Grandstream Networks GXW42 series User Manual

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Grandstream Networks GXW42 series User Manual | Manualzz

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Grandstream Networks, Inc.

Analog IP Gateway GXW42XX

16, 24 or 32 FXS Ports

GXW4200 series User Manual

Firmware Version 1.0.0.49

http://arttel.ru/shop/voip-shlyuzy/golosovoj-shlyuz-grandstream-gxw4216 www.grandstream.com

[email protected]

http://arttel.ru

GXW42XX USER MANUAL

INDEX

GNU GPL INFORMATION .......................................................................... 5

CHANGE LOG ........................................................................................... 6

WELCOME ................................................................................................. 7

GATEWAY GXW42XX OVERVIEW ....................................................................................................... 7

SAFETY COMPLIANCE ........................................................................................................................ 7

WARRANTY ........................................................................................................................................... 7

CONNECT YOUR GXW42XX GATEWAY .................................................. 9

EQUIPMENT PACKAGING ................................................................................................................... 9

CONNECT THE GXW42XX ................................................................................................................... 9

GXW42XX FEATURES ............................................................................ 13

SOFTWARE FEATURES OVERVIEW ................................................................................................ 13

HARDWARE SPECIFICATION ............................................................................................................ 15

GXW42XX LCD MENU ........................................................................................................................ 17

FIGURE 3: LCD MENUBASIC OPERATIONS ......................................... 17

BASIC OPERATIONS .............................................................................. 18

UNDERSTANDING GXW VOICE PROMPTS ..................................................................................... 18

PLACING A PHONE CALL .................................................................................................................. 19

PHONE OR EXTENSION NUMBERS .......................................................................................... 19

DIRECT IP CALLS ........................................................................................................................ 20

CALL HOLD ......................................................................................................................................... 21

CALL WAITING .................................................................................................................................... 21

CALL TRANSFER ................................................................................................................................ 21

BLIND TRANSFER ....................................................................................................................... 21

ATTENDED TRANSFER .............................................................................................................. 22

3-WAY CONFERENCING .................................................................................................................... 22

HUNTING GROUP .............................................................................................................................. 22

INTER-PORT CALLING ....................................................................................................................... 24

SENDING AND RECEIVING FAX ....................................................................................................... 25

SUPPORT RADIUS PROTOCOL ........................................................................................................ 25

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CALL FEATURES .................................................................................... 26

CONFIGURATION GUIDE ........................................................................ 28

CONFIGURING GXW42XX VIA VOICE PROMPT ............................................................................. 28

CONFIGURING GXW42XX WITH WEB BROWSER .......................................................................... 28

ACCESS THE WEB CONFIGURATION MENU ........................................................................... 29

IMPORTANT SETTINGS ..................................................................................................................... 29

NAT SETTINGS ............................................................................................................................ 29

DTMF METHODS ......................................................................................................................... 30

PREFERRED VOCODER (CODEC) ............................................................................................ 30

SAVING THE CONFIGURATION CHANGES...................................................................................... 48

REBOOTING FROM REMOTE ........................................................................................................... 48

CONFIGURATION THROUGH A CENTRAL SERVER ....................................................................... 48

SOFTWARE UPGRADE ........................................................................... 50

FIRMWARE UPGRADE THROUGH TFTP/HTTP/HTTPS .................................................................. 50

INSTRUCTIONS FOR LOCAL FIRMWARE UPGRADE: .................................................................... 51

CONFIGURATION FILE DOWNLOAD ................................................................................................ 51

FIRMWARE AND CONFIGURATION FILE PREFIX AND POSTFIX .................................................. 52

MANAGING FIRMWARE AND CONFIGURATION FILE DOWNLOAD .............................................. 52

RESTORE FACTORY DEFAULT SETTING ............................................. 53

FACTORY RESET ............................................................................................................................... 53

RESET BUTTON .......................................................................................................................... 53

IVR COMMAND ............................................................................................................................ 53

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T ABLE OF F IGURES

GXW42XX User Manual

Figure 1: Diagram of GXW4216/24/32 Panel ............................................................................................. 10

Figure 2: 50 Pin Telco Connector ................................................................................................................ 12

Figure 3: LCD Menu .................................................................................................................................... 17

T ABLE OF T ABLES

GXW42XX User Manual

Table 1: Definitions of the GXW Connectors .............................................................................................. 11

Table 2: Definitions of the GXW Display Panel .......................................................................................... 11

Table 3: GXW42XX Software Features ...................................................................................................... 13

Table 4: Hardware Specification of GXW42XX series gateways ............................................................... 15

Table 5: Definitions of the GXW Voice Prompts ......................................................................................... 18

Table 6: Call Features Table (Star Code) ................................................................................................... 26

Table 7: Basic Settings ............................................................................................................................... 31

Table 8: Status Page .................................................................................................................................. 33

Table 9: Advanced settings ........................................................................................................................ 33

Table 10: Profile Settings ........................................................................................................................... 38

Table 11: FXS Ports Settings ..................................................................................................................... 47

C

ONFIGURATION

GUI I

NTERFACE

E

XAMPLES

GXW42XX User Manual http://www.grandstream.com/products/gxw_series/gxw42xx/documents/gxw42xx_gui.zip

1. S CREENSHOT OF A DVANCED S ETTINGS C ONFIGURATION P AGE

2. S CREENSHOT OF B ASIC S ETTINGS C ONFIGURATION P AGE

3. S CREENSHOT OF FXS P ORTS C ONFIGURATION L OGIN P AGE

4. S CREENSHOT OF P ROFILE 1 C ONFIGURATION P AGE

5. S CREENSHOT OF P ROFILE 2 C ONFIGURATION P AGE

6. S CREENSHOT OF S TATUS P AGE

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GNU GPL INFORMATION

GXW42xx firmware contains third-party software licensed under the GNU General Public License (GPL).

Grandstream uses software under the specific terms of the GPL. Please see the GNU General Public

License (GPL) for the exact terms and conditions of the license.

Grandstream GNU GPL related source code can be downloaded from Grandstream web site from: http://www.grandstream.com/support/faq/gnu_gpl .

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CHANGE

LOG

This section documents significant changes from previous versions of GXW42xx user manuals. Only major new features or major document updates are listed here. Minor updates for corrections or editing are not documented here.

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WELCOME

Thank you for purchasing the Grandstream GXW42XX Analog FXS IP Gateway. The GXW42XX offers an easy to manage, easy to configure IP communications solution for any business with virtual and/or branch locations. The GXW42XX supports popular voice codecs and is designed for full SIP compatibility and interoperability with third party SIP providers, thus enabling you to fully leverage the benefits of VoIP technology, integrate a traditional phone system into a VoIP network, and efficiently manage communication costs.

This manual will help you learn how to operate and manage your GXW FXS Analog IP Gateway and make the best use of its many upgraded features including simple and quick installation, multi-party conferencing, and direct IP-IP Calling. This IP Analog Gateway is very easy to manage and scalable, specifically designed to be an easy to use and affordable VoIP solution for the small – medium business or enterprise.

GATEWAY GXW42XX OVERVIEW

The new GXW42XX series has a compact and quiet design and offers superb audio quality, rich feature functionality, strong security protection, and good manageability. It is auto-configurable, remotely manageable and scalable.

The GXW42XX features 16,24 or 32 port FXS interface for analog telephones, dual 10/100/1000Mbps network ports, and RJ21 analog port (1 for 4216 and 4224, 2 for 4232). In addition, it supports the option of 4 SIP Server profiles, caller ID for various countries/regions, T.38 fax, flexible dialing plans, security protection (SIPS/TLS), comprehensive voice codec including G.711 (a/u-law), G.723.1, G.726(16/24/32/40 bit rates), iLBC and G.729.

SAFETY COMPLIANCE

The GXW42XX is compliant with various safety standards including FCC/CE. Its power adapter is compliant with UL standard. Warning: use only the power adapter included in the GXW42XX package.

