Grandstream HandyTone 802 - HT802 - VoIP.ms Wiki

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Grandstream HandyTone 802 - HT802

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===Configuring the HandyTone 802===
===Configuring the HandyTone 802===
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'''Step 1 - Initial Configurations'''<br>
These instructions are based on HandyTone 802 software version 1.0.3.2 if you are running a different software version some menus and settings may be different.
These instructions are based on HandyTone 802 software version 1.0.3.2 if you are running a different software version some menus and settings may be different.
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====Plugging the HT802====
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'''Step #2 - Plugging the HT802'''<br>
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Now connect your phone to the HandyTone. Plugging the cable into the correct FXS Port that you configure.
Now connect your phone to the HandyTone. Plugging the cable into the correct FXS Port that you configure.
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'''''In most cases, you will only be setting yourself up with FXS PORT 1, so please make sure your telephone is connected to LINE #1.'''''
Finally plug the supplied power cable into the HandyTone.
Finally plug the supplied power cable into the HandyTone.
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====Getting IP address for the GUI====
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'''Step #3 - Getting IP address for the GUI'''<br>
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====Loging into the device====
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'''Step #4 - Logging into the device'''<br>
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You should now see a page that looks like this:
You should now see a page that looks like this:
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[[File:HT802_FirstPage.PNG|thumb|none|600px]]
[[File:HT802_FirstPage.PNG|thumb|none|600px]]
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====Configuring device's port FXS====
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'''Step #5 - Configuring device's port FXS'''<br>
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Now, click on FXS PORT1 and configure your settings accordingly (as shown below):
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Now, click on FXS PORT1 and configure your settings accordingly (as shown below).
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Please use the same server in the Failover SIP Server as your Primary SIP Server or leave the Failover SIP Server field Blank.
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* '''Primary SIP Server:''' toronto.voip.ms (Pick one of VoIP.ms multiple [http://wiki.voip.ms/article/Choosing_Server VoIP Servers])
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* '''Prefer Primary SIP Server:''' Set to Yes
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*'''Outbound Proxy:''' Set the same server you've configured at the '''Primary SIP Server''' field
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*'''NAT Traversal:''' Keep-Alive
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* '''SIP User ID:''' Replace with your Main SIP account or Subaccount UserID, e.g. 100000 or 100000_sub
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*'''Authenticate ID:''' Replace with your Main SIP account or Subaccount UserID, e.g. 100000 or 100000_sub
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* '''Authenticate Password:''' ********* (Use your SIP account password - by default this is the same as the Customer Portal)
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* '''''Name''''': Outbound CallerID Name (Optional)* '''See the requirements below.'''
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*'''DNS Mode:''' Set to "A Record"
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* '''SIP Registration:''' Set to Yes
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* '''Unregister On Reboot:''' Set to No
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* '''Outgoing Call Without Registration:''' Set to Yes
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* '''Register Expiration:''' Set to 5
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* '''Local RTP Port:''' Set to 10000
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* '''''Enable SIP OPTIONS/NOTIFY Keep Alive''''': OPTIONS
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* '''Allow Incoming SIP Messages from SIP Proxy Only:''' Set to Yes
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* '''Preferred DTMF method:''' In-audio, RFC2833
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* '''''Use P-Access-Network-Info Header (if present)''''': Set to No
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* '''''Use P-Emergency-info Header (if present)''''': Set to No
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* '''Enable Call Features:''' No
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* '''Dial Plan:''' {[x*]+}
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*'''Preferred Vocoder:''' PCMU, PCMA, G729
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Use the following settings to configure your VoIP.ms account: Configuration Page Settings:
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'''IMPORTANT''': Outbound CallerID Name
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  - We suggest entering your outbound Caller ID Name must be in '''capital letters'''. This will appears more clearly/visible on some devices.
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  - You must '''NOT''' use any special characters, they will not be displayed.
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  - Some of regular Canadian providers will not show more than '''15 characters max'''. We suggest shrinking or adapt your caller ID.
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  - Spaces are allowed in a caller id name.
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* '''''Primary SIP Server''''': servername.voip.ms (one of VoIP.ms multiple [[Choosing_Server#Choosing_a_Server | '''''servers''''']], you can choose the one closest to your location.)
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[[File:HT802_FXS_Port.jpg|thumb|none|600px]]
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'''Notice that it is necessary to use the same server for both the device and the DID number in order to get incoming calls correctly'''
 
