interval to retry registration if the process is failed due to 403 Forbidden. Valid range is 0 to 3600 in second. 0 second means stop retry registration.

Reregister Before Expiration

Determines how many seconds before the previous registration expires that the port should reregister.

Enable SIP OPTIONS/NOTIFY Keep Alive

Enable SIP OPTIONS or SIP NOTIFY keep-alive. Default is No.

SIP OPTIONS/NOTIFY Keep Alive Interval

Time interval for OPTIONS/NOTIFY Keep Alive feature in Second.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.

Local SIP Port

Defines the local SIP port the GXW42XX will listen and transmit. The default value for Profile 1 is 5060 and 6060 for Profile 2.

Use Random SIP Port

Default is No. If set to Yes, the device will pick randomly-generated SIP ports. This is usually necessary when multiple GXW42XX /HT50X are behind the same NAT.

Local RTP port

Defines the local RTP port used to listen and transmit RTP packets. The default value is 50000 for Profile 1, 51000 for Profile 2, 52000 for Profile 3 and 53000 for Profile 4.

SIP T1 Timeout

T1 is an estimate of the round-trip time between the client and server transactions.

If the network latency is high, select a larger value for more reliable usage..

SIP T2 Timeout

Maximum retransmission interval for non-INVITE requests and INVITE responses.

Remove OBP from Route Header

Default is No. If set to Yes, the Outbound Proxy will be removed from the route header.

Support SIP Instance ID

Default is Yes. If set to Yes, the contact header in REGISTER request will contain SIP Instance ID as defined in IETF SIP Outbound draft.

Hold Target Before Refer

Default is Yes. Select whether the gateway will send ReINVITE to hold transfer target before sending REFER to transferee.

Refer-To Use Target Contact

Default is No. If set to YES, then for Attended Transfer, the “Refer-To” header uses the transferred target’s Contact header information.  

SUBSCRIBE for MWI

Default is No. When set to Yes, a SUBSCRIBE for Message Waiting Indication will be sent periodically.

Enable 100rel

Enables the use of PRACK (Provisional Acknowledgment) method.

TEL URI

The default setting is “Disabled”. If the phone has an assigned PSTN Number, this field should be set to “User=Phone” then a “User=Phone” parameter will be attached to the “From header” in the SIP request to indicate the E.164 number. If server supports TEL URI format, then this option needs to be selected.

Use Request Routing ID in SIP Headers

Default is No. If set to Yes, device will use the configured [Request URI Routing ID] in the SIP Header. This option is usually used under a SIP trunk account’s configuration.

Do Not Escape ‘#’ as %23 in SIP URI

If set to “Yes”, device will use ‘#’ instead of %23 in the send URI.

Disable Multiple m Line in SDP

Default is No. If set to Yes, device will send only one m line in SDP, regardless how many m field in the incoming SDP.

Use Privacy Header

If set to Default, it will only add Privacy or PPI header when special feature is not Telkom SA or CBCOM.

Use P-Preferred-Identity Header

Controls whether the P-Preferred-Identity Header will present in the SIP INVITE message.
If set to Default, it will only add Privacy or PPI header when special feature is not Telkom SA or CBCOM.
Set to Default.

Use P-Access-Network-Info Header

Controls whether the P-Access-Network-Info Header will be present in the SIP INVITE message.
Deafult value is Yes.

Use P-Emergency-Info Header

Controls whether the P-Emergency-Info Header will present in the SIP INVITE message.
Default value is Yes.

Use P-Asserted-Identity Header

Controls whether the P-Asserted-Identity Header will present in the SIP INVITE message.
Default is No.

Use P-Early-Media Header

Controls whether the P-Early-Media Header support is enabled.
Default is no.

SIP REGISTER Contact Header Uses

Default is LAN Address. If set to WAN Address, device will detect its WAN address and use it in SIP REGISTER Contact Header.

