VoIP.ms Wiki - User contributions [en]

Grandstream GXP1630

2026-05-17T00:27:48Z

RogueScholar: Wikilink some terms to English Wikipedia and use the new Button template where helpful

[[File:Grandstream GXP1630 - Front View.png|frameless|right|440px|alt=Front-facing photo of a Grandstream GXP1630 IP phone|Grandstream GXP1630, front view]]
[[File:Grandstream GXP1630 - Side View.png|frameless|right|176px|alt=Side-facing photo of a Grandstream GXP1630 IP phone|Grandstream GXP1630, side view]]
The '''GXP1630''' is the most feature-rich model in [[wikipedia:Grandstream Networks|Grandstream Networks']] "Basic [[IP Phones]]" product line (identified by model numbers that begin with <code style="border: thin solid #ccc; padding: 1px 4px; white-space: collapse nowrap;">GXP16…</code>) of [[wikipedia:Sound-powered telephone|corded-handset]] [[wiktionary:deskphone#English|desk models]] marketed to [[wikipedia:Small and medium enterprises|small and medium-sized businesses]]. Directly quoting the manufacturer's website, the devices are described as:
<blockquote>"Basic IP phones designed for simple, intuitive, and effective communication in the hands of users who need it. Our [https://www.grandstream.com/products/ip-voice-telephony/gxp-series-ip-phones/gxp-series-basic-ip-phones GXP1600 series] of basic IP phones deliver interactive communications to a desktop for the user who needs access to VoIP but does not require advanced features. They are ideal for hotel rooms, information booths, and warehouse deployments for access to quick call control. Delivering a vibrant and clear user-interface, this device is a perfect solution for those who handle low-to-medium call volume and require access to key call efficiency functionalities."</blockquote>

Some common features of all models in this series are:
* Support for at least two lines with individual SIP registrations
* A [[wikipedia:Fast Ethernet#Copper|10/100Mbps Ethernet]] network connection
* HD audio
* Speakerphone
* Three-way [[Audio Conferencing|call conferencing]]

The GXP1630 improves on those by adding:
* Support for three (3) SIP accounts, each with a dedicated line button and programmable function key
* Four-way [[Audio Conferencing|call conferencing]]
* Two (2) switched [[wikipedia:Gigabit Ethernet|Gigabit Ethernet]] ports with integrated [[Power over Ethernet|Power over Ethernet (PoE)]]
* A backlit [[wikipedia:Liquid-crystal display|LCD display]], and
* Eight (8) [[Busy Lamp Field]] indicators
<div style="margin: 1em 0;"></div>

== Configuration ==
[[File:General GXP1630.png|thumb|right|360px|alt=Screenshot of GXP1630 General Settings page|'''Account''' ⇒ '''Account #''' ⇒ '''General Settings''']]
[[File:SIP Basic Grandstream GXP1630.png|thumb|right|360px|alt=Screenshot of GXP1630 Basic SIP Settings page|'''Account''' ⇒ '''Account #''' ⇒ '''SIP Settings''' ⇒ '''Basic Settings''']]
: ''Click the images at right to expand them, if needed.''
=== SIP Account ===
Once you have connected your device to your local network, it is first necessary to determine the IP address that was assigned to it in order to access the web browser-based administration controls.
# Using the physical navigation buttons on the phone, navigate through the menu using this sequence of choices:
#: '''<tt>Menu</tt>''' ⇒ '''<tt>Status</tt>''' ⇒ '''<tt>Network Status</tt>''' ⇒ '''<tt>IPv4 Address</tt>'''
# Enter the IP address shown on the device's LCD display into the address field of your preferred web browser.
# Use the following credentials at the login page which is displayed to access the available configuration options:
#: '''Username: <code>admin</code>'''
#: '''Password: <code>admin</code>'''

: '''''Note:''' Grandstream is gradually phasing out the use of ''<tt>admin</tt>'' as the default password for new models. As of 2023, it appears to still be working for this model, but should this become deprecated at some later date, check for a model/serial number sticker on the underside or back (or, for self-contained wireless handsets like the [[Grandstream WP810, WP820, WP822 and WP825|Grandstream WP8xx series]], in the battery compartment under the battery). The password will be identified by a preceding ''<tt>P/W:</tt>'', followed by a random string of characters unique to that individual device which is the factory default password for the ''<tt>admin</tt>'' account.''

=== General settings ===
Once you have authenticated and are shown the settings page, proceed to the indicated page and once there, change the following settings to match the details of your VoIP.ms account:
:: '''<tt>Account</tt>''' ⇒ '''<tt>Account #</tt>''' ⇒ '''<tt>General Settings</tt>'''
* '''Account Name:''' The name that will be displayed on the LCD screen of your phone.
* '''SIP Server:''' One of the VoIP.ms [[Choosing_Server#Choosing_a_Server|servers]], typically the one closest to your physical location.<br />&emsp;'''''Note: It is necessary to use the same server for both the device and the DID number in order to receive incoming calls.'''''
* '''SIP User ID:''' The account or sub-account number that you wish to use with this device.
* '''Authenticate Password:''' The password for that account or sub-account.
* '''Name:''' The name that will be transmitted as the Caller ID for outgoing calls ('''additional requirements below''')
*; &emsp;<span style="text-decoration: underline;">IMPORTANT:</span> : <ul style="margin: 0.3em 0 0 0.7em;"><li>We suggest entering your outbound Caller ID Name in '''capital letters'''. This will appear more clearly on most devices.</li><li>You must use only alpha-numeric characters; special characters, including standard punctuation, will not be displayed.</li><li>Enter a '''maximum of 15 characters''', as some Canadian providers will not show more than that; shrink or adapt your Caller ID accordingly.</li><li>Spaces are permitted in a Caller ID name, and count against the 15 character limit.</li></ul>
Click the {{Button|Save and Apply}} button.

=== Basic SIP settings ===
Once this is done, the time on the device will need to be set before it can send valid information for its registration status. To do that, navigate to the indicated page and make the following changes:
:: '''<tt>Account #</tt>''' ⇒ '''<tt>SIP Settings</tt>''' ⇒ '''<tt>Basic Settings</tt>'''
* '''SIP Registration:''' Yes
* '''Register Expiration:''' 5 (value specified in minutes)
* '''Enable OPTIONS Keep Alive:''' Yes
* '''SIP Transport:''' UDP or TCP
Finally, click {{Button|Save and Apply}}.

== Common issues ==
[[File:SIP Custom SIP Headers Grandstream GXP1630.png|thumb|right|360px|alt=Screenshot of GXP1630 Custom SIP Headers page|'''Accounts''' ⇒ '''Account #''' ⇒ '''SIP''' ⇒ '''Custom SIP Headers''']]

=== Outgoing calls ===
If you are able to receive incoming calls but outgoing calls are failing with error "No response", it is likely that the issue stems from either missing SIP headers or an unsupported audio codec. Login into your device's web-based configuration and navigate to:
:: '''<tt>Accounts</tt>''' ⇒ '''<tt>Account X</tt>''' ⇒ '''<tt>SIP</tt>''' ⇒ '''<tt>Custom SIP Header</tt>'''
On that page, ''deactivate'' these options:
* '''Use X-Grandstream-PBX Header'''
* '''Use P-Access-Network-Info Header'''
* '''Use P-Emergency-Info Header'''
Then navigate to the page below and change:
:: '''<tt>Accounts</tt>''' ⇒ '''<tt>Account X</tt>''' ⇒ '''<tt>SIP</tt>''' ⇒ '''<tt>Audio Settings</tt>'''
* Preferred Vocoder to '''G729A/B'''
* All of the remaining codec choices to '''PCMU'''

== External links ==
* [https://www.grandstream.com/products/ip-voice-telephony-gxp-series-ip-phones/gxp-series-basic-ip-phones/product/gxp1630 Official product page]
* [https://documentation.grandstream.com/article-categories/gxp16xx Official product documentation]

The full range of user and administrator guides for this device as well as other support documents are available at the [https://documentation.grandstream.com/ Grandstream documentation website]. The direct links below are believed to be the most useful and included here solely for your convenience.
* Basic information:
** [https://documentation.grandstream.com/knowledge-base/gxp1630-quick-installation-guide/ Grandstream GXP1630 Quick Installation Guide] ([https://documentation.grandstream.com/wp-content/uploads/2021/11/GXP1630_QIG-1.pdf PDF])
** [https://documentation.grandstream.com/knowledge-base/gxp16xx-quick-user-guide/ Grandstream GXP16XX Series Quick User Guide]
* Detailed information:
** [https://documentation.grandstream.com/knowledge-base/gxp16xx-series-user-guide/ Grandstream GXP16XX Series User Guide] ([https://documentation.grandstream.com/wp-content/uploads/2026/03/GXP16xx-Series-%E2%80%93-User-Guide.pdf PDF])
** [https://documentation.grandstream.com/knowledge-base/gxp16xx-series-administration-guide/ Grandstream GXP16XX Series Administration Guide] ([https://documentation.grandstream.com/wp-content/uploads/2026/03/GXP16xx-Series-%E2%80%93-Administration-Guide.pdf PDF])
** [https://documentation.grandstream.com/knowledge-base/firmware-upgrade-guide/ Firmware upgrade guide for all Grandstream devices]

[[Category:Grandstream|GXP1630]]
[[Category:SIP phones]]

Template:Button

2026-05-16T23:58:19Z

RogueScholar: Create template for styling UI buttons in configuration guides

<span style="background-color: #e3e6e8; border: thin solid #babfc5; border-radius: 4px; box-shadow: 0 1px 1px 0 #0c0d0e26, inset 0 1px 0 0 #fff; color: #0c0d0e; display: inline-block; font: 500 0.975em/1.5 sans-serif,sans-serif; margin: 0 0.1em; overflow-wrap: break-word; padding: 0 0.6em; text-shadow: 0 1px 0 #fff; white-space: collapse nowrap;">{{{text|{{{1|<noinclude>Button text</noinclude>}}}}}}</span><noinclude>
{{Documentation}}</noinclude>

Grandstream GXP1630

2026-05-08T16:19:49Z

RogueScholar: Add side view image

[[File:Grandstream GXP1630 - Front View.png|frameless|right|440px|alt=Grandstream GXP1630 front view|Grandstream GXP1630 corded IP desk phone, front view]]
[[File:Grandstream GXP1630 - Side View.png|frameless|right|176px|alt=Grandstream GXP1630 side view|Grandstream GXP1630 corded IP desk phone, side view]]
The '''Grandstream GXP1630''' is the most feature-rich model of the company's "Basic [[IP Phones|IP Phones]]" product line of corded-handset desk models (identified by model numbers that begin with <code style="border: thin solid #ccc; padding: 1px 4px; white-space: collapse nowrap;">GXP16…</code>) marketed to small and medium-size businesses. Directly quoting the manufacturer's website:
<blockquote>"Basic IP phones designed for simple, intuitive, and effective communication in the hands of users who need it. Our [https://www.grandstream.com/products/ip-voice-telephony/gxp-series-ip-phones/gxp-series-basic-ip-phones GXP1600 series] of basic IP phones deliver interactive communications to a desktop for the user who needs access to VoIP but does not require advanced features. They are ideal for hotel rooms, information booths, and warehouse deployments for access to quick call control. Delivering a vibrant and clear user-interface, this device is a perfect solution for those who handle low-to-medium call volume and require access to key call efficiency functionalities."</blockquote>

Some common features of all models in this series are:
* Support for at least two lines with individual SIP registrations
* A wired 10/100BaseT Ethernet connection
* HD audio
* Speakerphone
* Three-way conferencing

The GXP1630 improves on that baseline with:
* Support for three (3) SIP accounts, each with a dedicated line button and programmable function key
* Four-way conferencing
* Dual switched Gigabit Ethernet ports with integrated [[Power over Ethernet|Power over Ethernet (PoE)]]
* A backlit LCD display, and
* Eight (8) [[Busy Lamp Field]] indicators

== Configuration ==
[[File:General GXP1630.png|border|right|400px|alt=Screenshot of GXP1630 General Settings page|GXP1630 Configuration: Account ⇒ Account # ⇒ General Settings]]
[[File:SIP Basic Grandstream GXP1630.png|border|right|400px|alt=Screenshot of GXP1630 Basic SIP Settings page|GXP1630 Configuration: Account ⇒ Account # ⇒ SIP Settings ⇒ Basic Settings]]
[[File:SIP Custom SIP Headers Grandstream GXP1630.png|border|right|400px|alt=Screenshot of GXP1630 Custom SIP Headers page|GXP1630 Configuration: Accounts ⇒ Account X ⇒ SIP ⇒ Custom SIP Header]]
: ''Click the images at right to expand them, if needed.''
=== SIP Account ===
Once you have connected your device to your local network, it is first necessary to determine the IP address that was assigned to it in order to access the web browser-based administration controls.
# Using the physical navigation buttons on the phone, navigate through the menu using this sequence of choices:
#: '''<tt>Menu</tt>''' ⇒ '''<tt>Status</tt>''' ⇒ '''<tt>Network Status</tt>''' ⇒ '''<tt>IPv4 Address</tt>'''
# Enter the IP address shown on the device's LCD display into the address field of your preferred web browser.
# Use the following credentials at the login page which is displayed to access the available configuration options:
#: '''Username: <code>admin</code>'''
#: '''Password: <code>admin</code>'''

: '''''Note:''' Grandstream is gradually phasing out the use of <code>admin</code> as default password programmed into new models. As of 2023, it appears to still be working for this model, but should this become deprecated at some later date, check for a model/serial number sticker on the underside or back (or, for self-contained wireless handsets like the [[Grandstream WP810, WP820, WP822 and WP825|Grandstream WP8XX series]], in the battery compartment under the battery). The password will be identified by a preceding <tt>P/W:</tt>, followed by a random string of characters unique to that individual device that .''

=== General settings ===
Once you have authenticated and are shown the settings page, proceed to the indicated page and once there, change the following settings to match the details of your VoIP.ms account:
:: '''<tt>Account</tt>''' ⇒ '''<tt>Account #</tt>''' ⇒ '''<tt>General Settings</tt>'''
* '''Account Name:''' The name that will be displayed on the LCD screen of your phone.
* '''SIP Server:''' One of the VoIP.ms [[Choosing_Server#Choosing_a_Server|servers]], typically the one closest to your physical location.<br />&emsp;'''''Note: It is necessary to use the same server for both the device and the DID number in order to receive incoming calls.'''''
* '''SIP User ID:''' The account or sub-account number that you wish to use with this device.
* '''Authenticate Password:''' The password for that account or sub-account.
* '''Name:''' The name that will be transmitted as the Caller ID for outgoing calls ('''additional requirements below''')
*; &emsp;<span style="text-decoration: underline;">IMPORTANT:</span> : <ul style="margin: 0.3em 0 0 0.7em;"><li>We suggest entering your outbound Caller ID Name in '''capital letters'''. This will appear more clearly on most devices.</li><li>You must use only alpha-numeric characters; special characters, including standard punctuation, will not be displayed.</li><li>Enter a '''maximum of 15 characters''', as some Canadian providers will not show more than that; shrink or adapt your Caller ID accordingly.</li><li>Spaces are permitted in a Caller ID name, and count against the 15 character limit.</li></ul>
Click the '''Save and Apply''' button.

=== Basic SIP settings ===
Once this is done, the time on the device will need to be set before it can send valid information for its registration status. To do that, navigate to the indicated page and make the following changes:
:: '''<tt>Account #</tt>''' ⇒ '''<tt>SIP Settings</tt>''' ⇒ '''<tt>Basic Settings</tt>'''
* '''SIP Registration:''' Yes
* '''Register Expiration:''' 5 (value specified in minutes)
* '''Enable OPTIONS Keep Alive:''' Yes
* '''SIP Transport:''' UDP or TCP
Finally, click '''Save and Apply'''.

== Common issues ==
=== Outgoing calls ===
If you are able to receive incoming calls but outgoing calls are failing with error "No response", it is likely that the issue stems from either missing SIP headers or an unsupported audio codec. Login into your device's web-based configuration and navigate to:
:: '''<tt>Accounts</tt>''' ⇒ '''<tt>Account X</tt>''' ⇒ '''<tt>SIP</tt>''' ⇒ '''<tt>Custom SIP Header</tt>'''
On that page, deactivate these options:
* '''Use X-Grandstream-PBX Header'''
* '''Use P-Access-Network-Info Header'''
* '''Use P-Emergency-Info Header'''
Then navigate to the page below and change:
:: '''<tt>Accounts</tt>''' ⇒ '''<tt>Account X</tt>''' ⇒ '''<tt>SIP</tt>''' ⇒ '''<tt>Audio Settings</tt>'''
* Preferred Vocoder to '''G729A/B'''
* All of the remaining codec choices to '''PCMU'''

== External links ==
* [https://www.grandstream.com/products/ip-voice-telephony-gxp-series-ip-phones/gxp-series-basic-ip-phones/product/gxp1630 Official product page]
* [https://documentation.grandstream.com/article-categories/gxp16xx Official product documentation]

The full range of user and administrator guides for this device as well as other support documents are available at the [https://documentation.grandstream.com/ Grandstream documentation website]. The direct links below are believed to be the most useful and included here solely for your convenience.
* Basic information:
** [https://documentation.grandstream.com/knowledge-base/gxp1630-quick-installation-guide/ Grandstream GXP1630 Quick Installation Guide] ([https://documentation.grandstream.com/wp-content/uploads/2021/11/GXP1630_QIG-1.pdf PDF])
** [https://documentation.grandstream.com/knowledge-base/gxp16xx-quick-user-guide/ Grandstream GXP16XX Series Quick User Guide]
* Detailed information:
** [https://documentation.grandstream.com/knowledge-base/gxp16xx-series-user-guide/ Grandstream GXP16XX Series User Guide] ([https://documentation.grandstream.com/wp-content/uploads/2026/03/GXP16xx-Series-%E2%80%93-User-Guide.pdf PDF])
** [https://documentation.grandstream.com/knowledge-base/gxp16xx-series-administration-guide/ Grandstream GXP16XX Series Administration Guide] ([https://documentation.grandstream.com/wp-content/uploads/2026/03/GXP16xx-Series-%E2%80%93-Administration-Guide.pdf PDF])
** [https://documentation.grandstream.com/knowledge-base/firmware-upgrade-guide/ Firmware upgrade guide for all Grandstream devices]

[[Category:Grandstream|GXP1630]]
[[Category:SIP phones]]

Grandstream CloudUCM

2026-05-08T09:21:14Z

RogueScholar: Reorganize content

[[File:VoIPms Grandstream CloudUCM Logo.jpg|frameless|right|312px]]
'''[https://www.grandstream.com/products/ip-pbxs/ucm-series-ip-pbxs/product/clouducm Grandstream CloudUCM]''' is a cloud PBX that provides a scalable business communication and collaboration platform with powerful features and integrations that enable teams to be more productive than ever before.

== Configuration ==
=== Logging in ===
[[File:VoIPms Grandstream CloudUCM Login.jpg|frameless|right|400px]]
You can log directly to your instance URL or access it from your [[Grandstream Device Management Cloud System (GDMS)]] account.

=== Creating a trunk ===
[[File:New SIP trunk.png|border|right|576px]]
Once logged into your admin console, create a new SIP trunk for VoIP.ms by navigating to the '''<tt>Extension</tt>''' page and in the '''<tt>Trunk</tt>''' section select '''<tt>VoIP trunks</tt>''' from the left panel and making the following selections:
* '''Type:''' Register SIP trunk
* '''Provider Name:''' Enter a convenient name, such as "VoIP.ms" or "Main Trunk"
* '''Host Name:''' Enter your preferred POP server to connect to, i.e., <code style="border: thin solid #ccc; padding: 1px 4px; white-space: collapse nowrap;">toronto5.voip.ms</code>
* '''Keep trunk CID:''' Enable if you want the trunk to send its Caller ID for all outgoing calls or disable it if your extensions are going to send their own individual Called ID information
* '''Username:''' Enter your six-digit VoIP.ms account number or the full name of a sub-account
* '''Password:''' Enter the password for the account above

Once stored, these trunks are displayed with details such as Provider Name, Type, Hostname/IP and Username, plus options for editing and detecting the trunk.
* To add a new VoIP trunk, simply click on "Add SIP Trunk."
* For detailed configuration parameters, click on the ''edit'' icon.
* To set up Direct Outward Dialing (DOD) for the SIP Trunk, click on the corresponding option.
* To delete a VoIP trunk, click on the delete icon.
: '''''Note:''' Ensure to select the same VoIP server that corresponds to your DID number. You can verify the server associated with your VoIP number through your VoIP.ms customer portal under '''DID Numbers''' >> '''Manage DIDs''', in the '''POP''' section. It's crucial to match the server chosen here with the one in your portal to ensure incoming calls ring properly.''
* Click the '''Save''' button, but do not click '''Apply Changes''' yet.

