Grandstream Networks GXW4216 VOIP GATEWAY User Manual GXW42xx User Manual Draft

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Grandstream Networks, Inc.
Analog IP Gateway GXW4200 Series
16, 24 or 32 FXS Ports
GXW42XX USER MANUAL
GXW42XX USER MANUAL
INDEX
GNU GPL INFORMATION...........................................................................5
CHANGE LOG................................. ERROR! BOOKMARK NOT DEFINED.
CHANGES FROM 1.0.5.10 USER MANUAL .......................... ERROR! BOOKMARK NOT DEFINED.
WELCOME ..................................................................................................6
GATEWAY GXW42XX OVERVIEW....................................................................................................... 6
SAFETY COMPLIANCE ........................................................................................................................ 6
WARRANTY........................................................................................................................................... 6
CONNECT YOUR GXW42XX GATEWAY ...................................................8
EQUIPMENT PACKAGING ................................................................................................................... 8
CONNECT THE GXW42XX................................................................................................................... 8
GXW42XX FEATURES.............................................................................. 12
SOFTWARE FEATURES OVERVIEW ................................................................................................ 12
HARDWARE SPECIFICATION............................................................................................................ 14
BASIC OPERATIONS ............................................................................... 15
UNDERSTANDING GXW VOICE PROMPTS ..................................................................................... 15
PLACING A PHONE CALL .................................................................................................................. 16
PHONE OR EXTENSION NUMBERS.......................................................................................... 16
DIRECT IP CALLS........................................................................................................................ 17
CALL HOLD ......................................................................................................................................... 18
CALL WAITING .................................................................................................................................... 18
CALL TRANSFER................................................................................................................................ 18
BLIND TRANSFER....................................................................................................................... 18
ATTENDED TRANSFER .............................................................................................................. 19
3-WAY CONFERENCING.................................................................................................................... 19
HUNTING GROUP .............................................................................................................................. 20
INTER-PORT CALLING....................................................................................................................... 22
PSTN PASS THROUGH/LIFE LINE ........................................ ERROR! BOOKMARK NOT DEFINED.
SENDING AND RECEIVING FAX ....................................................................................................... 22
SUPPORT RADIUS PROTOCOL........................................................................................................22
CALL FEATURES ..................................................................................... 23
CONFIGURATION GUIDE......................................................................... 25
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CONFIGURING GXW42XX VIA VOICE PROMPT ............................................................................. 25
CONFIGURING GXW42XX WITH WEB BROWSER.......................................................................... 25
ACCESS THE WEB CONFIGURATION MENU........................................................................... 26
IMPORTANT SETTINGS ..................................................................................................................... 26
NAT SETTINGS............................................................................................................................ 26
DTMF METHODS......................................................................................................................... 27
PREFERRED VOCODER (CODEC)............................................................................................ 27
SAVING THE CONFIGURATION CHANGES...................................................................................... 46
REBOOTING FROM REMOTE ........................................................................................................... 46
CONFIGURATION THROUGH A CENTRAL SERVER ....................................................................... 46
SOFTWARE UPGRADE............................................................................ 48
CONFIGURATION FILE DOWNLOAD ................................................................................................ 49
FIRMWARE AND CONFIGURATION FILE PREFIX AND POSTFIX .................................................. 49
MANAGING FIRMWARE AND CONFIGURATION FILE DOWNLOAD .............................................. 50
RESTORE FACTORY DEFAULT SETTING .............................................. 51
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TABLE OF FIGURES
GXW42XX User Manual
FIGURE 1: DIAGRAM OF GXW4204-8 AND GXW4224 BACK PANEL ..................................................... 9
FIGURE 2: 50 PIN TELCO CONNECTOR ................................................................................................. 10
FIGURE 3: DIAGRAM OF GXW4204-8 AND GXW4224 DISPLAY PANEL .... ERROR! BOOKMARK NOT
DEFINED.