Caller ID Display

When set to “Auto”, the phone will look for the caller ID in the order of P-Asserted Identity Header, Remote-Party-ID Header and From Header in the incoming SIP invite. When set to “Disabled”, All incoming calls are displayed with “Unavailable”.

Use Actual Ephemeral Port in Contact with TCP/TLS

Controls the port information in the Via header and Contact header when TCP/TLS is selected for SIP Transport: If set to No (Default), these port numbers will use the permanent listening port on the device. Otherwise, they will use the ephemeral port for the connection.

Allow SIP Factory Reset

This feature Allows to factory reset the device directly through SIP NOTIFY.
Disabled by Default.

Ignore Alert-Info Header

This feature allows users to configure Ignore Alert-Info Header. It is configured to play the default ringtone by ignoring the Alert-Info header.
Set to "No" By Default.

SIP User-Agent

Allow users to configure SIP User-Agent and SIP User-Agent Postfix. The User-Agent header is constructed from these two, if not configured, the User-Agent header will use the default one.

SIP User-Agent Postfix

Allow users to configure SIP User-Agent Postfix.

Add MAC in User-Agent

If set to "Yes except REGISTER", all outgoing SIP messages will include the gateway MAC address in the User-Agent header, except for REGISTER and UNREGISTER. If set to "Yes to All SIP", all outgoing SIP messages will include the gateway MAC address in the User-Agent header. If set to "No", the gateway MAC address will not be included in the User-Agent header in any outgoing SIP messages.

The default value is "No"

Use MAC Header

If set to "Register Only", all outgoing register and unregister SIP message will include the MAC header. If set to "Yes to all request SIP", all outgoing request SIP messages will include the MAC header. If set to "No", the MAC header will not be included in any outgoing SIP messages.
Defaullt value is "No"

SIP Settings – Session Timer

Session Expiration

The session timer extension enables SIP sessions to be periodically “refreshed” via a SIP request (UPDATE, or re-INVITE. When the session interval expires, if there is no refresh via a UPDATE or re-INVITE message, the session will be terminated.Session Expiration is the time (in seconds) at which the session is considered timed out, if no successful session refresh transaction occurs beforehand. The default value is 180 seconds.

Min-SE

The minimum session expiration (in seconds). The default value is 90 seconds.

Caller Request Timer

If selecting “Yes” the phone will use session timer when it makes outbound calls if remote party supports session timer.

Callee Request Timer

If selecting “Yes” the phone will use session timer when it receives inbound calls with session timer request.

Force Timer

If selecting “Yes” the phone will use session timer even if the remote party does not support this feature. Selecting “No” will allow the phone to enable session timer only when the remote party support this feature. To turn off Session Timer, select “No” for Caller Request Timer, Callee Request Timer, and Force Timer.

UAC Specify Refresher

As a Caller, select UAC to use the phone as the refresher, or UAS to use the Callee or proxy server as the refresher.

UAS Specify Refresher

As a Callee, select UAC to use caller or proxy server as the refresher, or UAS to use the phone as the refresher.

Force INVITE

Session Timer can be refreshed using INVITE method or UPDATE method. Select “Yes” to use INVITE method to refresh the session timer.

SIP Settings – Security Settings

Validate Incoming Messages

Default is No. Defines whether the incoming messages will be validated.

Check SIP User ID for Incoming INVITE

Default is No. If set to Yes, SIP User ID will be checked in the Request URI of the incoming INVITE. If it doesn’t match the phone’s SIP User ID, the call will be rejected. Direct IP calling will also be disabled.

Accept Incoming SIP from Proxy Only

Checks SIP address of the Request URI in the incoming SIP message; if it doesn’t match SIP server address of the account, the call will be rejected. Default is No.

Authenticate Incoming INVITE

Default is No. If set to Yes, the phone will challenge the incoming INVITE for authentication with SIP 401 Unauthorized response.