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You can also find this information by logging into your Customer portal.
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'''Step 6 - Savings the changes'''<br>
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* '''''Failover SIP Server''''': (Please leave this Blank)
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Once you have configured the settings above, click the '''Update''' button and then the '''Reboot''' button to save the configurations. Your HT802 will power cycle after you click the reboot button. Please wait at least 30 seconds for the unit to finish power cycling. If you see that the Phone 1 LED (or phone 2 LED, depending on which FXS port you've configured our service for) is a solid blue color, then your unit is configured and ready to make calls.
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* '''''Outbound Proxy''''': servername.voip.ms (Use  the same server you used as '''''Primary SIP Server'''''
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'''That's it!''' You can now make a phone call.
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* '''''NAT Traversal''''': Keep-Alive
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The area code + the number for calls to the US & Canada
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* '''''SIP User ID''''': (Replace with your Main SIP account or Subaccount UserID, e.g. 100000 or 100000_sub)
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Or
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* '''''Authenticate ID''''': (Replace with your Main SIP account or Subaccount UserID, e.g. 100000 or 100000_sub)
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011 + the country code, area code, and number for calls worldwide (you may also use 00 instead of 011).
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* '''''Authenticate Password''''': ****** (Use the SIP account password - By default this is the same as the Customer Portal)
 
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* '''''DNS Mode''''': A Record
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== Call Encryption - TLS/SRTP ==
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* '''''SIP Registration''''': Yes
 
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* '''''Unregister On Reboot''''': No
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To use [[Call_Encryption_-_TLS/SRTP#Configuration_on_SIP_Client | encrypted calls (TLS)]], the following setting on the FXS Port must be changed:
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* '''''Outgoing Call Without Registration''''': Yes
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* '''''SIP Transport''''': TLS
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* '''''SRTP Mode''''': Enabled and forced
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* '''''Register Expiration''''': 5
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Also make sure '''''Local SIP Port''''' is now at “5061”.
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* '''''Allow Incoming SIP Messages from SIP Proxy Only''''': Yes
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-
 
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* '''''Preferred DTMF method''''': In-audio, RFC2833
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-
 
+
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* '''''Enable Call Features''''': No
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-
 
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* '''''Dial Plan''''': {[x*]+}
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-
 
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* '''''Preferred Vocoder''''': PCMU, PCMA, G729
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-
 
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[[File:HT802_FXS_Port.jpg|thumb|none|600px]]
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-
 
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====Saving the changes====
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Once you have configured the settings above, click the Update button and then the Reboot button to save the configurations.
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Your HT802 will power cycle after you click the reboot button. Please wait at least 30 seconds for the unit to finish power cycling. If you see that the Phone 1 LED (or phone 2 LED, depending on which FXS port you've configured our service for) is a solid blue color, then your unit is configured and ready to make calls.
+
-
 
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That's it! You can now make a phone call.
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-
 
+
-
The area code + the number for calls to the US & Canada
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-
 
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Or
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-
 
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-
011 + the country code, area code, and number for calls worldwide (you may also use 00 instead of 011).
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== Preventing Direct IP calls like 100 & 1000 ==
== Preventing Direct IP calls like 100 & 1000 ==
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'''Allow Incoming SIP Messages from SIP Proxy Only''' - Default is No. Check the incoming SIP messages. If they don’t come from the SIP proxy, they will be rejected. If this option is enabled, the device will not be able to make direct IP calls.
'''Allow Incoming SIP Messages from SIP Proxy Only''' - Default is No. Check the incoming SIP messages. If they don’t come from the SIP proxy, they will be rejected. If this option is enabled, the device will not be able to make direct IP calls.
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==Auto Provisioning==
 
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Some newer models of the HT802 now have Auto Provisioning and will delete the changes you make in setting up the device to use our service. Please go to your Graphical user interface and go to 'Advanced Settings' tab and look for "Firmware Upgrade and Provisioning" and disable it.
 
==Guide Links==
==Guide Links==
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In the event where you need the guides directly from Positron, you may find the user manual guides below:
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In the event where you need the guides directly from Grandstream, you may find the user and admin manual guides below:
User Manual : [http://www.grandstream.com/sites/default/files/Resources/ht80x_user_guide.pdf Download PDF]
User Manual : [http://www.grandstream.com/sites/default/files/Resources/ht80x_user_guide.pdf Download PDF]
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Admin Manual : [http://www.grandstream.com/sites/default/files/Resources/ht80x_administration_guide.pdf Download PDF]
Admin Manual : [http://www.grandstream.com/sites/default/files/Resources/ht80x_administration_guide.pdf Download PDF]
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[[Category:Analog Telephone Adapters]]

Latest revision as of 17:32, 6 March 2026

HT802 Device.jpg

The Grandstream HandyTone 802 is a reliable, inexpensive telephone adapter which works with the VoIP.ms service when placed after your broadband internet router.

Websites: Grandstream HT802

Help / Support: Grandstream Support


Contents



Configuring the HandyTone 802

Step 1 - Initial Configurations

These instructions are based on HandyTone 802 software version 1.0.3.2 if you are running a different software version some menus and settings may be different.

These instructions are also based on using the HandyTone in its factory default configuration, which obtains a dynamic IP address automatically from your router using DHCP. For information on configuring your HandyTone with a Static IP Address, please refer to the HandyTone user´s manual.

Each step is important in assuring that your device works properly.

We recommend that you read each step through in its entirety before performing the action indicated in the step.


