Product Related Questions

toughhawaiiΔίκτυα και Επικοινωνίες

26 Οκτ 2013 (πριν από 4 χρόνια και 6 μήνες)

287 εμφανίσεις

Product Related Questions

HandyTone Series FAQs

Q: What are the main differences between the 7 models of Handytones?









Ethernet Ports

1 RJ45


1 RJ45


2 RJ45


2 RJ45


2 RJ45


2 RJ45









FXS Port







FXO Port







through Port















Can you

explain the use of 'PSTN pass through' and 'FXO port'?

PSTN Pass through port:

What it can do:

Local manual switching between PSTN and IP mode on a per call basis.

User can switch to PSTN line by pressing *00 (or the configured strings) for each call
re they are placed. The device will revert back to the default IP mode once the phone
is hung up.

It can allow a PSTN call to ring/call the phone connected to the FXS port.

It also serves as a life line in case of power outage.

What it


e a VoIP call into the PSTN port

Allow a call from PSTN to route other VoIP devices (different from the FXS phone) over
the IP network

Automatically route calls made by the local user to PSTN line

On the HT486
Rev 1.0,
the PSTN port is only a life li
ne port that switches to the PTSN
network only when there is a loss of power.

FXO port:

It can support all the functions of a PSTN pass through plus:

Terminate a VoIP call into the PSTN port

Allow a PSTN call to call either the FXS phone or other VoIP devi
ces over the IP

Route call automatically and transparently to PSTN line according to user configuration


Can I call from FXS1 port to FXS2 port?

Yes, they can communicate with each other by dialing the respective extension number. Both
FXS po
rts need a valid sip account registered on the server.


How can I disable hook/flash?

On the Advanced Settings page, there is a field On
Hook Threshold. One on the selections for
this field is 'Hook/Flash OFF',

this option will disable hook/flash on the phone connected to the
ATA. To switch to a second channel, press

button on the phone, instead of doing

Note: This feature is not available on older hardware revision models.


What hardware revision is my Handytone?

At the bottom of the device, there will be a white sticker. On this sticker will be a note:

'Rev: x.0' where x is the Hardware Revision

You can also get this information under the Status tab of the Web Configurati
on pages.

Your hardware revision is very important information. Depending on your hardware revision, it
may/may not be able to upgrade to the latest release.

Ex. HT286 Rev 2.0 upgrades up to firmware, while Rev 3.0 can upgrade to the latest

Note: Once there is a new hardware revision out in the market, the older revision is no more
manufactured or sold.


How do I access the Web Configuration pages (for HT486/496/488)?

Please disconnect all connections to the HT486/HT496/HT488 and follow

the instructions


Connect the analog touch
tone phone to the HT.


Connect power supply.


Connect the Ethernet cable between the INTERNET Source (ex. Router, Modem) and the
WAN port on the HT.


Connect another Ethernet cable between your PC and the LAN
port on the HT


Wait for 30 seconds till your PC gets an IP ADDRESS (


Now, open Internet Explorer and type in, you should see Grandstream Login
Screen pop up.


Enter 'admin' as the password


Go to Advanced Settings page and switch "Ena
ble WAN port HTTP access" to YES, hit
'Update' and then 'Reboot'.


Disconnect your PC from the LAN port and connect it to any other port on your Router
within the same LAN Segment


Type in the actual IP ADDRESS of the HT (You can look this up by pressin
g *** on the
phone, and then 02) on Internet browser, access the Web Configuration page as you did earlier
and configure the device by filling in the information given by your Internet Telephony Service
Provider (ITSP).


How do I setup a 3
way confere
nce call?

Setting up a 3
way conference calling between parties using an HT, A and B is easy:


HT calls A


HT talks to A


HT presses FLASH or hook/flash and gets a new dialtone


A is on Hold


HT dials *23 and number for B


HT talks to B


HT presses FLASH or
hook/flash to initiate the 3
way calling


How do I perform attend transfer?

Attend Transfer from A to B through HT:


A calls HT


HT talks to A


HT presses FLASH or does hook/flash to get new dialtone.


A is on Hold


HT calls B


HT talks to B


HT hangs up to
perform the Attend Transfer.


A and B are in call now.


Which phone will ring if there is an incoming call on my wire (fixed) line (HT386)?

The Phone connected to FXS1 port will ring when there is an incoming call on the Fixed Line.


How do I make or
receive PSTN calls on my HT (for HT486/386/488)?

When receiving a call, the phone connected to the HT simply rings. When placing a call, dial the
'PSTN Line Access Code' first, as configured on the Web Configuration Page (by default it is
*00), and then
dial the desired PSTN number.

Q: How do I ensure my 911 calls are directed over the PSTN network and implement a dial
plan for specific area codes without dialing a (1) prefix?

Create the following string under Dial Plan Configuration option under FXS PO
RT configuration

{L: 404x+| L:770x+ | L:678x+| L:911 | x+}

Please note that only HT503 supports this feature.

Q: What is the led pattern of a HT502/503?