Authenticate server certificate domain

Default is No. If this is set to Yes, device will check the server TLS certificate to ensure that the Common Name matches the configured SIP server

Authenticate server certificate chain

Default is No. If this is set to Yes, device will check the server TLS certificate to ensure that it is authorized by a known Certificate Authority

Fax Settings

Fax Mode

T.38 (Auto Detect) FoIP by default, or Pass-Through (must use codec PCMU/PCMA)

Fax Tone Detection Mode

Default is Callee. This decides whether Caller or Callee sends out the re-INVITE for T.38 or Fax Pass Through.

Send Re-INVITE After Fax Completion

Default is No, If set to “Yes”, device will send an INVITE with audio vocoders upon competition of Fax to continue session in audio only.

Send Re-INVITE After Fax Tone

If set to “Yes”, device will send a Re-INVITE after Fax tone is detected, disabling will only work under Broadsoft feature.

Enable Silence Detection for Fax Disconnect

For fax machines that do not send a Disconnect when fax is done. This option Enables/Disables the detection of silence in order to know the fax has finished. The silence period is non-configurable and fixed to 7 seconds.

Audio Settings

Preferred DTMF method (in listed order)

The GXW42XX supports up to 3 different DTMF methods including in-audio, via RTP (RFC2833) and via Sip Info. The user can configure DTMF method in a priority list.

DTMF-RELAY Tag Respect SIP INFO

Default is No. If set to No, the DTMF-relay tag in Accept Header is always added in outbound INVITE. If set to Yes, it depends on if SIP INFO is chosen in Preferred DTMF Method.

Disable DTMF Negotiation

Default is No. If set to yes, use above DTMF order without negotiation

Enable Fast RFC2833

Default is No. DTMF will use 240 ms timing. If set to yes, DTMF will 100 ms instead.

DTMF Payload Type

Sets the payload type for DTMF using RFC2833.

Custom DTMF InBand Duration

This feature allows users to configure the Custom DTMF InBand Duration.

Inband DTMF Tx Gain

Defines the Transmission gain of inband DTMF.
Default value is 6dB.

DSP DTMF Detector Min Level

Defines the minimum level required for DTMF tone to be considered valid.
Default value is -25dBm from the valid range -45dBm to 0dBm

DSP DTMF Detector SNR

Defines the maximum signal-to-noise ratio required for DTMF tone to be considered valid.
Default value is set to 1, from the range 0-2

DSP DTMF Detector Deviation

Defines the deviation above and below the central frequency.
Default value is 25% from the valid range 0-25

DSP DTMF Detector Twist

Defines the maximum value by which one frequency energy level can surpass the another frequency energy level.
Default value is 5dB from the valid range 0-12

Preferred Vocoder

The GXW42XX supports up to 8 different Vocoder types including G.711 A-/U-law, G.726 (Supports bit rates 16, 24, 32 and 40), G.723.1, G.729A/B, iLBC. The user can configure Vocoders in a preference list that will be included with the same preference order in SDP message. The first Vocoder is entered by choosing the appropriate option in “Choice 1”. The last Vocoder is entered by choosing the appropriate option in “Choice 8”.

Voice Frames per TX

Number of the frame size when it transmits. Default is 2, from 1 -4 for G711/G726/G729

G723 Rate

Defines the encoding rate for G.723 vocoder. By default, 6.3kbps rate is chosen.

G726-32 Packing Mode

Choose the packing mode for G726-32

iLBC Frame Size

Sets the iLBC frame size in 20ms or 30ms

iLBC Payload type

Default value is 97. Defines payload type for iLBC. The valid range is between 96 and 127.

AAL2-G726-16 Payload type

Default value is100. Range is from 96 to 127.

AAL2-G726-24 Payload type

Default value is 99. Range is from 96 to 127.

AAL2-G726-32 Payload type

Default value is 104. Range is from 96 to 127.

AAL2-G726-40 Payload type

Default value is 103. Range is from 96 to 127.

