• Products
      • UCM6300 Ecosystem
          • Wave
          • UCM RemoteConnect
          • UCM6300 Series
      • Business Conferencing
          • Full HD Conferencing
          • Audio Conferencing
      • IP Video Telephony
          • IP Video Phones
          • Extension Modules
      • Networking Solutions
          • Wi-Fi Access Points
          • Wi-Fi Management
          • Gigabit Routers
      • Personal Collaboration Devices
          • Headsets
          • Webcams
      • IP PBXs
          • UCM Series IP PBXs
      • IP Voice Telephony
          • Carrier-Grade IP Phones
              • GRP Series Essential IP Phones
              • GRP Series Mid-Range IP Phones
          • GXP Series IP Phones
              • GXP Series High-End IP Phones
              • GXP Series Basic IP Phones
              • GXP Series Mid Range IP Phones
          • Softphone App
          • Extension Modules
          • DECT Cordless
          • WiFi Cordless
      • Facility Management
          • Full HD IP Cameras
          • Encoders/Decoders
          • Video Management
          • Intercoms & Paging
          • Facility Access Systems
          • Control Stations
      • Device Management
          • GDMS
      • Gateways & ATAs
          • VoIP Gateways
          • Analog Telephone Adaptors
    • null
  • Solutions
      • Business Solutions
          • Industry Vertical
          • Unified Communications
          • WiFi Networking
          • WiFi Voice & Video
          • Remote Work and Collaboration
          • Personal Collaboration
          • UCM6300 Ecosystem
      • Case studies
    • Grandstream Networks Solutions

  • Support
      • Resources
          • Find tools and documents for your products
      • Faq
          • Find answers to all your questions
      • Forum
          • Get help from the community
      • Helpdesk
          • Submit and manage your tickets
      • Product Archive
          • Our Product Archive
      • Firmware
          • Get the most up-to-date firmware for your devices
      • Tools
          • Get useful tools
    • Grandstream Networks Support

  • Events
      • Webinars
          • Learn more about our products and partnership programs
      • Trainings
          • Get to know how our products work
      • Trade Shows
          • Visit us at an upcoming trade show to see a full working demo of our products
  • Education
      • Grandstream Academy
          • Central portal for official Grandstream Certification Trainings
      • Learning Center
          • Find educational resources on our products and features
  • Company
      • About Grandstream
          • Contact us
      • Partners
          • Become A Partner
          • Platform Partners
          • Service Providers
          • Technology Partners
          • Unauthorized Grandstream Reseller List
      • Careers
      • Press Releases
      • Awards
      • Legal
          • Grandstream Warranty Policy
          • Minimum Advertise Price Policy
          • Online Marketplace Warranty Limitations Policy
          • Online Marketplace Seller Validation

    • Grandstream

  • Blog
close×

Search form

  • Home
  • Support
  • FAQ
  • Product Related Questions
  • HandyTone Series

HandyTone Series

Can I call from FXS1 port to FXS2 port?

Yes, they can communicate with each other by dialing the respective extension number. Both FXS ports need a valid sip account registered on the server. Route call automatically and transparently to PSTN line according to user configuration

Can I use a repeater with the DP715 to increase the coverage?

The DP715 cannot be used with a repeater to increase the coverage range.

Can we prevent random calls from any IP by limiting to accept calls from the SIP server only?

Yes. A user can enable a feature called "Allow Incoming SIP Messages from SIP Proxy Only". This field can be found under the FXS port configuration page in the web-gui.

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:

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

Does Grandstream allow HTTPS Provisionig with Grandstream Internal Certificate?

Yes. Grandstream allows HTTPS provisioning with Grandstream's very own certificate. If the server is configured to require a certificate for authentication, our devices can send an internal certificate that is flashed on all Grandstream devices. If you would like to use this feature please get in contact with our Grandstream Support for the GS Certificate.
Note: This is not the same as the "HTTP/HTTPS" user/password, which in this case this option is for authentication only once the connection has been established.

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 FLASH button on the phone, instead of doing hook/flash. Note: This feature is not available on older hardware revision models.

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 below:
:
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 (192.168.2.2):
Now, open Internet Explorer and type in 192.168.2.1, you should see Grandstream Login Screen pop up.:
Enter 'admin' as the password:
Go to Advanced Settings page and switch "Enable 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:
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).

How do I configure my HT503 to call me when my Grandstream IP camera's alarm is triggered?

Click here to learn how!

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.

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.

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.

How do I setup a 3-way conference 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 does the DTMF negotation work on the HT5xx series?

How the DTMF negotiation works: DTMF method negotiation: As a Caller/Callee: 1. If disable DTMF negotiation, DUT will use the first dtmf method from webUI. 2. If enable DTMF negotiation, DUT will check if there is RFC2833 from the SDP of 200OK (for Caller) or INVITE (for Callee). If it has RFC2833, DUT will check if itself has RFC2833 in DTMF list. If DUT also has RFC2833, then RFC2833 is chosen. Otherwise, DUT will check if there is SIP INFO supported by callee/caller. And then check itself. If DUT also support SIP INFOin DTMF list, then DUT uses SIP INFO. Otherwise, DUT directly use IN_AUDIO.