TABLE OF TABLES
GXW42XX User Manual
TABLE 1: DEFINITIONS OF THE GXW CONNECTORS.......................................................................... 10
TABLE 2: DEFINITIONS OF THE GXW DISPLAY PANEL........................................................................ 11
TABLE 3: GXW42XX SOFTWARE FEATURES ........................................................................................ 12
TABLE 4: HARDWARE SPECIFICATION OF GXW42XX SERIES GATEWAYS ...................................... 14
TABLE 5: DEFINITIONS OF THE GXW VOICE PROMPTS ..................................................................... 15
TABLE 6: CALL FEATURES TABLE (STAR CODE).................................................................................. 23
TABLE 7: BASIC SETTINGS ..................................................................................................................... 28
TABLE 8: STATUS PAGE........................................................................................................................... 29
TABLE 9: ADVANCED SETTINGS ............................................................................................................ 30
TABLE 10: FXS PORTS SETTINGS.......................................................................................................... 35
TABLE 11: PROFILE SETTINGS............................................................................................................... 37
CONFIGURATION GUI INTERFACE EXAMPLES
GXW42XX User Manual
http://www.grandstream.com/products/gxw_series/GXW42XX/documents/GXW42XX_gui.zip
1. SCREENSHOT OF ADVANCED SETTINGS CONFIGURATION PAGE
2. SCREENSHOT OF BASIC SETTINGS CONFIGURATION PAGE
3. SCREENSHOT OF FXS PORTS CONFIGURATION LOGIN PAGE
4. SCREENSHOT OF PROFILE 1 CONFIGURATION PAGE
5. SCREENSHOT OF PROFILE 2 CONFIGURATION PAGE
6. SCREENSHOT OF STATUS PAGE
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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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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 (no fans) 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 10M/100Mbps
network ports, and 50pins RJ21 analog port (1 for 4216 and 4224, 2 for 4236). 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.729A/B.
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.
•
This document is contains links to Grandstream GUI Interfaces. Please download these examples
http://www.grandstream.com/products/gxw_series/GXW42XX/documents/GXW42XX_gui.zip for your
reference.
•
This document is subject to change without notice. The latest electronic version of this user manual
is located at
http://www.grandstream.com/products/gxw_series/GXW42XX/documents/GXW42XX_usermanual_en
glish.pdf
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:
•
one device unit
•
one RJ45 Ethernet cable
•
one 12V 5A universal power adapter
•
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/gxw_series/GXW42XX/documents/GXW42XX_gui.zip.
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FIGURE 1: DIAGRAM OF GXW42XX FRONT & BACK PANEL
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Pin
Signal
Pin
Signal
RING1
26
TIP1
RING2
27
TIP2
RING3
28
TIP3
RING4
29
TIP4
RING5
30
TIP5
RING6
31
TIP6
RING7
32
TIP7
RING8
33
TIP8
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
25
RING25
50
TIP25
FIGURE 2: 50 PIN TELCO CONNECTOR
TABLE 1: DEFINITIONS OF THE GXW CONNECTORS
Ethernet Port
Connect to the internal LAN network, router or PC.
RESET
Factory Reset button. Press for 7 seconds to reset factory default settings.
DC 12V
Power adapter connection
Analog Ports
Connect to analog phones / fax machines with an RJ21 to RJ11 cable
FXS Ports
FXS port to be connected to analog phones / fax machines.
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Note: Once the GXW42XX is turned on and configured, the front display panel indicates the status of the
unit.
TABLE 2: DEFINITIONS OF THE GXW DISPLAY PANEL
Power LED
Indicates Power. Remains ON when Power is connected and turned ON.
Act LED
Remains ON after boot-up.
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.
NOTE:
All LED lights display green when ON.
Slow blinking of READY, WAN and LAN LED together indicates a firmware upgrade or
provisioning state.
LED POWER, READY and WAN lights are ON when device is up and running and successfully
registered to the SIP Server.
Slow blinking of READY LED indicates that device has not registered with any SIP Service
provider.
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GXW42XX FEATURES
The GXW4000 series is a next generation IP voice gateway that is interoperable and compatible with
leading IP-PBXs, Softswitches and SIP platforms. The GXW4200 series FXS gateway is autoconfigurable, remotely manageable and scalable. The GXW4200 series gateways come in three models
- the GXW4216, GXW4224 and GXW4036, 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
•
16, 24 or 32 FXS ports
•
one RJ-45 Ethernet port
•
4 configurable SIP profiles)
•
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
•
Comprehensive Dial Plan support for Outgoing calls.