Power led ON indicates power is connected

WAN led ON indicates port activity

LAN led ON indica
tes PC (or LAN) port activity

Phone1/2 or Line led indicates status of the respective FXS port or FXO in case of
HT503. ON is Busy, OFF is available and slow blinking there is a voice mail for that

Slow blinking of WAN and LAN together means the prod
uct’s firmware is in upgrading or
provisioning state.

BudgeTone Series FAQs

Q: What are the differences between BudgeTone
101, 102 and 200 models?

The BT model 101 (also called BT100) and 102 have the same software functions and the only
difference is tha
t model 101 has one Ethernet interface and model 102 has two Ethernet
interfaces i.e. the PC port (to be connected to your computer) and normal LAN port (to be
connected to your Internet Source.). These 2 ports are bridged together similar to a HUB. BT

is higher end version with Router feature and capable of supporting 100Mbps connections.

BudgeTone Model




LAN Interface

1 x RJ45

2 x RJ45

2 x RJ45





Router Feature





What Codecs

are supported by BudgeTone Series Phones?

BudgeTone Phones support a wide range of Codecs:

BT101 and BT102 support:


PCMU or g711(a/µ law)


G.729 A/B




G.722 (wideband)





BT200 supports PCMU, PCMA, g729 A/B, g723.1 and GSM codecs.


How do I configure a Public or Static IP address using the phone menu?

This is a very crucial configuration. Generally we recommend configuring via DHCP since it’s
the easiest method of configuration. But nevertheless that is the reason configuring Publi
IP on Grandstream Products is as easy as it gets.

Press the MENU button and go through options 1 to 5 carefully. For each option make the
necessary change, for ex. option 1 should be dhcp off, option 2 should be the static IP address,
option 3 is
the subnet mask, 4 is the gateway/router IP address and lastly and most importantly 5
is the DNS Server IP address. All these IP addresses need to be entered in 12 digit format i.e.
192168001029 for

Absolutely! Here is a simple example of
performing 3 way conferencing on BT phones:

A, B and C need to talk in a conference call. A will be performing the 3
way conference call
using his new BudgeTone Phone.

1. A calls B

2. A presses CONFERENCE and gets a new dialtone while B is on Hold

3. A cal
ls C

4. A presses CONFERENCE again to initiate the Conference Call

5. A, B and C are in conference!


How do I perform attend transfer and blind transfer using BudgeTone Phones?

Example: B needs to transfer a call from A to C.

Attend Transfer:


A an
d B are in a call


B needs to transfer the call to C


B presses FLASH to get new dialtone


B calls C while A is on Hold


Now, B can transfer the call from A to C by pressing TRANSFER and then hanging up.


A and C are in a call now!

Blind Transfer:

1. A and B ar
e in a call

2. B needs to Blind transfer the call to C

3. B presses TRANSFER and gets new dialtone; A is on Hold.

4. B dials number for C

5. Call is transferred to C


How do I configure voice
mail on the BudgeTone?

There is a field called Voice Mail
User ID on the ‘Advanced Settings’ page. You need to
configure this field with your voice mail account number ex.8500. Now, if you need to access
Voice Mail simply press the Message Button on the device and it will connect to the Voice Mail
Inbox. In case
of a new voice mail left in your inbox, the LED will start to blink RED.


Does the BudgeTone have the Off hook Auto
Dial and Auto
Answer feature?

BudgeTone 100 and BT200 models support both these features.


Does the BudgeTone support distinctive rin
gtone feature using ALERT
INFO Method?

Currently, BT phones support Distinctive Ringing for 3 specific Phone
Numbers/Extensions/User IDs. You will find this setting on Advanced Settings Page. Basically
you can configure 3 extensions with a Ringtone each an
d then whenever incoming call is from
one of these numbers, that configured Ringtone will play. The BT200 model supports Distinctive
Ringtones using SIP ALERT
INFO header.

BT200 supports the Alert
Info mapping to the 3 custom ring tone files. For example,

if you
configure the custom ring tone 1 user id to “priority” (instead of a real user ID), that ring tone
will be used if we receive INVITE with Alert
Info header in the following format:



Does the BudgeTone support Syslog Se
rver setting?

Yes, under Advanced Settings page, you will see Syslog Server and Syslog Level fields to setup
Syslog message retrieval. Syslog messages are very helpful in debugging any errors with the


Where do I view my Call History ex. Misse
d/Incoming/Outgoing Calls?

You can check Received/Missed Calls along with Outgoing calls made, on the Phone by
unhooking the receiver and then pressing CALLERS and CALLED resp. For more detailed Call
information you can log onto the web configuration pages
, and check the Status page.


How do I make a call using only the IP Address?

BT101 and BT102 models:


hook the receiver or press speakerphone.


Press the MENU button


Now dial the IP Address in 12 digit format ex.192168001029


Press SEND.


This model has the ability to dial an IP address under the same LAN segment by simply pressing
the last octet in the IP address.