Use of alternative power adapter may permanently damage the unit.

WARRANTY

Grandstream has a reseller agreement with our reseller customers. End users should contact the

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company from whom the product was purchased, for replacement, repair or refund.

If you purchased the product directly from Grandstream, contact your Grandstream Sales and Service

Representative for an RMA (Return Materials Authorization) number. Grandstream reserves the right to change the warranty policy without prior notification.

Caution: Changes or modifications to this product not expressly approved by Grandstream, or operation of this product in any way other than as detailed by this User Manual, could void your manufacturer

warranty.

Reproduction or transmittal of the entire or any part, in any form or by any means, electronic or print, for any purpose without the express written permission of Grandstream Networks, Inc. is not permitted.

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CONNECT YOUR GXW42XX GATEWAY

Connecting the GXW42XX gateway is easy. Before you begin, please verify the contents of the

GXW42XX package.

EQUIPMENT PACKAGING

Unpack and check all accessories. Equipment includes: x one device unit x one RJ45 Ethernet cable x one 12V 5A universal power adapter x mount

CONNECT THE GXW42XX

Follow these four (4) steps to connect your GXW42XX gateway to the Internet and access the unit’s configuration pages.

1. Connect standard touch-tone analog phones to the GXW42XX’s RJ21 port with a RJ11 to R21 cable.

2. Insert an RJ45 Ethernet cable into the WAN port of GXW42XX and connect the other end to an uplink port (a router or a modem, etc.)

3. Plug the power adapter into the GXW42XX gateway into a power outlet.

Follow the instructions from the topic “ Configuring GXW 42XX with Web Browser ” for initial configuration.

The GUI pages will guide you through the remaining steps to set-up your gateway. Examples of the GUI

Interfaces can be downloaded from : http://www.grandstream.com/products/ip-voice-telephony/enterprise-analog-gateways/gxw42xx/GUI.zip

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FIGURE 1: DIAGRAM OF GXW4216/24/32 PANEL

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Ethernet port

RESET

DC 12V 5A

Analog port

FXS ports

TABLE 1: DEFINITIONS OF THE GXW CONNECTORS

Connect to the internal LAN network or router

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

Power adapter connection

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

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

Note: Once the GXW42XX is turned on and configured, the front display panel indicates the status of the unit.

Act LED

TABLE 2: DEFINITIONS OF THE GXW DISPLAY PANEL

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

5

6

7

8

1

2

3

4

9

10

11

12

13

14

Signal

RING1

RING2

RING3

RING4

RING5

RING6

RING7

RING8

RING9

RING10

RING11

RING12

RING13

RING14

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Pin

31

32

33

34

35

36

37

38

39

26

27

28

29

30

Signal

TIP1

TIP2

TIP3

TIP4

TIP5

TIP6

TIP7

TIP8

TIP9

TIP10

TIP11

TIP12

TIP13

TIP14

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19

20

21

22

15

16

17

18

23

24

RING15

RING16

RING17

RING18

RING19

RING20

RING21

RING22

RING23

RING24

40

41

42

43

44

45

46

47

48

49

FIGURE 2: 50 PIN TELCO CONNECTOR

TIP15

TIP16

TIP17

TIP18

TIP19

TIP20

TIP21

TIP22

TIP23

TIP24

NOTE: y Pins 25 and 50, the last pair, are not used. y All LED lights display green when ON. y LINK LED and ACT LED blinking in following status indicate different network speed.

10M: LINK and ACT all blink, ACT blinks faster then LINK

100M: LINK and ACT all blink, ACT blinks the same fast as LINK

1000M: LINK does not blink, only ACT blink y Slow blinking of FXS LED together indicates a firmware upgrade or provisioning state.

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GXW42XX FEATURES

The GXW42XX series is a next generation IP voice gateway that is interoperable and compatible with leading IP-PBXs, Softswitches and SIP platforms. The GXW42XX series FXS gateway is autoconfigurable, remotely manageable and scalable. The GXW42XX series gateways come in three models

- the GXW4216, GXW4224 and GXW4232, each offering superb voice quality, traditional telephony functionality, easy deployment, and 16, 24 and 32 FXS ports respectively. Each model features flexible dialing plans, integrated call routing to support a pure IP network call and an external power supply.

SOFTWARE FEATURES OVERVIEW

x 16, 24 or 32 FXS ports x RJ-45 Ethernet ports x 4 configurable SIP profiles x Supports Voice Codecs:

G711(a/µ, Annex I & II), G723.1A, G726 (ADPCM with 16/24/32/40 bit rates), G729 A/B, iLBC,

T.38 Fax x Comprehensive Dial Plan support for Outgoing calls. x G.168 Echo Cancellation x Voice Activation Detection (VAD), Comfort Noise Generation (CNG), and Packet Loss

Concealment (PLC) x Supports PSTN/PBX analog telephone sets or analog trunks

TABLE 3: GXW42XX SOFTWARE FEATURES

GXW4216

Telephone Interfaces 16 FXS ports

GXW4224

24 FXS ports

16 SIP accounts, 4 profiles 24 SIP accounts,4 profiles SIP Provisioning

Network Interface 10/100/1000 Mbps, RJ-45

16 Concurrent Calls

10/100/1000 Mbps, RJ-45

24 Concurrent Calls Number of

Concurrent Calls

(except when using

SRTP)