•
G.168 Echo Cancellation
•
Voice Activation Detection (VAD), Comfort Noise Generation (CNG), and Packet Loss
Concealment (PLC)
•
Supports PSTN/PBX analog telephone sets or analog trunks
TABLE 3: GXW42XX SOFTWARE FEATURES
GXW4216
GXW4224
GXW4232
Telephone Interfaces
16 FXS ports
24 FXS ports
32 FXS ports
SIP Provisioning
16 SIP accounts, 4 profiles
24 SIP accounts, 4 profiles
32 SIP accounts, 4 profiles
Network Interface
10M/100 Mbps, RJ-45
10M/100 Mbps, RJ-45
10M/100 Mbps, RJ-45
4 Concurrent Calls
8 Concurrent Calls
24 Concurrent Calls,
Number
Concurrent
of
Calls
(except when using
SRTP)
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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 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.729A/B, 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
DHCP Server/Client
DHCP Client only
Fax over IP
T.38 compliant Group 3 Fax Relay up to 14.4kpbs and auto-switch to G.711 for Fax Pass-
DHCP Client only
DHCP Client only
through, Fax Datapump V.17, V.21, V.27ter, V.29 for T.38 fax relay
QoS
Diffserv, TOS, 802.1 P/Q
Diffserv, TOS, 802.1 P/Q
Diffserv, TOS, 802.1 P/Q
VLAN tagging
VLAN tagging
VLAN tagging
Transport Protocol
RTP/RTCP
RTP/RTCP
RTP/RTCP
DTMF Method
In-audio, RFC2833, and/or
In-audio, RFC2833, and/or
In-Audio, RFC2833, and/or
SIP Info
SIP Info
SIP Info
IP Signaling
SIP (RFC 3261)
SIP (RFC 3261)
SIP (RFC 3261)
Provisioning
TFTP, HTTP, HTTPS
TFTP, HTTP, HTTPS
TFTP, HTTP, HTTPS
Security
SRTP, IPSEC, TLS/SIPS,
SRTP, IPSEC, TLS/SIPS,
SRTP, IPSEC, TLS/SIPS,
HTTPS, 802.1x
HTTPS, 802.1x
HTTPS, 802.1x
Syslog support, HTTP and
Syslog support, HTTP and
Syslog support, HTTP and
Telnet access
Telnet access
Telnet access
Dial Plan
Yes
Yes
Yes
3-Way Conference
3-Way conference with local
3-Way conference with local
3-Way conference with local
mixing
mixing
mixing
Bellcore Type 1 & 2, ETSI,
Bellcore Type 1 & 2, ETSI,
Bellcore Type 1 & 2, ETSI,
BT, NTT, and DTMF-based
BT, NTT, and DTMF-based
BT, NTT, and DTMF-based
CID
CID
CID
Yes
Yes
Yes
Management
Caller ID
Polarity Reversal /
Wink
EMC
Network
Connectivity
EN55022/EN55024
and
EN55022/EN55024
and
EN55022/EN55024
and
FCC part15 Class B
FCC part15 Class B
FCC part15 Class B
IPv4,
IPv6,
TCP/UDP,
RTP/RTCP, HTTP/HTTPS,
ARP/RARP, ICMP, DNS,
DHCP,
NTP,
TFTP,
TELNET, PPPoE, STUN
IPv4,
IPv6,
TCP/UDP,
RTP/RTCP, HTTP/HTTPS,
ARP/RARP, ICMP, DNS,
DHCP,
NTP,
TFTP,
TELNET, PPPoE, STUN
IPv4,
IPv6,
TCP/UDP,
RTP/RTCP, HTTP/HTTPS,
ARP/RARP, ICMP, DNS,
DHCP,
NTP,
TFTP,
TELNET, PPPoE, STUN
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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
FXS LEDs
16
24
32
Telco connector
One (1) RJ21 (50 pins)
One (1) RJ21 (50 pins)
Two (2) RJ21 (50 pins)
Network interface
Single 10/100/1000 BASE-TX,RJ45
Universal
Input: 100-240VAC, 50/60Hz
Switching
Output: 12V DC, 5.0A
Telephone
Interface
Power Adaptor
Short Haul Loop
2REN: Up to 1km on 24
2REN: Up to 1km on
AWG wire
AWG wire
AWG wire
Mounting
Wall mount, Desktop
Wall mount, Desktop
Rack, Wall mount
Environmental
Operation: 0°C to 45°C
Storage:
24
2REN: Up to 1km on 24
-20°C to 80°C
Humidity: 10％ to 90% Non-condensing
Safety
UL
UL
UL
Compliance
FCC, CE, C-Tick
FCC, CE, C-Tick
FCC, CE, C-Tick
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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.
TABLE 5: DEFINITIONS OF THE GXW VOICE PROMPTS
Menu
Main Menu
Voice Prompt
User’s Options
“Enter a Menu Option”
Enter “*” for the next menu option
Enter “#” to return to the main menu
Enter 01 – 05, 07,10 - 17, 47, 86 or 99 Menu option
01
“DHCP Mode”, “PPPoE
Enter ‘9’ to toggle the selection
Mode ”or “Static IP Mode”
If user selects “Static IP Mode”, user need configure all the IP
address information through menu 02 to 05. If user selects
“Dynamic IP Mode”, the device will retrieve all IP address
information from DHCP server automatically when user reboots
the device.
02
“IP Address “ + IP address
The current WAN IP address is announced
Enter 12-digit new IP address if in Static IP Mode.