In the Advanced Settings page there is an option "Use Quick IP
call mode", by default it is set to
No. When this option is set
to YES, and #XXX is dialed, where X is 0
9 and XXX <=255, phone
will make direct IP call to aaa.bbb.ccc.XXX where aaa.bbb.ccc comes from the local IP address
REGARDLESS of subnet mask.

#XX or #X are also valid so leading 0 is not required (but OK).

For exa
: calling just dial #3 follow by SEND or # calling just dial #23 follow by SEND or # calling just dial #123 follow by SEND or # dial #3 and #03 and #003 has same eff
> call


If you have a SIP Server configured, Direct IP
IP call will still work. However, if you are
using STUN, Direct IP
IP call will also use STUN.


How do I ignore an incoming call?

Yes, if you press the MUTE/DEL key whilst
receiving an incoming call, it will be rejected and
forwarded to your Voice Mail (if configured).


This feature is available only on BT200 model.

GXP Series FAQs

Q: The GXP Series supports multiple languages. Which languages are available?

eries has a language pak that supports English, Spanish, German, French, Polish,
Italian, others as requested. The language pack is available on the
Firmware Section

and is
updated regul


For the Call Log, how many entries are allowed for dialed, missed, etc.?

The call log stores up to 50 entries in each call log (incoming|outgoing|missed). All GXP models
are the same.


Does the availability of some features depend on the
associated PBX? Which ones?

The end
user can choose to support advanced call features locally (on the phone) or choose the
PBX (or server) to support them. Please use the GUI Account Page settings to select or deselect
this feature. Call features include c
all forward (all types), transfer (all types), hold, mute, DND
and conference.


Is call recording supported? Does this require an add
on device?

If the server or PBX supports call recording, we can support it. The GXP Series does not require
al add
on devices to support this feature. Call recording is supported by all models if the
PBX or server supports it.


For the address book, how many entries can be stored?

Up to 500 entries in the phonebook.

XML sync is dependent on the size of th
e XML file and is
limited by current memory allocation.

The address book is supported on all GXP models.


Does the display show the call duration or a voice mail count?

Call duration is displayed during an active call. Each line has an individual voi
cemail and
number of voicemails is displayed for each line.

Call duration and number of voicemails is
supported by all GXP models.


How do I use the paging or intercom feature on the GXP Series?

For GXP20xx, on the Client side, the following two fiel
ds (Account Page) need to be set to Yes.

Allow Auto Answer by Call

Turn Off speaker on Remote disconnect

For GXP21xx/GXP1450, on the Client side, the following field (Accoung Page) needs to be set
to Yes.

Allow Auto Answer by Call


Server has to support this feature to make it work.


Is the GXP2020 Expansion Module compatible with the GXP2120/GXP2110 models?

Yes, it is compatible. However, the GXP2120/2110 has a special EXT port on the back which
requires a different extension cab
le (Included with the GXP2020 Expansion Module).

Q: Why am I getting a strange icon with a down arrow going into a basket [_]?

This symbol is stating that the phone is writing files to the call record detail file.

This occurs
when the phone is either id
le for 5 minutes or when the call records reach greater than 100 calls.


What configuration changes on the web
gui require a reboot?

Under Basic and Account settings you are not required to reboot the phone.

If changes were
made under Advanced settings

only the network changes require a reboot.

The following
parameters do not require a reboot:

Admin Password

STUN Server

Phonebook XML Download

Offhook Auto Dial

Call Progress Tones

Custom ring tone 1

3, used if incoming caller ID is set

Intercom User

Disable Call

Disable Call
Waiting Tone

Disable Direct IP Calls

Use Quick IP
call mode

Disable Conference

Enable MPK sending DTMF

Disable Transfer

Auto Attended Transfer

Display Language

GXP2000 FAQs


What are the differences between the
BudgeTone Series and the GXP2000?

The GXP2000 SIP Enterprise phone has more feature functionality than the BT Series.

Functionally, the GXP2000 has multiple accounts, GUI Interface, Busy Lamp Field (BLF), speed
dial, dual 10/100Mbps Ethernet ports, POE,

Please reference the GXP2000 User Manual for
more information.


What Codecs are supported by the GXP2000?

The GXP2000 supports a wide range of Codecs:


PCMU or G.711(a/µ law)


G.729 A/B






How many VoIP accounts can I have on the

The GXP2000 IP phone is capable of having a maximum of 4 different SIP (VoIP) Accounts.
These accounts may be different extensions on the same Server or different Servers from
different Service Providers.

The 7 functional buttons on the right sid
e of the phone are speed
dial buttons.


When Line1 to Line4 are busy with calls, next incoming call will light up on the first
function key (Speed Dial
1). You can pick the call up by pressing the function key like a Line
button. For outgoing calls, p
ress the button under the cradle (i.e. on
hook and off
hook) to get a
new dialtone.


How can I configure BLF (Busy Lamp Field) on the 7 speed dial buttons of the

The Speed Dial buttons can also be configured for Asterisk™ BLF feature which al
lows you to
realize if a particular extension is busy with a call or idle.