GXW4232

32 FXS ports

32 SIP accounts, 4 profiles

10/100/1000 Mbps, RJ-45

32 Concurrent Calls,

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Voice over Packet

Capabilities

Voice Activity Detection

(VAD) with CNG (comfort noise generation) and PLC

(packet loss concealment),

LEC with NLP Packetized

Voice Protocol Unit

(supports RTP/RTCP and

AAL2 protocol), G.168 compliant Echo

Cancellation, Dynamic Jitter

Buffer, Modem detection & auto-switch to G.711

Voice Activity Detection

(VAD) with CNG (comfort noise generation) and PLC

(packet loss concealment),

LEC with NLP Packetized

Voice Protocol Unit

(supports RTP/RTCP and

AAL2 protocol), G.168 compliant Echo

Cancellation, Dynamic Jitter

Voice Activity Detection

(VAD) with CNG (comfort noise generation) and PLC

(packet loss concealment),

LEC with NLP Packetized

Voice Protocol Unit

(supports RTP/RTCP and

AAL2 protocol), G.168 compliant Echo

Cancellation, Dynamic Jitter

Buffer, Modem detection & auto-switch to G.711

Buffer, Modem detection & auto-switch to G.711

Voice Compression G.711 + Annex I (PLC),

Annex II (VAD/CNG format) encoder and decoder, G.722

(pending), G.723.1A,

G.726(ADPCM with

16/24/32/40 bit rates),

G.729, iLBC, G.726 provides proprietary VAD, CNG, and signal power estimation,

Voice Play Out unit

(reordering, fixed and adaptive jitter buffer, clock synchronization), AGC

(automatic gain control),

G.711 + Annex I (PLC),

Annex II (VAD/CNG format) encoder and decoder, G.722

(pending), G.723.1A,

G.726(ADPCM with

16/24/32/40 bit rates),

G.729, iLBC, G.726 provides proprietary VAD, CNG, and signal power estimation,

Voice Play Out unit

(reordering, fixed and adaptive jitter buffer, clock synchronization), AGC

(automatic gain control),

G.711 + Annex I (PLC),

Annex II (VAD/CNG format) encoder and decoder, G.722

(pending), G.723.1A,

G.726(ADPCM with

16/24/32/40 bit rates),

G.729, iLBC, G.726 provides proprietary VAD, CNG, and signal power estimation,

Voice Play Out unit

(reordering, fixed and adaptive jitter buffer, clock synchronization), AGC

(automatic gain control),

Status output, Decoder controlling via voice packet header

Status output, Decoder controlling via voice packet header

Status output, Decoder controlling via voice packet header

DHCP Server/Client DHCP Client only DHCP Client only DHCP Client only

Fax over IP

QoS

Transport Protocol

T.38 compliant Group 3 Fax

Relay up to 14.4kpbs and auto-switch to G.711 for Fax

Pass-through,

T.38 compliant Group 3 Fax

Relay up to 14.4kpbs and auto-switch to G.711 for Fax

Pass-through,

T.38 compliant Group 3 Fax

Relay up to 14.4kpbs and auto-switch to G.711 for Fax

Pass-through,

Fax Datapump V.17, V.21, Fax Datapump V.17, V.21,

V.27ter, V.29 for T.38 fax relay

V.27ter, V.29 for T.38 fax relay

Fax Datapump V.17, V.21,

V.27ter, V.29 for T.38 fax relay

DiffServ, TOS, 802.1P/Q

VLAN tagging

DiffServ, TOS, 802.1P/Q

VLAN tagging

DiffServ, TOS, 802.1P/Q

VLAN tagging

RTP/RTCP RTP/RTCP RTP/RTCP

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DTMF Method

IP Signaling

Provisioning

In-audio, RFC2833, and/or SIP Info

SIP (RFC 3261)

TFTP, HTTP, HTTPS

SRTP, IPSEC, TLS/SIPS, HTTPS, 802.1x Security

Management

Dial Plan

3-Way Conference

Caller ID

Syslog support, HTTP and Telnet access

Yes

3-Way conference with local mixing

Bellcore Type 1 & 2, ETSI, BT, NTT, FSK and DTMF-based CID

Polarity Reversal /

Wink

Yes

Network

Connectivity

IPv4, IPv6, TCP/UDP, RTP/RTCP, HTTP/HTTPS, ARP/RARP, ICMP, DNS,

DHCP, NTP, TFTP, TELNET, PPPoE, STUN

HARDWARE SPECIFICATION

The hardware specifications of the GXW FXS series are detailed in Table 4.

TABLE 4: HARDWARE SPECIFICATION OF GXW42XX SERIES GATEWAYS

GXW4216 GXW4224 GXW4232

16 FXS Ports 24 FXS Ports 32 FXS Ports Telephone

Interface

FXS LEDs

Network interface

Power Input

32

LCD screen

Telco connector

RJ-11 connectors

NAND Flash

DRAM

Function Buttons

Environmental

16 24

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

Input:100-240VAC, 50/60Hz

Output: 12V DC, 5.0A

128×32 pixel

1 RJ21 (50 pins)

Yes

64MB

1 RJ21 (50 pins)

128MB (+ separate 8MB if with a slave processor)

1 button for Reset/Factory Reset

Operation: 0°C to 45°C

2 RJ21 (50 pins)

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Storage: -20°C to 60°C

Humidity: 10 Á to 90% Non-condensing

Mounting Desktop and Rack mount

Fixed, 48V Onhook Voltage

Ring Voltage

Ring Frequency

Short Haul Loop 2REN, up to 6000ft on 24 AWG wire

Outdoor Protection Over-voltage Protection and surge immunity

Signaling

50Vrms (balanced ringing)

20-50Hz

EMC

FXS Loop-start

EN55022/EN55024 and FCC part15 Class B

Safety UL

FCC, CE, C-Tick Compliance

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GXW42XX LCD MENU

The GXW42XX gateway series includes a small LCD screen for the display of basic information. The LCD has a display area of 128x32 pixels, which will allow for 2 lines of text with a 16px height limitation per line.

The LCD menu is showed as Figure 3: LCD Menu. Menu is navigated by the Down-arrow and OK button.

FIGURE 3: LCD MENU

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BASIC OPERATIONS

UNDERSTANDING GXW VOICE PROMPTS

GXW42XX has a built-in voice prompt menu for simple device configuration. To enter the voice prompt menu, press *** on the standard analog phone connected to any FXS port.

Menu Voice Prompt

TABLE 5: DEFINITIONS OF THE GXW VOICE PROMPTS

User’s Options

003

004

005

007

Main Menu

001

“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 a option

“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.

“Subnet “ + IP address Same as Menu option 002

“Gateway “ + IP address Same as Menu option 002

“DNS Server “ + IP address Same as Menu option 002

Preferred Vocoder Enter “9” to go to the next selection in the list: x PCM U x PCM A x iLBC x G-726 x G-723 x G-729

010

013

014

“MAC Address”

Firmware

Address

Announces the Mac address of the unit.

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

Configuration Server IP

Address

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

015 Upgrade Protocol

FIRMWARE 1.0.0.49

Upgrade protocol for firmware and configuration update. Enter

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“9” to toggle between TFTP and HTTP

016

017

047

086

099

700-732

Firmware Version

Firmware Upgrade

“Direct IP Calling”

Firmware version information.

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

1. always check

2. check when pre/suffix changes

3. never upgrade

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

Voice Mail

“RESET”

Number of voice mails

Enter “9” to reboot the device; or

Enter MAC address to restore factory default setting

(See Restore Factory Default Setting section)

Phone calls between GXW42xX support inter-port calling from voice menu for easy different ports of the same

GW42xx 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

Five Success Tips when using the Voice Prompt

1. “*” shifts down to the next menu option

2. “#” returns to the main menu

3. “9” functions as the ENTER key in many cases to confirm an option

4. All entered digit sequences have known lengths - 2 digits for menu option and 12 digits for IP address. For IP address, add 0 before the digits if the digits are less than 3 (i.e. - 192.168.0.26 should be key in like 192168000026. No decimal is needed).

5. Key entry cannot be deleted but the phone may prompt error once it is detected

PLACING A PHONE CALL

PHONE OR EXTENSION NUMBERS

1. Dial the number directly and wait for 4 seconds (To change the default value, modify the following setting - “No Key Entry Timeout”)

or

2. Dial the number directly and press # (Use # as dial key” must be configured in web configuration).

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Examples:

1. Dial a number (e.g. (626) 666-7890), first enter the prefix number (usually 1+ or international code) followed by the phone number. Press # or wait for 4 seconds. Check with your VoIP service provider for further details on prefix numbers.

DIRECT IP CALLS

Direct IP calling allows two parties, that is, a FXS Port with an analog phone and another VoIP Device, to talk to each other in an ad hoc fashion without a SIP proxy.

Elements necessary to completing a Direct IP Call:

1. Both GXW42XX and other VoIP Device, have public IP addresses, or

2. Both GXW42XX and other VoIP Device are on the same LAN using private IP addresses, or

3. Both GXW42XX and other VoIP Device can be connected through a router using public or private

IP addresses (with necessary port forwarding or DMZ).

GXW42XX supports two ways to make Direct IP Calling:

Using IVR

1. Pick up the analog phone then access the voice menu prompt by dial “***

2. Dial “047” to access the direct IP call menu

3. Enter the IP address using format ex. 192*168*0*160 after the dial tone.

Using Star Code

1. Pick up the analog phone then dial “*47

2. Enter the target IP address using same format as above.

Note: NO dial tone will be played between step 1 and 2.

Destination ports can be specified by using “*” (encoding for “:”) followed by the port number.

Examples: a) If the target IP address is 192.168.0.160, the dialing convention is

*47 or Voice Prompt with option 047, then 192*168*0*160. followed by pressing the “#” key if it is configured as a send key or wait 4 seconds. In this case, the default destination port 5060 is used if no port is specified. b) If the target IP address/port is 192.168.1.20:5062, then the dialing convention would be:

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*47 or Voice Prompt with option 047, then 192*168*0*160*5062 followed by pressing the “#” key if it is configured as a send key or wait for 4 seconds.

CALL HOLD

Place a call on hold by pressing the “flash” button on the analog phone (if the phone has that button).