03
“Subnet “ + IP address
Same as Menu option 02
04
“Gateway “ + IP address
Same as Menu option 02
05
“DNS
Server
“
IP
Same as Menu option 02
address
07
Preferred Vocoder
Enter “9” to go to the next selection in the list:
•
PCM U
•
PCM A
•
iLBC
•
G-726
•
G-723
•
G-729
10
“MAC Address”
Announces the Mac address of the unit.
12
WAN Port Web Access
Enter “9” to toggle between enabled and disabled
13
Firmware
14
Server
IP
Announces current Firmware Server IP address. Enter 12 digit
Address
new IP address.
Configuration Server IP
Announces current Config Server Path IP address. Enter 12 digit
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15
Address
new IP address.
Upgrade Protocol
Upgrade protocol for firmware and configuration update. Enter
“9” to toggle between TFTP and HTTP
16
Firmware Version
Firmware version information.
17
Firmware Upgrade
Firmware upgrade mode. Enter “9” to rotate among the following
three options:
47
“Direct IP Calling”
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”.)
700-
between
User can make internal calls between different ports inside the
different ports of the same
same GW even without being registered to SIP server. By dialing
GW42xx
*** and 7xx, user can reach colleague connected to port “xx”.
86
Voice Mail
Number of voice mails
99
“RESET”
Enter “9” to reboot the device; or
716/724/732
Phone
calls
Enter MAC address to restore factory default setting
(See Restore Factory Default Setting section)
“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 can not 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
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2. Dial the number directly and press # (Use # as dial key” must be configured in web configuration).
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 “47” 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 47, 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.
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b) If the target IP address/port is 192.168.1.20:5062, then the dialing convention would be:
*47 or Voice Prompt with option 47, then 192*168*0*160*5062 followed by pressing the “#” key
if it is configured as a send key or wait for 4 seconds.
NOTE: When completing direct IP call, the “Use Random Port” should set to “NO”. You can not make
direct IP calls between FXS ports since they are using same IP.
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
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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.
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.
3­WAY 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 for all three parties when configuration
“Transfer on Conference Hangup” is set to “No”. If the configuration is set to “Yes”, A will
transfer B to C so that B and C can continue the conversation.
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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.
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 4008 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"
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FXS Port #8: SIP UserID and Authenticate ID left blank, Hunting Group set to "7"
…
FXS Port #24: SIP UserID and Authenticate ID left blank, Hunting Group set to "2"
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
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.
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INTER­PORT 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 stand alone 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.
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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CALL FEATURES
GXW42XX supports the traditional telephony features available in a PBX as well as additional advanced
telephony features.
TABLE 6: CALL FEATURES TABLE (STAR CODE)
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 (G726r40), *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.
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*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.
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.
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CONFIGURATION GUIDE
CONFIGURING GXW42XX VIA VOICE PROMPT
DHCP MODE
Select voice menu option 01 to enable GXW42XX to use DHCP.
STATIC IP MODE
Select voice menu option 01 to enable GXW42XX to use STATIC IP mode, then use option 02, 03, 04, 05
to set up IP address, Subnet Mask, Gateway and DNS server respectively.
FIRMWARE SERVER IP ADDRESS
Select voice menu option 13 to configure the IP address of the firmware server.
CONFIGURATION SERVER IP ADDRESS
Select voice menu option 14 to configure the IP address of the configuration server.
UPGRADE PROTOCOL
Select voice menu option 15 to choose firmware and configuration upgrade protocol. User can choose
between TFTP and HTTP.
FIRMWARE UPGRADE MODE
Select voice menu option 17 to choose firmware upgrade mode among the following three options:
1) always check, 2) check when pre/suffix changes, and 3) never upgrade
WAN PORT WEB ACCESS (Note: This option is not present on GXW4224 gateway)
Select voice menu option 12 to enable WAN Port Wed Access of the device configuration pages.
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:
1. Follow table 5 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 IP address for the GXW42XX).
NOTE:
•
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/gxw_series/GXW42XX/documents/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
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.
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2. Use Random Ports (under Advanced Settings webpage)
It really depends on your network settings, so set this parameter to Yes or No, whichever works.
Generally if you have multiple IP devices under the same network, it should be set to Yes. If using a
Public IP address, set this parameter to No.
3. 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.
•
DTMF in-audio
•
DTMF via RTP (RFC2833)
•
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:
•
PCMU/A (or G711µ/a)
•
G729 A/B
•
G723
•
G726 (16/24/32/40)
•
iLBC
•
AAL2 (all G726)
DEFINITIONS
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:
•
STATUS: Displays the network status, account status, software version and MAC-address of the
phone
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•
BASIC SETTINGS: Basic preferences such as date and time settings, multi-purpose keys and
LCD settings can be set here.