To configure this on the GXP2000,
select Asterisk™ BLF from the drop down box next to Key Mode for a configured Speed Dial
contact on the Basic Settings page. Now, on the Phone, if
the LED next to the respective button
lights up Red, it means the extension is Busy; if it flashes Red it indicates an incoming call on
that extension.


Can I display any custom text along with my extension number on the phone LCD?

Yes, any text enter
ed on the ‘Name’ field along with the ‘User ID’ will be displayed on the LCD
in Bold (for Account 1 only). Under Basic Settings, field ‘Display Clock instead of Date’ should
be set to No.


Can I enable/disable recording of missed calls?

Yes, on each Ac
count Page, there is a field for ‘Disabling Missed Calls’. If set to Yes, ‘Missed
Calls’ for that Account will not be recorded anymore.


Does the GXP2000 have ‘intuitive dialing’ and call log for previously dialed and
received calls?

The GXP2000 has a
call log with only last 20 records stored and will be reset once the phone
reboots. Just off hook and press the left and right arrow key you should see that. There is no
intuitive dialing yet but we hope to implement address book function in the new firmwa


How do I use the paging or intercom feature on the GXP2000?

On the Client side, the following two fields (Account Page) need to be set to Yes.

Allow Auto Answer by Call

Turn Off speaker on Remote disconnect


The PBX Server has
to support this feature to make it work.


How do I make a call using the IP Address only?

This phone has the ability to dial an IP address under the same LAN segment by simply pressing
the last octet in the IP address.

In the Advanced Settings page the
re is an option "Use Quick IP
call mode", by default it is set to
No. When this option is set to YES, and #XXX is dialed, where X is 0
9 and XXX <=255, phone
will make direct IP call to aaa.bbb.ccc.XXX where aaa.bbb.ccc comes from the local IP address
RDLESS of subnet mask.

#XX or #X are also valid so leading 0 is not required (but OK).

For Example
: calling just dial #3 follow by SEND or # calling just dial #23 follow by SEND or # calling 192.
168.0.123 just dial #123 follow by SEND or # dial #3 and #03 and #003 has same effect
> call


If you have a SIP Server configured, Direct IP
IP call will still work. However, if you are
using STUN, Direct IP
IP call will a
lso use STUN.


How do I ignore an incoming call?

Yes, if you press the MUTE/DEL key whilst receiving an incoming call, it will be rejected and
forwarded to your Voice Mail (if configured).


Does the GXP2000 support PoE?

Yes, the GXP2000 supports

Power over Ethernet It can accept either pins 7/8(+), 4/5(
) or 7/8(
4/5(+). The injector voltage must above 48VDC. When you use PoE power you must take away
5V power adaptor.


Does the GXP2000 support Distinctive Ringtone using ALERT INFO method?

The GXP2000 supports the Alert
Info mapping to the 3 custom ring tone files. For example, if
you configure the custom ring tone 1 user id to “priority” (instead of a real user ID), that ring
tone will be used if we receive INVITE with Alert
Info header in
the following format:

Info: ;info=priority


How do I configure my GXP2000 for Daylight Savings?

The "Automatic Daylight Saving Time Rule" shall have the following syntax:


Both start
time and end
time have the same


month: 1,2,3,..,12 (for Jan, Feb, .., Dec)

day: [+|

weekday: 1, 2, 3, .., 7 (for Mon, Tue, .., Sun), or 0 which means the daylight saving rule is not
based on week days but based on the day of the month.

hour: hour (0

minute: minute (0

If "weekday" is 0, it means the date to start or end daylight saving is at exactly the given date. In
that case, the "day" value must not be negative.

If "weekday" is not zero and "day" is positive, then the daylig
ht saving starts on the first "day"th
iteration of the weekday (1st Sunday, 3rd Tuesday etc).

If "weekday" is not zero and "day" is negative, then the daylight saving starts on the last "day"th
iteration of the weekday (last Sunday, 3rd last Tuesday etc).

The ‘savings time’ is in the unit of minutes. The saving time may also be preceded by a negative
) sign if subtraction is desired instead of addition.

The default value for "Automatic Daylight Saving Time Rule" shall be set to
;60" which is the rule for US.


Which headset models do you recommended to work with your GXP Series Phone?

Most “noise canceling” enabled headsets are compatible with the GXP series.

We recommend
VXI, Jabra, HelloDirect and Plantronics headsets. We h
ave tested with Jabra GN9350,
Plantronics CS70N, Plantronics M12, M10, M22 and VXI10V Tuffset, etc.


How do you configure "event list BLF" on the GXP Series?

You will need to configure a "event list BLF" URI on your server side if the server supports i

On the GXP, under account page, fill in the ""event list BLF" field with the URI without the
domain. (i.e.: BLF1006).

Under Basic Settings, please select "event list BLF", choose
account number, monitored number,


Is the GXP2000 extension module compatible with the other GXP series phones?

No, the GXP2000 extension module is only compatible with the GXP2000.