Press the “flash” button again to release the previously held Caller and resume conversation. If no “flash” button is available, use “hook flash” (toggle on-off hook quickly). You may drop a call using hook flash.

CALL WAITING

Call waiting tone (2 short beeps) indicates an incoming call, if the call waiting feature is enabled. Toggle between incoming call and current call by pressing the “flash” button. First call is placed on hold. Press the “flash” button to toggle between two active calls.

CALL TRANSFER

BLIND TRANSFER

Assume that call Caller A and B are in conversation. A wants to Blind Transfer B to C:

3. Caller A presses FLASH on the analog phone to hear the dial tone.

4. Caller A dials *87 then dials caller C’s number, and then # (or wait for 4 seconds).

5. Caller A will hear the confirm tone. Then, A can hang up.

NOTE: Enable Call Feature” must be set to “Yes” in web configuration page.

Caller A can place a call on hold and wait for one of three situations:

1. A quick confirmation tone (similar to call waiting tone) followed by a dial tone. This indicates the transfer is successful (transferee has received a 200 OK from transfer target). At this point, Caller

A can either hang up or make another call.

2. A quick busy tone followed by a restored call (on supported platforms only). This means the transferee has received a 4xx response for the INVITE and we will try to recover the call. The busy tone is just to indicate to the transferor that the transfer has failed.

3. Continuous busy tone. The phone has timed out. Note: continuous busy tone does not indicate the transfer has been successful, nor does it indicate the transfer has failed. It often means there was a failure to receive second NOTIFY – check firmware for most recent release.

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ATTENDED TRANSFER

Assume that Caller A and B are in conversation. Caller A wants to Attend Transfer B to C:

1. Caller A presses FLASH on the analog phone for dial tone.

2. Caller A then dials Caller C’s number followed by # (or wait for 4 seconds).

3. If Caller C answers the call, Caller A and Caller C are in conversation. Then A can hang up to complete transfer.

4. If Caller C does not answer the call, Caller A can press “flash” to resume call with Caller B.

NOTE: When Attended Transfer fails and A hangs up, the GXW42XX will ring back user A to remind A that B is still on the call. A can pick up the phone to resume conversation with B.

3-WAY CONFERENCING

The GXW42XX supports Bellcore style 3-way Conference.

Instructions for 3-way conference:

Assuming that call party A and B are in conversation. A (GXW42XX) wants to bring C in a conference:

1. A presses FLASH (on the analog phone, or Hook Flash for old model phones) to get a dial tone.

2. A dials C’s number then # (or wait for 4 seconds).

3. If C answers the call, then A presses FLASH to bring B, C in the conference.

4. If C does not answer the call, A can press FLASH back to talk to B.

5. If A presses FLASH during conference, C will be dropped out.

6. If A hangs up, the conference will be terminated or transfer B to C if “Transfer on Conference

Hangup” set to yes

HUNTING GROUP

This feature allows the user to setup a single SIP account on the gateway and have the ability to use all

FXS ports to make/receive calls. Using this feature, all ports active in same Hunting Group will have the same phone number and incoming calls will be distributed in a Linear or Circular manner among the ports active in that Hunting Group. The number of hunting groups is limited by the number of ports each GXW model has - i.e. each port can be its own Hunting Group. The most practical and efficient way to use

Hunting Groups is to assign 2 or 3 ports to separate Hunting Groups.

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One additional and popular way to use the Hunting Group feature is called “multiplexed analog lines”. In this configuration, a legacy PBX system with 8 FXO trunks can be connected to 8 GXW 42xx ports configured as a Hunting Group. The GXW can be registered to a SIP server provider using only one phone number. If the SIP service provider allows multiple calls to the same number, the GXW will allow 8 concurrent calls to the same SIP number. All office members can be reached remotely using the same phone number in a round-robin fashion.

Example Configuration of a typical Hunting Group:

1. Configure the SIP account from your VoIP Service Provider on FXS port 1 under FXS Ports webpage.

2. Select Active under the Hunting Group drop box for FXS port 1.

3. For the remaining ports (say 2, 3 and 4) select 1 for Hunting Group. Ports 2, 3 and 4 are now active members of the hunting group associated with port 1.

This configuration will route all calls directed to FXS port 1 to ports 2, 3 and/or 4 in round robin fashion respectively if port 1 is busy or times out. You can configure the ring timeout on the Profile page.

Example configuration of a multiple Hunting Group:

FXS Port #1: SIP UserID and Authenticate ID entered, Hunting group set to "Active"

FXS Port #2: SIP UserID and Authenticate ID left blank, Hunting Group set to "1"

FXS Port #3: SIP UserID and Authenticate ID left blank, Hunting Group set to "1"

FXS Port #4: SIP UserID and Authenticate ID entered, Hunting group set to "Active"

FXS Port #5: SIP UserID and Authenticate ID left blank, Hunting Group set to "4"

FXS Port #6: SIP UserID and Authenticate ID left blank, Hunting Group set to "4"

FXS Port #7: SIP UserID and Authenticate ID entered, Hunting group set to "Active"

FXS Port #8: SIP UserID and Authenticate ID left blank, Hunting Group set to "7"

Hunting Group 1 contains ports 1, 2, 3. Hunting Group 4 contains ports 4, 5, 6.

Hunting Group 7 contains ports 7, 8.

Please be aware, the choice of 1 for ports 2 and 3, the choice of 4 for ports 5 and 6, the choice 7 for port

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8 is required to indicate that the SIP account tied to port marked as “Active” will be used for all members of the same Hunting group. Needless to say, those members of the same Hunting group may not be sequential ports. In following example ports 3, 5 and 7 tied to SIP Account configured in Port #1 marked as “Active”, and ports 4,6,8 tied to SIP Account configured in Port #2 marked as “Active” as well.

Example of not sequential configuration of a multiple Hunting Group:

FXS Port #1: SIP UserID and Authenticate ID entered, Hunting group set to "Active"

FXS Port #2: SIP UserID and Authenticate ID entered, Hunting Group set to "Active"

FXS Port #3: SIP UserID and Authenticate ID left blank, Hunting Group set to "1"

FXS Port #4: SIP UserID and Authenticate ID left blank, Hunting group set to "2"

FXS Port #5: SIP UserID and Authenticate ID left blank, Hunting Group set to "1"

FXS Port #6: SIP UserID and Authenticate ID left blank, Hunting Group set to "2"

FXS Port #7: SIP UserID and Authenticate ID left blank, Hunting group set to "1"

FXS Port #8: SIP UserID and Authenticate ID left blank, Hunting Group set to "2"

FXS Port #24: SIP UserID and Authenticate ID left blank, Hunting Group set to "2"

Note: A single call directed to the SIP account will NOT result in all ports ringing at the same time. They will ring in the hunting group only. This feature is applicable to incoming calls only.

There are two types of hunting groups, Linear and Circular. Linear style will sort the call to the lowestnumbered available line, this is also called “serial hunting”. Circular style will distribute the calls "roundrobin". 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.

INTER-PORT CALLING

In some cases a user may want to make phone calls between the phones connected to multiple ports of the same gateway when it is used as a standalone unit, without the use of a SIP server. This feature will also be applicable when the gateway is used with Hunting Groups and is registered to SIP server only with one master number. In such cases users still will be able to make inter-port calls by using the IVR feature.

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For example, on the GXW42xx inter-port calling is achieved by dialing *** and 7 plus two extra digits corresponding to the port number. For example the user connected to port 1 can be reached by dialing *** and 701; the user connected to port 24 can be reached by dialing *** 724.

SENDING AND RECEIVING FAX

GXW42XX supports fax in two modes: 1) T.38 (Fax over IP) and 2) Fax Pass through. T.38 is the preferred method because it is more reliable and works well in most network conditions. If the service provider supports T.38, please use this method by selecting T.38 as fax mode (default). If the service provider does not support T.38, pass-through mode may be used. If you have problems with sending or receiving Fax, toggle the Fax Tone Detection Mode setting.

