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Grandstream Networks HandyTone HT286 Product Related Questions

Grandstream Networks HandyTone HT286 Product Related Questions

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HandyTone Series FAQs
Q: What are the main differences between the 7 models of Handytones?
Features
HT286
1 RJ45
Ethernet Ports
(LAN)
DHCP/NAT/Route
No
r
FXS Port
1
FXO Port
No
PSTN Pass-
No
through Port
Remote
TFTP/HTT
Configuration
P
Q: 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
before 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 CANNOT do:
Terminate 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
Note: On the HT486 Rev 1.0, the PSTN port is only a life line 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:
Product Related Questions
HT386
HT486
2 RJ45
1 RJ45
(LAN/WAN
(LAN)
)
No
Yes
2
1
No
No
Yes
Yes
TFTP/HTT
TFTP/HTTP TFTP/HTTP TFTP/HTTP TFTP/HTTP
P
HT488
HT496
HT503
2 RJ45
2 RJ45
(LAN/WAN
(LAN/WAN
)
)
Yes
Yes
1
2
1
No
Yes
No
HT502
2 RJ45
(LAN/WAN
)
Yes
2
No
No

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Summary of Contents for Grandstream Networks HandyTone HT286

  • Page 1 Product Related Questions HandyTone Series FAQs Q: What are the main differences between the 7 models of Handytones? HT488 Features HT286 HT386 HT486 HT496 HT502 HT503 2 RJ45 2 RJ45 2 RJ45 2 RJ45 1 RJ45 1 RJ45 Ethernet Ports (LAN/WAN (LAN/WAN (LAN/WAN...
  • Page 2  Terminate a VoIP call into the PSTN port  Allow a PSTN call to call either the FXS phone or other VoIP devices over the IP network  Route call automatically and transparently to PSTN line according to user configuration Q: Can I call from FXS1 port to FXS2 port? Yes, they can communicate with each other by dialing the respective extension number.
  • Page 3 9. Disconnect your PC from the LAN port and connect it to any other port on your Router within the same LAN Segment 10. Type in the actual IP ADDRESS of the HT (You can look this up by pressing *** 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).
  • Page 4 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 PORT configuration page: {L: 404x+| L:770x+ | L:678x+| L:911 | x+} Please note that only HT503 supports this feature.
  • Page 5 1. PCMU or g711(a/µ law) 2. G.729 A/B 3. G.723.1 4. G.722 (wideband) 5. G.726-32 6. iLBC BT200 supports PCMU, PCMA, g729 A/B, g723.1 and GSM codecs. Q: How do I configure a Public or Static IP address using the phone menu? This is a very crucial configuration.
  • Page 6 Attend Transfer: 1. A and B are in a call 2. B needs to transfer the call to C 3. B presses FLASH to get new dialtone 4. B calls C while A is on Hold 5. Now, B can transfer the call from A to C by pressing TRANSFER and then hanging up. 6.
  • Page 7 Alert-Info:;info=priority Q: Does the BudgeTone support Syslog Server 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 device. Q: Where do I view my Call History ex.
  • Page 8 Note: 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. Q: 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).
  • Page 9 Allow Auto Answer by Call-Info 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-Info Note: The PBX Server has to support this feature to make it work. Q: Is the GXP2020 Expansion Module compatible with the GXP2120/GXP2110 models? Yes, it is compatible.
  • Page 10 GXP2000 FAQs Q: 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, etc.
  • Page 11 Yes, any text entered 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. Q: Can I enable/disable recording of missed calls? Yes, on each Account Page, there is a field for ‘Disabling Missed Calls’.
  • Page 12 192.168.0.2 calling 192.168.0.123 just dial #123 follow by SEND or # 192.168.0.2 dial #3 and #03 and #003 has same effect --> call 192.168.0.3 Note: 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.
  • Page 13 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 (0-23), minute: minute (0-59) 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"...
  • Page 14 No, the GXP2000 extension module is only compatible with the GXP2000. Extension modules for other GXP phones will be available in 2008. GXV3000 FAQS Q: 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™...
  • Page 15 call. This is very useful in an enclosed LAN environment where you can assign static IP based on, for example, room number in a building. Q: How do I set up the video phone when my internet bandwidth connection is very low? You need at least 128kbps bandwidth in both directions (uplink and downlink) to make the GXV-3000 work with good picture quality.
  • Page 16 Yes. Q: 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”...
  • Page 17 Q: 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 on 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...
  • Page 18 Q: 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 the HTTP port is 88, then the RTSP port of the device that you configure on GSurf / mobile phone should be 2088.
  • Page 19 Q: How do you connect Alarm equipment to the Alarm-Out connection on the DVS? A Sample connection diagram is shown:...
  • Page 20 Q: 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...
  • Page 21 2. Configure the PTZ related parameters. Log in to the web GUI of 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. NOTE: 1.
  • Page 22 up to 5 minutes to read them. Please do not refresh 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. Q: Why is there a black / flashing bar at the bottom of the video feed? This can occur if the DVS does not recognize the standard of the connected camera.
  • Page 23 GXW IP Analog Gateway Series FAQs Q: How do I specify different settings for different channels? Check the syntax for your required setting. For example, off-hook dial setting: if you want all incoming PSTN calls off-hook auto dial to the same extension 200, use the syntax: "ch1- 8:200;".
  • Page 24 Two-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 the first stage by calling a pre-programmed number (on the GXW410x) and receive the PSTN dial tone in return.
  • Page 25  Contact Grandstream Support for the latest firmware or visit www.grandstream.com.  Set the following values in the FXO Lines web configuration page: Enable Current Disconnect to Yes (if the PSTN provider utilizes Current Disconnect). Current Disconnect Threshold: 300 Min Delay Before Dial PSTN: 750 Q: 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 up first;...
  • Page 26 Q: How many concurrent calls can be made on the GXW4024? The GXW4024 will support 24 concurrent calls for all codecs. Q: Does the GXW series support RFC 3960? Yes. The GXW400x supports RFC 3960 - early-media / ring-tone Q: Does the GXW series support RFC 3264? Yes.
  • Page 27 Q: 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 recommendation to gxebetatest@grandstream.com. Q: When there is a call coming from PSTN, I can hear some humming noise, what’s wrong? 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 switch/hub’s PSU that connected to the GXE5000...
  • Page 28 Q: 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. Q: Can you prepend an area code as a prefix on the FXO ports for outgoing calls? No, at this time you can only put a prepend prefix on SIP trunks.
  • Page 29 Q: 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 default(-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.
  • Page 30 No, IPSEC is not supported on the GXE or Grandstream phones at this time. Q: Is there a recommended STUN server? You can find a list of free public STUN servers at: http://www.voip-info.org/wiki-STUN Q: Will 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.