Extension modules
for other GXP phones will be available in 2008



What service providers or platform support the GXV3000?

ITSPs with pure SIP proxy (e.g. Iptel’s SER); or ITSPs with hybrid SIP proxy (e.g. Digium
Asterisk™ R1.4, SBC or OBP) but supporting H.264 and H.263 when manipulating media.


What are the functions of the USB ports?

The 2 USB interfaces are USB 2.0 host ports.

The USB interface can be used for external USB
device like flash to store captured image, ring tones, phone address books (not implemented yet),

When a USB flash

drive is connected to the USB 2.0 port, an icon will display in the lower
section of the LCD.


Do I need a service provider? Can I make IP
IP call between video phones?

Yes, you need to have an Internet Telephony Service Provider (ITSP). You can als
o use “Direct
IP Call” to talk to each other in an ad hoc fashion without a SIP proxy.

“Direct IP Call” can be
made between two phones if:

both phones have public IP addresses, or

both phones are in a same LAN/VPN using private or public IP addresses, or

both phones can be connected through a router using public or private IP addresses (with
necessary port forwarding or DMZ)

To make a direct IP call, press OK button to bring up MAIN MENU.

Select “Direct IP Call”.

Press OK key to bring up the input inte
rface and then key in the 12
digit target IP address. Press
OK key to initiate call.

For example:

If the target IP address is and the port is 5062 (e.g., input the following: 192*168*1*60#5062

“*” key represent “.” and
key represent “:”.

Press OK to dial out.

Or, you can select:

Use Quick IP
call mode:



Then use “#xxx#” to call where xxx is the last octet of the IP address if both phones are in the
same LAN.

For example:

If the target IP address is and use default port 5060, then you can
just off hook the phone and press “#60#” or “#60” followed by SEND key to make the quick IP
call. This is very useful in an enclosed LAN environment where you can assign sta
tic IP based
on, for example, room number in a building.


How do I set up the video phone

when my internet bandwidth connection is very low?

You need at least

bandwidth in both directions (uplink and downlink) to make the
3000 work with g
ood picture quality.

If using less than
, for example 96kbps bandwidth, you can configure:
Frame Rate to 5
(lowest) and Video Bit Rate to 64kbps or even 32kbps (lowest)
to see if the video quality is
acceptable in your network environment. Otherwis
e, please upgrade your bandwidth as you don’t
have enough physical data pipe to run the video data stream.

Detailed raw data rate (without using video codec compression) can be roughly calculated as

If the LCD is 320x240 for QVGA and one pixel nee
ds one bit (black or white) to present
(actually it is more because of color), then the data capacity for ONE frame is:
320 x 240 = 76.8

If the frame rate is 5 frames per second, then the total data stream (raw data for video without

76.8 x 5 = 384 kbps

Also, you need to add the additional audio data rate (details please refer to
Codec FAQ


With the H.264 codec, the video data can be
compressed. But to ensure good picture quality
having the
necessary bandwidth is the KEY

to a good picture because so much data that needs
to be transmitted simultaneously in both directions.


What is the meaning of the letters displayed on the LCD scree
n when a call is made?

U: This means the Audio / Video codec used in the call. It shows audio is using
G.711 µlaw (PCMU) and video is using H.264 video codec.

U: This shows

video bandwidth used in the call (OSD always shows
r side bandwidth used and suppose you should know your own sending video bandwidth as
this video bit rate is configured by you in the configurations).

For example: “86K/15F” means you are receiving video bit rate using 86Kbps and 15F means it
is using 15 f
rame per second to send the video.

The top left and right time stamp shows
“current time”

“call duration”


Can I connect a TV monitor to the GXV



Can I broadcast the video from the GXV3000 into my local network?

No, if you mean
live call broadcasting or multicasting in the LAN using the phone as source.

Yes, if you mean a video conference using server or MCU as source.

Also, you can configure the phone in “Video Surveillance” mode and get the video stream
packets in the same LAN

via the monitor PCs. For more detailed information, please refer to the
“Video Surveillance” section of GXV
3000 User Manual.


Can I take a snapshot of the image in the LCD monitor?

Yes. Once a USB flash drive is connected and ready for use (an icon
will show up in lower part
of the LCD when it is ready), the user can save a snap shot or capture a picture on the LCD and
stored into the flash drive in YUV format. For detailed information, please refer to the
“Auxillary Ports” section of GXV
3000 User M



Why is the plug
in not displayed in Microsoft Internet Explorer?

Trouble Shooting 1: Is the Plug
in properly installed?

How to solve: You must install the plug
in when you access the web GUI of the GXV3501/3504
for the first time
. here are two ways to install it:

A: Install the certified ActiveX control

B: Download the program or copy it from a disk, unzip the files to a temporary directory, go into
the directory, close Internet Explorer, double click on install.bat to install t
he Active X control


Why can’t I access the GXV 3501/3504 web configuration interface?

Trouble Shooting 1: Is your internet service down?

How to solve: Connect a PC to the internet to test the connection

Trouble Shooting 2: Are the PC and the encoder o
n different subnets?