=== Additional SIP settings ===
Once you've set up your basic SIP trunk, it's time to optimize some settings. Begin by clicking on the edit icon for your trunk.

==== SIP headers ====
[[File:SIP-basic settings.png|border|right|576px]]
To ensure the proper sending of the '''"FROM"''' header, follow these steps:
* Set '''<code>fromuser:</code>''' to your main VoIP.ms account number or sub-account name (replacing <code>100000</code>)

==== Audio codecs ====
Navigate to "Advanced Settings" and under "Codec Preference," prioritize the codecs supported by VoIP.ms in the order listed:
* '''G.722'''
* '''PCMU'''
* '''G.729'''

==== Trunk Caller ID ====
If you wish to transmit Called ID name and number information from your trunk, utilize the '''<tt>PAI (P-Asserted-Identity)</tt>''' header according to this format:
: '''<code>"CallerIDName" <CallerIDNumber></code>'''

: '''Note:''' Enabling the PAI header without specifying values will default to the CID priority rules from Grandstream, allowing you to pass the Caller ID information through from extensions. To prevent overriding by your trunk, ensure that no Caller ID Number is set in your VoIP.ms portal.

==== NAT Keep Alive ====
To prevent your modem from closing local SIP ports, set the following configuration options as indicated:
* '''Enable Heartbeat Detection:''' Enabled
* '''Heartbeat Frequency:''' <code>50</code>

After completing the configuration, click on the "Save" button, and optionally, click "Apply Changes" to enact them. Your trunk status should display as "Registered" on your VoIP.ms dashboard. Keep in mind that calls will not be operational until you establish your outbound and inbound routes.

=== Creating an outbound route ===
Outbound routes play a crucial role in matching your dialing patterns and directing calls through the appropriate trunk. VoIP.ms recommends including the following patterns in your outbound route:
''Ensure that each of your dial patterns is prefixed by the underscore character (<code>_</code>)''
* <code>_1NXXXXXXXXX</code>
* <code>_NXXXXXXXXX</code>
* <code>_4XXX</code>
* <code>_00.</code>
* <code>_011.</code>
* <code>_033.</code>
* <code>_044.</code>

To create your outbound routes, navigate to '''<tt>Extension/Trunk</tt>''' ⇒ '''<tt>Outbound routes</tt>''' from the left panel, then click on "Add". In this section, you only need to complete the following fields:
* '''Calling Rule Name:''' Select any desired name for this route.
* '''Pattern:''' Define the pattern that callers need to dial; you can utilize the suggested dial patterns provided by VoIP.ms, as mentioned above.
* '''Trunk:''' Choose the VoIP.ms trunk through which your call will be routed.
* '''Privilege Level:''' Determine the privilege level required for extensions to utilize this route. If you follow our suggested dial pattern, we recommend setting it to "International" to enable placing international calls. Note that extensions with lower privilege levels won't have access to this route. Hover over the field name for additional information.

: '''''Note:''' If you wish to incorporate a dial-out prefix, simply type it after the underscore (<code>_</code>) character in your dial patterns. To remove this prefix during dialing, utilize the "Strip" field, where you can specify the number of digits to strip after the underscore character. For example, if you intend to use <code>9</code> as a dial-out prefix, your pattern should be <code>_9NXXXXXXXXX</code>. To remove the <code>9</code> when dialing out, set the "Strip" field to <code>1</code>; this ensures that only one character (in this case, the number <code>9</code>) will be stripped off.''

=== Creating an inbound route ===
[[File:Inbound routes setup.png|border|right|576px]]
With your inbound routes configured, you can efficiently manage all incoming calls from multiple phone numbers using just one trunk. This eliminates the need for multiple trunks, as the inbound routes can effectively receive and direct calls to their respective destinations within your UCM.

: '''''Note:''' You should not use more than one VoIP.ms trunk on the same device.''

To set up inbound routes, navigate to '''<tt>Extension/Trunk</tt>''' ⇒ '''<tt>Inbound Routes</tt>''' from the left panel, and then click on '''<tt>Add</tt>'''.
: [[File:Inbound routes main.png|frameless|320px]]

In this section, complete the following fields:
* '''Trunks:''' Select your VoIP.ms trunk to specify where incoming calls should be directed.
* '''Pattern:''' Enter your VoIP.ms DID phone number exactly as it appears in the "Manage DIDs" section of your customer portal. Ensure it is prefixed by the underscore (<code>_</code>) character. Avoid using wildcards and ensure the full number is used to prevent conflicts in the inbound pattern.
* '''Default Destination:''' Choose the default destination to which your PBX should route incoming calls; this could include an internal extension, an IVR, a ring group, or other locally managed options within your PBX.

== External links ==
=== Guides and manuals ===
* [https://www.grandstream.com/products/ip-pbxs/ucm-series-ip-pbxs/product/clouducm Official product page]
* [https://documentation.grandstream.com/article-categories/cloudUCM Official documentation listing]
** [https://documentation.grandstream.com/knowledge-base/clouducm-user-manual/ CloudUCM user manual]
** [https://documentation.grandstream.com/knowledge-base/clouducm-endpoint-configuration-guide/ CloudUCM endpoint configuration]
** [https://documentation.grandstream.com/article-categories/deployment-clouducm/ CloudUCM deployment scenarios]

[[Category:Grandstream|CloudUCM]]
[[Category:PBXes]]

File:Grandstream GXP1630 - Side View.png

2026-05-08T08:07:52Z

RogueScholar: == Description == Side-view photo of a Grandstream GXP1630 corded desktop IP phone showing the desktop stand. === Original source === This image was retrieved from the [https://www.grandstream.com/products/ip-voice-t

== Description ==
Side-view photo of a [[Grandstream GXP1630|Grandstream GXP1630]] corded desktop [[IP Phones|IP phone]] showing the desktop stand.

=== Original source ===
This image was retrieved from the [https://www.grandstream.com/products/ip-voice-telephony-gxp-series-ip-phones/gxp-series-basic-ip-phones/product/gxp1630 official Grandstream product page] image gallery (accessible at [https://content.grandstream.com/hubfs/Product%20Images/GXP/gxp1630_side_web.png content.grandstream.com/hubfs/Product Images/GXP/]) with the filename <code style="border: thin solid #ccc; font-size: 1em; padding: 1px 4px; white-space: collapse nowrap;">gxp1630_side_web.png</code>, which was [https://web.archive.org/web/20260508080458if_/https://content.grandstream.com/hubfs/Product%20Images/GXP/gxp1630_side_web.png archived by The Wayback Machine] on May 8, 2026. It was subsequently resampled and optimized for reduced file size before being uploaded to [[Welcome|VoIP.ms Wiki]].

[[Category:Grandstream]]

Grandstream GXP1630

2026-05-08T07:42:04Z

RogueScholar: Refresh, reorganize and clarify content

[[File:Grandstream GXP1630 - Front View.png|thumb|right|648px|alt=Grandstream GXP1630 SIP desk phone|Grandstream GXP1630 corded IP desk phone]]
The '''Grandstream GXP1630''' is the most feature-rich model of the company's "Basic [[IP Phones|IP Phones]]" product line of corded-handset desk models (identified by model numbers that begin with <code style="border: thin solid #ccc; padding: 1px 4px; white-space: collapse nowrap;">GXP16…</code>) marketed to small and medium-size businesses. Directly quoting the manufacturer's website:
<blockquote>"Basic IP phones designed for simple, intuitive, and effective communication in the hands of users who need it. Our [https://www.grandstream.com/products/ip-voice-telephony/gxp-series-ip-phones/gxp-series-basic-ip-phones GXP1600 series] of basic IP phones deliver interactive communications to a desktop for the user who needs access to VoIP but does not require advanced features. They are ideal for hotel rooms, information booths, and warehouse deployments for access to quick call control. Delivering a vibrant and clear user-interface, this device is a perfect solution for those who handle low-to-medium call volume and require access to key call efficiency functionalities."</blockquote>

Some common features of all models in this series are:
* Support for at least two lines with individual SIP registrations
* A wired 10/100BaseT Ethernet connection
* HD audio
* Speakerphone
* Three-way conferencing

The GXP1630 improves on that baseline with:
* Support for three (3) SIP accounts, each with a dedicated line button and programmable function key
* Four-way conferencing
* Dual switched Gigabit Ethernet ports with integrated [[Power over Ethernet|Power over Ethernet (PoE)]]
* A backlit LCD display, and
* Eight (8) [[Busy Lamp Field]] indicators

== Configuration ==
[[File:General GXP1630.png|border|right|384px|alt=Screenshot of GXP1630 General Settings page|GXP1630 Configuration: Account ⇒ Account # ⇒ General Settings]]
[[File:SIP Basic Grandstream GXP1630.png|border|right|384px|alt=Screenshot of GXP1630 Basic SIP Settings page|GXP1630 Configuration: Account ⇒ Account # ⇒ SIP Settings ⇒ Basic Settings]]
[[File:SIP Custom SIP Headers Grandstream GXP1630.png|border|right|384px|alt=Screenshot of GXP1630 Custom SIP Headers page|GXP1630 Configuration: Accounts ⇒ Account X ⇒ SIP ⇒ Custom SIP Header]]
: ''Click the images at right to expand them, if needed.''
=== SIP Account ===
Once you have connected your device to your local network, it is first necessary to determine the IP address that was assigned to it in order to access the web browser-based administration controls.
# Using the physical navigation buttons on the phone, navigate through the menu using this sequence of choices:
#: '''<tt>Menu</tt>''' ⇒ '''<tt>Status</tt>''' ⇒ '''<tt>Network Status</tt>''' ⇒ '''<tt>IPv4 Address</tt>'''
# Enter the IP address shown on the device's LCD display into the address field of your preferred web browser.
# Use the following credentials at the login page which is displayed to access the available configuration options:
#: '''Username: <code>admin</code>'''
#: '''Password: <code>admin</code>'''

: '''''Note:''' Grandstream is gradually phasing out the use of <code>admin</code> as default password programmed into new models. As of 2023, it appears to still be working for this model, but should this become deprecated at some later date, check for a model/serial number sticker on the underside or back (or, for self-contained wireless handsets like the [[Grandstream WP810, WP820, WP822 and WP825|Grandstream WP8XX series]], in the battery compartment under the battery). The password will be identified by a preceding <tt>P/W:</tt>, followed by a random string of characters unique to that individual device that .''

=== General settings ===
Once you have authenticated and are shown the settings page, proceed to the indicated page and once there, change the following settings to match the details of your VoIP.ms account:
:: '''<tt>Account</tt>''' ⇒ '''<tt>Account #</tt>''' ⇒ '''<tt>General Settings</tt>'''
* '''Account Name:''' The name that will be displayed on the LCD screen of your phone.
* '''SIP Server:''' One of the VoIP.ms [[Choosing_Server#Choosing_a_Server|servers]], typically the one closest to your physical location.<br />&emsp;'''''Note: It is necessary to use the same server for both the device and the DID number in order to receive incoming calls.'''''
* '''SIP User ID:''' The account or sub-account number that you wish to use with this device.
* '''Authenticate Password:''' The password for that account or sub-account.
* '''Name:''' The name that will be transmitted as the Caller ID for outgoing calls ('''additional requirements below''')
*; &emsp;<span style="text-decoration: underline;">IMPORTANT:</span> : <ul style="margin: 0.3em 0 0 0.7em;"><li>We suggest entering your outbound Caller ID Name in '''capital letters'''. This will appear more clearly on most devices.</li><li>You must use only alpha-numeric characters; special characters, including standard punctuation, will not be displayed.</li><li>Enter a '''maximum of 15 characters''', as some Canadian providers will not show more than that; shrink or adapt your Caller ID accordingly.</li><li>Spaces are permitted in a Caller ID name, and count against the 15 character limit.</li></ul>
Click the '''Save and Apply''' button.

=== Basic SIP settings ===
Once this is done, the time on the device will need to be set before it can send valid information for its registration status. To do that, navigate to the indicated page and make the following changes:
:: '''<tt>Account #</tt>''' ⇒ '''<tt>SIP Settings</tt>''' ⇒ '''<tt>Basic Settings</tt>'''
* '''SIP Registration:''' Yes
* '''Register Expiration:''' 5 (value specified in minutes)
* '''Enable OPTIONS Keep Alive:''' Yes
* '''SIP Transport:''' UDP or TCP
Finally, click '''Save and Apply'''.

== Common issues ==
=== Outgoing calls ===
If you are able to receive incoming calls but outgoing calls are failing with error "No response", it is likely that the issue stems from either missing SIP headers or an unsupported audio codec. Login into your device's web-based configuration and navigate to:
:: '''<tt>Accounts</tt>''' ⇒ '''<tt>Account X</tt>''' ⇒ '''<tt>SIP</tt>''' ⇒ '''<tt>Custom SIP Header</tt>'''
On that page, deactivate these options:
* '''Use X-Grandstream-PBX Header'''
* '''Use P-Access-Network-Info Header'''
* '''Use P-Emergency-Info Header'''
Then navigate to the page below and change:
:: '''<tt>Accounts</tt>''' ⇒ '''<tt>Account X</tt>''' ⇒ '''<tt>SIP</tt>''' ⇒ '''<tt>Audio Settings</tt>'''
* Preferred Vocoder to '''G729A/B'''
* All of the remaining codec choices to '''PCMU'''

== External links ==
* [https://www.grandstream.com/products/ip-voice-telephony-gxp-series-ip-phones/gxp-series-basic-ip-phones/product/gxp1630 Official product page]
* [https://documentation.grandstream.com/article-categories/gxp16xx Official product documentation]

The full range of user and administrator guides for this device as well as other support documents are available at the [https://documentation.grandstream.com/ Grandstream documentation website]. The direct links below are believed to be the most useful and included here solely for your convenience.
* Basic information:
** [https://documentation.grandstream.com/knowledge-base/gxp1630-quick-installation-guide/ Grandstream GXP1630 Quick Installation Guide] ([https://documentation.grandstream.com/wp-content/uploads/2021/11/GXP1630_QIG-1.pdf PDF])
** [https://documentation.grandstream.com/knowledge-base/gxp16xx-quick-user-guide/ Grandstream GXP16XX Series Quick User Guide]
* Detailed information:
** [https://documentation.grandstream.com/knowledge-base/gxp16xx-series-user-guide/ Grandstream GXP16XX Series User Guide] ([https://documentation.grandstream.com/wp-content/uploads/2026/03/GXP16xx-Series-%E2%80%93-User-Guide.pdf PDF])
** [https://documentation.grandstream.com/knowledge-base/gxp16xx-series-administration-guide/ Grandstream GXP16XX Series Administration Guide] ([https://documentation.grandstream.com/wp-content/uploads/2026/03/GXP16xx-Series-%E2%80%93-Administration-Guide.pdf PDF])
** [https://documentation.grandstream.com/knowledge-base/firmware-upgrade-guide/ Firmware upgrade guide for all Grandstream devices]

[[Category:Grandstream]]
[[Category:SIP phones]]

File:Grandstream GXP1630 - Front View.png

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RogueScholar: == Description == Photo of a Grandstream GXP1630 corded desktop IP phone. === Original source === This image was retrieved from the [https://www.grandstream.com/products/ip-voice-telephony-gxp-series-ip-phones/gxp-se

== Description ==
Photo of a [[Grandstream GXP1630|Grandstream GXP1630]] corded desktop [[IP Phones|IP phone]].

=== Original source ===
This image was retrieved from the [https://www.grandstream.com/products/ip-voice-telephony-gxp-series-ip-phones/gxp-series-basic-ip-phones/product/gxp1630 official Grandstream product page] image gallery (accessible at [https://content.grandstream.com/hubfs/Product%20Images/GXP/gxp1630_front_web.png content.grandstream.com/hubfs/Product Images/GXP/]) with the filename <code style="border: thin solid #ccc; font-size: 1em; padding: 1px 4px; white-space: collapse nowrap;">gxp1630_front_web.png</code>, which was [https://web.archive.org/web/20260508053907if_/https://content.grandstream.com/hubfs/Product%20Images/GXP/gxp1630_front_web.png archived by The Wayback Machine] on May 7, 2026. It was subsequently resampled and optimized for reduced file size before being uploaded to [[Welcome|VoIP.ms Wiki]].

[[Category:Grandstream]]

File:Grandstream HandyTone HT802v2 - Top View.png

2026-05-08T04:09:06Z

RogueScholar: Add image description and original source details

== Description ==
High-angle photo of the top of a [[Grandstream HT802v2|Grandstream HandyTone HT802v2]] [[ATA Devices|analog telephone adapter]], showing all lighted device status indicators.

=== Original source ===
This image was retrieved from the [https://www.grandstream.com/products/gateways-and-atas/analog-telephone-adaptors/product/ht802v2 official Grandstream product page] image gallery (accessible at [https://www.grandstream.com/hubfs/HT802%20v2_2.png grandstream.com/hubfs/]) with the filename <code style="border: thin solid #ccc; font-size: 1em; padding: 1px 4px; white-space: collapse nowrap;">HT802 v2_2.png</code>, which was [https://web.archive.org/web/20250214110604if_/https://www.grandstream.com/hubfs/HT802%20v2_2.png archived by The Wayback Machine] on February 14, 2025. It was subsequently resampled and optimized for reduced file size before being uploaded to [[Welcome|VoIP.ms Wiki]].

[[Category:Grandstream]]

File:Grandstream HandyTone HT802v2 - Rear View.png

2026-05-08T04:03:57Z

RogueScholar: Add description of image and original source details

== Description ==
Photo of rear of [[Grandstream HT802v2|Grandstream HandyTone HT802v2]] [[ATA Devices|analog telephone adapter]], showing all available connection ports.

=== Original source ===
This image was retrieved from [https://www.grandstream.com/products/gateways-and-atas/analog-telephone-adaptors/product/ht802v2 official Grandstream product page] image gallery (accessible at [https://www.grandstream.com/hubfs/HT802%20v2_3.png grandstream.com/hubfs/]) with the filename <code style="border: thin solid #ccc; font-size: 1em; padding: 1px 4px; white-space: collapse nowrap;">HT802 v2_3.png</code>, which was [https://web.archive.org/web/20241108193323/https://www.grandstream.com/hubfs/HT802%20v2_3.png archived by The Wayback Machine] on November 8, 2024. It was subsequently resampled and optimized for reduced file size before being uploaded to [[Welcome|VoIP.ms Wiki]].

[[Category:Grandstream]]

Grandstream WP810, WP820, WP822 and WP825

2026-04-17T22:56:55Z

RogueScholar: Add to :Category:Grandstream

[[File:Grandstream WP810.png|thumb|right|upright=0.5|Grandstream WP810]]
The '''Grandstream WP810, WP820, WP822 and WP825''' are two-line cordless [[IP Phone]]s which, instead of communicating through a compatible DECT cordless base, operate in a self-contained manner by connecting to wi-fi directly. The handset is able to register directly with any of the VOiP.ms servers.

== Configuration ==
The package includes a handset and a charger; no other components are needed.