SUPPORT RADIUS PROTOCOL

GXW42XX supports RADIUS for authentication, authorization and billing purposes. Primary and secondary RADIUS server configurations are available to provide redundancy to this feature. In case

Primary Radius server becomes unusable, RADIUS requests will be automatically sent to the secondary server. When at least one RADUIS server was configured, the device will allow users to make phone calls only after authorization from RADIUS server has been received. CDR (Call Detail Record) is also sent to the RADIUS server for billing purposes. RAIDUS server can send requests to terminate calls when run out of pre-paid credit.

The GXW42xx will be able to work in VoIP billing environment using redundant double server configuration. User will be able to configure primary and secondary RADUIS server IP Addresses or

FQDNs. Once at least one RADUIS server was configured, the device will allow users to make phone calls only after permission from RADIUS server has been received. In case Primary Radius server becomes unusable, secondary will take role of primary and will manage credit recourses in the network.

Imbedded RADIUS client also supports request generated by Radius server to terminate calls when run out of pre-paid credit.

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*03

*16

*17

*30

*31

*67

*82

*47

CALL FEATURES

GXW42XX supports the traditional telephony features available in a PBX as well as additional advanced telephony features.

Key Call Features

TABLE 6: CALL FEATURES TABLE (STAR CODE)

*02

*50

*51

*69

*70

*71

*72

*73

*74

*78

*79

*87

Forcing a Codec (per call) *027110 (PCMU), *027111 (PCMA), *02723 (G723), *02729

(G729), *0272616 (G726-r16), *0272624 (G724-r24), *0272632 (G726-r32), *0272640 (G726r40), *027201 (iLBC)

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

Enable SRTP

Disable SRTP

Block CallerID (for all-config change)

Send CallerID (for all-config change)

Block CallerID (per call)

Send CallerID (per call)

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.

Disable Call Waiting (for all-config change)

Enable Call Waiting (for all-config change)

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

Disable Call Waiting (Per Call)

Enable Call Waiting (Per Call)

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

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

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

Paging mode.

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

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

Blind Transfer

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#

*90

*91

*92

*93

Flash/Hook

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

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

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

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

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.

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CONFIGURATION GUIDE

CONFIGURING GXW42XX VIA VOICE PROMPT

DHCP M ODE

Select voice menu option 001 to enable GXW42XX to use DHCP.

STATIC IP M ODE

Select voice menu option 001 to enable GXW42XX to use STATIC IP mode, then use option 002, 003,

004, 005 to set up IP address, Subnet Mask, Gateway and DNS server respectively.

PPP O E M ODE

Select voice menu option 001 to enable GXW42XX to use PPPoE mode.

F IRMWARE S ERVER IP A DDRESS

Select voice menu option 013 to configure the IP address of the firmware server.

C ONFIGURATION S ERVER IP A DDRESS

Select voice menu option 014 to configure the IP address of the configuration server.

U PGRADE P ROTOCOL

Select voice menu option 015 to choose firmware and configuration upgrade protocol. User can choose between TFTP, HTTP, and HTTPS.

F IRMWARE U PGRADE M ODE

Select voice menu option 017 to choose firmware upgrade mode among the following three options:

1) always check, 2) check when pre/suffix changes, and 3) never upgrade

CONFIGURING GXW42XX WITH WEB BROWSER

The GXW42XX series gateway has an embedded Web server that allows users to configure the

GXW42XX through a web browser.

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ACCESS THE WEB CONFIGURATION MENU

The GXW42XX HTML configuration menu can be accessed via Ethernet port:

To access the HTML configuration menu from the Ethernet port:

1. Follow table 4 to find the Ethernet port IP address.

2. Open a web browser, type in the IP address – for example: http://GXW42XX -IP-Address (the

GXW42XX IP-Address is the Ethernet IP address for the GXW42XX).

NOTE: x The IVR announces 12 digits IP address, you need to strip out the leading “0” in the IP address. For ex.

IP address: 192.168.001.014, you need to type in http://192.168.1.14 in the web browser.

Once the HTTP request is entered and sent from a web browser, the user will see a log-in screen. There are two default passwords for the login page:

User Level: Password: Web pages allowed:

End User Level 123 Only Status and Basic Settings

Administrator Level admin Browse all pages

The password is case sensitive with maximum length of 25 characters. The factory default password for End

User and administrator is “123” and “admin” respectively. Only an administrator can access the “ADVANCED

SETTING”, “Profile 1~4”, and “FXS PORTs” configuration pages. Please reference the GUI pages using the following link: http://www.grandstream.com/products/ip-voice-telephony/enterprise-analog-gateways/gxw42xx/GUI.zip.

IMPORTANT SETTINGS

The end-user must configure the following settings according to the local environment.

NOTE: Most settings on the web configuration pages are set to the default values.

NAT SETTINGS

If you plan to keep the gateway within a private network behind a firewall, we recommend using STUN

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Server. The following three (3) settings are useful in the STUN Server scenario:

1. STUN Server (under Advanced Settings webpage)

Enter a STUN Server IP (or FQDN) that you may have, or look up a free public STUN Server on the internet and enter it on this field. If using Public IP, keep this field blank.

2. NAT Traversal (under the Profile web pages)

Set this to Yes when gateway is behind firewall on a private network.

DTMF METHODS

DTMF Settings are in Profile pages. x DTMF in-audio x DTMF via RTP (RFC2833) x DTMF via SIP INFO

You can enable set priority of DTMF methods according to your preference, from Priority 1 to 3. This setting should be based on your server DTMF setting.

PREFERRED VOCODER (CODEC)

The GXW42XX supports a broad range of voice codecs. Under Profile web pages, choose your preferred order of different codecs: x PCMU/A (or G711µ/a) x G729 x G723 x G726 (16/24/32/40) x iLBC x AAL2 (all G726)

D EFINITIONS

This section will describe the options in the Web configuration user interface. As mentioned, a user can log in as an administrator or end-user.

Functions available for the end-user are: x STATUS: Displays the network status, account status, software version and MAC-address of the phone

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x BASIC SETTINGs: Basic settings such as basic network, date and time settings, and web/telnet access settings can be set here. x PROFILE - AUDIO SETTINGS: DTMF, Vocoder, and Analog Line settings can be configured here for each port.

Additional functions available to administrators are: x BASIC SETTINGS: Full settings for network, upgrade/provisioning, TR-069, Security and Syslog. x ADVANCED SETTINGS: To set advanced Ring Tongs, FXO Failover, and System Features. x PROFILE X: To configure General Settings, Network Settings, SIP Settings, Fax Settings, Audio

Settings, Call Settings, Ring Tones x FXS PORTS: To configure each of the FXS ports and Hunting Groups etc.

TABLE 7: BASIC SETTINGS

IP Address Mode Chose to configure with DHCP, Static IP or PPPoE

Preferred DNS

Server

DHCP Settings

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

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.

Used by clients and servers to exchange vendor class ID. Vendor Class ID

(Option 60)

Static IP Settings

Configure the IP address, Subnet Mask, Default Router IP address, DNS Server 1 (primary),

DNS Server 2 (secondary) fields. These fields are set to zero by default.

QoS Settings

Configure the Layer 3 QoS, Layer 2 QoS 802.1Q/VLAN Tag, and Layer 2 QoS 802.1p Priority

Value.

STUN Settings

Configure the STUN server parameters, including: Use STUN (Yes/No), STUN server (IP or

Domain), Number of STUN Response Misses Allowed, and Keep-Alive Interval.

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DHCP hostname This option specifies the name of the client. This field is optional but may be required by some

Internet Service Providers. Default is blank.

DHCP domain

DHCP vendor class ID

This option specifies the domain name that client should use when resolving hostnames via the

Domain Name System. Default is blank.

Used by clients and servers to exchange vendor-specific information. Default is GXW42XX.