Additional functions available to administrators are:
•
ADVANCED SETTINGS: To set advanced network settings, codec settings and XML
configuration settings.
•
PROFILE X: To configure each of the SIP accounts.
•
FXS PORTS: To configure each of the FXS ports and Hunting Groups etc.
TABLE 7: BASIC SETTINGS
End User Password
Password to access the Web Configuration Menu. This field is case sensitive with a
maximum length of 25 characters.
Web Port
By default, HTTP uses port 80. This field is for customizable web port.
Telnet Server
Default is Yes.
IP Address
There are two modes to operate the GXW42XX :
DHCP mode: The GXW42XX acquires its IP address from the first DHCP server it
discovers from the LAN It is on.
PPPoE: Set the PPPoE account settings. The GXW42XX will establish a PPPoE
session if any of the PPPoE fields is set.
Static IP mode: 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.
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
This option specifies the domain name that client should use when resolving
hostnames via the Domain Name System. Default is blank.
DHCP vendor class ID
Used by clients and servers to exchange vendor-specific information. Default is
HT500.
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
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Controls how the date/time is displayed according to the specified time zone.
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Self Defined Time Zone
The syntax is std offset dst [offset],start[/time],end[/time]
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 has access to the Device Status page.
TABLE 8: STATUS PAGE
MAC Address
The device ID in hexadecimal format. This is needed for Internet Service Provider
troubleshooting. Note: there are separate MAC addresses for the WAN side and the
LAN side. The LAN 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.
IP Address
Shows IP address of GXW42XX
Product Model
Contains the product model info.
Software Version
Program: This is the main software release. Boot and Loader are not changed often.
System Up Time
Shows system uptime since the last reboot.
PPPoE Link Up
Shows whether the PPPoE connection is running if connected to DSL modem.
NAT
Shows type of NAT the GXW42XX is connected to via its WAN port. It is based on
STUN protocol.
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Port Status
Displays relevant information regarding the individual FXS ports. Example:
Port
Hook
Registration
DND
FXS1
On Hook
Registered
No
FXS2
Off Hook
Registered
No
FXS3
On Hook
Not
No
Forward
Busy
Delayed
Forward
Forward
613
614
Registered
…
…
…
FXS24
On Hook
Registered
Yes
615
FXS port 24 user has set Do Not Disturb.
FXS port 1 user has set his calls to be forwarded unconditionally to ext 613
FXS port 2 user has set his calls to be forwarded to 614 when his phone is busy.
FXS port 3 user is not registered with his SIP Server.
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.
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 48.
Layer 2 QoS
Value used for layer 2 VLAN tag. Default setting is blank.
STUN Server
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.voip-info.org/wiki-STUN
Keep-alive interval
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.
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Use STUN to detect
Use STUN keep-alive to detect WAN side network problems. If keep-alive request does
network connectivity
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.
Firmware
Upgrade
& Enables GXW42XX to download firmware or configuration file using either the TFTP or
Provisioning
HTTP/S 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 The URL for the HTTP server used for firmware upgrade and configuration via HTTP.
Server
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
XML
Config
File The password used for encrypting the XML configuration file using OpenSSL.
Password
This is required for the phone to decrypt the encrypted XML configuration file.
HTTP/HTTPS User Name
The user name needed to authenticate withthe HTTP/HTTPS server.
HTTP/HTTPS Password
The password needed to authenticate with the HTTP/HTTPS server.
Firmware File Prefix
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.
Firmware File Postfix
Default is blank. This setting is useful for Internet Telephone Service Providers. End
users should keep it blank.
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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 If set to “Yes”, configuration and upgrade server’s information can be obtained using
override server
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 “Yes, daily at hour ___” or “Yes, weekly on day ___” is checked, user can
specify “Hour of the day (0-23)” or “Day of the week (0-6)”. Default time is Monday
1AM.
When “Yes, Yes, every ___ minutes” is checked, user can specify the time interval
between each upgrade check. Default is 10080 minutes.
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
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.
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.
SSL
Private
Key Enter SSL Private Key password here.
Password
ACS URL
User specify the Auto Configuration Server’s URL (TR-069 protocol)
ACS Username
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
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Periodic Inform Interval
Connection
Frequency that the inform packets will be sent out to the ACS
Request Set a user name for the ACS to connect to this device
Username
Connection
Request Set a password for the ACS to connect to this device
Password
System Ring Cadence
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)
Call Progress Tones
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
Lock Keypad Update
If set to “Yes”, the configuration update via keypad is disabled.
Disable Voice Prompt
Disables the voice prompt configuration. Default is “No”. If set to “Yes” accessing
integrated voice menu will be impossible.