The handsets are similar in look-and-feel to the cordless [[Grandstream DP750/DP720|Grandstream DP720]] series, except for the absence of the cordless base station.
# Insert the supplied lithium-ion battery and place the handset in the charger.
# Once the handset is recharged, connect it to your local wi-fi by pressing [menu] (the centre button) → [settings] (the gear icon) → "Network" → "Wi-Fi Networks".
# The handset will show a list of available wi-fi networks. Select one, then select "Password" and enter your local wi-fi password.
# Press [Connect]; the display should show "Wi-Fi Networks" with the list of network names and "Connected" for the selected network.
# Return to the main menu. Select [status] (the ''i'' icon) then "Network Status". The handset will display IPv4 info, which includes the IPv4 address.[[File:Grandstream WP810 Account 1 General Settings.png|border|right|800px|Account 1 - General Settings]]
# Open a web browser to the displayed address, which will look like <nowiki>https://192.168.1.123</nowiki> (or whatever address your network dynamically assigns). If using HTTPS the browser will display a warning that the device is using an "unsigned certificate"; select "proceed anyway" to go to the login screen.
# On the current version of this product, the "admin" password is hidden in the lower-left corner of the model/serial number sticker inside the handset's battery compartment. It's under the battery. (Older versions had username "admin" and password "admin" as the default - which the user is prompted to immediately change.) Look for "P/W" followed by s0m3g1bb3r1sh as the random password.
# Put the battery back in the handset and the handset back on the charger. Try a web login using username "admin" and the password from the sticker in the battery compartment.
# A configuration menu should appear. Navigate to "Accounts" → "Account 1" → "General Settings".
#* Account Active: '''Yes'''
#* Account Name: Create a name to identify this line. This name (up to eight characters) will appear on the handset screen when selecting a line appearance to make a call.
#* SIP Server: One of VoIP.ms multiple [[Choosing Server|servers]]. For inbound calls to work the server chosen must match the one used by your DID Number. For example: '''toronto1.voip.ms''' if your numbers are assigned to that server.
#* SIP User ID: (your user number, plus the subaccount if applicable.) For example: 123456_1 is voip.ms user #123456 subaccount #1
#* Authenticate Password: (the password for your voip.ms account or subaccount)
#* Name: Display name for outbound caller ID (15 characters or less)
#* Voice Mail Access Number: '''*97'''
#* Account Display: '''User Name'''
#* "Authenticate ID", "Secondary SIP Server", "Outbound Proxy", "Backup Outbound Proxy" may be left blank
# Click on the [Save and Apply] button (for the WP820, click [Save] then click [Apply]). The handset should now be registered directly to the selected VoIP.ms server.
# Verify that the registration is successful by returning to "Status" → "Account Status" on the web interface, or by checking the main account or subaccount registration status on the "Customer Portal: Home Portal" of your VoIP.ms account.
# Go to "Accounts" → "Account 1" → "SIP Settings" and set the "Basic Settings" and "Audio Settings" as described for the [[Grandstream DP750/DP720]] or other similar Grandstream [[IP Phones]]. Verify that "Accounts" → "Account 1" → "SIP Settings" → "Basic Settings" has "Register Expiration" set to a small value (such as "3" for three minutes) as the default (one hour) may not be often enough to keep you connected if you're behind a NAT.
# If the time display on the handset is incorrrect, go to "Settings" → "Preferences" → "Date and time" on the WP810 (or "System settings" → "Time and language" → "Time zone" on the WP820) to manually select an appropriate time zone.
# If you have a second account or [[Sub Accounts|subaccount]] with VoIP.ms or another provider, setup "Account 2" in the same manner as above.
# Try a test call (for example '''4443''' and hit the [green] button connects to the VoIP.ms echo test). The handset should be ready for use.

== Notes ==
# The [flash] button is not supported by Grandstream's cordless handsets. (It is supported by their analog telephone adapter line.) This may result in inability to access certain of the VoIP.ms [[features]], such as [[Call Parking|call parking]].
# Grandstream's firmware update server ("Maintenance" → "Upgrade" → "Firmware server path" on the WP820) appears as "fm.grandstream.com/gs" on some older versions of the code. This needs to be "firmware.grandstream.com" or the handset will not be able to upgrade from network.
# The default configuration on these handsets is vulnerable to [[Sip Scanner Ghost Calls]], even if you are behind a uPnP router. To fix this issue (based on the WP820 settings), go to each account (1 and 2) and for each, change "Account #n" → "SIP settings" → "Local SIP Port" to be "0" (use random port) instead of "5060" or other well-known values. [Save] this and [Apply]. Then go to each account (1 and 2) and check the checkbox for "Account #n" → "Advanced settings" → "Only Accept SIP Requests from Known Servers". Again, [Save] and [Apply].

== Dial plan ==
The dial plan structure is described on pages 49-51 of the [https://www.grandstream.com/hubfs/Product_Documentation/wp820_administration_guide.pdf WP820 admin guide]. The corresponding page on the WP820 web interface is "Accounts" → [Account 1] → "Call settings" → "Dial plan". (The WP810's documentation lists "dial plan" as an available feature but does not explain how to configure it.)

While the normal operation of the WP810/WP820 series is to wait for the user to dial the entire number, then press the [green] button, it is possible to create dial plans which wait for specific fixed-length patterns (such as ''1[2-9]xx[2-9]xxxxxx'' for an eleven-digit North American +1-NXX-NXXXXXX format number) and send the call as soon as that number of digits is provided. This bypasses both the wait for the [green] button (or "call" soft key) to be pressed and (where applicable) any prompt for "audio call" vs. "video call".

For instance, ''1[2-9]xx[2-9]xxxxxx'' would send a North American domestic call out as soon as 1 plus ten digits have been dialled.

{| style="font-size:0.92rem;"
! scope="row" style="font-family: monospace; padding: 0.2em 1em; text-align: center;" | +,1,2,3,4,5,6,7,8,9,0,<br />*,#,A,a,B,b,C,c,D,d
| Individual digit, to be passed through verbatim
|-
! scope="row" style="font-family: monospace; padding: 0.2em 1em; text-align: center;" | x
| Matches any digit from 0-9
|-
! scope="row" style="font-family: monospace; padding: 0.2em 1em; text-align: center;" | xx+
| At least 2-digit number
|-
! scope="row" style="font-family: monospace; padding: 0.2em 1em; text-align: center;" | xx
| Exactly 2-digit number
|-
! scope="row" style="font-family: monospace; padding: 0.2em 1em; text-align: center;" | xx?
| 1 or 2-digit numbers from 0-9
|-
! scope="row" style="font-family: monospace; padding: 0.2em 1em; text-align: center;" | ^
| Exclude
|-
! scope="row" style="font-family: monospace; padding: 0.2em 1em; text-align: center;" | .
| Wildcard, matches one or more characters
|-
! scope="row" style="font-family: monospace; padding: 0.2em 1em; text-align: center;" | [3-5]
| Any digit of 3, 4, or 5.
|-
! scope="row" style="font-family: monospace; padding: 0.2em 1em; text-align: center;" | [147]
| Any digit 1, 4, or 7
|-
! scope="row" style="font-family: monospace; padding: 0.2em 1em; text-align: center;" | <2=011>
| Replace digit 2 with 011 when dialing
|-
! scope="row" style="font-family: monospace; padding: 0.2em 1em; text-align: center;" | <=1>
| Add a 1 to dialed number
|-
! scope="row" style="font-family: monospace; padding: 0.2em 1em; text-align: center;" | <1=>
| Remove 1 from the number dialed
|-
! scope="row" style="font-family: monospace; padding: 0.2em 1em; text-align: center;" | |
| Logical "or" operator; most dial plans will consist of { followed by multiple rules separated by the | pipe character; the last character in the dial plan will be }
|-
! scope="row" style="font-family: monospace; padding: 0.2em 1em; text-align: center;" | \+
| + sign
|-
! scope="row" style="font-family: monospace; padding: 0.2em 1em; text-align: center;" | T
| Flag - when adding a “T” at the end of a dial plan rule, the phone *should* wait for 3 seconds before calling out. In theory, this should allow for rules like '' <=1>[2-9]xx[2-9]xxxxxx | <=1555>[2-9]xxxxxxT '' which signal that a ten-digit North American number should go out immediately (just adding the leading +1) but a seven-digit local call should wait in case the user attempts to enter more digits. A user's call to 234-5678 will wait before being redirected to 1-555-234-5678 (you may want to replace 555 with the original area code for your region) while a user's call to 555-234-5678 will be sent to 1-555-234-5678 immediately. In practice, the WP820 appears not to send calls with the 'T' flag until the user presses the [green] button - a behaviour which contradicts the three-second delay then send which is described in the manual.
|}

The factory default Grandstream dial plan appears to be <code style="font-size: 0.92rem;">{ x+ | \+x+ | *x+ | *xx*x+ | x+*x+*x+*x+ | x+*x+*x+*x+#x+ }</code>

This plan will wait until the user dials the entire number and presses the [green] button; no translations or modifications are made to the dialled numbers.

== See also ==
* The [[Grandstream HandyTone 502 - HT502#Dial plans|Grandstream 502]] guide page has additional information on user-configured dial plans which is applicable to multiple Grandstream device models.
* The [[Grandstream DP750/DP720]] is also similar in configuration to this series, except for the use of one DP750 cordless base for multiple handsets.

== External links ==
The manufacturer's guides and manuals are available directly from Grandstream:
* [https://www.grandstream.com/hubfs/Product_Documentation/wp810_user_guide.pdf WP810 User Manual] (PDF)
* [https://www.grandstream.com/hubfs/Product_Documentation/wp810_administration_guide.pdf WP810 Admin Manual] (PDF)

* [https://www.grandstream.com/hubfs/Product_Documentation/wp820_user_guide.pdf WP820 User Manual] (PDF)
* [https://www.grandstream.com/hubfs/Product_Documentation/wp820_administration_guide.pdf WP820 Admin Manual] (PDF)

[[Category:Grandstream]]
[[Category: SIP phones]]

File:Grandstream WP810 Account 1 General Settings.png

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RogueScholar: Add to :Category:Grandstream

Screenshot of "Account 1" → "General Settings" tab on Grandstream WP810 admin page.

[[Category:Grandstream]]

File:Grandstream WP810.png

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RogueScholar: Add to :Category:Grandstream

Grandstream WP810 product photo, from manufacturer.

[[Category:Grandstream]]

Grandstream Device Management Cloud System (GDMS)

2026-04-17T22:35:39Z

RogueScholar: Reorganize page and add to :Category:Grandstream

Grandstream Device Management System (GDMS) is a free zero-touch cloud provisioning and management system that provides easy-to-use enterprise-level tools to manage Grandstream products before, during and after deployment. It provides a centralized interface to provision, manage, monitor and troubleshoot Grandstream products.

GDMS provides a '''FREE''' centralized interface to provision, manage, monitor and troubleshoot Grandstream products, including device management, account management, device configuration, firmware upgrades, device monitoring, intelligent alarm, and statistical analysis, individually or in batches of devices by site, group and model. This powerful cloud management platform adds robust security features to protect enterprise networks, including bank-grade TLS encryption from end-to-end, two-factor login authentication to protect your data security, two-way certificate verification and encryption of all device information. By offering a centralized, secure and zero-touch management for Grandstream endpoints, GDMS is ideal for enterprises, service providers and multi-site businesses as it allows them to easily deploy and manage large networks of Grandstream solutions.
[[File:GDMS.png|thumb|right|400px|link=https://www.youtube.com/watch?v=iq8BhrkSTWY|Tutorial video]]

== Resources ==
'''[https://www.gdms.cloud/ GDMS Portal]''' – https://gdms.cloud/
: 📚 [https://documentation.grandstream.com/article-categories/gdms/ GDMS Documentation Center]
: 📖 [https://documentation.grandstream.com/article-categories/user-manual-gdms/ User Manual]
:* [https://documentation.grandstream.com/article-categories/administration-gdms/ Administration Guide]
:* [https://documentation.grandstream.com/knowledge-base/gdms-user-guide/#product-overview Product Overview]
:* [https://documentation.grandstream.com/knowledge-base/gdms-user-guide/#gdms-technical-specifications GDMS Technical Specifications]
:* [https://blog.grandstream.com/faq/product-related-questions/gdms-grandstream-device-management-system FAQ Blog]

== Add Organization ==
[[File:GDMS Add-Organization Details.png|border|right|528px]]
You can create multiple Organization under your GDMS platform, each Devices and SIP accounts for each organization will be managed separately.

Click on '''[Organization]''' on the left menu,
: [[File:GDMS_Organization.png|200px]]

Then at the bottom of the page '''[+ Create Organization]'''.
: [[File:GDMS_Add-Organization.png|200px]]

You will be prompted to add an "Organization Name" and assign a User if you need to have someone separate to manage and access the GDMS platform. You can leave it at yourself in order to simply separate the organization you manage, such différent clients/entities.

== Add Sites ==
[[File:GDMS Site Add Details.png|border|right|440px]]
Organize your deployment and management. You can create multiple sites (Eg: departments) under your or into a specific organization. Its also possible to create template config per sites in order to have different parameters that will affect a group of devices at the same time. Example, images, Key config/Layout/Speed dial etc. but only affected to the specific device under this site.

Go to '''"Site"''' on the left menu.
: [[File:GDMS_SiteMenu.png|200px]]

Then on the upper tab, click on '''"Add Site"'''.
: [[File:GDMS_Site_Add.png|200px]]

You can either choose to create individual sites or have one that has Parent.
:* '''Name''': Enter the site name.
:* '''Parent Site''': If its related to the default one or choose the parent. ''(Generally click on default if you they are all under your/the '''main organization)''
:* '''Description''': Enter a description for your own usage.

== Add an SIP Server ==
[[File:GDMS SIP Account Details.png|border|right|680px]]
Go to '''"VoIP Account"''' on the left menu, then click on '''"SIP Account"'''.
: [[File:GDMS_SIP_Server_Menu.png|200px]]

Now click on the upper button, click on '''"Add Server"'''.
: [[File:GDMS_SIP_Server_Add_Button.png|200px]]
* '''Server Name''': Enter a name per instance Montreal3
* '''SIP Server''': Enter your preferred pop server, eg: YourPreferredPoP.voip.ms ''Note that you can also add a one of our alternative ports such as 5080 or 42872 by typing :5080 after the sip server.''
* '''Voice Mail Access Number''': *97
* '''DNS Mode''': Select '''"A RECORD"'''
* '''NAT Traversal''': Select '''"Keep-Alive"'''.
* '''SIP Transport''': Select '''"UDP"''' ''(Note that you can also use "TCP". You can also use "TLS" if you want to use call encryption between the device and our server. Note that in your sub-account you need to have this option enabled, and on the SIP Server you will need to add ''':5061''' after the pop server address.)''
* '''Additional Settings''': Click on "Add" then select '''"BLF Server"''', reenter the same information you have indicated on the field '''SIP Server'''.

Click '''[Save]'''

== Add an user SIP Account ==
# Go to "VoIP Account" on the left menu, then click on "SIP Account".
# Click on the corner right '''[Add Account]''' button.
#* '''Account Name''': The name you would like to name this account
#* '''SIP User ID''': Enter your VoIP.ms sub-account username ######_SubAccount
#* '''Authenticate ID''': Enter your VoIP.ms sub-account username ######_SubAccount
#* '''Authenticate Password''': Enter your sub-account password
#* '''Name'''*: This is your outbound "CALLERID NAME", enter it in '''Capital letters, Not longer than 15 characters, spaces are allowed and count as a character.'''
#* '''Voice Mail Access Number''': Enter *97 ''(Star 97)''
#* '''Outbound Proxy''': Leave it empty
#* '''SIP Server''': Select the POP Server you have created previously.

If you already have a device into your organization/site, you can assign it.

== Adding a device ==
Go to "Device Management" on the left menu, then "VoIP Device" and click on the corner right '''[Add Device]''' button.
* '''Device Name''': Enter a device name (for your own reference and for searchability in the GDMS plateforme)
* '''MAC Address''': Enter the MAC address under the phone
* '''S/N''': Enter the Serial Number under the phone
* '''Site''': Choose where you want to deploy this phone, if its on your main SITE (Default) or any other site you have created.

Click [Save]

== Creating a common config ==
Since all the device you will link on VoIP.ms will need to have the same config. It's also possible to create individual config per IP Phone if you want to customise it.

Go to "Device Template" on the left menu, and "By Site", then click on the corner right '''[Add Site Template]''' button.
* '''Name''': Name it the way you want eg: "Global Config"
* '''Auto Provision to Devices in''': You can choose the "Site" where you want to create this template and have it affecting all sites or specific sites.
* '''Description''': Enter a description for your own usage.

Note that if you create multiple sites, you will need to add them under this template by editing it with the edit button.

Then click "Save". Once done, where the template you have created, click on the "Gear" option.

At the corner right, where "General Series" select "GRP" or "GXP" Series depending the devices you have. This will provide you more config you can edit.

Click on the upper tab "Account", then on the left menu, click on "SIP Settings".

{| style="width: 100%;"
| style="vertical-align: top; width: 50%;" | In the same page where '''SIP Basic Settings'''
* '''Enable OPTIONS Keep Alive''': Check the box and set the toggle to '''"Yes"'''.
* '''SIP Registration''': Check the box and set the toggle to '''"Yes"'''.
* '''Register Expiration''': enter '''1''', ''for 60 seconds''.
* '''Use Privacy Header''': Check the box and set the toggle to '''"No"'''.

In the same page where '''SIP Advanced Settings'''
* '''PUBLISH for Presence''': Check the box and set the toggle to '''"Yes"'''.
| style="width: 50%;" | In the same page where ''' Custom SIP Headers'''
* '''Use P-Access-Network-Info Header''': Check the box and set the toggle to '''"No"'''.
* '''Use P-Emergency-Info Header''': Check the box and set the toggle to '''"No"'''.
* '''Use X-Grandstream-PBX Header''': Check the box and set the toggle to '''"No"'''.

On the left menu click '''Audio Settings''' on the same page where '''Preferred Vocoder'''
* '''Preferred Vocoder - choice 1''': Check the box and choose "'''G722'''".<br />''(Be sure the codec G722 is enabled on your '''sub-account setting''' under '''Advanced settings''')''

On the left menu click '''Advanced Settings''' on the same page where '''Security Settings'''
* '''BLF Call-pickup Prefix''': Check the box and type "*8" (Star 8).
|}
Replicate these for all accounts you may use.

[[Category:Grandstream]]

Grandstream CloudUCM

2026-04-14T20:22:54Z

RogueScholar: Add to :Category:Grandstream

[[File:VoIPms_Grandstream_CloudUCM_Logo.jpg|250px|right]]
CloudUCM is a cloud PBX that provides a scalable business communication and collaboration platform with powerful features and integrations that enable teams to be more productive than ever before.

'''Website''': [https://www.grandstream.com/products/ip-pbxs/ucm-series-ip-pbxs/product/clouducm CloudUCM]

== Login into your device ==

You can log directly to your instance URL or accessing it directly from your GDMS account.
[[File:VoIPms_Grandstream_CloudUCM_Login.jpg|none|550px]]

== Creating a trunk ==
: Once Logged into your admin console, to create a VoIP.ms trunk, navigate to the '''"Extension"''' then '''"Trunk"''' section and select '''"VoIP trunks"''' from the left panel. Within this interface, you'll create a SIP trunk.

:: [[File:VoIPms_Granstream_CloudUCM_trunk_Menu.png|none|650px]]

===SIP Trunk===
----
VoIP trunks can be configured within CloudUCM's Web GUI interface by navigating to '''Extension/Trunk''' >> '''VoIP Trunks'''.

:: '''Type''': Register SIP trunk
:: '''Provider Name:: ''': Enter a name for your conviencen, it could also be "VoIP.ms" or "Main Trunk"
:: '''Host Name:: ''': Type your preferred POP server that you would like to be connected on. i.e. toronto5.voip.ms
:: '''Keep trunk CID:: ''': Enable it, if you want the trunk to send its own CID number or disable it if your extensions are going to send their own CID number.
:: '''Username:: ''': 100000 (replace with your main VoIP.ms account number or sub-account name)
:: '''Password:: ''': ********* (Enter your Sub Account password.