Use First Matching Vocoder in 200OK SDP

Default is No. If set to “Yes”, device will include only the first match vocoder in its 200OK response, otherwise it will include all match vocoders in same order received in INVITE.

SRTP Mode

Default is Disabled. Other options are Enabled but not forced, and Enabled and forced.It uses SDP Security Description to exchange key. Please refer to the following link : https://tools.ietf.org/html/rfc4568SRTP: https://datatracker.ietf.org/doc/html/rfc3711

SRTP Key Length

Allow Users to configure the cipher method or key length if SRTP is enabled The Value of the SRTP Key length can be :

By default, it is set to AES 128&256 bit

Crypto Life Time

Default setting is Yes. Enable or disable the crypto life time when using SRTP. If users set to disable this option, phone does not add the crypto life time to SRTP header.

Silence Suppression (VAD)

Default is No. VAD allows detecting the absence of audio and conserve bandwidth by preventing the transmission of “silent packets” over the network.

Jitter Buffer Type

Select either Fixed or Adaptive based on network conditions.

Jitter Buffer Length

Select Low, Medium or High based on network conditions.High (initial 200ms, min 40ms, max 600ms) Note: not all vocoders can meet the high requirementMedium (initial 100ms, min 20ms, max 200ms)Low (initial 50ms, min 10ms, max 100ms)

RTCP

With this feature enabled, the device will send RTCP Sender Report during the conversation until it’s ended.

SLIC Setting

Depends on standard phone type (and location)

Ringing Frequency

This feature allows to customize ringing frequency.

Caller ID Scheme

Select the value according to the local Telco standard where the GXW42XX is installed.Please refer to the pull-down list to select.

Polarity Reversal

Default is No. If set to “Yes”, polarity will be reversed upon call establishment and termination.

Loop Current Disconnect

Set to Yes if the traditional PBX you are using with GXW42XX uses this method for signaling call termination. Default is No.

Play busy/reorder tone before Loop Current Disconnect

Default is No. If set to yes, it will play busy/reorder tone before loop current disconnect upon call fail.

Loop Current Disconnect duration

Default is 200.In 100 – 10000 milliseconds range.

Enable Hook Flash

If no, flash will be treated as an on-hook event.

Hook Flash timing(Minimum)/ (Maximum)

Sets the minimum/maximum time the phone is on hook before being detected as a hook flash. The range is 40 to 2000 milliseconds

On Hook Timing

Sets the time required to detect that the phone is on hook. The range is 40 to 2000 milliseconds. Default is 400 milliseconds.

Gain

Handset volume adjustment. RX is for receiving volume (direction FXS🡪to analog phone), TX is for transmission volume (Analog phone🡪 to FXS). Default values are 0dB for both parameters. Loudest volume: +12dB Lowest volume: -12dB.RX Gain can be increased by using “Force + x dB” options. But note, when using the Force options, may introduce echo issue.

Disable Line Echo Canceller (LEC)

Default is No. If set to “Yes”, device will not use LEC to remove echo from a voice communication.

SLIC Setting

Depends on standard phone type (and location)

Caller ID Scheme

Select the value according to the local Telco standard where the GXW42XX is installed. Please refer to the pull-down list to select.

Call Settings

Early Dial

Default is No. Use only if proxy supports 484 response.This parameter controls whether the phone will send an early INVITE each time a key is pressed when a user dials a number. If set to “Yes”, an INVITE is sent using the dial-number collected thus far; Otherwise, no INVITE is sent until the “(Re-)Dial” button is pressed or after about 5 seconds have elapsed if the user forgets to press the “Re-Dial” button. The “Yes” option should be used ONLY if there is a SIP proxy configured and the proxy server supports 484 Incomplete Address response. Otherwise, the call will likely be rejected by the proxy (with a 404 Not Found error).This feature is NOT designed to work with and should NOT be enabled for direct IP-to-IP calling.

Dial Plan Prefix

Sets the prefix added to each dialed number.