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
This feature allows users to place an outbound PSTN call in case there is a loss of an
active registration (SIP server unreachable) of all FXS profiles. If set to “YES”, when
GXW42XX recognizes a loss of registration, all outgoing calls will be routed to an FXO
gateway.
The use of this option presumes a configured GXW410x or another FXO gateway with
an active PSTN line connection.
FXO Gateway
IP Address or URI of the FXO gateway.
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
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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
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
Primary
IP Address or FQDN of the primary RADIUS Server
RADIUS
Auth Primary Radius server authentication port. Default value is 1812
RADIUS
Acct Primary Radius server accounting port. Default value is 1813
Port
Primary
Port
Primary Radius Server Special secret string should be preconfigured according to RADIUS Server
Secret
Secondary
configuration
RADIUS IP Address or FQDN of the secondary RADIUS Server
Server
Secondary RADIUS Auth Secondary Radius server authentication port. Default value is 1812
Port
Secondary RADIUS Acct Secondary Radius server accounting port. Default value is 1813
Port
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Secondary
Radius Special secret string should be preconfigured according to RADIUS Server
Server Secret
configuration
RADIUS Timeout
Default value is 2 seconds. The time between retries the GXW will send “AccessRequest” 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
Configuration
Device 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: FXS PORTS SETTINGS
FXS Port
FXS Port Number
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 ID
Select the corresponding Profile ID (1/2/3/4)
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Hunting Group
This feature enables the gateway to register all existing FXS ports with the same
phone number. Each incoming call will be routed to first available port in Linear or
Circular mode. User may configure all ports as members of the same Hunting Group
or it may configure different port combinations for more than one Hunting Group.
For example: Ports 1, 3 and 5 are members of the same Hunting Group, the rest of
the ports may have separate numbers and may be reached independently.
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.
Select appropriate value for Hunting Group feature. The original SIP account should
be set to Active while the group members should be set to the port number of the
Active Port.
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.
Enable Port
If set to No, FXS port will become inactive (Default is set to Yes)
Port#
FXS Port Number
Offhook Auto-dial
This feature allows you to automatically dial the number specified in this field as soon
as the port is offhooked, i.e. when the receiver on the phone connected to Port# is
picked up.
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Map to FXO Port#
This is used only when peering with a Grandstream GXW410x. Default is 1,
Supported values are 1-8, meaning line 1 to line 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 the call will be sent to.
and Port
Sip port that will be annexed to the IP address above.
TABLE 11: PROFILE SETTINGS
Profile Active
When set to Yes the SIP Profile is activated.
Primary SIP Server
Primary 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 Default is no. If set to yes it will register to Primary Server if registration with Failover
Server
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.
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.
NAT Traversal
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.
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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.
Outgoing Call without Default is No. If set to “Yes,” user can place outgoing calls even when not registered (if
Registration
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
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).
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.
Local RTP Port
Defines the local RTP-RTCP port pair the GXW42XX will listen and transmit. It is the
base RTP port for channel 0. When configured, channel 0 will use this port _value for
RTP and the port_value+1 for its RTCP; channel 1 will use port_value+2 for RTP,
port_value+3 for its RTCP and so on. The default value for Profile 1 is 5004 and 6004 for
Profile 2.
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Use random port
Default is No. If set to Yes, the device will pick randomly-generated SIP and RTP ports.
This is usually necessary when multiple GXW42XX/HT50X are behind the same NAT.
Refer to Use Target Default is No. If set to YES, then for Attended Transfer, the “Refer-To” header uses the
Contact
transferred target’s Contact header information.
Transfer
on Default is No. In which case if conference originator hangs up the conference will be
Conference
terminated. When option YES is chosen, originator will transfer other parties to each
Hang up
other so that B and C can choose either to continue the conversation or hang up.
Disable Bellcore Style Default is No. you can make a Conference by pressing ‘Flash’ key. If set to Yes, you
3-Way Conference
Remove
OBP
need to dial *23 + second callee number.
from Default is No. If set to Yes, the Outbound Proxy will be removed from the route header.
Route Header:
Support SIP Instance Default is Yes. If set to Yes, the contact header in REGISTER request will contain SIP
ID
Instance ID as defined in IETF SIP Outbound draft.
Validate incoming SIP Default is No. If set to yes all incoming SIP messages will be strictly validated according
message
to RFC rules. If message does not pass validation process, call will be rejected.
Check SIP user ID for Default is No. Check the SIP User ID in Request URI. If they don’t match, the call will
incoming INVITE
Allow
be rejected.
Incoming
Messages
from
SIP Default is No. If incoming SIP message does not match with SIP Server, it will be
SIP rejected.