Note: If negotiation is enabled, the priority of DTMF order doesn’t matter. DUT will first try to match RFC2833 and then SIP INFO, the least priority is In-Audio. If negotiation is disabled, DUT will use the first DTMF method configured in webUI.

What are the main differences between the 9 models of Handytones?

Features

HT286

HT386

HT486

HT488

HT503

HT496

HT502

HT701 HT702 HT704

Ethernet Ports

1 RJ45
(LAN)

1 RJ45
(LAN)

2 RJ45
(LAN/WAN)

2 RJ45
(LAN/WAN)

2 RJ45
(LAN/WAN)

2 RJ45
(LAN/WAN)

1 RJ45
(LAN)
1 RJ45
(LAN)
1 RJ45
(LAN)

DHCP/NAT/Router

No

No

Yes

Yes

Yes

Yes

No No No

FXS Port

1

2

1

1

2

2

1 2 4

FXO Port

No

No

No

1

No

No

No No No

PSTN Pass-through Port

No

Yes

Yes

Yes

No

No

No No No

Remote Configuration

TFTP/HTTP

TFTP/HTTP

TFTP/HTTP

TFTP/HTTP

TFTP/HTTP

TFTP/HTTP

TFTP/HTTP


TFTP/HTTP


TFTP/HTTP


What format does the Grandstream Config Generator encrypt the file into?

Our configuration files are encypted with 128 AES. As for the new XML format, it will be decrypted with 256 AES.

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 Configuration 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 1.0.7.19 firmware, while Rev 3.0 can upgrade to the latest release.:
Note: Once there is a new hardware revision out in the market, the older revision is no more manufactured or sold.

What is a dial-plan and how do I configure it?

The dial-plan is a set of rules that governs the call-routing behavior of a device. When a user dials a sequence of number the device will refer to the rules in the dial-plan in order to determine how best to connect the call. Users can refer to the user manual for more details on how to configure the dial-plan. All user manual for your products can be found here

=

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 indicates 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 port.
Slow blinking of WAN and LAN together means the product’s firmware is in upgrading or provisioning state.

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.

Need more help?


Resources
Firmwares, tools and documents

FAQ
Find answers to all your questions

Forums
Get help from the community

Helpdesk
Submit and manage your tickets
  • Common Questions
    • Informational
    • Firmware
    • Video/Voice/Speech Codecs
    • Custom Ringtone FAQ
    • Common Configuration Questions
    • Typical Service Provider Configurations
    • Direct IP Call
    • Headset Compatibility
  • Product Related Questions
    • GDMS Grandstream Device Management System
    • IPVideoTalk Service
      • GVC Related FAQ
      • IPVideoTalk Portal Related FAQ
      • Plan Related FAQ
      • Security Related FAQ
      • WebRTC Related FAQ
    • UCM6300 Series IP PBX
      • GDMS Related FAQ
      • PBX Related FAQ
      • Wave Web/Mobile App Related FAQ
    • WP810 WiFi Cordless
    • WP820 WiFi Cordless
    • GVC3210
    • GWN series
    • GAC2500
    • Wave Lite
    • GVC3200/GVC3202
    • GVR355X NVR
    • GVR3552 NVR
    • GVR3550 NVR
    • GXV3240/3275 IP Multimedia Phones
    • GXP2130/2140/2160 IP Phones
    • DP715/710 Series
      • DP720/750Series
    • HandyTone Series
    • BudgeTone Series
    • GXP Enterprise Phone Series
    • GXP2200 Enterprise Multimedia Phone for Android
    • GXV3140 IP Multimedia Phone
      • Basic Installation and Settings
      • Registering the Device
      • Basic Features
      • Personalize
      • External Devices
      • Making/Receiving Calls
      • Call Features
      • Voice and Video Mail
    • GXV3175 IP Multimedia Phone Touchscreen
    • GXV350x IP Video Encoder
    • GXW IP Analog Gateway Series
    • Surveillance Series
    • UCM6100 Series
    • GXP2000 Enterprise Phone
  • Troubleshooting
    • Troubleshooting Questions
  • Provisioning
    • Device Provisioning
  • Interoperability
    • Interoperability Questions
    • Open Source Configuration
  • GNU General Public License
    • GNU GPL Information & Download

Information

  • Products
  • Solutions
  • Support
  • Events
  • Partners
  • Privacy Policy

Contact

  • About Grandstream
  • Contact us

Newsletter Signup

Get the latest news in exclusivity

Subscribe to our newsletter

Follow Us

Follow us on social media

© 2020 Grandstream Networks, Inc.
Site Map

Corporate Headquarters
126 Brookline Ave, 3rd Floor Boston, MA 02215