How to solve: Check the subnet mask and default gateway of the encoder and PC

Trouble Shooting 3: Is there conflict with another IP address?

How to solve: Try to change the IP address of the device

Trouble Shooting 4: Has the HTTP po
rt been changed?

How to solve: Contact the administrator of the device for more information


Why is video playback not working in the web interface?

Trouble shooting 1: The plug
in is not installed or is not installed properly

How to solve: Install the

in again

How can I use a cell phone to watch the video stream on the DVS?

You must set the video resolution to QCIF if you want to watch the DVS video stream from a
cell phone. Make sure to set the bit rate to 64kbps to ensure the best video


Why doesn’t the IP address of the device reset when I click the “Restore” button on

the Maintenance page?

The DVS could be installed in areas that are not easy to access. For example, it could be installed
on the roof of a building or the cei
ling of an office. This makes it difficult to reinstall the device,
therefore the “Restore” function will not clear the IP address. Press the RESET button on the
device for at least 6 seconds until you hear a beep to perform a factory reset of all paramete
(including the IP address).


Why can’t users watch the live video stream using a mobile phone or GSurf after
changing the HTTP Port of the device?

Make sure that the RTSP port of the device is set to 2000 plus the HTTP Port number. For
example, if t
he HTTP port is 88, then the RTSP port of the device that you configure on GSurf /
mobile phone should be 2088.


How do you connect different types of detectors (such as infrared & smoke detectors) to
the DVS?

Sample connection diagrams are shown belo

Figure 1: Sample Alarm
in Connection Diagram 1

Figure 2: Sample Alarm
in Connection Diagram 2


How do you connect Alarm equipment to the Alarm
Out connection on the DVS?

A Sample connection diagram is shown:


How does PTZ work with the DVS

Follow these steps to configure the PTZ function on the DVS:

1. Connect the PTZ device to the DVS. A sample connection diagram is shown below.

Figure: Sample PTZ Device Connection Diagram

2. Configure the PTZ related parameters. Log in to the web GUI o
f the DVS, go to the PTZ page
and configure the PTZ protocol and Baud rate according to the specs of the PTZ device.

3. Reboot the DVS.


1. The FOCUS function may not work as many dome cameras support auto focus.

2. Press and hold the corresponding
control button to adjust the pan, tilt, zoom and speed.


Some notes when using SD cards / USB drives.


1. The DVS only supports FAT32 formatted USB drives

2. The DVS supports SD and SDHC

3. It takes 10
15 seconds to read SDHC and USB drives with
large memory capacities. Please
wait 15 seconds to unplug the SD/USB drive after you plug them into the device.

4. If there are many files (ie. 1800 or more image batch files) on the SD/USB drive, it can take
up to 5 minutes to read them. Please do not ref
resh the web interface at this time as the DVS will
restart reading the SD/USB drive. Grandstream is currently working on a fix for this issue.


Why is there a black / flashing bar at the bottom of the video feed?

This can occur if the DVS does not recog
nize the standard of the connected camera. If you
experience this issue, please restart the DVS. The DVS will detect the standard of the connected
camera and use that standard when the DVS boots up. To avoid this make sure to connect analog
cameras before
the DVS boots up.


Port forwarding

Two ports must be forwarded on your router to watch video from a DVS that is located on a
private network from a PC in a public network. The web port (HTTP) and the RTSP port. Please
make note that the RTSP port number
is changes according to the web port. If the web port is 80,
then the RTSP port is 554. If the web port is not 80, then the RTSP port equals the web port
+2000. For example, if the web port is 88, then the RTSP port will be



How do you connect Alar
m equipment to the Alarm
Out connection on the DVS?

GXW IP Analog Gateway Series FAQs


How do I specify different settings for different channels?

Check the syntax for your required setting.

For example

hook dial setting

if you want all
incoming PSTN calls off
hook auto dial to the same extension 200, use the syntax:



if you prefer channel 1 to off
hook auto dial to extension 200, channels 2
to extension 201, and channel 8 to extension 202, use the following synt



What’s the difference between one
stage dialing and two
stage dialing

One stage dialing

means the end user hears a dial tone immediately and can place a call.

stage dialing

means the end user has to dial twice

once to reach a second dial tone, and
again to reach the final destination.

In other words, a call traversing from PSTN to VoIP or VoIP
to PSTN must go through two dialing stages to reach the intended recipient.

For example
, a
VoIP user will complete t

first stage

by calling a pre
programmed number (on the GXW410x)
and receive the PSTN dial tone in return.

The user will then complete the

second stage

dialing a PSTN number.

The reverse happens for PSTN to VoIP calls.

If one
stage dialing is
used, a call from either the VoIP or PSTN side will pass through the
gateway to the intended recipient.

With one
stage dialing, a PSTN number dialed by a VoIP user
will reach the GXW410x, and the GXW410x will immediately dial the call on the PSTN side.

or PSTN to VoIP, an incoming PSTN caller will be directly connected with a VoIP extension.

stage dialing offers convenient and streamlined dialing, while two
stage dialing offers
flexibility and complete access to the VoIP network.