Proxy Only
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
Maximum retransmission interval for non-INVITE requests and INVITE responses.
DTMF Payload Type
Sets the payload type for DTMF using RFC2833.
Preferred
method
DTMF The GXW42XX supports up to 3 different DTMF methods including in-audio, via RTP
(in
listed (RFC2833) and via Sip Info. The user can configure DTMF method in a priority list.
order)
Disable
DTMF Default is No. If set to yes, use above DTMF order without negotiation
Negotiation
Send
Hook
Flash Default is No. If set to yes, flash will be sent as a DTMF event.
Event
Enable Call Features
Default is Yes. (If Yes, call features using star codes will be supported locally)
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Proxy Require
SIP Extension to notify SIP server that the unit is behind the NAT/Firewall.
Use NAT IP
NAT IP address used in SIP/SDP message. Default is blank.
Distinctive Ring Tone
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 Default is No. If set to YES Call Waiting caller ID will not be provided to analog phone
Caller ID
connected to this FXS port.
Disable Call Waiting Default is No. This is to disable the stutter Call Waiting Tone when a Call Waiting call
Tone
arrives. The CWCID will still be displayed.
Disable
Receiver Default is No. If YES, ROH tone will not be played after off hook for 60 seconds.
Offhook Tone
Disable
Reminder Default is No. This is to disable the Reminder Ring that is played when a call is waiting
Ring for On-Hold Call
on hold and the analog phone goes on-hook.
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.
Note: This is only supported in model GXW4224 ver 4.A or newer.
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Ring Timeout
Incoming call will stop ringing when not picked up given a specific period of time.
Hunting Group Ring Default is 20 seconds. If call is not answered within this designated time period, the call
Timeout
will be forwarded to the next member of a Hunt Group.
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 Default value is 20 seconds. In case this feature activated using * codes (*92 code), the
Wait Timeout
call will be forwarded after this preconfigured amount of time.
No Key Entry Timeout
Default is 4 seconds. Call will be completed within this time interval if no additional key
entry occurs.
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.
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.
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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
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.
3.
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 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*]+ }.
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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 Default is No. If set to Yes, incoming calls with anonymous Caller ID will be rejected with
Rejection
486 Busy message.
Special Feature
Default is Standard. Choose the selection to meet some special requirements from
Softswitch vendors. Example of vendors - CBCOM, RNK.
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
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
Refresher
UAS
Specify As a Caller, select UAC to use the phone as the refresher, or UAS to use the Callee or
proxy server as the refresher.
Specify As a Callee, select UAC to use caller or proxy server as the refresher, or UAS to use the
Refresher
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.
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Send Re-INVITE After Default is No, If set to “Yes”, device will send an INVITE with audio vocoders upon
Fax
completition of Fax to continue session in audio only.
Enable
Silence For fax machines that do not send a Disconnect when fax is done. This option
Detection
for
Fax Enables/Disables the detection of silence in order to know the fax has finished. The
Disconnect
silence period is non-configurable and fixed to 7 seconds.
Enable 100rel
Enables the use of PRACK (Provisional Acknowledgment) method.
Use
First
Vocoder
Matching Default is No. If set to “Yes”, device will include only the first match vocoder in its 200OK
in
200OK response, otherwise it will include all match vocoders in same order received in INVITE.
SDP
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/E and 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
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 Choose the packing mode for G726-32
option
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
Default value is 100. Range is from 96 to 127.
Payload type
AAL2-G726-24
Default value is 99. Range is from 96 to 127.
Payload type
AAL2-G726-32
Default value is 104. Range is from 96 to 127.
Payload type
AAL2-G726-40
Default value is 103. Range is from 96 to 127.
Payload type
VAD
Default is No. VAD allows detecting the absence of audio and conserve bandwidth by
preventing the transmission of "silent packets" over the network.
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Symmetric RTP
Default is No. When set to Yes the device will change the destination to send RTP
packets to the source IP address and port of the inbound RTP packet last received by
the device.
Fax Mode
Fax
Tone
T.38 (Auto Detect) FoIP by default, or Pass-Through (must use codec PCMU/PCMA)
Detection Default is Callee. This decides whether Caller or Callee sends out the re-INVITE for
Mode
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.
•
High (initial 200ms, min 40ms, max 600ms) Note: not all vocoders can meet the
high requirement
SRTP Mode
•
Medium (initial 100ms, min 20ms, max 200ms)
•
Low (initial 50ms, min 10ms, max 100ms)
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 Set to Yes if the traditional PBX you are using with GXW42XX uses this method for
Disconnect
Loop
signaling call termination. Default is No.