Dial Plan

Dial Plan Rules:Accept Digits: 1,2,3,4,5,6,7,8,9,0 , *, #, A,a,B,b,C,c,D,dGrammar: x – any digit from 0-9;xx+ – at least 2 digits number;xx – at least 2 digits number;^ – exclude;[3-5] – any digit of 3, 4, or 5;[147] – any digit 1, 4, or 7;<2=011> – replace digit 2 with 011 when dialing< =1> – add a leading 1 to all numbers dialed, vice versa will remove a 1 from the number dialed| – orExample 1: {[369]11 | 1617xxxxxxx} –Allow 311, 611, 911, and any 10 digit numbers of leading digits 1617Example 2: {^1900x+ | <=1617>xxxxxxx} –Block any number of leading digits 1900 and add prefix 1617 for any dialed 7 digit numbersExample 3: {1xxx[2-9]xxxxxx | <2=011>x+} –Allow any length of number with leading digit 2 and 10 digit-numbers of leading digit 1 and leading exchange number between 2 and 9; If leading digit is 2, replace leading digit 2 with 011 before dialing.Default: Outgoing – {x+}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 dialing 7 numbers and 1617 area code will be added automatically1[2-9]xx[2-9]xxxxxx |- allows dialing to any US/Canada Number with 11 digits length011[2-9]x. – allows international calls starting with 011[3469]11 – allow dialing special and emergency numbers 311, 411, 611 and 911Note: In some cases user wishes to dial strings such as *123 to activate voice mail or other application provided by service provider. In this case * should be predefined inside dial plan feature and the Dial Plan will be: { [x*]+}.

Use # as Dial Key

Allows users to configure the “#” key as the “Send” (or “Dial”) key. If set to “Yes”, “#” will send the number. In this case, this key is essentially equivalent to the “Dial” key. If set to “No”, this “#” key can be included as part of number.

Disable # as Redial key

When “Use # as dial key” is enabled, and this option is set to NO, the “#” key will function as redial key but If it’s set to “Yes” “#” key will not function as redial key.

No Key Entry Timeout

Default is 4 seconds. Call will be completed within this time interval if no additional key entry occurs.

Off-Hook Auto-Dial Delay

Determines how many seconds after off-hook to wait before autodialing the extension set under Advanced Port Settings. The range is 0 to 60 seconds.

Allow Auto-Dial Config Per Port

Allow users to configure Auto-Dial feature per port when hunting group is enabled. This feature is disabled by default.

RFC2543 Hold

Allows users to toggle between RFC2543 hold and RFC3261 hold. RFC2543 hold (0.0.0.0) allows user to disable the hold music sent to the other side. RFC3261 (a line) will play the hold music to the other side.

Disable Call Waiting

Default is No. If set to Yes Call Waiting indication information will not be provided to analog phone connected to this FXS port.

Disable Call Waiting Caller ID

Default is No. If set to Yes Call Waiting caller ID will not be provided to analog phone connected to this FXS port.

Disable Call Waiting Tone

Default is No. This is to disable the stutter Call Waiting Tone when a Call Waiting call arrives. The CWCID will still be displayed.

Disable Connected Line ID

Default is No. If set to Yes, Connected Line ID will not be displayed even received.

Disable Receiver Offhook Tone

Default is No. If set to Yes, ROH tone will not be played after off hook for 60 seconds.

Disable Reminder Ring for On-Hold Call

Default is No. This is to disable the Reminder Ring that is played when a call is waiting on hold and the analog phone goes on-hook.

Disable Reminder Ring for DND

This feature allows users to disable reminder ring when FXS port is on DND mode.

Disable Voicemail Reminder Tone

If this feature is set to yes, the reminder tone will not be played when there is voicemail.

Disable Visual MWI

Visual message indicator is a special on-hook caller ID type message that enables and disables the message waiting light on certain phones. GXW42XX has this feature enabled by default. However, certain phones (rare) that do not support it may mistakenly treat this CID signal as an incoming call. A configuration option is needed to turn on MWI in this case.