Why does the

GXW410x not pick up incoming PSTN calls using one
stage dialing?

Make sure that you have a valid extension entered in the

Unconditional Call Forward to
setting under the FXO Lines web configuration page.

Also, make sure that that extension is
le by the GXW410x.

For example, if using Asterisk™ and you have a SIP account with
an ID of 200 but there is no extension 200 in the context the GXW410x resides in, the
GXW410x will not be able to reach that SIP user and will not pick up the incoming PSTN



Why can I receive incoming PSTN calls, but outgoing calls (one
stage or two
can't grab a PSTN line?

You may have compatibility issues with some Verizon, Qwest, or certain other PSTN lines.

The following issues may occur:

stage VoIP
to PSTN calls cause the line LED to turn on and hang up immediately

stage VoIP to PSTN calls, GXW410x does not pick up a PSTN line

Incoming PSTN calls do not have caller ID information (if caller ID information is

To fix the above issues:

Contact Grandstream Support for the latest firmware or visit

Set the following values in the FXO Lines web configuration page:


Enable Current Disconnect to Yes (if the PSTN provider utilizes



Current Disconnect Threshold: 300


Min Delay Before Dial PSTN: 750


What should I do if I encounter port hang after a few calls?

Generally, port hang is caused by inability to detect the disconnect signal from the side that
hangs u
p first; usually it is when the PSTN side hangs up first.

Follow these steps to fix this

Call your PSTN carrier and ask what kind of PSTN disconnect signal is used −


(or current disconnect) or

tone disconnect


If it is

power dis

(common in most developed countries), ask for the
duration of power loss as a disconnect signal.



tone disconnect

is used, ask for the disconnect tone frequency and cadence, as
well as AC Termination impedance.


configure the GXW410x to reflect

the disconnect signal used by the PSTN

Power Disconnect:

navigate to FXO Lines in the web configuration
pages and set Enable Current Disconnect to



Make sure the correct
duration of power loss is set in Current Disconnect Threshold.


Tone Disconnect to “


Tone Disconnect:

navigate to FXO Lines in the web configuration pages
and set Enable Current Disconnect to



configure the AC
Termination Impedance and the disconnect tone frequency and cadence
(usually Reorder Tone

or Busy Tone under Channels in the web
configuration pages) to match your PSTN provider.


Reboot the GXW and place some test calls. If the port hang issue is not fixed,
contact Grandstream Technical Support.


What is the difference between an FXO and
an FXS Gateway?

FXO Gateway (GXW410x)


Its allows IP networks to talk to PSTN networks through the
gateway by simply connecting analog lines to it. Primarily used to allow remote IP endpoints to
be able to use local main office PSTN lines.

FXS Gateway (


It allows your traditional telephone system to function as an IP
System by simply connecting analog FXS trunks or analog handsets to the gateway. Primarily
used to allow traditional handsets/PABX to be used in an IP environment.

Q: How many co
ncurrent calls can be made on the GXW4024?

The GXW4024 will support 24 concurrent calls for all codecs.


Does the GXW series support RFC 3960?

Yes. The GXW400x supports RFC 3960

media / ring


Does the GXW series support RFC 3264?


The GXW400x supports RFC 3264



Does the GXW series support RFC 3515?

Yes. The GXW400x supports RFC 3515

refer method


Does the GXW series support RFC 3262?


The GXW400x supports RFC 3263

DNS locating

Q: How many simultaneous
fax calls can the GXW40xx handle?

The GXW40xx series can handle up to two T.38 simultaneous sessions and G.711 Pass
on the remaining ports.

GXE502x Series FAQs


How do I make my own IVR for Auto

You will need to record your IVR with
8000Hz/MONO/16bit in wav format, then upload the
IVR file to the GXE5000 under Auto
Attendant setting.

You can preview the recording.


Can I use phone to record IVR for Auto

Not with the current firmware.

The future firmware will allow use
r to record IVR via any


What is the USB port for?

Currently the USB port is used for internal troubleshooting file dump.

Grandstream will provide
more usage for USB in the future, if you have any suggestions, please send your
ion to


When there is a call coming from PSTN, I can hear some humming noise, what’s

In general, the humming noise come from the none
grounded power source, it could be from the
PSU to GXE5000, or could be from the s
witch/hub’s PSU that connected to the GXE5000
WAN/LAN port.

It is recommended to use a Ground PSU for most of the network environment.
Grandstream is working on improving the PSU for GXE5000.

If you have any questions, please

for troubleshooting.


When there is a call between PSTN and internal extension via build
in FXO port, the
VoIP side can hear echo, what’s wrong?

Echo is generated because of incompatible AC Impedance o
n FXO lines.

In general, the
Impedance is chosen base on the country where the GXE5000 is, if, however, it’s not working
well, you can chose using “Model” and select the corresponding Impedance number till it
matches and does not generate echo.