: [[File:New SIP trunk.png||none|650px]]

Once created, these trunks are displayed with details such as Provider Name, Type, Hostname/IP, Username, and options for editing and detecting the trunk.

: To add a new VoIP trunk, simply click on "Add SIP Trunk."
: For detailed configuration parameters, click on the ''edit'' icon.
: To set up Direct Outward Dialing (DOD) for the SIP Trunk, click on the corresponding option.
: To delete a VoIP trunk, click on the delete icon.

'''Note:''' Ensure to select the same VoIP server that corresponds to your DID number.
You can verify the server associated with your VoIP number through your VoIP.ms customer portal under '''DID Numbers''' >> '''Manage DIDs''', in the '''POP''' section.
It's crucial to match the server chosen here with the one in your portal to ensure incoming calls ring properly.

Click '''Save''' button, do not click '''Apply Changes''' yet.

====Extra SIP settings====
----
====Additional SIP Configuration====

Once you've set up your basic SIP trunk, it's time to optimize some settings. Begin by clicking on the edit icon for your trunk.

'''SIP Headers'''
: To ensure the proper sending of the '''"FROM"''' header, follow these steps:

: Set '''"fromuser:"''' to your main VoIP.ms account number or sub-account name (replace "100000").

[[File:SIP-basic settings.png|thumb|none|650px|Click to enlarge]]

'''Audio Codecs'''
Navigate to "Advanced Settings" and under "Codec Preference," prioritize the codecs supported by VoIP.ms: '''G.722''', '''PCMU''', and '''G.729''', in that order.

'''Trunk Caller ID'''
If you wish to transmit CID name and number from your trunk, utilize the '''PAI (P-Asserted-Identity)''' header in the following format: '''"CallerIDName" <CallerIDNumber>'''.

: '''Note:''' Enabling the PAI header without specifying values will default to the CID priority rules from Grandstream, allowing you to pass through the CID from extensions.

: '''Note:''' Ensure no Caller ID Number is set in your VoIP.ms portal to prevent overriding by your trunk.

'''NAT Keep Alive'''
: To prevent your modem from closing local SIP ports, activate:

: '''Enable Heartbeat Detection:''' Enabled
: '''Heartbeat Frequency:''' 50

After completing the configuration, click on the "Save" button, and optionally, click "Apply Changes" to enact them. Your trunk status should display as "Registered" on your VoIP.ms dashboard. Keep in mind that calls will not be operational until you establish your outbound and inbound routes.

== Creating your outbound route ==
Outbound routes play a crucial role in matching your dialing patterns and directing calls through the appropriate trunk.

: '''VoIP.ms recommends including the following patterns in your outbound route:'''

:* _1NXXXXXXXXX
:* _NXXXXXXXXX
:* _4XXX
:* _00.
:* _011.
:* _033.
:* _044.
: Ensure that each of your dial patterns is prefixed by the underscore character "_".

To create your outbound routes, navigate to '''Extension/Trunk >> Outbound routes''' from the left panel, and click on "Add".

::[[File:CloudUCM_Outbound_routes_general.png|none|650px]]

In this section, you only need to complete the following fields:

'''''Calling Rule Name''''': Select any desired name for this route.
'''''Pattern''''': Define the pattern that callers need to dial. You can utilize the suggested dial patterns provided by VoIP.ms, as mentioned above.
'''''Trunk''''': Choose the VoIP.ms trunk through which your call will be routed.
'''''Privilege Level''''': Determine the privilege level required for extensions to utilize this route. If you follow our suggested dial pattern, we recommend setting it to "International" to enable placing international calls. Note that extensions with lower privilege levels won't have access to this route. Hover over the field name for additional information.

[[File:VoIPms_CloudUCM_Outbound_routes_setup.png|none|650px]]

Note: If you wish to incorporate a dial-out prefix, simply type it after the underscore "_" character in your dial patterns. To remove this prefix during dialing, utilize the "Strip" field, where you can specify the number of digits to strip after the underscore "_" character.

<pre>
For example: If you intend to use "9" as a dial-out prefix, your pattern should be _9NXXXXXXXXX.
To remove this "9" when dialing out, set the "Strip" field to "1". This ensures that only one character (in this case, the number "9") will be stripped off.
</pre>

== Creating your inbound route ==
: With your inbound routes configured, you can efficiently manage all incoming calls from multiple phone numbers using just one trunk. This eliminates the need for multiple trunks, as the inbound routes can effectively receive and direct calls to their respective destinations within your UCM.

'''Note:''' You should not use more than one VoIP.ms trunk on the same device.

: To set up inbound routes, navigate to '''Extension/Trunk >> Inbound Routes''' from the left panel, and then click on "Add".

: [[File:Inbound routes main.png|thumb|none|550px|Click to enlarge]]

In this section, complete the following fields:

:* '''Trunks:''' Select your VoIP.ms trunk to specify where incoming calls should be directed.
:* '''Pattern:''' Enter your VoIP.ms DID phone number exactly as it appears in the "Manage DIDs" section of your customer portal. Ensure it is prefixed by the underscore "_" character. Avoid using wildcards and ensure the full number is used to prevent conflicts in the inbound pattern.
:* '''Default Destination:''' Choose the default destination to which your PBX should route incoming calls. This could include an internal extension, an IVR, a ring group, or other locally managed options within your PBX.

[[File:Inbound routes setup.png|thumb|none|800px|Click to enlarge]]

== External links ==
=== Guides and manuals ===
= Resources =
: Product Documentation
:: https://documentation.grandstream.com/article-categories/cloudUCM

: Administration
::* CloudUCM - User Manual
::* Manage CloudUCM Services Through GDMS
::* Cloud UCM Multi-Tenant Management User Guide
:: https://documentation.grandstream.com/knowledge-base/clouducm-user-manual/

: Using CloudUCM
::* CloudUCM - Endpoint Configuration
::* CloudUCM - Quick User Guide
:: https://documentation.grandstream.com/knowledge-base/clouducm-endpoint-configuration-guide/

: Deployment
::* CloudUCM Deployment Scenarios
:: https://documentation.grandstream.com/article-categories/deployment-clouducm/

[[Category:Grandstream]]
[[Category:PBXes]]

Grandstream HT802v2

2026-04-14T20:21:10Z

RogueScholar: Polish up some of the language and add to :Category:Grandstream

This guide is intended to assist with the process of configuring a '''Grandstream HandyTone HT802v2''' for use with [[VoIP.ms]] services. This device and its single-line counterpart, the HT801v2—along with their "v1" predecessors—have been the [[ATA Devices#Most Popular ATA Devices|most widely-used ATAs]] by our customers in recent years, according to internal network analytics.

== Background ==
[[wikipedia:Grandstream Networks|Grandstream]] announced the release of a new lineup of VoIP [[ATA Devices|Analog Telephone Adapters (ATAs)]] and hardware gateways on October 30, 2024 that would arrive to market at the start of 2025; included in the announcement were details about a replacement for the venerable [[Grandstream HandyTone 802 - HT802|HandyTone 802]] ATA named the HT802v2.<sup>{[[#CITEREF1|1]]|</sup><span id="REFCITE1" style="display: hidden;"></span> The clear intention was to refresh both product lines with upgraded hardware while avoiding any disruption to the widespread popularity the existing models already enjoyed, and so the new replacements generally targeted their predecessors' feature sets and form factors with exacting precision. The bulk of the announced changes were located "under the hood," such as being built atop modern SoCs that operated at faster clock speeds, had more onboard volatile memory and conformed to an expanded set of compliance standards. The physical designs and even the earlier model names/numbers were largely reused in their entirety (though retail SKUs and internal product numbers were, of course, changed), and for most the only visible change was a <code style="background-color: #eee; border: thin solid #ccc; color: #000; font-family: monospace; padding: 1px 4px;">v2</code> appended to their name (for "version 2").

=== Device details ===
The HT802v2 is an entry-level model intended for residential and light [[wikipedia:Small office/home office|SOHO (Small Office/Home Office)]] installations that is capable of leveraging a single network link into two simultaneously operable analog phone ports. Its form factor is very compact, measuring 10×10×2.95cm and weighing 114g, even utilizing one of the small, ubiquitous [[wikipedia:USB hardware#Micro connectors|USB Micro-B ports]] for the AC power input (expecting 5VDC/1A), which allows it to be powered by all but the most feeble of mobile phone chargers manufactured in the last decade. Yet despite its diminutive footprint, it nevertheless manages to provide a range of features that is both deep and broad, including:
* Support for two SIP profiles, compatible with both [[Order a DID Number|VoIP.ms DIDs]] and [[Sub Accounts|sub-accounts]], each mapped to a dedicated FXS (RJ11) phone port
* Incoming and outgoing [[Caller ID]] with support for custom formatting and ringtones
* A [[wikipedia:Fast Ethernet#Copper|10/100Mbps Ethernet over twisted pair]] port
* Optional [[Call Encryption - TLS/SRTP|TLS and SRTP encryption protocols]] to secure calls and accounts
* [[Audio Conferencing|Three-way call conferencing]]
* Call transfers, [[Call Forwarding|forwarding]], hold and do-not-disturb
* SIP server failover to ensure your device stays connected to the service even if the default server goes offline
* [[Virtual Fax|Fax over IP capability]] (using either T.38 or G.711 passthrough)

<gallery caption="Grandstream HandyTone HT802v2 photos (click to expand)" heights="224" perrow="3" widths="224" style="margin: 1.5em auto; text-align: center; width: fit-content;">
File:Grandstream HandyTone HT802v2 - Front View.png|Front view
File:Grandstream HandyTone HT802v2 - Top View.png|Top view, showing status LEDs for power, network connection, and in-use indicator for both analog phone ports
File:Grandstream HandyTone HT802v2 - Rear View.png|Rear view with Ethernet and dual phone ports, USB Micro-B power input and pinhole reset
</gallery>

== Configuring your device ==
[[File:HT802v2 FXS1 General Settings AccountRegistration.png|thumb|right|512px|FXS Port 1 General Settings > Account Registration]]

=== Prerequisites ===
* Grandstream HT802 V2
:* Latest firmware installed ([https://www.grandstream.com/support/firmware|check Grandstream site]) ''(We recommend that users update their device firmware to at least version 1.0.5.7.)''
:* VoIP.ms account with a configured DID and a sub-account
:* Internet access via Ethernet
:* Phone or fax device connected to FXS Port 1

=== Logging into the web interface ===
: Connect the HT802 to your network and power it on.
::* Pick up the connected phone, dial: <code style="background-color: #eee; border: thin solid #ccc; color: #000; font-family: monospace; padding: 1px 4px;">* * * 0 2</code> (Star Star Star 0 2)

The ATA will read its IP address.
Open a browser and go to that IP (e.g., http://192.168.1.50).
: Login
:* Username: <code>admin</code>
:* Password: found on device label (or as set by you)

=== General Settings (FXS PORT1) ===
[[File:HT802v2 FXS1 General Settings NetworkSettings.png|thumb|right|512px|FXS Port 1 General Settings > Network Settings]]

==== Account Registration ====
:* '''Account Active:''' ✅ Checked
:* '''Primary SIP Server''': popserver.voip.ms (or your nearest VoIP.ms POP) ''eg: montreal1.voip.ms''
:* '''Prefer Primary SIP Server''' Yes
:* '''Outbound Proxy''' popserver.voip.ms (Same as the Primary SIP Server)
:* ''Backup Outbound Proxy This is optional you can enter a different VoIP.ms PoP Server location. eg: toronto2.voip.ms''
::* ''Prefer Primary Outbound Proxy: Check the box, only if you entered a backup Outbound Proxy''
:* '''SIP User ID''' Your VoIP.ms sub-account username
:* '''SIP Authenticate ID''' Same as SIP User ID
:* '''SIP Authentication Password''' Your sub-account password

==== Network Settings ====
:* '''DNS Mode''': A Record
:* '''NAT Traversal''' Keep-Alive
: Click '''Save and Apply'''.

=== SIP Settings Tab ===
[[File:FXS Port1 Advanced Settings.png|thumb|right|512px|FXS Port 1 Advanced Settings > Security Settings]]

==== SIP Basic settings ====
:* '''SIP Registration''': Yes
:* '''Register Expiration''': 2 (minutes)
:* '''Enable SIP OPTIONS Keep Alive''': Yes
:* Local SIP Port: 5060 (change if multiple ATAs)

: Disable extra headers:
:* '''Privacy Header''' → No
:* '''P-Preferred-Identity Header''' → No
:* '''P-Access-Network-Info''' → No
:* '''P-Emergency-Info''' → No

==== Codec Settings Tab ====
: '''Preferred Vocoder''' (in all sessions): G722, (G.711 μ-law) then set G711 for the rest.

==== Advanced Settings ====
: '''Check SIP User ID for Incoming INVITE''': Enabled
: '''Allow incoming SIP messages from SIP proxy only''': Enabled

==== Faxing ====
[[File:FXS Port1 Call Freatures Settings.png|thumb|right|512px|FXS Port 1 Call Features Settings]]
: Ensure options like Enable‑Call‑Waiting, Caller‑ID, and Tone are unchecked — avoiding disruptions during fax sessions.
: Security Settings: Enable Allow incoming packets from SIP proxy only to restrict access to VoIP.ms servers.

: Under '''Call Settings'''
: '''Disable Local Call Features''': Yes
:* '''Enable-Call-Waiting''': uncheck
:* '''Enable-Call-Waiting-Caller-ID''': uncheck
:* '''Enable-Call-Waiting-Tone''': uncheck

: Under '''Call Features Settings''':
:* Set ''' Enable Local Call Features''' '''No''' to prevent unintended feature code usage via the fax machine.

: Under '''Codec Settings''':
::* '''Fax Mode''': T.38 (if you are using it for faxing with a VoIP.ms FAX POP server)
::* '''T.38 Max Bit Rat'''e: 14400 bps
::* '''Jitter Buffer''': Fixed, High

: Under '''Analog Signal Line Configuration'''
::* '''Enable Line Echo Canceller (LEC)''' → Check

== Testing ==
: Reboot the ATA after applying settings.
: Check '''Status tab''' → Registration should be '''Yes'''.
: Dial: <code style="background-color: #eee; border: thin solid #ccc; color: #000; font-family: monospace; padding: 1px 4px;">4443</code> for a Voice Echo Test.

== References ==
# [[#REFCITE1|^]] <span class="plainlinks" id="CITEREF1">Bowers, Phil. (October 30, 2024) "[https://blog.grandstream.com/press-releases/grandstream-releases-new-atas-and-voip-gateways Grandstream Releases New ATA's and VoIP Gateways]." ''[https://blog.grandstream.com/ Grandstream Networks, Inc.]'' [https://web.archive.org/web/20241101213744if_/https://blog.grandstream.com/press-releases/grandstream-releases-new-atas-and-voip-gateways Archived] from the original on Nov. 1, 2024. Last accessed on Feb. 23, 2026.</span>

== External links ==
* [https://www.grandstream.com/products/gateways-and-atas/analog-telephone-adaptors/product/ht802v2 Official product page]
* [https://www.grandstream.com/hubfs/Product_Documentation/HT802%20v2/Datasheet_HT802_V2_English.pdf Product datasheet and specifications] (PDF)
* [https://documentation.grandstream.com/article-categories/ht8xx-series Grandstream Documentation Center]
** [https://documentation.grandstream.com/knowledge-base/ht802-v2-quick-installation-guide/ HT802v2 Quick Installation Guide]
** [https://documentation.grandstream.com/knowledge-base/ht80x-v2-administration-guide/ HT802v2 Administration Guide]
** [https://documentation.grandstream.com/knowledge-base/ht80x-v2-user-guide/ HT802v2 User Guide]
* [https://www.grandstream.com/support/firmware Firmware download site]

[[Category:Analog Telephone Adapters]]
[[Category:Grandstream]]

Grandstream GXP2120 IP Phone

2026-04-14T19:31:05Z

RogueScholar: Add to :Category:Grandstream and :Category:SIP phones

[[File:Gxp2110.png|300px|thumb|left|Grandstream GXP2120 IP Phone]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Configuration Detail==

'''Step 1:'''

You must first determine what IP address the IP Phone was assigned. To do this, press the round menu button, then the down arrow next to it until the display shows:

[2] IP Addr

Now, press the round menu button again and you will be shown the IP address of your device (e.g. 192.168.0.100).

'''Step 2:'''

Open your web browser and go to the IP you just found: https://[DeviceIP]

'''Step 3:'''

You will be prompted to enter a password, the default is "admin", then click the "Login" button.

'''Step 4:'''

Click on '''ACCOUNT''', followed by the '''ACCOUNT 1''' (to configure your line 1). You need to modify only a few settings from the factory default.

Fill in the followings fields.

* '''SIP Server:''' atlanta.voip.ms (You can enter the address of any of our multiple servers, this is just an sample with Atlanta server)
* '''SIP User ID:''' 100000 (Replace with your VoIP.ms username, it is a six digit number)
* '''Authenticate Password:''' ********* (Type your VoIP.ms account Password)
* '''SIP Registration:''' Yes
* '''Register Expiration:''' 5
* '''Name:''' Your Outbound CallerID Name '''(See the requirements below)'''.

'''IMPORTANT for 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.
- '''Enter a max of 15 characters.''' 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.

'''Step 5:'''

Finally, please click the '''Update''' button and then '''Reboot''' at the bottom of the configuration screen.

[[File:Grandstream2021final1.jpg|800px|]]

==Ghost Calls==

If you receive Weird Calls such as from CallerID 100 or in the middle of the Night not showing in your CDR, please note that these calls are not going through our Network but rather through the internet directly to your IP Phone without authentication.

Users have the ability to deny calls that are not authenticated. You can find the option by navigating the phone web interface, clicking on Account X-->SIP Settings-->Security Settings and enabling "Authenticate Incoming invite"
You can also go to Account X -> SIP Settings -> Security Settings -> and enable "Accept Incoming SIP from Proxy Only"

==Guide Links==
In the event where you need the guide directly from Grandstream, you may find the user manual guide below:

User Manual : [http://www.grandstream.com/sites/default/files/Resources/gxp21xx_14xx_usermanual_english.pdf Download PDF]

[[Category:Grandstream]]
[[Category:SIP phones]]

Grandstream HandyTone 486

2026-04-14T19:28:47Z

RogueScholar: Add to :Category:Grandstream

[[File:HT486.jpg|300px|thumb|left|Grandstream HandyTone 486]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Configuration Detail==

'''STEP 0'''

Before powering the device, plug in an Ethernet cable from your Cable or DSL modem into the WAN port of the Handytone 486.

Doing this before powering the device facilitates the device getting an IP address via DHCP.

'''STEP 1'''

Connect an Ethernet cable from your PC to the Handytone 486 LAN port. Make sure that your PC Ethernet Internet Protocol (TCP/IP) is set to “Obtain an IP address automatically” .

'''STEP 2'''

Go to any browser equipped computer on your network and enter the address: http://192.168.2.1

'''STEP 3'''

You will be prompted to enter a password, the default is "admin", then click the "Login" button.

'''STEP 4'''

Go to BASIC SETTINGS tab.

If you wish to fall back on PPPOE if DHCP does not work and if your ISP assigns you IP number by PPPOE, you can enter the information assigned by your ISP into the circled areas.

[[File:Grandstream-install8 (1).gif|400px]]

'''STEP 5'''

Go to ADVANCED SETTINGS tab.

'''SIP Server''': atlanta.voip.ms (one of our multiple servers, you can choose the one closer to your location.)

'''SIP User ID''': 100000 (Replace with your 6 digits Main SIP account UserID or Sub Account user name, i.e. 123456 or 123456_sub)

'''Authenticate ID:''' 100000 (Replace with your 6 digits Main SIP account UserID or Sub Account user name, i.e. 123456 or 123456_sub)

'''Authenticate Password''': ********* (SIP Account Password)

'''Name:''' Enter your full name, this will show up as part of your caller ID.