PPPoE account

ID

PPPoE username. Necessary if ISP requires you to use a PPPoE (Point to Point Protocol over

Ethernet) connection.

PPPoE password PPPoE account password.

PPPoE Service

Name

This field is optional. If your ISP uses a service name for the PPPoE connection, enter the service name here. Default is blank.

Time Zone Controls how the date/time is displayed according to the specified time zone.

Self Defined Time The syntax is std offset dst [offset],start[/time],end[/time]

Zone Default is set to : MTZ+6MDT+5,M3.2.0,M11.1.0

MTZ+6MDT+5,

It indicates a time zone with 6 hours offset with 1 hour ahead which is the US central time.

It is positive (+) if the local time zone is west of the Prime Meridian and negative

(-) if it is east.

Prime Meridian (International or Greenwich Meridian)

M3.2.0,M11.1.0

The 1st number indicates Month: 1,2,3,..,12 (for Jan, Feb, .., Dec)

The 2nd number indicates the nth iteration of the weekday: (1st Sunday, 3rd

Tuesday etc)

The 3rd number indicates Weekday: 0,1, 2, ..,6(for Sun, Mon, Tue, .., Sat)

Therefore, this example is the DST which starts from the second Sunday of

March to the 1st Sunday of November.

Language Default is English. Languages supported with voice prompt.

In addition to the Basic Settings configuration page, end users also have access to the Device Status page.

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MAC Address

TABLE 8: STATUS PAGE

The device ID in hexadecimal format. This is needed for Internet Service Provider troubleshooting. The MAC address will be used for provisioning and can be found on the label on original box and on the label located on the bottom panel of the device.

Shows WAN IP address of GXW42XX WAN IP

Address

Product Model

Software

Version

System Up

Time

Contains the product model info.

Program: This is the main software release. Boot and Loader are not changed often.

Shows system uptime since the last reboot.

PPPoE Link Up

NAT

Shows whether the PPPoE connection is running if connected to DSL modem.

Shows type of NAT the GXW42XX is connected to via its WAN port. It is based on STUN protocol.

Port Status Displays relevant information regarding the individual FXS ports. Example:

Port Hook Registration DND Forward Busy

Forward

Delayed

Forward

FXS1

FXS2

FXS3

FXS4

FXS…

On Hook

Off Hook

Registered

Registered

On Hook Registered

… …

No

No

On Hook Not Registered No

Yes

613

614

615

Advanced User configuration includes not only the end user configuration, but also advanced configurations such as: SIP configuration, Codec selection, NAT Traversal Setting and other miscellaneous configuration.

TABLE 9: ADVANCED SETTINGS

Admin Password Administrator password. Only the administrator can configure the “Advanced Settings” page.

Password field is purposely left blank for security reasons after clicking update and saved. The maximum password length is 25 characters.

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Layer 3 QoS This field defines the layer 3 QoS parameter which can be the value used for IP Precedence or

Diff-Serv or MPLS. Default value is 12.

Value used for layer 2 VLAN tag. Default setting is blank. Layer 2 QoS

STUN Server

Keep-alive interval

IP address or Domain name of the STUN server.

STUN server assists devices behind a NAT firewall or a router to set up calls via VoIP.

STUN server is often provided by the VoIP service provider or a public STUN server can be used. For a sample list of public free STUN servers please refer to: http://www.voipinfo.org/wiki-STUN

This parameter specifies the number of seconds after which a new blank UDP packet will be sent out to the proxy/server port in order to have the port stay open and the device remain reachable. Default is 20 seconds.

Use STUN to detect network connectivity

Firmware

Upgrade &

Provisioning

Use STUN keep-alive to detect WAN side network problems. If keep-alive request does not yield any response for configured number of times, the device will restart the TCP/IP stack. If the STUN server does not respond when the device boots up, the feature is disabled.

Enables GXW42XX to download firmware or configuration file using the TFTP, HTTP or HTTPS protocols.

Via TFTP Server This is the IP address of the configured TFTP server. If selected and it is non-zero or not blank, the GXW42XX retrieves the new configuration file or new code image from the specified TFTP server at boot time. After 5 attempts, the system will timeout and then it will start the boot process using the existing code image in the Flash memory. If a TFTP server is configured and a new code image is retrieved, the new downloaded image is saved into the Flash memory.

Note: Do NOT interrupt the firmware upgrade process (especially the power supply) as this will damage the device . Depending on the network environment this process may take up to 15 or 20 minutes.

Via HTTP or

HTTPS Server

The URL for the HTTP server used for firmware upgrade and configuration via HTTP.

Note: If Auto Upgrade is set to No, GXW42XX will only do HTTP download once at boot up.

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

Config Server

Path

IP address or domain name of 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

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XML Config File

Password

The password used for encrypting the XML configuration file using OpenSSL.

This is required for the phone to decrypt the encrypted XML configuration file.

HTTP/HTTPS

User Name

HTTP/HTTPS

Password

The user name needed to authenticate withthe HTTP/HTTPS server.

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

Firmware File

Prefix

Firmware File

Postfix

Default is blank. If configured, GXW42XX will request firmware file with the prefix. This setting is useful for Internet Telephone Service Providers. End users should keep it blank.

Default is blank. This setting is useful for Internet Telephone Service Providers. End users should keep it blank.

Config File Prefix Default is blank. This setting is useful for Internet Telephone Service Providers. End users should keep it blank.

Config File

Postfix

Default is blank. This setting is useful for Internet Telephone Service Providers. End users should keep it blank.

Allow DHCP

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.

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.

There are three options to choose from: “Always check for New Firmware at Boot up”, “Check

New Firmware only when F/W pre/suffix changes”, and “Always Skip the Firmware Check”.

Authenticate

Conf File

Firmware Key

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

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

SSL Certificate The GXW42XX series supports SIP over TLS. It has built-in private key and SSL certificate. The user specified SSL certificate used for SIP over TLS is in X.509 format.

SSL Private Key You may also customize the SSL Private Key. The user specified SSL private key used for SIP over TLS is in X.509 format.

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SSL Private Key

Password

Enter SSL Private Key password here.

ACS URL

ACS Username

User specify the Auto Configuration Server’s URL (TR-069 protocol)

User specify the ACS Username

ACS Password User specify the ACS password

Periodic Inform

Enable

Default is No. If set to YES, device will send inform packets to the ACS

Periodic Inform

Interval

Frequency that the inform packets will be sent out to the ACS

Connection

Request

Username

Set a user name for the ACS to connect to this device

Set a password for the ACS to connect to this device Connection

Request

Password

System Ring

Cadence

Call Progress

Tones

Configuration option for all FXS ports ring cadence for all incoming calls.

(Syntax: c=on1/off1-on2/off2-on3/off3; [...]) Default is set to c=2000/4000; (US standards)

Using these settings, 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. ONis 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.

• “Dial tone”

• “Ringback tone”

• “Busy/Re-order tone”

• “Confirmation tone”

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

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

Update

Disable Voice

Prompt

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

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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.

Failover to FXO

Gateway

Enable or disable the Failover FXO Gateway

FXO Gateway IP Address or URI of the FXO gateway.

NTP server

Syslog 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

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 DEBUG,

INFO, WARNING or ERROR. 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

Primary RADIUS

Server

IP Address or FQDN of the primary RADIUS Server

Primary RADIUS

Auth Port

Primary Radius server authentication port. Default value is 1812

Primary RADIUS

Acct Port

Primary Radius server accounting port. Default value is 1813

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Primary Radius

Server Secret

Special secret string should be preconfigured according to RADIUS Server configuration

Secondary

RADIUS Server

IP Address or FQDN of the secondary RADIUS Server

Secondary

RADIUS Auth

Port

Secondary Radius server authentication port. Default value is 1812

Secondary

RADIUS Acct Port

Secondary Radius server accounting port. Default value is 1813

Secondary

Radius Server

Secret

Special secret string should be preconfigured according to RADIUS Server configuration

RADIUS Timeout Default value is 2 seconds. The time between retries the GXW will send “Access-Request” message to RADIUS server in purpose to authenticate it.