Visual MWI Type

Specifies how voicemail notifications are visually indicated on analog phones, with options like "FSK," "Neon," or "PR."

It is set to FSK by Default.

MWI Tone

This feature allows users to configure the MWI tone for the voice mail, the device will play the MWI tone when the user picks up. you can set it to either "Default" or "Special Proceed Indication Tone".

The default is the Stuttgart dial tone.

Transfer on Conference Hangup

Defines whether the call is transferred to the other party if the conference initiator hangs up.

Send Hook Flash Event

Default is No. If set to yes, flash will be sent as a DTMF event.

Send Hook Flash Event Method

When Send Hook Flash Event Method is set to SIP INFO, hook flash event will be sent via “SIP INFO”, if set to Default, hook flash event will be sent as DTMF event.

Flash Digit Control

Considering the following Conditions are met : 

Send Hook Flash Event = No
Flash Digit Control = Yes

Note: Gibtelecom ITSP is now supported for Flash digit control actions mentioned above.
Default Value is set to "No"

Callee Flash to 3WC

When this feature is set to Yes, the device would be able to set up a 3-way conference call even if it’s the callee in the second call overriding the default behavior for call control when a call is established and held, port as callee establishes another call. Default behavior: pressing hook flash can toggle between two calls. If this setting is set to “YES”, pressing hook flash will bridge two calls as behavior of a 3-way conference.

Ring Timeout

Incoming call will stop ringing when not picked up given a specific period of time.

If set to yes, the phone will play ringback tone when there is no receiving early media RTP packets.
Default value is set to "No"

Delayed Call Forward Wait Timeout

Default value is 20 seconds. In case this feature activated using * codes (*92 code), the call will be forwarded after this preconfigured amount of time.

Send Anonymous

If this parameter is set to “Yes”, the “From” header along with Privacy andP_Asserted_Identity headers in outgoing INVITE message will be set to anonymous, blocking Caller ID.

Anonymous Call Rejection

Default is No. If set to Yes, incoming calls with anonymous Caller ID will be rejected with 486 Busy message.

Replace Beginning ‘+’ in Caller ID with

Allows users to configure the Replace the beginning '+' in Caller ID with the configured string when sending the Caller ID to the analog phone.

Hunting Group Type

Linear and Circular. Linear style will sort the call to the lowest-numbered available line, this is also called “serial hunting”. Circular style will distribute the calls “round-robin”. If a call is assigned to line 1, the next call goes to 2 and the next to 3. The succession throughout each of the lines continues even if one of the previous lines becomes available. When the end of the hunt group is reached, the hunting starts over at the first line. Lines are skipped if they are still busy on a previous call. These two hunting styles can be configured from the Profile X page.

Hunting Group Ring Timeout

Default is 20 seconds. If call is not answered within this designated time period, the call will be forwarded to the next member of a Hunt Group.

Call Features Settings

Enable Call Features

When enabled, Do not Disturb, Call Forward and other call features can be used via the local feature code on the phone. Otherwise, the ITSP feature code will be used.

Enable All will override all individual features enable setting.

Note: When call features on GXW42xx are enabled (UCF, DND…), if picking up the handset, GXW42xx will play stutter tone to indicate that a call feature is active then the dial tone will be played after that.

Reset Call Features

Allows users to reset call features via Web UI. Default is No.

SRTP Feature

Default is Yes, and star code is Enable-16, Disable-17.

SRTP per call Feature

Default is Yes, and star code is Enable-18, Disable-19.

CID Feature

Default is Yes, and star code is Enable-31, Disable-30.

CID per call Feature

Default is Yes, and star code is Enable-82, Disable-67.

Direct IP calling Feature

Default is Yes, and active star code is 47.

CW Feature

Default is Yes, and star code is Enable-51, Disable-50.

CW per call Feature

Default is Yes, and star code is Enable-71, Disable-70.