'''User ID is phone number''': No

'''SIP Registration:''' Yes

'''NAT transversal''' : 10

[[File:Step5.jpeg]]

'''STEP 6'''

Click on the '''"Update"''' button at the bottom of the form.
[[File:Example.jpg]]

'''STEP 7'''

You will be brought to a confirmation screen on which you will click on the only option available: '''"Click to view your changes"'''.

'''STEP 8'''

Click on the '''"Reboot"''' button at the bottom of the form.
[[File:Example2.jpg]]

== See also ==
* [[Grandstream HandyTone 286]]

[[Category: Analog Telephone Adapters]]
[[Category:Grandstream]]

Grandstream HandyTone 286

2026-04-14T19:27:21Z

RogueScholar: Add to :Category:Grandstream

[[File:Grandstream286.gif|300px|thumb|left|Grandstream HandyTone 286]]
<div style="width:100%;overflow:hidden;clear:both"></div>

The HandyTone 286 manual can be [http://www.grandstream.com/support/ht_series/general/ht_support.html found here].

==Configuration Details==

'''STEP 1''' - Connect your router with the supplied Ethernet network cable to the HandyTone. Now connect your phone to the HandyTone. Finally plug the supplied power cable into the HandyTone.

'''STEP 2''' - Wait 60 seconds after plugging your HandyTone in.

'''STEP 3''' - Pick up the phone connected to the HandyTone and dial the * key on your phone 3 times (* * *).

'''STEP 4''' - 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 HandyTone such as - "192.168.001.010" write this number down.

'''STEP 5''' - Open a web browser on your computer such as Internet Explorer and enter the IP address you heard in step 4 as the address (I.E. where you would normally enter www.yahoo.com).
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".

'''STEP 6''' - You should now see a page that looks like this:

[[File:Gstream login.gif]]

'''STEP 7''' - Enter the password for the HandyTone in the password field. The default administrator password for the HandyTone is "admin" (without quotes).

'''STEP 8''' - After entering the password you should see a screen that looks roughly like this:

[[File:Gstream config ht286.gif]]

'''Step 9''' - Click on '''Account 1''' to configure your first line.

'''Step 10''' - Fill the followings fields.

* '''SIP Server:''' atlanta.voip.ms (or use one of our multiple servers)
* '''SIP User ID:''' 100000 (Replace with your 6 digits Main SIP account UserID or Sub Account user name, i.e. 123456 or 123456_sub)
* '''Authenticate ID:''' ********* (Replace with your 6 digits Main SIP account UserID or Sub Account user name, i.e. 123456 or 123456_sub)
* '''Authenticate Password:''' ********* (SIP Account Password)
* '''User ID is phone number:''' No
* '''Name:''' (Enter any name you like)
* '''SIP Registration:''' Yes
* '''Register Expiration:''' 3 (This setting is in minutes not seconds)
* '''NAT Traversal :''' No
** '''Notes''': NAT Traversal can be set to '''Yes''' if you are behind a router (even if you do not set a value in the STUN server field). Indeed according to the reference guide, with no specified STUN server, then the phone will only periodically (every 20 seconds by default) send a blank UDP packet (with no payload data) to the SIP server to keep the mapped port open on the NAT.
* '''Home NPA''': Empty (Some tests show that this field should be kept empty or you will encountered difficulties with some numbers ending with a busy signal [tested with 418 in home NPA, and a 1-877 number] )

----

===Message Waiting Indicator===

To enable this feature with your '''Grandstream HandyTone 286''' you need to associate a [[Voicemail|voicemail]] with the account (or subaccount) you have registered in the device and set the '''SUBSCRIBE for MWI''' option to '''YES'''.

----

===Known Issues===

When the device hangs up the call after 17-20 minutes into the call.

In both '''FXS''' and '''FXO''' config pages change these values:

'''UAC Specify refresher''': UAS

'''UAS Specify refresher''': UAS

'''Force INVITE''': Yes

[[Category:Analog Telephone Adapters]]
[[Category:Grandstream]]

Grandstream HandyTone 502 - HT502

2026-04-14T19:24:49Z

RogueScholar: Add to :Category:Grandstream

[[File:Ht502.jpg|300px|thumb|left|Grandstream HandyTone 502 - HT502]]
The '''Grandstream HandyTone 502''' is a two-line analogue telephone adapter with a built-in single port wired 10/100Mbps LAN router and network address translation (NAT). It is reported to support both tone and [[pulse dial]] (a capability which was removed from the later [[Grandstream HandyTone 702 - HT702|HT702]]).

A [http://www.grandstream.com/products/gateways-and-atas/analog-telephone-adaptors/product/handytone-502 product description] and manual is available from the manufacturer. You can access your User Manual here and it includes your device's star codes on page 25: http://www.grandstream.com/sites/default/files/Resources/ht502_user_manual.pdf <br clear="all"/>

==Connecting the HandyTone==

1. Connect a standard touch-tone analog telephone to PHONE port (or PHONE1,
PHONE2 port for HT386/496/502).

2. Connect a PSTN telephone line to LINE port (optional, applies to HT386/486/488/503 only).

3. Insert the Ethernet cable into the Ethernet port (HT286/386) or WAN port (HT486/488/496/502/503) of
HandyTone and connect the other end of the Ethernet cable to an uplink port (a router or a modem, etc.).

4. Connect a PC to the LAN port of HandyTone (optional, applies to HT486/488/496/502/503 only).

5. Insert the power adapter into the HandyTone and connect it to an electrical outlet.

6. Using the HandyTone embedded web server or IVR (Interactive Voice Prompt) menu, you can further
configure the phone using either a static IP or DHCP.

[[File:502.jpg]]

==Accessing to the Web Configuration==

1. From the analog phone, press *** to get into the IVR menu. Enter option 02 to obtain the HandyTone’s IP
address.

2. For HT486/488/496/502/503, please enable the “WAN side HTTP access” option by entering IVR
option 12 and press 9. A reboot or power cycle of the HandyTone is required after this change. You can
also access the HandyTone’s web configuration from a PC connected to the LAN port via 192.168.2.1.

3. Type the HandyTone’s IP address in your PC browser.

4. Log in using password “admin” to configure the HandyTone.

==Settings==

'''Step 1:'''

* Click on '''FXS 1''' to configure your first line.
* '''''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.'''''

'''Step 2:'''

Fill the followings fields.

*'''Account Active:''' Yes
* '''Primary SIP Server:''' atlanta.voip.ms (Pick one of VoIP.ms multiple [http://wiki.voip.ms/article/Choosing_Server VoIP Servers])
*'''Outbound Proxy:''' Set the same server you've configured at the '''Primary SIP Server''' field
*'''NAT Traversal:''' No, But 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:''' ********* (account password)

* '''SIP Registration:''' Yes
* '''Register Expiration:''' 5 minutes
*'''Preferred Vocoder:''' Select as primary option the PCMU codec and as secondary option the G729 codec
*'''Allow Incoming SIP Messages from SIP Proxy Only:''' Yes
[[File:HT502-02.png|600px]]

'''Step 3:'''
* (Optional) Select a customised dial plan.

'''Step 4:'''
* Click on '''Basic settings''' to configure your time zone.
* Set ''Time Zone'' to one of the predefined values (such as Eastern Time: GMT-5) or a self-defined time zone (for example: MTZ+6MDT+5,M4.1.0,M11.1.0) for your area.

: ''Don't forget to reboot your device after you've applied the recommended settings.''

== Dial plans ==
The use of a customised dial plan is optional.

A dial plan contains a series of digit sequences, separated by the | character, entirely enclosed within { curly brackets }. Each time a phone button is pressed, the ATA will attempt to match the digit sequence to the dial plan.

===Grandstream's default dial plan===

{ x+ | *x+ | *xx*x+ }

===VoIP.ms===

We recommend the use of the dial plan presented below, however, you can build your own dial plan depending on your needs
{911|[3469]11|*xx.|<=1>[2-9]xxxxxxxxx|1[2-9]xxxxxxxxx|<=1555>[2-9]xxxxxx|011xxxxxxxxxxxx.|011[2-9]x.| 4xxx | ***xx}
Replace the 555 with your area code if you want to be able to dial 7 digit numbers.

===Building your own dial plan===
In theory, a standard [[Dial Plan for Linksys ATAs|VoIP.ms dial plan]] could be adapted to Grandstream as:
: {911|310xxxx|<=1555>[2-9]xxxxxx|1[2-9]xx[2-9]xxxxxx|[2-9]xx[2-9]xxxxxx|*xx|***xxx|*xx.|[3468]11|822|0|00|4xxx|**275*x.|xxxxxxxxxxxx.}
This would attempt the following translations:
: 911 (the North American emergency number) is passed verbatim
: 310xxxx passes any seven-digit 310-xxxx numbers verbatim
: <=1555>[2-9]xxxxxx adds area code 1-555 to seven-digit local calls (which begin with anything other than 0 or 1)
: 1[2-9]xx[2-9]xxxxxx passes any full eleven-digit North American format (1-NXX-NXX-XXXX) telephone number verbatim
: [2-9]xx[2-9]xxxxxx passes any ten-digit North American format (NXX-NXX-XXXX) telephone number verbatim. This is not compatible with the 7-digit rule above.
: *xx passes any 'star' and two digit codes (like *73 to disable call forwarding)
: *xx. is similar but passes any code with 'star' followed by at least two digits
: ***xxx passes three stars followed by any three numeric digits
: [3468]11 passes 311, 411, 611 and 811 verbatim
: 822|0|00 passes 822, 0 or 00 verbatim
: 4xxx passes any four-digit number in the 4000-4999 range (such as 4443, an [[Dialing Codes|echo test]] for [[Troubleshooting Outgoing Calls|troubleshooting]])
: **275*x. passes any number with **275* followed by at least one digit (a [[Dialing Codes|dialling code]] to reach [http://www.sipbroker.com/sipbroker/action/providerWhitePages SIP Broker] numbers)
: xxxxxxxxxxxx. passes any number which consists of a dozen or more consecutive numeric digits

Unfortunately, in the Grandstream 502 dial plans, as soon as one rule is matched (except for x+ or x. which allow an arbitrary number of additional digits) the call is sent out immediately. (By contrast, the Linksys dial plan will wait for a short [typically five second] delay unless 'S0' is specified explicitly; later Grandstream models such as the [[Grandstream HandyTone 802 - HT802|HT802]] also support a three-second delay by placing a 'T' flag at the end of a rule). It's therefore not possible to deal with multiple conflicting alternatives on this ATA by waiting for the user to dial additional digits which could potentially match some other valid pattern.

For instance 444-3xxx (a seven-digit local call) and 4443 (an echo test) might both be desired as valid for some dial plans, but can't both be supported at once. Seven and ten-digit local calls also couldn't both be supported in the same dial plan.

For North American 7-digit dial (adds 1-555- to local calls):
: { 0x+ | *x+ | [2-9]11 | <=1555>[2-9]xxxxxx | 1[2-9]xx[2-9]xxxxxx |***xxx | **275*x+ | *xx*x+ }
For North American 10-digit dial:
: { 0x+ | *x+ | [2-9]11 | [2-9]xx[2-9]xxxxxx | 1[2-9]xx[2-9]xxxxxx |***xxx | **275*x+ | *xx*x+ }
For no translation of domestic numbers:
: { x+ | *x+ | ***xxx | **275*x+ | *xx*x+ }

Note: In the seven digit dial plan, replace 555 by the area code of your choice. The 4443 (Echo Test) and 4747 (DTMF Test) [[Dialing Codes|dialling codes]] will not be supported.

The dial plan structure (described on [http://grandstream.com/products/ht_series/ht502/documents/ht502_usermanual_english.pdf page 44 of the manual] is otherwise very similar to that described for the [[Dial Plan for Linksys ATAs]] with minor discrepancies:

{| class="wikitable"
|-
! Grandstream 502
! Cisco/Linksys ATA's
! Function
|-
| style="text-align:center;" colspan="2"|0 1 2 3 4 5 6 7 8 9 * # a b c d
| You can use any of these characters to represent a pressed phone digit.
|-
| style="text-align:center;" colspan="2"| x
| Any phone digit, [0-9]
|-
| style="text-align:center;" colspan="2"|[sequence]
| You can enter characters between brackets to create a list of acceptable digits. <br>For example, if you enter the range [1-5], the user may only press the digits from 1 to 5. <br>You can also use individual numbers, and certain other characters, in combination. For example [35-8*] allows the user to press 3, 5, 6, 7, 8 or *. [369]11 allows calls to 311, 611 or 911.
|-
| style="text-align:center;"|+ (plus) or . (period)
| style="text-align:center;"|. (period)
| You can use a period to accept zero or more entries of a give digit. <br>For example, '''01.''' allows the user to enter 0, 01, 011 and so on, while '''xx+''' allows any two or more numeric digits to be dialled.
|-
| style="text-align:center;"|<dialed=substituted>
| style="text-align:center;"|<dialed:substituted>
| This is used for sequence substitution, you can use this to indicate that certain numbers dialled are replaced by other characters. The ''dialled'' digits can be zero or more characters. <br>For example, with '''<=1555>xxxxxxx''' if the user dials a 7 digit number, the number 1555 is added to the beginning of the sequence. If the user presses 6782345, the system transmits 15556782345. ''<2=011>'' replaces digit 2 with 011 when dialling and ''< =1>'' adds a leading 1 to all numbers dialled
|-
| style="text-align:center;"|^ (leading carat)
| style="text-align:center;"|! (trailing exclamation point)
| You can use this character to prohibit a dial sequence. <br/>For example '''^1900x+''' will reject any sequence that starts with 1900.
|-
| style="text-align:center;"|{} (curly brackets)
| style="text-align:center;"|() (parentheses)
| These enclose the entire dial plan text
|-
| style="text-align:center;" colspan="2"| | (pipe character)
| These separate each of the individual rules within the dial plan
|}

There are no supported Grandstream 502 equivalents to these codes:
{| class="wikitable"
|-
! Cisco/Linksys ATA's
! Function
|-
| style="text-align:center;"|, (comma)
| No supported equivalent. Used between digits to play an “outside line” dial tone after a user-entered sequence.
|-
| style="text-align:center;"|S0 or L0
| No supported equivalent. Overrides the Short or Long inter-digit timer to 0 seconds
|-
| style="text-align:center;"|P#
| No supported equivalent. Pauses # seconds.
|}

The factory default Grandstream dial plan is: { x+ | *x+ | *xx*x+ } where:
: x matches any numeric digit
: x+ matches one or more digits
: 0 1 2 3 4 5 6 7 8 9 * # a b c d match those specific, individual digits only
Each rule is separated from the others by | and the full dial plan is enclosed in { } curly braces.

To ensure that user-dial strings such as *123 are passed through to activate features provided by VoIP.ms (as service provider) instead of being handled locally by the Grandstream ATA, the * sequence should be defined inside the dial plan; the resulting dial plan rule would contain: { *x+ }

== Known Issues and Resolutions ==

'''Hearing an Echo on the Line:'''
: Please go to your FXS port setting screen, the one you are using with our service, and verify that the option 'Disable Line Echo Canceller' is set to NO.
: You can also adjust the setting Gain, where the Rx (Other Person) is a gain level for signals transmitted by FXS and Tx (You) is a gain level for signals received by FXS.

'''Receiving Weird Calls such as from CallerID 100 or in the middle of the Night not showing in your CDR:'''
: These calls are not going through our Network but rather through the internet directly to your ATA Device.
: Please check in the Web GUI under each FXS port tab that this option is enabled: Allow Incoming SIP Messages from SIP Proxy Only: Set to YES.
: This will make sure you can only receive calls from VoIP.ms.

==Guide Links==
In the event where you need the guide directly from Grandstream, you may find the company guide below:

User Manual : [http://www.grandstream.com/sites/default/files/Resources/ht502_user_manual.pdf Download PDF]

[[Category: Analog Telephone Adapters]]
[[Category:Grandstream]]

Category:Grandstream

2026-04-14T19:23:21Z

RogueScholar: Create category page to aggregate device articles by manufacturer

This category contains articles about devices manufactured by [[wikipedia:Grandstream Networks|Grandstream Networks]], a Boston, USA-based communications technology company founded in 2002.

== External links ==
* [https://www.grandstream.com/ Official website]

[[Category:Devices]]

ATA Devices

2026-04-14T19:16:25Z

RogueScholar: Improve listings for Grandstream devices

{| cellspacing="0" style="width: 100%; background: linear-gradient(135deg, #fef2f2 0%, #fee2e2 100%); border: 1px solid #fca5a5; border-radius: 8px; padding: 25px; margin: 25px 0; box-shadow: 0 2px 10px rgba(248, 113, 113, 0.1);"
|-
| style="text-align: left;" |
<div style="font-size: 1.1em; font-weight: 500; color: #b91c1c; margin-bottom: 15px;">What is an ATA?</div>

An '''Analog Telephone Adapter (ATA)''' is a device used to connect one or more standard analog telephones to a digital telephone system (such as Voice over IP) or a non-standard telephone system.

'''Learn More:''' [https://wiki.voip.ms/article/Back_to_Basics_-_What_is_an_ATA%3F Back to Basics - What is an ATA?]<br>

'''Looking for an IP Phone?''' [[IP_Phones|Click here]] to see all models.
|}

__TOC__

== Compatibility ==
{| class="wikitable" style="width: 100%; margin: 20px 0; border-collapse: collapse; border: none; border-radius: 8px; overflow: hidden; box-shadow: 0 2px 8px rgba(0,0,0,0.06);"
! style="background: linear-gradient(135deg, #d0382d 0%, #b8261a 100%); color: white; padding: 18px 16px; font-weight: 700; text-align: center;" | '''Device Name'''
! style="background: linear-gradient(135deg, #d0382d 0%, #b8261a 100%); color: white; padding: 18px 16px; font-weight: 700; text-align: center;" | '''T.38 Faxing'''
! style="background: linear-gradient(135deg, #d0382d 0%, #b8261a 100%); color: white; padding: 18px 16px; font-weight: 700; text-align: center;" | '''SIP TLS'''
|-
| style="padding: 16px; background: #ffffff; border-bottom: 1px solid #f1f5f9;" | [[Atcom AG188N|Atcom AG188N]]
| style="padding: 16px; background: #ffffff; border-bottom: 1px solid #f1f5f9; text-align: center;" | ✅
| style="padding: 16px; background: #ffffff; border-bottom: 1px solid #f1f5f9; text-align: center;" | ❌
|-
| style="padding: 16px; background: #fafbfc; border-bottom: 1px solid #f1f5f9;" | [[Auerswald COMpact 5010|Auerswald COMpact 5010]]
| style="padding: 16px; background: #fafbfc; border-bottom: 1px solid #f1f5f9; text-align: center;" | ✅
| style="padding: 16px; background: #fafbfc; border-bottom: 1px solid #f1f5f9; text-align: center;" | ❌
|-
| style="padding: 16px; background: #ffffff; border-bottom: 1px solid #f1f5f9; font-weight: 600;" | [[Grandstream HandyTone 802 - HT802|Grandstream HT802]]
| style="padding: 16px; background: #ffffff; border-bottom: 1px solid #f1f5f9; text-align: center;" | ✅
| style="padding: 16px; background: #ffffff; border-bottom: 1px solid #f1f5f9; text-align: center;" | ✅
|-
| style="padding: 16px; background: #fafbfc; border-bottom: 1px solid #f1f5f9; font-weight: 600;" | [[Grandstream_HT802v2|Grandstream HT802V2]]
| style="padding: 16px; background: #fafbfc; border-bottom: 1px solid #f1f5f9; text-align: center;" | ✅
| style="padding: 16px; background: #fafbfc; border-bottom: 1px solid #f1f5f9; text-align: center;" | ✅
|-
| style="padding: 16px; background: #ffffff; border-bottom: 1px solid #f1f5f9;" | [[Media5 Mediatrix C7 and 4100|Mediatrix C7/4100]]
| style="padding: 16px; background: #ffffff; border-bottom: 1px solid #f1f5f9; text-align: center;" | ✅
| style="padding: 16px; background: #ffffff; border-bottom: 1px solid #f1f5f9; text-align: center;" | ✅
|-
| style="padding: 16px; background: #fafbfc; border-bottom: 1px solid #f1f5f9;" | [[OBi300|Polycom OBi300]]
| style="padding: 16px; background: #fafbfc; border-bottom: 1px solid #f1f5f9; text-align: center;" | ✅
| style="padding: 16px; background: #fafbfc; border-bottom: 1px solid #f1f5f9; text-align: center;" | ✅
|-
| style="padding: 16px; background: #ffffff;" | [[ReadyNet AC1000MS and AC1300MS|ReadyNet AC1000MS/AC1300MS]]
| style="padding: 16px; background: #ffffff; text-align: center;" | ✅
| style="padding: 16px; background: #ffffff; text-align: center;" | ❌
|}

== Devices ==
=== Atcom AG188N ===
{| cellspacing="0" style="width: 100%; background: white; border: 1px solid #e2e8f0; border-radius: 12px; padding: 25px; margin: 20px 0; box-shadow: 0 2px 10px rgba(0,0,0,0.06);"
|-
| style="width: 300px; vertical-align: top; padding-right: 25px;" |
[[File:Atcom-ag188n.jpg|300px|thumb|center|Atcom AG188N]]

| style="vertical-align: top;" |
<div style="font-size: 1.4em; font-weight: 600; color: #1e293b; margin-bottom: 15px;">Atcom AG188N</div>

'''Company:''' Atcom<br>
'''Ports:''' 1 FXS port<br>
'''Overview:''' Single-port voice gateway with built-in router and lifeline port support.