RADIUS Retry Default value is 3 times. Number of times the device will try to authenticate itself with preconfigured RADIUS server during initialization process.

Download Device

Configuration

This setting allows user to download a text file containing all the P values of each setting as configured on the unit. (Note: For Security Reasons, any Password will not be Downloaded)

TABLE 10: PROFILE SETTINGS

When set to Yes the SIP Profile is activated. Profile Active

Primary SIP

Server

Failover SIP

Server

Prefer Primary

SIP Server

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

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.

Default is No. If set to yes it will register to Primary Server if registration with Failover server expires

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.

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SIP transport

NAT Traversal

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.

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

If activated (by choosing “Yes”) and a STUN server is also specified, then the GXW42XX performs according to the STUN client specification. Under this mode, the embedded STUN client will detect if and what type of firewall/NAT is being used. If the detected NAT is a Full

Cone, Restricted Cone, or a Port-Restricted Cone, the GXW42XX will use its mapped public

IP address and port in all of its SIP and SDP messages.

If the NAT Traversal field is set to “Yes” with no specified STUN server, the GXW42XX will periodically (every 20 seconds) send a blank UDP packet (with no payload data) to the SIP server to keep the “hole” on the NAT open.

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)

One mode can be chosen for the client to look up server.

The default value is “A Record”

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.

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.

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Outgoing Call without

Registration

Register

Expiration

Default is No. 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.

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

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 Allows the user to specify the time frequency (in seconds) for the GXW42XX to re-register

Failure Retry Wait after registration failure. The default interval is 20 seconds. The maximum interval is 3600

Time seconds (1 hour).

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 port. This is usually necessary when multiple Gateway or ATA are behind the same NAT.

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

Refer to Use

Target Contact

Transfer on

Conference

Hang up

Default is No. In which case if conference originator hangs up the conference will be terminated. When option YES is chosen, originator will transfer other parties to each other so that B and C can choose either to continue the conversation or hang up.

Disable Bellcore

Style 3-Way

Conference

Default is No. you can make a Conference by pressing ‘Flash’ key. If set to Yes, you need to dial *23 + second callee number.

Remove OBP from Route

Header:

Support SIP

Instance ID

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

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.

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Validate incoming Default is No. If set to yes all incoming SIP messages will be strictly validated according to

SIP message RFC rules. If message does not pass validation process, call will be rejected.

Check SIP user ID Default is No. Check the SIP User ID in Request URI. If they don’t match, the call will for incoming

INVITE be rejected. (no direct IP calling if Yes)

Allow Incoming

SIP Messages from SIP Proxy

Only

Default is No. If incoming SIP message does not match with SIP Server, it will be rejected.

(no direct IP calling if Yes)

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 larger value for more reliable usage.

SIP T2 Interval

DTMF Payload

Type

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

Sets the payload type for DTMF using RFC2833.

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.

Disable DTMF

Negotiation

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

Send Hook Flash

Event

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

Enable Call

Features

Default is Yes. (If Yes, call features using star codes will be supported locally)

Proxy Require

Use NAT IP

SIP Extension to notify SIP server that the unit is behind the NAT/Firewall.

NAT IP address used in SIP/SDP message. Default is blank.

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Distinctive

Ringtone

Custom Ring Tone 1 to 3 with associate Caller ID: when selected, if Caller ID is configured, then the device will ONLY uses 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.

Distinctive ring tones can be configured not only for matching whole number, but also for matching prefixes. In this case symbol * (star) will be used.

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.

For example:

If configured as *617, Ring Tone 1 will be used in case of call arrived from Massachusetts.

Any other incoming call will ring using cadence defined in parameter System Ring Cadence located under Advanced Settings Configuration page.

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 Receiver

Offhook Tone

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

Disable Reminder Default is No. This is to disable the Reminder Ring that is played when a call is waiting on

Ring for On-Hold hold and the analog phone goes on-hook.

Call

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 This is the type of signal sent to the analog phone to make it turn the lamp ON upon receiving a Voice mail. Check the phone’s manual to find out what signal is supported, FSK (default) or

Neon.

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

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.

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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.

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.

No Key Entry

Timeout

Early Dial

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

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.

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.

Dial Plan Dial Plan Rules:

1. Accept 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; a. xx+ - at least 2 digits number; b. xx. – at least 2 digits number; c. ^ - exclude; d. [3-5] - any digit of 3, 4, or 5; e. [147] - any digit 1, 4, or 7; f. <2=011> - replace digit 2 with 011 when dialing

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g. < =1> - add a leading 1 to all numbers dialed, vice versa will remove a 1 from the number dialed h. | - or

Example 1: {[369]11 | 1617xxxxxxx} –

Allow 311, 611, 911, and any 10 digit numbers of leading digits 1617

Example 2: {^1900x+ | <=1617>xxxxxxx} –

Block any number of leading digits 1900 and add prefix 1617 for any dialed 7 digit numbers

Example 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+} 3.

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 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 - allow dialing special and emergency numbers 311, 411, 611 and 911

Note: 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*]+ }.

Subscribe for

MWI

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

Send Anonymous If this parameter is set to “Yes”, the “From” header along with Privacy and

P_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.

Special Feature Default is Standard. Choose the selection to meet some special requirements from

Softswitch vendors. Example of vendors - CBCOM, RNK.

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Session

Expiration

Grandstream implemented SIP Session Timer. 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

Caller Request

Timer

Callee Request

Timer

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

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

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

Force Timer

UAC Specify

Refresher

UAS Specify

Refresher

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.

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

Send Re-INVITE

After Fax

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

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.

Enable 100rel Enables the use of PRACK (Provisional Acknowledgment) method.

Use First Default is No. If set to “Yes”, device will include only the first match vocoder in its 200OK

Matching Vocoder response, otherwise it will include all match vocoders in same order received in INVITE. in 200OK SDP

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Preferred

Vocoder

The GXW42XX supports up to 5 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 option

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.

Default value is 100. Range is from 96 to 127. AAL2-G726-16

Payload type

AAL2-G726-24

Payload type

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

AAL2-G726-32

Payload type

AAL2-G726-40

Payload type

VAD

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

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

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

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.

Jitter Buffer Type Select either Fixed or Adaptive based on network conditions.

Jitter Buffer

Length

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

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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

SDES: http://www.apps.ietf.org/rfc/rfc4568.html

SRTP: http://www.apps.ietf.org/rfc/rfc3711.html

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.

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.

Loop Current

Disconnect duration

Default is 200. In 100 - 1 0000 milliseconds range.

Hook Flash timing

Time period when the cradle is pressed (Hook Flash) to simulate FLASH. To prevent unwanted activation of the Flash/Hold and automatic phone ring-back, adjust this time value.

On Hook Timing On-hook timing is the minimum time for an on-hook event to be validated.

Gain Handset volume adjustment. RX is for receiving volume (direction FXS Æ to analog phone),

TX is for transmission volume (Analog phone Æ to FXS). Default values are 0dB for both parameters. Loudest volume: +6dB Lowest volume: -6dB.

Disable Line

Echo Canceller

(LEC)

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

Configure ring cadences according to preference. Ring Tones

FXS Port

SIP User ID

TABLE 11: FXS PORTS SETTINGS

FXS Port Number

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

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

Hunting Group Assign this port to a certain hunting group

SAVING THE CONFIGURATION CHANGES

After user makes a change to the configuration, press the “Update” button in the Configuration Menu. The web browser will then display a message window to confirm saved changes.

Grandstream recommends reboot or power cycle the gateway after saving changes.