Step #2 - Plugging the HT802


Connect your HandyTone to your router with the supplied Ethernet network cable.

Now connect your phone to the HandyTone. Plugging the cable into the correct FXS Port that you configure.

In most cases, you will only be setting yourself up with FXS PORT 1, so please make sure your telephone is connected to LINE #1.

Finally plug the supplied power cable into the HandyTone.


Step #3 - Getting IP address for the GUI


Wait 60 seconds after plugging your HT802 in. Pick up the phone connected to the HT802 and dial *** on it.

Please have a pen and paper ready. You will hear a message - "Enter a menu option", then enter 0 2 on your phone. You will now hear a message giving you the IP address of your HT802 such as - "192.168.001.010" and write this number down.

Open a web browser on your computer such as Chrome or Firefox and enter the IP address you heard in step 4 as the address (I.E. where you would normally enter www.voip.ms).

Please note: Some browsers will require you to remove leading zero's ( 0 's ) in the IP address. For example if you heard "192.168.001.010" you should change this to "192.168.1.10".

The Interface has a timeout so please make changes quickly or save/update your settings every couple of minutes.


Step #4 - Logging into the device


You should now see a page that looks like this:

HT802 Login.jpg

Enter the password for the HT802 in the password field. The default administrator password for the HT802 is admin

After entering the password you should see a screen that looks similar to the one below:

HT802 FirstPage.PNG

Step #5 - Configuring device's port FXS

Now, click on FXS PORT1 and configure your settings accordingly (as shown below).

  • Primary SIP Server: toronto.voip.ms (Pick one of VoIP.ms multiple VoIP Servers)
  • Prefer Primary SIP Server: Set to Yes
  • Outbound Proxy: Set the same server you've configured at the Primary SIP Server field
  • NAT Traversal: Keep-Alive
  • SIP User ID: Replace with your Main SIP account or Subaccount UserID, e.g. 100000 or 100000_sub
  • Authenticate ID: Replace with your Main SIP account or Subaccount UserID, e.g. 100000 or 100000_sub
  • Authenticate Password: ********* (Use your SIP account password - by default this is the same as the Customer Portal)
  • Name: Outbound CallerID Name (Optional)* See the requirements below.
  • DNS Mode: Set to "A Record"
  • SIP Registration: Set to Yes
  • Unregister On Reboot: Set to No
  • Outgoing Call Without Registration: Set to Yes
  • Register Expiration: Set to 5
  • Local RTP Port: Set to 10000
  • Enable SIP OPTIONS/NOTIFY Keep Alive: OPTIONS
  • Allow Incoming SIP Messages from SIP Proxy Only: Set to Yes
  • Preferred DTMF method: In-audio, RFC2833
  • Use P-Access-Network-Info Header (if present): Set to No
  • Use P-Emergency-info Header (if present): Set to No
  • Enable Call Features: No
  • Dial Plan: {[x*]+}
  • Preferred Vocoder: PCMU, PCMA, G729
IMPORTANT: Outbound CallerID Name
  - We suggest entering your outbound Caller ID Name must be in capital letters. This will appears more clearly/visible on some devices.
  - You must NOT use any special characters, they will not be displayed. 
  - Some of regular Canadian providers will not show more than 15 characters max. We suggest shrinking or adapt your caller ID. 
  - Spaces are allowed in a caller id name.
HT802 FXS Port.jpg


Step 6 - Savings the changes

Once you have configured the settings above, click the Update button and then the Reboot button to save the configurations. Your HT802 will power cycle after you click the reboot button. Please wait at least 30 seconds for the unit to finish power cycling. If you see that the Phone 1 LED (or phone 2 LED, depending on which FXS port you've configured our service for) is a solid blue color, then your unit is configured and ready to make calls.

That's it! You can now make a phone call.

The area code + the number for calls to the US & Canada

Or

011 + the country code, area code, and number for calls worldwide (you may also use 00 instead of 011).


Call Encryption - TLS/SRTP

To use encrypted calls (TLS), the following setting on the FXS Port must be changed:

  • SIP Transport: TLS
  • SRTP Mode: Enabled and forced

Also make sure Local SIP Port is now at “5061”.

Preventing Direct IP calls like 100 & 1000

To Prevent Direct IP calls to your device and only allow calls from our service please enable the following 2 options in your FXS Port Configuration Page.

Check SIP User ID for incoming INVITE - Default is No. Check the incoming SIP User ID in Request URI. If they don’t match, the call will be rejected. If this option is enabled, the device will not be able to make direct IP calls.


Allow Incoming SIP Messages from SIP Proxy Only - Default is No. Check the incoming SIP messages. If they don’t come from the SIP proxy, they will be rejected. If this option is enabled, the device will not be able to make direct IP calls.


Guide Links

In the event where you need the guides directly from Grandstream, you may find the user and admin manual guides below:

User Manual : Download PDF

Admin Manual : Download PDF

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