Call return Feature

Default is Yes, and active star code is 69.

Unconditional forward Feature

Default is Yes, and star code is Enable-72, Disable-73.

Busy forward Feature

Default is Yes, and star code is Enable-90, Disable-91.

Delayed forward Feature

Default is Yes, and star code is Enable-92, Disable-93.

Paging Feature

Default is Yes, and active star code is 74.

DND Feature

Default is Yes, and star code is Enable-78, Disable-79.

Blind Transfer Feature

Default is Yes, and active star code is 87.

Disable LEC per call Feature

Default is Yes, and active star code is 03.

3WC Feature

Disable Bellcore-style 3-Way Conference default is No. If set to Yes, flash on 2nd line will not bring 2nd line in to the conference with 1st line. Instead, flash will switch between lines.

Star Code 3WC Feature default is Yes.

3WC active star code is 23.

Provision start Feature

Default is Yes, and active star code is 99.

Play registration id Feature

Default is Yes, and active star code is 98.

Play port number

Enable the port playing feature

Enable playing port number

This feature allows users to use the star code to check the port number the analog phone is connected

Forced Codec Feature

Default is Yes, and active star code is 02.

For enabling each codec feature, default is Yes.

Default code of codec:

Ring Tones

Distinctive Ringtone

Customizes the Ring Tone 1 to 3 with associate caller ID: when selected, if caller ID is configured, then the device will ONLY use this ring tone when the incoming call is from the Caller ID. System Ring Tone is used for all other calls. When selected but no Caller ID is configured, the selected ring tone will be used for all incoming calls using the FXS port. Distinctive ring tones can be configured not only for matching a whole number, but also for matching prefixes. In this case the symbol “x+” will be used.

For example: if configured as 617x+, Ring Tone 1 will be used in case of call arrived from the area code 617. Any other incoming call will ring using cadence defined in parameter System Ring Cadence located under Advanced Settings Configuration page.

Note: If server supports Alert-Info header and standard ring tone set (Bellcore) or distinctive ring tone 1-10 is specified, then the ring tone in the Alert-Info header from server will be used. Bellcore rings and tones are independent from custom ring tones.

The custom ring tones can also be specified by alert-info header, for example Alert-Info: ;info=ring5

Ring Tones

Configure ring cadences according to preference.

Call Waiting Tones

Distinctive Call Waiting Tones

Allow users to configure distinctive call waiting tone. The selected call waiting tone will be used if the incoming caller ID matches the specified number

Call Waiting Tone Cadences

Configure ring cadences according to preference.

Port Settings

SIP User ID

User account information, provided by VoIP service provider (ITSP). Usually in the form of digit similar to phone number or actually a phone number.

Authenticate ID

SIP service subscriber’s Authenticate ID used for authentication. Can be identical to or different from SIP User ID.

Password

SIP service subscriber’s account password for GXW42XX to register to (SIP) servers of ITSP.

Name

Any name to identify this specific user.

Profile

Select the corresponding Profile ID (1/2/3/4)

Enable FXS (TR-069)

Enable or disable the FXS port.

Advanced Port Settings

Offhook Auto-dial

Configure an auto dial number when Offhook.

Hunting Group

Assign this port to a certain hunting group

Request URI Routing ID

If configured, device will route the incoming call to designated port by request URI user ID in SIP INVITE.

Lock/Unlock Extension Code

Allows users to configure the locking extension for the GXW42xx FXS Ports.
The lock password has to be a 4 character digital string.

FXO Mapping

Map to FXO Port #

This is used only when peering with a Grandstream GXW410x. Default 1, Supported values are 1-8, meaning line 1 to 8 of the GXW410x device where the port will be mapped to.

Map to FXO Gateway IP

This is used when peering with an FXO gateway of any brand. You have to specifically mention the IP and sip port where call will be sent to.

Port

SIP port that will be annexed to the IP address above.

Key

Mapping

0-9

0-9

A