REBOOTING FROM REMOTE

Press the “Reboot” button at the bottom of the configuration menu to reboot the phone remotely. The web browser will then display a message window to confirm that reboot is underway. Wait 30 seconds to log in again.

CONFIGURATION THROUGH A CENTRAL SERVER

Grandstream GXW42xx can be automatically configured from a central provisioning system.

When GXW42xx boots up, it will send TFTP or HTTP/HTTPS requests to download configuration files,

“cfg000b82xxxxxx” and “cfg00082xxxxxx.xml”, where “000b82xxxxxx” is the LAN MAC address of the

GXW42xx. If the download of “cfgxxxxxxxxxxxx.xml” is not successful, the provision program will issue request a generic configuration file “cfg.xml”. Configuration file name should be in lower case letters.

The configuration data can be downloaded via TFTP or HTTP/HTTPS from the central server. A service provider or an enterprise with large deployment of GXW42xx can easily manage the configuration and service provisioning of individual devices remotely from a central server.

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Grandstream provides a central provisioning system GAPS (Grandstream Automated Provisioning

System) to support automated configuration of Grandstream devices. GAPS uses enhanced (NAT friendly) TFTP or HTTP (thus no NAT issues) and other communication protocols to communicate with each individual Grandstream device for firmware upgrade, remote reboot, etc.

Grandstream provides GAPS service to VoIP service providers. Use GAPS for either simple redirection or with certain special provisioning settings. At boot-up, Grandstream devices by default point to

Grandstream provisioning server GAPS, based on the unique MAC address of each device, GAPS provision the devices with redirection settings so that they will be redirected to customer’s TFTP or

HTTP/HTTPS server for further provisioning.

Grandstream also provides configuration tools (Windows and Linux/Unix version) to facilitate the task of generating device configuration files. The Grandstream configuration tools are free to end users. The

. configuration tools and configuration templates are available for download from http://www.grandstream.com/support/tools .

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SOFTWARE UPGRADE

Software upgrade can be done via either TFTP or HTTP/HTTPS. The corresponding configuration settings are in the ADVANCED SETTINGS configuration page.

FIRMWARE UPGRADE THROUGH TFTP/HTTP/HTTPS

To upgrade via TFTP or HTTP/HTTPS, the “Firmware Upgrade and Provisioning upgrade via” field needs to be set to TFTP HTTP or HTTPS, respectively. “Firmware Server Path” needs to be set to a valid URL of a TFTP or HTTP server, server name can be in either FQDN or IP address format. Here are examples of some valid URL.

e.g. firmware.mycompany.com:6688/Grandstream/1.0.7.6

e.g. firmware.grandstream.com

NOTES: y Firmware upgrade server in IP address format can be configured via IVR. Please refer to the

CONFIGURATION GUIDE section for instructions. If the server is in FQDN format, it must be set via the web configuration interface. y Grandstream recommends end-user use the Grandstream HTTP server. Its address can be found at http://www.grandstream.com/support/firmware . Currently the HTTP firmware server URL is firmware.grandstream.com. For large companies, we recommend to maintain their own TFTP/

HTTP/HTTPS server for upgrade and provisioning procedures. y Once a “Firmware Server Path” is set, user needs to update the settings and reboot the device. If the configured firmware server is found and a new code image is available, the GXW will attempt to retrieve the new image files by downloading them into the GXW420x ’s SRAM. During this stage, the GXW’s LEDs will blink until the checking/downloading process is completed. Upon verification of checksum, the new code image will then be saved into the Flash. If

TFTP/HTTP/HTTPS fails for any reason (e.g. TFTP/HTTP/HTTPS server is not responding, there are no code image files available for upgrade, or checksum test fails, etc), the GXW will stop the

TFTP/HTTP/HTTPS process and simply boot using the existing code image in the flash. y Firmware upgrade may take as long as 15 to 30 minutes over Internet, or just 5 minutes if it is performed on a LAN. It is recommended to conduct firmware upgrade in a controlled LAN environment if possible. For users who do not have a local firmware upgrade server,

Grandstream provides a NAT-friendly TFTP server on the public Internet for firmware upgrade.

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y Grandstream’s latest firmware is available http://www.grandstream.com/support/firmware .

Oversea users are strongly recommended to download the binary files and upgrade firmware locally in a controlled LAN environment. y Alternatively, user can download a free TFTP or HTTP server and conduct local firmware upgrade.

A free windows version TFTP server is available for download from http://support.solarwinds.net/updates/New-customerFree.cfm

. Our latest official release can be downloaded from http://www.grandstream.com/y-firmware.htm.

INSTRUCTIONS FOR LOCAL FIRMWARE UPGRADE:

1. Unzip the file and put all of them under the root directory of the TFTP server.

2. Put the PC running the TFTP server and the GXW42XX device in the same LAN segment.

3. TFTP server's security settings should be changed from "Receive Only" to "Transmit Only" for the firmware upgrade.

4. Configure the Firmware Server Path with the IP address of the PC

5. Update the change and reboot the unit.

CONFIGURATION FILE DOWNLOAD

Grandstream SIP Device can be configured via Web Interface as well as via Configuration File through

TFTP or HTTP/HTTPS. “Config Server Path” is the TFTP or HTTP/HTTPS server path for configuration file. It needs to be set to a valid URL, either in FQDN or IP address format. The “Config Server Path” can be same or different from the “Firmware Server Path”.

A configuration parameter is associated with each particular field in the web configuration page. A parameter consists of a Capital letter P and 1 to 3 (Could be extended to 4 in the future) digit numeric numbers. i.e., P2 is associated with “Admin Password” in the ADVANCED SETTINGS page. For a detailed parameter list, please refer to the corresponding firmware release configuration template.

When a Grandstream device boots up or reboots, it will issue a request for a configuration file

“cfgxxxxxxxxxxxx”, where “xxxxxxxxxxxx” is the MAC address of the device, i.e., “cfg000b820102ab”. In addition, device will also requests a XML configuration file “cfgxxxxxxxxxxxx.xml”. If the download of

“cfgxxxxxxxxxxxx.xml” is not successful, the provision program will issue a request for a generic configuration file “cfg.xml”. Configuration file name should be in lower case letters.

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RESTORE FACTORY DEFAULT SETTING

FACTORY RESET

WARNING! Restoring the Factory Default Setting will DELETE all configuration information of the phone.

Please BACKUP or PRINT out all the settings before you approach to following steps. Grandstream will not take any responsibility if you lose all the parameters of setting and cannot connect to your VoIP service provider.

There are two (2) methods for resetting your unit:

RESET BUTTON

Reset default factory settings following these four (4) steps:

1. Unplug the Ethernet cable.

2. Locate a needle-sized hole on the back panel of the gateway unit next to the power connection.

3. Insert a pin in this hole, and press for more than 4 seconds.

4. Take out the pin. All unit settings are restored to factory settings.

IVR COMMAND

Reset default factory settings using the IVR Prompt (Table 5):

1. Dial “***” for voice prompt.

2. Enter “099” and wait for “reset” voice prompt.

3. Enter the encoded MAC address (Look below on how to encode MAC address).

4. Wait 15 seconds and device will automatically reboot and restore factory settings.

Encode the MAC Address

1. Locate the MAC address of the device. It is the 12 digit HEX number on the bottom of the unit.

2. Key in the MAC address. Use the following mapping:

0-9: 0-9

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A: 22 (press the “2” key twice, “A” will show on the LCD)

B: 222

C: 2222

D: 33 (press the “3” key twice, “D” will show on the LCD)

E: 333

F: 3333

For example: if the MAC address is 000b8200e395, it should be keyed in as “0002228200333395”.

NOTE:

1. Factory Reset will be disabled if the “Lock keypad update” is set to “Yes”.

2. If still have difficulties to get access to the device, pushing the reset button during booting up will trigger the RECOVERY MODE, and device will use static IP 192.168.1.234/24. You may need a direct connection to access the configuration page.

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