<div style="margin-top: 20px;">
[[Atcom AG188N|<span style="background: linear-gradient(135deg, #d0382d, #b8261a); color: white; padding: 12px 24px; border-radius: 20px; text-decoration: none; font-weight: 600; display: inline-block; box-shadow: 0 2px 10px rgba(208, 56, 45, 0.3);">Configuration</span>]]
</div>
|}

=== Auerswald COMpact 5010 ===
{| cellspacing="0" style="width: 100%; background: white; border: 1px solid #e2e8f0; border-radius: 12px; padding: 25px; margin: 20px 0; box-shadow: 0 2px 10px rgba(0,0,0,0.06);"
|-
| style="width: 300px; vertical-align: top; padding-right: 25px;" |
[[File:Auerswald_5010.jpg|300px|thumb|center|Auerswald COMpact 5010]]

| style="vertical-align: top;" |
<div style="font-size: 1.4em; font-weight: 600; color: #1e293b; margin-bottom: 15px;">Auerswald COMpact 5010</div>

'''Company:''' Auerswald<br>
'''Type:''' Communication center<br>
'''Overview:''' Multi-line communication system supporting analog, ISDN, and VoIP connections.

<div style="margin-top: 20px;">
[[Auerswald COMpact 5010|<span style="background: linear-gradient(135deg, #d0382d, #b8261a); color: white; padding: 12px 24px; border-radius: 20px; text-decoration: none; font-weight: 600; display: inline-block; box-shadow: 0 2px 10px rgba(208, 56, 45, 0.3);">Configuration</span>]]
</div>
|}

=== Grandstream HT802 ===
{| style="background-color: #fff; border: thin solid #e2e8f0; border-collapse: collapse; border-radius: 12px; box-shadow: 0 2px 10px #0000000f; margin: 1.5em 0; width: 100%;"
| rowspan="2" style="background-color: #f8f9fa; padding: 2.75rem 1rem; width: 300px;" | [[File:HT802 Device.jpg|frameless|center|middle|225px|link=Grandstream HandyTone 802 - HT802|Grandstream HandyTone HT802]]
| style="color: #1e293b; font-size: 1.45em; font-weight: 600; padding: 1.25rem 0 0 3.25em; text-decoration: underline;" | [[Grandstream HandyTone 802 - HT802|HandyTone HT802]]
|-
| style="padding-left: 3em; vertical-align: top;" | <ul style="font-size: 0.9rem; list-style-image: unset; list-style-type: none; margin: 0.75em 0 1.5em 2.75em;">
<li>'''Manufacturer:''' [[:Category:Grandstream|Grandstream]]</li>
<li>'''Ports:'''
* Two [[wikipedia:Registered jack#RJ11, RJ14, RJ25 wiring|<abbr title="Single-line telephone jack">RJ11C</abbr>]] [[wikipedia:FXO and FXS#Use in voice-over-IP systems|<abbr title="Foreign Exchange Station, signifying that the device generates a dial tone on a port">FXS</abbr>]] for analog phones
* One [[wikipedia:Fast Ethernet#Copper|Fast Ethernet]] network</li>
<li>'''Features:''' [[Virtual Fax|Virtual fax (T.38)]], [[Call Encryption - TLS/SRTP|call encryption (TLS & SRTP)]]</li>
<li>'''Overview:''' Two-port analog telephone adapter with encryption and automated provisioning.</li>
</ul>
<div style="margin-left: 1.5em;">[[Grandstream HandyTone 802 - HT802|<span style="background: linear-gradient(135deg, #d0382d, #b8261a); border-radius: 20px; box-shadow: 0 2px 10px #d0382d4d; color: #fff; display: inline-block; font-weight: 600; padding: 12px 24px; text-decoration: none;">Configuration</span>]]</div>
|}

=== Grandstream HT802v2 ===
{| style="background-color: #fff; border: thin solid #e2e8f0; border-collapse: collapse; border-radius: 12px; box-shadow: 0 2px 10px #0000000f; margin: 1.5em 0; width: 100%;"
| rowspan="2" style="background-color: #f8f9fa; padding: 2.75rem 1rem; width: 300px;" | [[File:Grandstream HandyTone HT802v2 - Top View.png|frameless|center|middle|225px|link=Grandstream HT802v2|Grandstream HandyTone HT802v2]]
| style="color: #1e293b; font-size: 1.45em; font-weight: 600; padding: 1.25rem 0 0 3.25em; text-decoration: underline;" | [[Grandstream HT802v2|HandyTone HT802v2]]
|-
| style="padding-left: 3em; vertical-align: top;" | <ul style="font-size: 0.9rem; list-style-image: unset; list-style-type: none; margin: 0.75em 0 1.5em 2.75em;">
<li>'''Manufacturer:''' [[:Category:Grandstream|Grandstream]]</li>
<li>'''Ports:'''
* Two [[wikipedia:Registered jack#RJ11, RJ14, RJ25 wiring|<abbr title="Single-line telephone jack">RJ11C</abbr>]] [[wikipedia:FXO and FXS#Use in voice-over-IP systems|<abbr title="Foreign Exchange Station, signifying that the device generates a dial tone on a port">FXS</abbr>]] for analog phones
* One [[wikipedia:Gigabit Ethernet#1000BASE-T|Gigabit Ethernet]] network
* One [[wikipedia:USB hardware#Micro connectors|USB Micro-B]] for AC power</li>
<li>'''Features:''' [[Virtual Fax|Virtual fax (T.38)]], [[Call Encryption - TLS/SRTP|call encryption (TLS & SRTP)]], touchtone configuration, automated provisioning, self-updating firmware, custom ringtones, [[Pulse dial|pulse dialing]], [[wikipedia:Dynamic DNS|dynamic DNS]]</li>
<li>'''Overview:''' Updated version of the HT802 with enhanced performance and security features.</li>
</ul>
<div style="margin-left: 1.5em;">[[Grandstream HT802v2|<span style="background: linear-gradient(135deg, #d0382d, #b8261a); border-radius: 20px; box-shadow: 0 2px 10px #d0382d4d; color: #fff; display: inline-block; font-weight: 600; padding: 12px 24px; text-decoration: none;">Configuration</span>]]</div>
|}

=== Mediatrix C7 and 4100 Series ===
{| cellspacing="0" style="width: 100%; background: white; border: 1px solid #e2e8f0; border-radius: 12px; padding: 25px; margin: 20px 0; box-shadow: 0 2px 10px rgba(0,0,0,0.06);"
|-
| style="width: 300px; vertical-align: top; padding-right: 25px;" |
[[File:Mediatrix_C7.jpg|300px|thumb|center|Mediatrix C7]]

| style="vertical-align: top;" |
<div style="font-size: 1.4em; font-weight: 600; color: #1e293b; margin-bottom: 15px;">Mediatrix C7 and 4100 Series</div>

'''Company:''' Media5 Corporation<br>
'''Ports:''' Up to 24 analog ports<br>
'''Overview:''' Enterprise VoIP adaptors with SIP over TLS and SRTP security features.

<div style="margin-top: 20px;">
[[Media5 Mediatrix C7 and 4100|<span style="background: linear-gradient(135deg, #d0382d, #b8261a); color: white; padding: 12px 24px; border-radius: 20px; text-decoration: none; font-weight: 600; display: inline-block; box-shadow: 0 2px 10px rgba(208, 56, 45, 0.3);">Configuration</span>]]
</div>
|}

=== Polycom OBi300 ===
{| cellspacing="0" style="width: 100%; background: white; border: 1px solid #e2e8f0; border-radius: 12px; padding: 25px; margin: 20px 0; box-shadow: 0 2px 10px rgba(0,0,0,0.06);"
|-
| style="width: 300px; vertical-align: top; padding-right: 25px;" |
[[File:obi300.png|300px|thumb|center|Polycom OBi300]]

| style="vertical-align: top;" |
<div style="font-size: 1.4em; font-weight: 600; color: #1e293b; margin-bottom: 15px;">Polycom OBi300</div>

'''Company:''' Polycom<br>
'''Ports:''' 1 FXS port<br>
'''Overview:''' Single-port ATA with T.38 fax support for home office applications.

<div style="margin-top: 20px;">
[[OBi300|<span style="background: linear-gradient(135deg, #d0382d, #b8261a); color: white; padding: 12px 24px; border-radius: 20px; text-decoration: none; font-weight: 600; display: inline-block; box-shadow: 0 2px 10px rgba(208, 56, 45, 0.3);">Configuration</span>]]
</div>
|}

=== ReadyNet AC1000MS and AC1300MS ===
{| cellspacing="0" style="width: 100%; background: white; border: 1px solid #e2e8f0; border-radius: 12px; padding: 25px; margin: 20px 0; box-shadow: 0 2px 10px rgba(0,0,0,0.06);"
|-
| style="width: 300px; vertical-align: top; padding-right: 25px;" |
[[File:Readynet-ac1000mslogo.png|300px|thumb|center|ReadyNet AC1000MS]]

| style="vertical-align: top;" |
<div style="font-size: 1.4em; font-weight: 600; color: #1e293b; margin-bottom: 15px;">ReadyNet AC1000MS and AC1300MS</div>

'''Company:''' ReadyNet Solutions<br>
'''Type:''' Wi-Fi router with built-in ATA<br>
'''Overview:''' Dual-band Wi-Fi router with integrated two-line (AC1000MS) or one-line (AC1300MS) SIP ATA.

<div style="margin-top: 20px;">
[[ReadyNet AC1000MS and AC1300MS|<span style="background: linear-gradient(135deg, #d0382d, #b8261a); color: white; padding: 12px 24px; border-radius: 20px; text-decoration: none; font-weight: 600; display: inline-block; box-shadow: 0 2px 10px #d0382d4d;">Configuration</span>]]
</div>
|}

== Legacy devices ==
{| style="background: linear-gradient(135deg, #fef3c7 0%, #fde68a 100%); border: 2px solid #f59e0b; border-collapse: collapse; border-radius: 0.5em; box-shadow: 3px 3px 6px 6px #5e513d7a; margin: 1em 4% 2.5em; width: 92%;"
|+ style="color: #92400e; font-size: 1rem; font-weight: 700; margin-bottom: 0.5em;" | ⚠️ The following ATA devices are listed here for reference purposes, but may be subject to limited support, discontinued development or have reached end-of-life status ⚠️
| style="padding: 1.5em;" |
* <span style="font-size: 1.04em; font-weight: 500; text-decoration: underline;">Cisco / Linksys</span>
** '''[[Cisco Linksys PAP2|PAP2]]''' - First released in 2004, this model featured two RJ-11 FXS ports and one 10BaseT/100BaseTX RJ-45 network port; it was declared to be end-of-life and support was discontinued in 2014.
** '''[[Cisco Linksys PAP2T|PAP2T]]''' - Discontinued two-port ATA with enhanced features
** '''[[Cisco SPA112|SPA112]]''' - Discontinued two-port analog telephone adapter
** '''[[Cisco SPA2100 Phone Adapter|SPA2100]]''' - Discontinued phone adapter
** '''[[Cisco SPA2102 Phone Adapter with Router|SPA2102]]''' - Discontinued phone adapter with router
** '''[[Cisco SPA941|SPA941]]''' - Discontinued phone adapter
** '''[[Cisco WRP400|WRP400/WRP500]]''' - Discontinued wireless routers with built-in ATA (discontinued 2017)
* <span style="font-size: 1.04em; font-weight: 500; text-decoration: underline;">Grandstream</span>
** '''[[Grandstream HandyTone 486|HandyTone HT486]]''' - Legacy HandyTone model superseded by newer versions
** '''[[Grandstream HandyTone 502 - HT502|HandyTone HT502]]''' - Legacy HandyTone model with router functionality
** '''[[Grandstream HandyTone 702 - HT702|HandyTone HT702]]''' - Legacy HandyTone model superseded by HT802 series
* <span style="font-size: 1.04em; font-weight: 500; text-decoration: underline;">Netgear</span>
** '''[[Netgear WGR615V|WGR615V]]''' - Discontinued wireless router with built-in ATA
* <span style="font-size: 1.04em; font-weight: 500; text-decoration: underline;">ObiHAI</span>
** '''[[OBi 100/110 & OBi 200|OBi 100/110/200]]''' - Legacy OBiHAI devices (company acquired by Polycom)
* <span style="font-size: 1.04em; font-weight: 500; text-decoration: underline;">Telco</span>
** '''[[Telco AC-211|AC-211]]''' - Legacy ATA from defunct SunRocket service
* <span style="font-size: 1.04em; font-weight: 500; text-decoration: underline;">TP-Link</span>
** '''[[TP-Link TD-VG3631|TD-VG3631]]''' - DSL modem/router combo with VoIP functionality

----

<span style="font-size: 1.06em; font-weight: 600;">ℹ️ For new deployments, we recommend avoiding these devices and choosing one of the supported devices '''[[#ATA Devices|listed above]]'''.</span>
|}

[[Category:Analog Telephone Adapters]]

Latency, the problem you might have not been aware of

2026-04-14T15:47:57Z

RogueScholar: Fix double redirect

#REDIRECT [[Latency... the Problem You Might Have Not Been Aware Of]]

Latency, the problem you might have not been aware of.

2026-04-14T15:46:22Z

RogueScholar: Fix double redirect

#REDIRECT [[Latency... the Problem You Might Have Not Been Aware Of]]

Agregar artículos

2026-03-25T19:40:23Z

RogueScholar: Cambiar el estilo de la página y convertir enlaces wiki interlingüísticos en enlaces de sección más apropiados

{| class="wikitable" style="border: medium groove #ddd; border-collapse: collapse; box-shadow: 3px 3px 2px 1px #ccc; caption-side: bottom; float: right; margin: 0.4em 4rem 1.6em 2rem; text-align: center;"
|+ '''''Traducciones de páginas'''''
| style="border-bottom: none; font-weight: 500; min-width: 14em;" | Artículo en inglés
| style="border-bottom: none; font-weight: 500; min-width: 14em;" | Artículo en francés
|-
! scope="col" style="border-top: none;" | [[Add Articles|Add Articles (English)]]
! scope="col" style="border-top: none;" | [[Ajouter un Article|Ajouter un Article (Français)]]
|}

== Solicitud de autorización ==
Si desea ayudar contribuyendo con un nuevo artículo o agregando información a los artículos existentes, escriba a [mailto:support@voip.ms support@voip.ms] solicitando permiso de edición para el nombre de usuario que usa en este wiki.

== Escriba un nuevo artículo ==
Debe [[Special:UserLogin|iniciar sesión o crear una cuenta]] si desea crear un nuevo artículo en la wiki de VoIP.ms.

=== Paso 1: Artículo de búsqueda ===
Intente buscar el artículo, es posible que alguien ya haya escrito uno similar.

# Vaya a la búsqueda general y escriba el título de su artículo, luego presione el botón <span style="background-color: #eaebec; border: thin solid #ccc; border-radius: 0.25em; font-size: 0.88rem; font-weight: 500; padding: 0 8px;">Ir</span> (Go) como se muestra:
#: [[File:Searchtop2.png|frameless|alt=Búsqueda para artículo "Add Articles"|Cuadro de búsqueda]]
# Si no hubo resultados que coincidan con su consulta, la wiki presentará una pantalla como la siguiente:
#: [[File:Createthepage.png|border|alt=Pagina de resultados de la búsqueda para artículo "Add Articles"|Resultados de la búsqueda]]

=== Paso 2: Crea tu artículo ===
Todo lo que necesita hacer ahora es hacer clic en las letras rojas del resultado de la búsqueda y estará en camino de crear su nuevo artículo. Debe tener en cuenta lo siguiente al crear su nuevo artículo:
* Tenga en cuenta que la búsqueda que ingresó se utilizará como título, con las letras mayúsculas y minúsculas que escribió.
* El guión bajo "_" se convertirá en un espacio en blanco en el título del artículo o en cualquier lugar dentro de la wiki que utilice URL, imágenes o contenido multimedia.
*: <big>Consulta de búsqueda: <code style="border: thin solid #ccc; margin: 0.75em 0.25em; padding: 2px 6px;">Seguridad_en_PBX</code> ⇒ Título del artículo: [[Seguridad en PBX]]</big>

=== Paso 3: Escribe tu artículo ===
En el Área de texto puede escribir sobre el tema relevante que ha elegido.
* Sería una buena idea leer la sección [[#Formato Wiki|§ Formato Wiki]] antes de comenzar a escribir su artículo para mantener el formato wiki estándar.

=== Paso 4: Guarda tu artículo ===
# Antes de guardar tu artículo, anote el título para que pueda crear enlaces a él desde otros artículos en este wiki.
# Haga clic en el botón <span style="background-color: #eaebec; border: thin solid #ccc; border-radius: 0.25em; font-size: 0.88rem; font-weight: 500; padding: 0 8px;">Guardar página</span> (Save page) en la parte inferior de la página.

== Editar un artículo existente ==
Debe [[Special:UserLogin|iniciar sesión o crear una cuenta]] si desea editar un artículo existente en la wiki de VoIP.ms.

=== Paso 1: Artículo de búsqueda ===
# Vaya a la búsqueda general y escriba el título de su artículo, luego presione el botón <span style="background-color: #eaebec; border: thin solid #ccc; border-radius: 0.25em; font-size: 0.88rem; font-weight: 500; padding: 0 8px;">Ir</span> (Go) como se muestra:
#: [[File:Searchtop2.png|frameless|alt=Búsqueda para artículo "Add Articles"|Cuadro de búsqueda]]
# Si el artículo aparece en la página de resultados, haga clic en el título del artículo.
# Será redirigido al artículo y podrá leer su contenido.

=== Paso 2: Edite el artículo ===
Haga clic en el enlace [[File:Editwiki.png|alt=Botón editar|Botón editar]] en la parte superior del artículo para ver y editar el código fuente del artículo. Ahora puede editar el artículo como desee.

=== Paso 3: Guarde el artículo ===
Haga clic en el botón <span style="background-color: #eaebec; border: thin solid #ccc; border-radius: 0.25em; font-size: 0.88rem; font-weight: 500; padding: 0 8px;">Guardar</span> (Save) página en la parte inferior de la página.