Current Default is 200. In 100 - 1 0000 milliseconds range.
Disconnect duration
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.
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Gain
Handset volume adjustment.
RX is for receiving volume (direction FXSÆto analog
phone), TX is for transmission volume (Analog phoneÆ to FXS). Default values are 0dB
for both parameters. Loudest volume: +6dB Lowest volume: -6dB.
Disable
Line
Echo Default is No. If set to “Yes”, device will not use LEC to remove echo from a voice
Canceller (LEC)
communication.
Ring Tones
Configure ring cadences according to preference.
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 IP phone 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 request to download configuration file,
“cfg000b82xxxxxx” or “cfg000b82xxxxxx.xml”, where “000b82xxxxxx” is the LAN MAC address of the
GXW42XX. It will first request “cfg000b82xxxxxx” then “cfg000b82xxxxxx.xml”
A service provider or an enterprise with large deployment of Grandstream devices can easily manage the
configuration and service provisioning of individual devices remotely from a central server.
Grandstream has a central provisioning system called GAPS (Grandstream Automated Provisioning
System). GAPS supports automatic 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.
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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
server for further provisioning. Grandstream also provide GAPSLITE software package which contains
our NAT friendly TFTP server and a configuration tool to facilitate the task of generating device
configuration files.
The GAPSLITE configuration tool is now free to end users. The tool and configuration template 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/HTTPS server, server name can be in either FQDN or IP address format. Here are
examples of some valid URL.
firmware.mycompany.com:6688/Grandstream/1.0.6.8
or
firmware.grandstream.com
NOTES:
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.
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.
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.
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.
Alternatively, user can download a free TFTP or HTTP server and conduct local firmware upgrade.
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.
parameter consists of a Capital letter P and 2 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 Grandstream Device boots up or reboots, it will issue request for configuration file named
“cfgxxxxxxxxxxxx”, where “xxxxxxxxxxxx” is the MAC address of the device, i.e., “cfg000b820102ab”. The
configuration file name should be in lower cases.
FIRMWARE AND CONFIGURATION FILE PREFIX AND POSTFIX
Firmware Prefix and Postfix allows device to download the firmware name with the matching Prefix and
Postfix. This makes it possible to store ALL of the firmwares with different version in one single directory.
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Similarly, Config File Prefix and Postfix allows device to download the configuration file with the matching
Prefix and Postfix. Thus multiple configuration files for the same device can be stored in one directory.
In addition, when the field “Check New Firmware only when F/W pre/suffix changes” is selected, the
device will only issue firmware upgrade request if there are changes in the firmware Prefix or Postfix.
MANAGING FIRMWARE AND CONFIGURATION FILE DOWNLOAD
When “Automatic Upgrade” is set to “Yes”, Service Provider can use P193 to have the devices
periodically check with either Firmware Server or Config Server, whenever they are defined. This allows
the device periodically check whether there is any new changes need to be taken, similar to the AntiVirus
Software to upgrade the Virus Definition files. Screenshot is below:
Automatic Upgrade:
No
Yes, every
Yes, daily at hour
10080
(0-23).
minutes(60-5256000).
Yes, weekly on day
(0-6).
If automatic upgrade is enabled, service provider can further customize the behavior and distribute server
load by setting hour of the day and/or day of the week for upgrade.
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RESTORE FACTORY DEFAULT SETTING
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.
FACTORY RESET
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 about 7 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 “99” 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
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)
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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”.
FCC Warning
This device complies with part 15 of the FCC Rules. Operation is subject to the following two conditions:
(1) This device may not cause harmful interference, and (2) this device must accept any interference
received, including interference that may cause undesired operation.
Any Changes or modifications not expressly approved by the party responsible for compliance could void
the user's authority to operate the equipment.
FCC 15.105 Class B
(b) For a Class B digital device or peripheral, the instructions furnished the user shall include the following
or similar statement, placed in a prominent location in the text of the manual:
Note: This equipment has been tested and found to comply with the limits for a Class B digital device,
pursuant to part 15 of the FCC Rules. These limits are designed to provide reasonable protection against
harmful interference in a residential installation. This equipment generates, uses and can radiate radio
frequency energy and, if not installed and used in accordance with the instructions, may cause harmful
interference to radio communications. However, there is no guarantee that interference will not occur in a
particular installation. If this equipment does cause harmful interference to radio or television reception,
which can be determined by turning the equipment off and on, the user is encouraged to try to correct the
interference by one or more of the following measures:
—Reorient or relocate the receiving antenna.
—Increase the separation between the equipment and receiver.
—Connect the equipment into an outlet on a circuit different from that to which the receiver is connected.
—Consult the dealer or an experienced radio/TV technician for help.
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