How d
o I use the GXE5000 to take incoming faxes?

The GXE5000 has a build
in FAX server that can take any incoming faxes. It allows users to
designate a particular extension for incoming faxes, it auto
detects incoming faxes and converts
the incoming fax to a PD
F file and sends it to the designate person via email.

GXE5000 also
allow different extension to send faxes to each other.


Do I need to reboot GXE5000 to turn on Syslog?


By default, the Syslog is already turned on to report at ERROR level, you
can choose any
other level including DEBUG, it will be activated immediately after selecting the level.


Does the GXE5000 supports multi
layer voice menu for Auto


Users can create multiple Voice Menus and choose “Other Voice Menu” to c
multiple voice menus.


Can you peer multiple GXEs through DDNS as oppose to using a static IP address?

Yes, it is possible to use a dynamic domain name. However the current firmware does not
support TZO so if the IP address changes the GXE does not have the client to update it.


Can you prepend an area code as a prefix on the FXO ports for outgoing cal

No, at this time you can only put a prepend prefix on SIP trunks.


When entering the voicemail system my extension # and password are not recognized by
the GXE.

How can I fix this?

The GXE502X uses RFC 2833 for its DTMF tones.

Make sure that all
registered SIP devices
have their DTMF mode set to RFC 2833.


When I enter the voicemail feature code:

*99 I do not hear an IVR?

What can I do?

Your GXE may not have come pre
loaded with the newest voice prompts.

Click here

download the newest voice prompts.


Why doesn’t anything happen when I try to dial in or out of my PSTN lines?

Most likely you did not set up the line call control for

the FXO port that the PSTN line is
connected to.

Click on the
Trunk/Phone Lines

section of the web UI and go to the
Internal PSTN
Trunk Line

configuration page.

Go to the
Line Call Control

field at the bottom the page and set
up the outbound prefix and
inbound call flow for each FXO port that is connected to a phone

Save your changes and reboot the GXE.

Your analog phone lines should now work


Why I can not read the PDF file for FAX?

You will need to upgrade Adobe
Read software to
V6.0 or V7.0. The PDF file generated for
GXE5000 cannot be opened with Adobe
Read V2.0.


How does feature code work?

Feature code is pretty much self

to use * code, make sure put a * after the feature
code., e.g, feature code *72 is * co
de for “Unconditional Call Forward”,

if you want to enable it
for extension 8003, then the dial sequence should be:



Why can't I see Caller ID from incoming PSTN calls?

You can try to increase the minimum RX level for FSK Caller ID from de
40 db) to higher
level (From
40 to 0 db) and it might solve the caller ID issue.

Please be advised that this may
also introduce the noise from PSTN line as well.


I am using an HT503 as an external PSTN trunk, why aren't the GXE’s feature code

For feature codes to work on an HT503 registered with the GXE, you must set the

Enable call

radio button to “No” and use the following syntax for the dial plan:

plan = {*x+}


After adding agents to a call queue, why are calls

coming into the queue not
automatically routed to the agents even when they are not busy?

You probably forgot to set the “Publish for Presence” field to YES on the agent’s account page
(on the phone web UI).


How do Call Park and Call Pick up work?


Park is done using an Attended

For example, if you have a GXP2000 that takes
an incoming call on Line 1, you can press Line 2(or any line in this case) to automatically put
Line 1 on hold.

You can then enter your Call
Park feature code (ie. *7
5), this will prompt a
parking announcement:

“Parked At 000.”

You can then press Transfer followed by the Line1
button to transfer the call on Line1 into parking extension 000.

To pick up the call, dial
*000 to pick up the call, assuming *76 is your

Pickup feature code.

The parking
extension number starts at 000.


Is there a reference table for defining the Time Zone?

Yes, please
click here


How do I setup multi
office trunking with keeping the Outbound prefix 9 for all?

You will need to peer two GXE units and make sure that the session keep alive feature is

This way you will be able to share trunks between the

two locations.


How do I configure a softphone on my computer that is connected to Wireless AP and
then GXE on LAN?

You will most likely have to forward the proper ports on your Wireless access point.


Do Grandstream

phones/GXE502X support IPSEC LAN to LAN VPNs?

No, IPSEC is not supported on the GXE or Grandstream phones at this time.


Is there a recommended STUN server?

You can find a list of free public STUN servers at:


port forwarding need to be configured for a remote extension?

Port forwarding may need to be configured in addition to DMZ a, UPnP and STUN.

This all
depends on how your network is configured.


Is the GXE able to have shared line appearances like regula
r a PBX?

Yes, the GXE does support shared line appearance/BLF/Presence watching.


Do you recommend using a double NAT with the GXE?

Grandstream does not recommend using a double NAT unless you are a very experienced
VoIP/Networking professional.

Some d
ouble NAT configurations can be traversed by using
STUN, UPnP and port forwarding.


How do I handle a SIP trunk for inbound and outbound call to another third party SIP
server when the GXE is inside a NAT?

You will most likely have to use STUN, UPnP or

port forward