== Formato Wiki ==
La siguiente imagen muestra las herramientas de edición proporcionadas por la wiki. Estas herramientas le permitirán utilizar un formato estándar en sus artículos.
: [[File:Wkitools.png|alt=Editar barra de herramientas|Editar barra de herramientas]]

=== Cómo utilizar la barra de herramientas (toolbar) ===
==== Texto en negrita ====
Utilice la herramienta de Texto en negrita ([[File:Toolboldtext.png|alt=Botón por texto en negrita|Botón Texto en negrita]]) para hacer que las palabras sean más notorias de lo normal y hacerlas más visibles que el resto del texto. Seleccione una parte del texto que le gustaría resaltar y luego haga clic en la herramienta texto en negrita para resaltarlos. Ahora haga clic en el botón <span style="background-color: #eaebec; border: thin solid #ccc; border-radius: 0.25em; font-size: 0.88rem; font-weight: 500; padding: 0 8px;">Avanzado</span> (Advanced).

'''Ejemplo:'''
<div style="border: thin solid #aaa; box-shadow: 1px 1px 1px 1px #0003; display: inline-block; font-family: monospace; font-size: larger; margin: 0.75em 0.25em 0.75em 2.5em; padding: 2px 6px 0; width: fit-content;"><nowiki>'''Esta es información importante que requiere énfasis.'''</nowiki></div> ⇒ '''Esta es información importante que requiere énfasis.'''

La salida del código debe ser el texto seleccionado rodeado de tres (3) comillas simples (<big><tt><nowiki>'''Texto seleccionado'''</nowiki></tt></big>) al principio y al final del texto. Puede usar las tres (3) citas simples para hacer partes de su artículo en negrita sin la necesidad de la herramienta texto en negrita.

==== Texto en cursiva ====
Utilice la herramienta de Texto en cursiva ([[File:Toolitalictext.png|alt=Botón por texto en cursiva|Botón Texto en cursiva]]) para hacer que parte del contenido se destaque de manera elegante. Seleccione el texto que le gustaría poner en cursiva y haga clic en la herramienta texto en cursiva.

'''Ejemplo:'''
<div style="border: thin solid #aaa; box-shadow: 1px 1px 1px 1px #0003; display: inline-block; font-family: monospace; font-size: larger; margin: 0.75em 0.25em 0.75em 2.5em; padding: 2px 6px 0; width: fit-content;"><nowiki>Su softphone: ''debería estar listo para hacer llamadas''</nowiki></div> ⇒ Su softphone: ''debería estar listo para hacer llamadas''

La salida del código debe ser el texto seleccionado rodeado de dos (2) comillas simples (<big><tt><nowiki>''Texto seleccionado''</nowiki></tt></big>) al principio y al final del texto. Puede utilizar las dos (2) citas simples para poner partes de su artículo en cursiva sin la necesidad de la herramienta texto en cursiva.

==== Enlace interno ====
Utilice la herramienta de Enlace interno ([[File:Toolinternallink.png|alt=Botón por enlace interno|Botón Enlace interno]]) para crear un enlace desde un texto seleccionado a un artículo de esta wiki. Seleccione el texto que le gustaría convertir en un enlace y presione la herramienta enlace interno. Tenga en cuenta que el texto seleccionado debe ser el nombre real del artículo para que la herramienta funcione.

'''Ejemplos:'''
* Enlace interno simple: <div style="border: thin solid #aaa; box-shadow: 1px 1px 1px 1px #0003; display: inline-block; font-family: monospace; font-size: larger; margin: 0.75em 0.25em; padding: 2px 6px 0; width: fit-content;"><nowiki>Consulte nuestro artículo [[Softphones]]</nowiki></div> ⇒ Consulte nuestro artículo [[Softphones]]
* Enlace interno con descripción: <div style="border: thin solid #aaa; box-shadow: 1px 1px 1px 1px #0003; display: inline-block; font-family: monospace; font-size: larger; margin: 0.75em 0.25em; padding: 2px 6px 0; width: fit-content;"><nowiki>Consulte nuestro artículo [[Softphones|Escriba una descripción aquí]]</nowiki></div> ⇒ Consulte nuestro artículo [[Softphones|Escriba una descripción aquí]]

==== Enlace externo ====
Utilice la herramienta de Enlace externo ([[File:Toolexternallink.png|alt=Botón por enlace externo|Botón Enlace externo]]) para crear un enlace desde un texto seleccionado a un artículo fuera de la wiki. Seleccione el texto que le gustaría convertir en un enlace y presione la herramienta enlace externo.

'''Ejemplos:'''
* Enlace externo simple: <div style="border: thin solid #aaa; box-shadow: 1px 1px 1px 1px #0003; display: inline-block; font-family: monospace; font-size: larger; margin: 0.75em 0.25em; padding: 2px 6px 0; width: fit-content;"><nowiki>Consulte https://es.wikipedia.com/</nowiki></div> ⇒ Consulte https://es.wikipedia.com/
* Enlace externo con descripción: <div style="border: thin solid #aaa; box-shadow: 1px 1px 1px 1px #0003; display: inline-block; font-family: monospace; font-size: larger; margin: 0.75em 0.25em; padding: 2px 6px 0; width: fit-content;"><nowiki>Consulte [https://es.wikipedia.com/ Escriba una descripción aquí]</nowiki></div> ⇒ Consulte [https://es.wikipedia.com/ Escriba una descripción aquí]
* Enlace de referencia externo: <div style="border: thin solid #aaa; box-shadow: 1px 1px 1px 1px #0003; display: inline-block; font-family: monospace; font-size: larger; margin: 0.75em 0.25em; padding: 2px 6px 0; width: fit-content;"><nowiki>Consulte [https://es.wikipedia.com/]</nowiki></div> ⇒ Consulte [https://es.wikipedia.com/]

La salida del código debe ser el texto seleccionado con un corchete (<big><tt><nowiki>[https://dominio.tld/artículo texto seleccionado]</nowiki></tt></big>) al principio y al final del texto. Puede utilizar los corchetes para crear enlaces externos sin la necesidad de la herramienta enlace externo. Recuerde agregar la parte http:<nowiki />// o https:<nowiki />// de los enlaces para que esto funcione como un enlace externo.

==== Título de nivel 2 ====
Utilice la herramienta de Título de nivel 2 ([[File:Toollevelheadline.png|alt=Botón por título de nivel 2|Botón Título de nivel 2]]) para crear subtítulos o definir un nuevo tema en su artículo. Seleccione el texto que le gustaría convertir en un subtítulo y haga clic en la herramienta título de nivel 2.

'''Ejemplo:'''
<div style="border: thin solid #aaa; box-shadow: 1px 1px 1px 1px #0003; display: inline-block; font-family: monospace; font-size: larger; margin: 0.75em 0.25em 0.75em 2.5em; padding: 2px 6px 0; width: fit-content;"><nowiki>== Detalles de configuración ==</nowiki></div> ⇓ <div style="border-bottom: thin solid #aaa; font-size: 1.5em; margin: 0 2.5em 0.6em 10em; padding: 0.5em 0 0.17em; width: auto;">1 Detalles de configuración</div>

La salida del código debe ser el texto seleccionado con dos (2) signos iguales al principio y al final (<big><tt><nowiki>== Texto seleccionado ==</nowiki></tt></big>). Puede utilizar los dos (2) signos iguales para crear subtítulos sin la necesidad de la herramienta encabezado de nivel 2.

==== Archivo incrustado ====
Utilice la herramienta de Archivo incrustado ([[File:Toolembeddedfile.png|alt=Botón por archivo incrustado|Botón Archivo incrustado]]) para vincular un texto seleccionado a una imagen o contenido multimedia dentro de la wiki. Seleccione el texto que le gustaría convertir en imagen o contenido multimedia y haga clic en la herramienta archivo incrustado. Tenga en cuenta que el texto seleccionado debe ser el nombre exacto del archivo o contenido multimedia que desea incrustar.

'''Ejemplos:'''
* Para usar la versión completa del archivo: <div style="border: thin solid #aaa; box-shadow: 1px 1px 1px 1px #0003; display: inline-block; font-family: monospace; font-size: larger; margin: 0.75em 0.25em; padding: 2px 6px 0; width: fit-content;"><nowiki>[[File:Toolembeddedfile.png]]</nowiki></div> ⇒ [[File:Toolembeddedfile.png]]
* Para usar una representación de 200 píxeles de ancho en un cuadro en el margen derecho con 'texto alternativo' y descripción: [[File:Wkitools.png|thumb|right|200px|alt=texto alternativo|Descripción]]
:: <div style="border: thin solid #aaa; box-shadow: 1px 1px 1px 1px #0003; display: inline-block; font-family: monospace; font-size: larger; margin: 0.75em 0.25em; padding: 2px 6px 0; width: fit-content;"><nowiki>[[File:Wkitools.png|thumb|right|200px|alt=texto alternativo|Descripción]]</nowiki></div> ⇒

El código de salida debe tener dos corchetes al principio y al final del texto seleccionado y la etiqueta de archivo. Puede utilizar los dos corchetes y la etiqueta Archivo para incrustar una imagen o contenido multimedia sin la necesidad de la herramienta archivo incrustado.

* Para agregar una imagen o contenido multimedia al artículo, ya debería haberse cargado.
* Siga la sección [[#Subir archivos|§ Subir archivos]] de este artículo para cargar su propio contenido en la wiki.

==== Nowiki ====
Utilice la herramienta de Nowiki ([[File:Toolnowiki.png|alt=Botón por nowiki|Botón Nowiki]]) para ignorar cualquier tipo de formato wiki y códigos wiki en un texto específico. Seleccione el texto que desea ignorar el formato wiki y luego haga clic en la herramienta nowiki.

'''Ejemplo:'''
<div style="border: thin solid #aaa; box-shadow: 1px 1px 1px 1px #0003; display: inline-block; font-family: monospace; font-size: larger; margin: 0.75em 0.25em 0.75em 2.5em; padding: 2px 6px 0; width: fit-content;"><nowiki><nowiki>'''Texto sin formato'''</nowiki></nowiki></div> ⇒ <nowiki>'''Texto sin formato'''</nowiki>

El código de salida debe ser para las etiquetas <big><code><nowiki></code></big> al principio y al final del texto seleccionado (<big><tt><nowiki>texto sin formato wiki</nowiki></tt></big>). Puede usar las etiquetas <big><code><nowiki></code></big> para evitar que el texto sea formateado como wiki sin la necesidad de la herramienta nowiki.

==== Línea horizontal ====
Utilice la herramienta de Línea horizontal ([[File:Toolhorizontalline.png|alt=Botón por línea horizontal|Botón Línea horizontal]]) para separar el contenido del artículo con una línea horizontal. Coloque el cursor donde desee la línea horizontal y luego haga clic en la herramienta línea horizontal.

'''Ejemplo:'''
<div style="border: thin solid #aaa; box-shadow: 1px 1px 1px 1px #0003; display: inline-block; font-family: monospace; font-size: larger; margin: 0.75em 0.25em 0.75em 2.5em; padding: 2px 6px 0; width: fit-content;"><nowiki>----</nowiki></div> ⇓<br /><br />
----

El código de salida debe tener quatro (4) guiones (<big><tt>----</tt></big>). Puede usar los quatro (4) guiones para crear una línea horizontal sin la necesidad de la herramienta línea horizontal.

== Subir archivos ==
Debe [[Special:UserLogin|iniciar sesión o crear una cuenta]] si desea cargar un archivo en la wiki de VoIP.ms.

=== Paso 1: Cargar archivo ===
# Vaya a la Caja de herramientas en el menú de la izquierda de la wiki y haga clic en el enlace '''[[Special:Upload|Cargar archivo]]''' ([[Special:Upload|Upload file]]).
# En la página de carga, haga clic en el botón <span style="background-color: #eaebec; border: thin solid #ccc; border-radius: 0.25em; font-size: 0.88rem; font-weight: 500; padding: 0 8px;">Elegir archivo</span> (Choose File). Tenga en cuenta que los formatos permitidos son *.png, *.gif o *.jpg/*.jpeg y el tamaño máximo de archivo es de 25 MiB.
#: [[File:Choosefile.png|frameless|alt=Botón para elegir un archivo|Botón Elegir archivo]]
#* Después de elegir el archivo de su computadora, puede ver que el nombre del archivo de destino ya está completado con el nombre del archivo que acaba de elegir.
#* Puede cambiar el nombre del archivo de destino a lo que desee.
#* Debe memorizar el nombre del archivo de destino para poder usarlo en un artículo.
#* Puede agregar un pequeño resumen del archivo si lo desea.
#* Si ya existe una imagen con el mismo nombre en la wiki, recibirá un mensaje similar al siguiente:
#*: <div style="border: thin solid #aaa; box-shadow: 1px 1px 1px 1px #0003; font-size: 0.9rem; margin: 0.75em 0.25em; padding: 0.4em 1em; width: fit-content;">Ya existe un archivo con este nombre, verifique [[:File:Example2.jpg|Archivo:Ejemplo2.jpg]] si no está seguro de querer cambiarlo.</div>
#* Si desea reemplazar el archivo anterior por el nuevo, solo necesita ignorar este mensaje.

=== Paso 2: Guardar archivo ===
# Haga clic en el botón <span style="background-color: #eaebec; border: thin solid #ccc; border-radius: 0.25em; font-size: 0.88rem; font-weight: 500; padding: 0 8px;">Cargar archivo</span> (Upload File) en la parte inferior de la página.
# Verá una vista previa del archivo en la pantalla, esto significa que el proceso de carga del archivo finalizó correctamente.

[[Category:Guías]]

User talk:Drouleau

2026-03-25T04:31:00Z

RogueScholar: Inquire about review of pending edits

== Drafts awaiting review ==
[[User:Drouleau|@Drouleau]]: I'm sorry to bother you, but I was wondering if I could prevail on you to review a couple of draft edits I submitted several weeks ago that are still stuck in the queue log, namely:
* [[Grandstream HT802v2]]
* [[SIP URI]]
I have a sizeable amount of experience with MediaWiki and VoIP and write a lot of documentation for other technical projects, all of which I hope will be readily apparent at first glance so that you won't be burdened with much time needed to review and approve them to go live. I didn't want to clog up the ticket system with such a minor request as this, so I hope it's okay to inquire with you directly like this simply from having seen you in the [[Special:RecentChanges|Recent Changes log]] often as of late and that you're a member of the Reviewers user rights group. Please inform me if there is someone else with whom I should liaise for Wiki-related matters like this or another mechanism by which they should be made. Also, is there a process for being added to the autoreviewed user group here, and if so, what are the prerequisites and how does one initiate it?

If you reply to me here, please do remember to include a wikilink to my username (as I did with you at the start of this message) so that I'll be notified of it on my next visit to the site. Thanks in advance for your assistance, <span style="float: right; margin: 0 1rem 0 2rem; white-space: collapse nowrap;">— [[User:RogueScholar|RogueScholar]] 00:31, 25 March 2026 (EDT)</span>

User:RogueScholar

2026-03-25T03:56:27Z

RogueScholar: Continue adding information

== About me ==
Oh boy, where to start? Alright, getting the basics out of the way…
{| class="wikitable" style="font-size: 1.05em;"
|+ style="font-size: 1.2rem; font-weight: 700; padding-bottom: 0.4rem;" | RogueScholar Vital Statistics
! scope="row" style="padding: 0.2em 1em;" | Gender
| style="padding: 0.2em 0.6em;" | Male
|-
! scope="row" style="padding: 0.2em 1em;" | Age
| style="padding: 0.2em 0.6em;" | Old enough to know better, but not so old that I don't still do it anyway just to reassure myself that I still can (a/k/a middle-aged)
|-
! scope="row" style="padding: 0.2em 1em;" | [[wikipedia:Ecoregion#Terrestrial|Terrestrial ecoregion]]
| style="padding: 0.2em 0.6em;" | [[wikipedia:Northern California coastal forests|Northern California coastal forests]] currently, but the [[wikipedia:Central Pacific coastal forests|Central Pacific coastal forests]] are the ultimate goal
|-
! scope="row" style="padding: 0.2em 1em;" | Languages spoken/written/understood
| style="padding: 0.2em 0.6em;" | [[wikipedia:American English|American English]] obviously, plus a long-standing (albeit rusty) fluency in [[wikipedia:European Portuguese|português europeu]] and a newly-acquired basic familiarity with [[wikipedia:Finnish language|suomi]] (enough to obviate any worries of starving to death or shitting myself for want of directions to the restroom while in Finland; claims of further skill are abundant but remain purely speculative)
|-
! scope="row" style="padding: 0.2em 1em;" | Formal education
| style="padding: 0.2em 0.6em;" | Dual (2) [[wikipedia:Bachelor's degree|baccalaureates]], both awarded by [[wikipedia:Public_university#United_States|state universities]] among the 79 classified as "[[wikipedia:Carnegie Classification of Institutions of Higher Education#Undergraduate/Graduate-Doctorate|Mixed Undergraduate/Graduate-Doctorate Large]]" and "Research 1: Very High Research Spending and Doctorate Production" in the 2022 [[wikipedia:Carnegie Classification of Institutions of Higher Education|Carnegie Classification of Institutions of Higher Education]], in the following disciplines:
* [[wikipedia:Engineering mathematics|Engineering mathematics]] and [[wikipedia:Engineering statistics|statistics]]
* [[wikipedia:Philosophy|Philosophy]] (with honors)
|-
! scope="row" style="padding: 0.2em 1em;" | Occupational expertise
| style="padding: 0.2em 0.6em;" | If they crunch bits and bytes, it's a near-certainty that I can operate, administer, architect interoperability at scale and (when needed) repair them, all with a high degree of skill. I've worked for the likes of Tektronix, Xerox, Boeing and others in addition to freelancing as a consultant on a number of large projects over the years. A rough chronology of the job titles I've held looks like:
* Technical Writer (2021–)
* Systems Analyst (2009–2021, seven years in Senior role)
* System Administrator (1997–2009, five years in Senior role)
|}

SIP URI

2026-03-03T07:48:25Z

RogueScholar: Reorganize lede section for improved coherency and add a more responsive layout to handle such a media-dense page

{| class="wikitable" style="border: medium groove #ddd; border-collapse: collapse; box-shadow: 3px 3px 2px 1px #ccc; caption-side: bottom; float: right; margin: 0.4em 4rem 1.6em 2rem; text-align: center;"
|+ '''''Page translations'''''
| style="font-weight: 500; min-width: 14em;" | Article en Français
| style="font-weight: 500; min-width: 14em;" | Artículo en Español
|-
! scope="col" | [[SIP URI FR|SIP URI (French)]]
! scope="col" | [[Dirección URI (SIP URI)|Dirección URI (Spanish)]]
|}
An '''SIP URI''' (Session Initiation Protocol Uniform Resource Identifier) identifies the connection target for calling another person via SIP. In other words, a SIP URI is a user's SIP phone number. The addressing schema resembles an email address and is always constructed according to this format:
<code style="background-color: #e8e9ea; border: thin solid #ccc; display: block; font-size: 1rem; margin: 1em 0 1em 15%; padding: 1px 4px; white-space: nowrap; width: fit-content;">sip:<var>user</var>@<var><host_domain/IP></var>[:<var>port</var>]</code>

One example of a valid URI would be <code style="background-color: #f4f4f4; border: thin solid #ccc; font-size: 1.25em; padding: 1px 4px;">sip:johnsmith@my-uri.com</code>.

A detailed treatment of the address schema and adjacent technologies can be found on English Wikipedia, specifically on the [[wikipedia:SIP URI scheme|SIP URI scheme]] and [[wikipedia:Telephone number mapping|telephone number mapping]] articles and those they link to. The addresses, which use the same <code style="background-color: #f4f4f4; border: thin solid #ccc; font-size: 1.25em; padding: 1px 4px;"><var>user</var>@<var>domain…</var></code> format as email addresses, allow an individual Internet telephony user to be called directly from another SIP client application/device without ever passing through the [[wikipedia:Public switched telephone network|public switched telephone network]] (PSTN) or incurring the tolls for doing so.

An SIP address may be used in a variety of ways, with some of the more common being:
* Receiving calls that originate from your SIP URI, using your DID number or an internal extension from a [[Sub Accounts|sub account]]
** We also offer "Virtual number" service for this use case, with the primary distinction that they can only connect to calls that are initiated using an SIP URI, and not traditional PSTN phone calls. By tightly defining their scope, we are able to price them very affordably at just 25¢/month plus 0.1¢/minute (one cent every ten minutes) for incoming calls.
* As a [[Call Forwarding|forwarding destination]] for a [[Manage DID|VoIP.ms DID number]]
* As a target for an individual speed dial entry (<div style="background-color: #f4f4f4; border: thin solid #ccc; display: inline-block; padding: 0 4px; vertical-align: top;">[[File:Material Design phone-dial icon (U+F1559).png|frameless|13px|link=]]<code style="font-size: 1.2em; font-weight: 500; margin: 0 0 0 0.4em; padding: 0;">＊75<var>xx</var></code></div>) in your [[Phone book#Create a Phone Book Entry with a SIP URI|VoIP.ms address book]]
* As a means of transferring incoming calls out to your various VoIP.ms extensions or numbers from outside Internet servers

Please note SIP URI is only functional through an external source, it is not intended to be used internally with VoIP.ms.

== Send calls to an external SIP URI address ==

You can route incoming calls to your DID numbers using a SIP URI address from other companies. You will need to create a string like youraccount@yourip to which you can route the DID(s).

: '''Make sure the other company or provider supports the use of SIP URI'''

=== Creating a new SIP URI ===
To make a SIP URI that can be used with multiple DIDs, use the {DID} tag. The {DID} expression will be replaced automatically by the DID number as it is listed in the "Manage DID section".
[[File:Forward.jpg|border|right]]
; Examples : <ul><li><code style="background-color: #f4f4f4; border: thin solid #ccc; font-size: 1.25em; margin: 1.5em 0; padding: 1px 4px;">1{DID}@128.144.122.12</code></li><li><code style="background-color: #f4f4f4; border: thin solid #ccc; font-size: 1.25em; margin: 1.5em 0; padding: 1px 4px;">12143221234@128.144.122.12</code></li><li><code style="background-color: #f4f4f4; border: thin solid #ccc; font-size: 1.25em; margin: 1.5em 0; padding: 1px 4px;">some_extension_name@128.144.122.12:5080</code></li><li><code style="background-color: #f4f4f4; border: thin solid #ccc; font-size: 1.25em; margin: 1.5em 0; padding: 1px 4px;">other_extension_name@voip.example.com</code></li></ul>

''Dialing out a SIP URI or using a SIP URI as forwarding does not generate cost (Outgoing call).''

==== Two newer options for SIP URIs ====
* CallerID Override: Permits you to override the callerID that will be received on the receiving end of the SIP URI.
* CallerID E164: Your CallerID will become E164 compliant and thus show on the receiving end as +12123262233 (+(countrycode)(areacode)(number)).

==== Enabling TLS on SIP URI ====
If you wish to proceed on having TLS enabled on your SIP URI, you will need to create an extended SIP URI, which you can do so with the information right below!

===== Extended SIP URI Format (BETA) - SIP URI TLS =====
In addition to the standard format outlined above, an Extended SIP URI Format can also be used. This Format contains additional options as indicated below:

<code style="background-color: #f4f4f4; border: thin solid #ccc; font-size: 1.25em; font-weight: 700; padding: 1px 4px;">username[:password[:md5secret[:authname[:transport]]]]@host[:port]</code>

To specify any of the additional parameters of the Extended SIP URI Format, you must introduce one or more colon (:) characters between the username value and the @ character. Parameters are optional and read from left to right. To supply a particular parameter, it is necessary to precede it with the appropriate number of colons to its left. For example, to specify a username, password and transport, the SIP URI would be as follows:

username:password:::transport@host

Specifying the username, password and authname would be done as follows:

username:password::authname@host

The additional parameters of the Extended SIP URI Format are described below:

'''password''': This is the plain text password to be used when authentication is required by the destination endpoint.

'''md5secret''': Alternatively an md5 representation of the password can also be used instead of the plain text version when authentication is required by the destination endpoint.

'''authname''': An optional authentication name can also be supplied as a parameter, which will be used instead of the username.

'''transport''': A specific transport type can be specified for the outbound connection. Valid values for this parameter are 'tcp' for the TCP transport, 'tls' for TLS encrypted signalling, as well as 'udp' for UDP transport (the default).

=== Creating a phone book entry ===
: ''More information: '''[[Phone book#Create a Phone Book Entry with a SIP URI|Phone book § Entries for an SIP URI]]'''''
A SIP URI may be associated with a [[phone book]] or speed dial entry in the same manner as any other telephone number.

[[File:Pb entry sipuri.jpg|800px]]

'''SIP URI''': Here you can either select '''Use Existing''' or '''Create New''' to assign the [[SIP URI]] to your Phone Book entry.

This replaces alphanumeric addresses (such as sip:user@provider.example.org) with numeric abbreviations (such as *7501) which can be easily dialled from [[devices|IP phones]] that only offer a numeric keypad.

=== Codec Negotation ===
By default when you route your incoming calls to an external SIP URI address, the system sends the INVITE allowing all VoIP.ms supported codecs (ulaw, g729a and GSM).
In that case if you want to use a specific codec (from the supported ones) you need to restrict that in your end. For instance, if you are using an Asterisk/PBX System and only wish to use ulaw codec, you will need to make sure to have the following settings in the trunk:

: disallow=all
: allow=ulaw

== Receiving incoming calls from an SIP URI ==
=== Using your DID number ===
You can receive SIP URI calls using the following format {Number}@sip.voip.ms, this can be used with your local US and Canada numbers, so they can be reached from outside.
[[File:Did.jpg|border|center]]

This format of a SIP address must follow this [[wikipedia:SIP URI scheme|SIP URI scheme]] as a means to reach VoIP.ms subscribers.

Another variant, also valid, is to specify the specific VoIP.ms server on which your DID is registered, ie:

:sip:4166471234@toronto.voip.ms

'''Please note that the option to dial sip.voip.ms is more reliable than using the server as you won't have to specify the Point of Presence. Using the server instead, you will have to dial the correct server that the number is using.'''

=== Using your sub account internal extension ===
When you assign an internal extension for a [[Sub Accounts|sub account]], it can also be used as an external SIP URI. For example, if your extension is 2, you could be reached directly via SIP from another network with a URI like this: 1000002@houston.voip.ms

:(Replace houston.voip.ms by the server you are registered to, 100000 by your account ID and the 2 by your internal extension).

Important: no call flow or filtering can be applied to calls make to the external SIP URI. Calls will immediately ring the device registered to this sub-account.
[[File:Extension.jpg|border|center]]

=== Using a Virtual number ===
Virtual SIP numbers are similar to standard DID numbers. The major difference is that virtual SIP numbers are not accessible via "PSTN". They can only be reached via "SIP URI" over internet. For example, if you have a DID number with another provider and they support SIP URI Forwarding, you could forward your number to a virtual number at voip.ms just like if it was one of our numbers.

All virtual numbers consist of the following digits: 11 + Accountcode + 3 digits of your choice for a total of 11 digits. The final uri will be that number followed by the @ sign at one of our server. If you intend to send the calls to a phone or adapter, you'll need to point it to the proper server.

: Example SIP URI: 11100000123@houston.voip.ms

[[File:Virtual SIP Number - Create new.png|border|center]]

== Reseller Configuration ==
[[File:ResellerClient SelectClient Only.png|frame|right]]
Also if you're using the [[Reseller Basic Guide|Reseller Interface]], you can associate each SIP URI feature with one of your reseller client.

'''Reseller Client''': Here you can select your reseller client that you want to associate this feature. You need first to create the account of your customer using the [[Reseller Basic Guide|Reseller section]] in your Customer Portal.<div style="clear: right;"></div>

=== SIP URI using the Reseller Interface ===
The feature is available for your client through the Reseller interface. You must enable this feature in your package in order to give them the ability to leverage this.

Note that the DID must be linked to your client. (Reseller > Manage Client's accounts > Click on '''Manage client''' where your client.)

# [[File:SIPURI Reseller 1.png|thumb|none|304x256px|Go under the navigation bar on '''[Reseller]''' then click on '''[Manage Rates & Packages]'''.]]
# [[File:SIPURI Reseller 2.png|thumb|none|880x96px|Click on the Edit button to edit your package, or click on '''[Create a new package]''' to create a new one.]]
# [[File:SIPURI Reseller 3.png|thumb|none|800x480px|Go under the '''[Reseller System Configuration]''' Tab, and on the section "Type of configuration" select: '''[Package Configuration]''']]
# [[File:SIPURI Reseller 4.png|thumb|none|720x352px|Then scroll down and find the feature "'''SIP URI'''", and enable it.]]

==== Add an SIP URI entry for a client ====
<div style="columns: 2;">
# [[File:SIPURI_Add.png|thumb|none|560x296px|To add a SIP URI entry for your client, or to help your client adding one. Go under the '''[Services]''' at the left navigation bar, then on '''[SIP URI]''']]
# [[File:SIPURI_Add2.png|thumb|none|760x320px|Click on the tab Add New SIP URI and enter the URI and a Description for the new entry. The description field will help the search bar or searchability.]]
# Click on the '''[Save SIP URI]''' button.
</div>

[[Category:Guides]]

File:Virtual SIP Number - Create new.png

2026-03-03T07:24:44Z

RogueScholar: Conversion of Media:Virtualsip.jpg to PNG image format as workaround to error message "Error creating thumbnail: Incomplete GD library configuration: missing function imagecreatefromjpeg".

Conversion of [[Media:Virtualsip.jpg]] to PNG image format as workaround to error message "Error creating thumbnail: Incomplete GD library configuration: missing function imagecreatefromjpeg".

File:Material Design phone-dial icon (U+F1559).png

2026-03-03T05:50:42Z

RogueScholar: * Source: [https://pictogrammers.com/library/mdi/icon/phone-dial/ pictogrammers.com] * License: [https://www.apache.org/licenses/LICENSE-2.0 Apache-2.0]

* Source: [https://pictogrammers.com/library/mdi/icon/phone-dial/ pictogrammers.com]
* License: [https://www.apache.org/licenses/LICENSE-2.0 Apache-2.0]

User talk:RogueScholar

2026-03-02T11:22:27Z

RogueScholar: Create my User talk page

If you've got a question for me or need to call my attention to something, create a new second-level heading at the bottom of this page named for the topic (<code>== <var>Message Topic</var> ==</code>) and write your message below it, making sure to include your wiki signature (<code><nowiki>~~~~</nowiki></code>, that's four consecutive tildes) at the very end. I don't visit this wiki very regularly so there may be a long delay before I respond, but I do check this page on every visit so the message delivery itself shouldn't be a concern.

User:RogueScholar

2026-03-02T11:11:24Z

RogueScholar: Create preliminary user page

Grandstream HT802v2

2026-02-24T10:02:06Z

RogueScholar: General improvements to page layout and formatting plus some introductory text with wikilinks to complementary documentation

This guide is intended to assist with the process of configuring a '''Grandstream HandyTone HT802v2''' for use with [[VoIP.ms]] services. This device and its single-line counterpart (HT801v2)—along with their "v1" predecessors—have been the [[ATA Devices#Most Popular ATA Devices|most widely-used ATAs]] by our customers in recent years, according to internal network analytics.

== Background ==
Grandstream announced the release of a new lineup of VoIP [[ATA Devices|Analog Telephone Adapters (ATAs)]] and hardware gateways intended to replace their original products of both types on October 30, 2024, intending them to be available for purchase through normal channels at the start of 2025; included in the announcement were details about the new HandyTone HT802v2 ATA.<sup>{[[#CITEREF1|1]]|</sup><span id="REFCITE1" style="display: hidden;"></span> Hoping to utilize the popularity of their existing crop of devices, the new lineup consisted of models that targeted the feature sets and form factors of each of the earlier models that were powered by modern SoCs operating at faster clock speeds and with more onboard volatile memory. Though retail SKUs and internal product numbers were of course changed, the new devices all bore the same names and model numbers as their predecessors, but with <code style="background-color: #eee; border: thin solid #ccc; color: #000; font-family: monospace; padding: 1px 4px;">v2</code> appended to each (for "version 2").

=== Device details ===
The HT802v2 is an entry-level model intended for residential and light SOHO (Small Office/Home Office) installations, capable of leveraging a single network link into two simultaneously usable analog phone ports. The device itself is very compact, measuring 10×10×2.95cm and weighing 114g, and utilizes a ubiquitous USB Micro-B port for AC power input (expecting 5VDC/1A)—allowing it to be powered by all but the most feeble of mobile phone chargers manufactured in the last decade. Despite its small form factor it provides a broad range of features, including:
* Support for two SIP profiles, compatible with both [[Order a DID Number|VoIP.ms DIDs]] and [[Sub Accounts|sub-accounts]], each mapped to a dedicated FXS (RJ11) phone port
* Incoming and outgoing [[Caller ID]] with support for custom formatting and ringtones
* A T-Base10/100 Ethernet (RJ45) port
* Optional [[Call Encryption - TLS/SRTP|TLS and SRTP encryption protocols]] to secure calls and accounts
* [[Audio Conferencing|Three-way call conferencing]]
* Call transfers, [[Call Forwarding|forwarding]], hold and do-not-disturb
* SIP server failover to ensure your device stays connected to the service even if the default server goes offline
* [[Virtual Fax|Fax over IP capability]] (using either T.38 or G.711 passthrough)

<gallery caption="Grandstream HandyTone HT802v2 photos (click to expand)" heights="224" perrow="3" widths="224" style="margin: 1.5em auto; text-align: center; width: fit-content;">
File:Grandstream HandyTone HT802v2 - Front View.png|Front view
File:Grandstream HandyTone HT802v2 - Top View.png|Top view, showing status LEDs for power, network connection, and in-use indicator for both analog phone ports
File:Grandstream HandyTone HT802v2 - Rear View.png|Rear view with Ethernet and dual phone ports, USB Micro-B power input and pinhole reset
</gallery>

== Configuring your device ==
[[File:HT802v2 FXS1 General Settings AccountRegistration.png|thumb|right|512px|FXS Port 1 General Settings > Account Registration]]

=== Prerequisites ===
* Grandstream HT802 V2
:* Latest firmware installed ([https://www.grandstream.com/support/firmware|check Grandstream site]) ''(We you use firmware version 1.0.5.7 or later.)''
:* VoIP.ms account with a configured DID and a sub-account
:* Internet access via Ethernet
:* Phone or fax device connected to FXS Port 1

=== Logging into the web interface ===
: Connect the HT802 to your network and power it on.
::* Pick up the connected phone, dial: <code style="background-color: #eee; border: thin solid #ccc; color: #000; font-family: monospace; padding: 1px 4px;">* * * 0 2</code> (Star Star Star 0 2)

The ATA will read its IP address.
Open a browser and go to that IP (e.g., http://192.168.1.50).
: Login
:* Username: <code>admin</code>
:* Password: found on device label (or as set by you)

=== General Settings (FXS PORT1) ===
[[File:HT802v2 FXS1 General Settings NetworkSettings.png|thumb|right|512px|FXS Port 1 General Settings > Network Settings]]

==== Account Registration ====
:* '''Account Active:''' ✅ Checked
:* '''Primary SIP Server''': popserver.voip.ms (or your nearest VoIP.ms POP) ''eg: montreal1.voip.ms''
:* '''Prefer Primary SIP Server''' Yes
:* '''Outbound Proxy''' popserver.voip.ms (Same as the Primary SIP Server)
:* ''Backup Outbound Proxy This is optional you can enter a different VoIP.ms PoP Server location. eg: toronto2.voip.ms''
::* ''Prefer Primary Outbound Proxy: Check the box, only if you entered a backup Outbound Proxy''
:* '''SIP User ID''' Your VoIP.ms sub-account username
:* '''SIP Authenticate ID''' Same as SIP User ID
:* '''SIP Authentication Password''' Your sub-account password

==== Network Settings ====
:* '''DNS Mode''': A Record
:* '''NAT Traversal''' Keep-Alive
: Click '''Save and Apply'''.

=== SIP Settings Tab ===
[[File:FXS Port1 Advanced Settings.png|thumb|right|512px|FXS Port 1 Advanced Settings > Security Settings]]

==== SIP Basic settings ====
:* '''SIP Registration''': Yes
:* '''Register Expiration''': 2 (minutes)
:* '''Enable SIP OPTIONS Keep Alive''': Yes
:* Local SIP Port: 5060 (change if multiple ATAs)

: Disable extra headers:
:* '''Privacy Header''' → No
:* '''P-Preferred-Identity Header''' → No
:* '''P-Access-Network-Info''' → No
:* '''P-Emergency-Info''' → No

==== Codec Settings Tab ====
: '''Preferred Vocoder''' (in all sessions): G722, (G.711 μ-law) then set G711 for the rest.

==== Advanced Settings ====
: '''Check SIP User ID for Incoming INVITE''': Enabled
: '''Allow incoming SIP messages from SIP proxy only''': Enabled

==== Faxing ====
[[File:FXS Port1 Call Freatures Settings.png|thumb|right|512px|FXS Port 1 Call Features Settings]]
: Ensure options like Enable‑Call‑Waiting, Caller‑ID, and Tone are unchecked — avoiding disruptions during fax sessions.
: Security Settings: Enable Allow incoming packets from SIP proxy only to restrict access to VoIP.ms servers.

: Under '''Call Settings'''
: '''Disable Local Call Features''': Yes
:* '''Enable-Call-Waiting''': uncheck
:* '''Enable-Call-Waiting-Caller-ID''': uncheck
:* '''Enable-Call-Waiting-Tone''': uncheck

: Under '''Call Features Settings''':
:* Set ''' Enable Local Call Features''' '''No''' to prevent unintended feature code usage via the fax machine.

: Under '''Codec Settings''':
::* '''Fax Mode''': T.38 (if you are using it for faxing with a VoIP.ms FAX POP server)
::* '''T.38 Max Bit Rat'''e: 14400 bps
::* '''Jitter Buffer''': Fixed, High

: Under '''Analog Signal Line Configuration'''
::* '''Enable Line Echo Canceller (LEC)''' → Check

== Testing ==
: Reboot the ATA after applying settings.
: Check '''Status tab''' → Registration should be '''Yes'''.
: Dial: <code style="background-color: #eee; border: thin solid #ccc; color: #000; font-family: monospace; padding: 1px 4px;">4443</code> for a Voice Echo Test.

== References ==
# [[#REFCITE1|^]] <span class="plainlinks" id="CITEREF1">Bowers, Phil. (October 30, 2024) "[https://blog.grandstream.com/press-releases/grandstream-releases-new-atas-and-voip-gateways Grandstream Releases New ATA's and VoIP Gateways]." ''[https://blog.grandstream.com/ Grandstream Networks, Inc.]'' [https://web.archive.org/web/20241101213744if_/https://blog.grandstream.com/press-releases/grandstream-releases-new-atas-and-voip-gateways Archived] from the original on Nov. 1, 2024. Last accessed on Feb. 23, 2026.</span>

== External links ==
* [https://www.grandstream.com/products/gateways-and-atas/analog-telephone-adaptors/product/ht802v2 Official product page]
* [https://www.grandstream.com/hubfs/Product_Documentation/HT802%20v2/Datasheet_HT802_V2_English.pdf Product datasheet and specifications] (PDF)
* [https://documentation.grandstream.com/article-categories/ht8xx-series Grandstream Documentation Center]
** [https://documentation.grandstream.com/knowledge-base/ht802-v2-quick-installation-guide/ HT802v2 Quick Installation Guide]
** [https://documentation.grandstream.com/knowledge-base/ht80x-v2-administration-guide/ HT802v2 Administration Guide]
** [https://documentation.grandstream.com/knowledge-base/ht80x-v2-user-guide/ HT802v2 User Guide]
* [https://www.grandstream.com/support/firmware Firmware download site]

[[Category:Analog Telephone Adapters]]

File:Grandstream HandyTone HT802v2 - Rear View.png

2026-02-24T07:52:35Z

RogueScholar:

File:Grandstream HandyTone HT802v2 - Top View.png

2026-02-24T07:52:09Z

RogueScholar:

File:Grandstream HandyTone HT802v2 - Front View.png

2026-02-24T07:51:33Z

RogueScholar: