VoIP.ms Wiki - User contributions [en]

Audacity

2025-06-24T18:38:14Z

Carlb: /* Editing */ - "such as" should refer to the concatenation, not the list of downlinks. The idea was to concatenate individual digits Example: a Tommy Tutone song from 1982 made 867-5309 unusable

'''Audacity''' is a free, third-party open-source audio editing tool, useful in creating [[recordings]] for upload to the system.

These [[recordings]], once uploaded, may be used as announcements section where you can select a greeting from uploaded audio recordings.
Such as example: [[Digital Receptionist (IVR)|IVR]]s, [[Calling Queues|calling queues]], [[Voicemail|voicemail greeting]] [[DID Troubleshooting#Failover options|failover options]], etc..., or even used as a destination to which to [[Manage DID|route a DID number]].

:'''IMPORTANT''': The sound file should be a non-compressed Windows .WAV sound file (extension .wav) with the format: '''PCM, 8kHz, 16 bits and Mono'''. Note that if its saved in '''MP3''', it will be converted in when you upload it through the portal.

== Download and installation ==
[[Image:Audacity-screenshot.png|388px|thumb|Audacity]]
Audacity is available for Windows, Linux PC and Macintosh computers; it may be downloaded freely at '''http://audacity.sourceforge.net/download/windows'''

Most Linux distributions include Audacity, which can be installed automatically from the package manager ''(usually "apt-get install audacity" or "yum install audacity")''.

== Recording ==
:# Select File → New
:# Select an input source (such as a microphone)
:# Adjust the recording levels using the microphone volume control at the top of the screen
:# Press "record" and record your message
:# Press "stop".

Your recorded audio is now ready to be edited or saved.

== Editing ==
Audacity has a wide range of capabilities, many of them overkill for simple IVR recording tasks or intended as special effects for editing music. A full manual is online at http://manual.audacityteam.org/o/ — here is a short introduction to a few commonly-used tasks:

;Trim a recording:
:# Use the mouse to select the portion of the audio track you wish to keep ''(go to the start, press and hold the button, drag the mouse to the end, release)''
:# Click the "trim audio" icon ''(which looks like a waveform with the beginning and end flatlined)''
:# All except the selected portion will be discarded, useful for removing blank/silent/unused segments before or after your actual recording

;Cut, copy and paste:
:# The standard scissors, clipboard and duplicate page icons are available:
:# Use the mouse to select the portion of the audio track you wish to copy
:# Click the "copy" icon
:# The audio may now be pasted into a different position in the same or another track ''(use File → Duplicate to create an identical second track for editing)''

;Concatenate two or more recordings:
:# Use File → Open to open the initial WAV file
:# Select Project → Import Audio to import a second WAV file
:# Click the Time Shift Tool ''(horizontal double-headed arrow button)''
:# Use the double-headed arrow to drag the start of the second WAV file to the right so that it starts after the first recording ends
:# Press Control-A ''(or Edit → Select all)'' to select all tracks at once
:# Select File → Export As WAV to save the combined WAV files as a single WAV

There are libraries with various stock phrases and standard announcements on http://downloads.asterisk.org/pub/telephony/sounds — these messages ''(two huge gzip archives):
are available in the correct format for the system but by design one often needs to concatenate or copy-and-paste multiple messages.

:* '''[http://downloads.asterisk.org/pub/telephony/sounds/asterisk-core-sounds-en-wav-current.tar.gz asterisk-core-sounds-en-wav-current.tar.gz]'''
:* '''[http://downloads.asterisk.org/pub/telephony/sounds/asterisk-extra-sounds-en-wav-current.tar.gz asterisk-extra-sounds-en-wav-current.tar.gz]''')''

Note that on each ZIP File you have a .txt file that will gives you the file name with the text script. Each file may contain a small snippet, which can be concatenated to create responses such as:

::* "The number you have reached..."
::* "...eight..."
::* "...six..."
::* "...seven..."
::* "...five..."
::* ".........."
::* "...has been disconnected."
::* "Please hang up..."

to identify individual extensions or individual error conditions. For these concatenations, audio editing software is invaluable.

== Export and upload ==
To save your recording in non-compressed Windows .WAV sound file (extension .wav) with the format: '''PCM, 8kHz, 16 bits and Mono''':
:# From the pull-down menu at the upper-left corner of the individual audio track, select "Mono"
:# From the "Project Rate (Hz)" pull-down box at the lower-left corner of the screen, select "8000"
:# From the "File → Export" menu item, select "WAV (Microsoft) signed 16-bit PCM" in the pull-down box above the "Save" button
:# Give your new .wav recording a descriptive name and click "Save"

To upload a recording, login to the customer portal and go to '''DID numbers''' → '''Recordings'''.

[[Image:Recording2.JPG]]

:* '''Name:''' you can set the name you want to identify this recording
:* '''File:''' here you need to select the recording you want to upload on your account.
:* '''Upload:''' click on this to start uploading procedure.

When you upload a new recording, the system can take '''up to 60 seconds''' to propagate this recording to all VoIP servers. It will not be playable until this process is complete. See [[Recordings]] for more information on using recordings once they have been uploaded to the system.

[[Category: Guides]]

Recordings

2025-06-24T18:33:12Z

Carlb: /* Using an existing recording */

{| class="wikitable"
|+
|-
! Article en Français !! Artículo en Español
|-
| [https://wiki.voip.ms/article/Enregistrements Français] ||
[https://wiki.voip.ms/article/Grabaciones_(Recordings) Español]
|}

'''How to use your own Recordings?'''
----

VoIP.ms allows you to upload an audio file and you can use it for different options that we have under DID Numbers Routing, [[Digital Receptionist (IVR)|Digital Receptionist]], [[Calling Queues|Calling Queues]] and others.

: '''IMPORTANT''': The sound file '''must''' be a non-compressed Windows .WAV sound file (extension .wav) with the '''Format: PCM, 8kHz, 16 bits and Mono.'''

__TOC__

== Setup Recording ==

=== Creating a Recording ===

If you are familiar with creating or editing sound files, please do so using sound editing software such as [[Audacity]] ([https://www.audacityteam.org/download free download]) for Windows, Linux, MacOS).

''' You can use an online converter such as https://g711.org that will allow you to upload your and convert your file. When you are on the page, simply leave all the settings as is and upload your file. Once uploaded, leave all settings as is and click on [submit]. When converted, Right-click on the link that it generates and click on Save As.'''

If you are unfamiliar with creating a sound file, please do not worry because there is an easy way to create your properly formatted recording. You can leave yourself a voicemail and use the file that is emailed to you as your recording to be uploaded under Recordings.

: Note: For this to work, please make sure you modify your [[Voicemail]] Mailbox in the [https://voip.ms/m/voicemail.php Voicemail Section] to change the format from compressed WAV49 format to uncompressed WAV format; also be sure to enable transmission of inbound messages as e-mail and save your settings.

=== Using an existing recording ===

There are various sound files readily available for frequently-used prompts; one common set recorded by Allison Smith [http://theivrvoice.com] is bundled with the free [[Asterisk (SIP)|Asterisk PBX]] software and matches the existing prompts on the system. This includes all of the standard phrases ("0" to "9", "is not available", "is not in service", "please leave a message"...) with various novelty prompts (including a few jokes) available as optional extras. [http://downloads.asterisk.org/pub/telephony/sounds/] Some corporate users purchase professional recordings from these or other sources; these customised prompts from name-brand voice talent tend to be expensive but can provide seamless integration for large-company [[DigitalReceptionist IVR|interactive voice response]] applications.

The stock Asterisk message libraries (asterisk-core-sounds-en-wav-current.tar.gz and asterisk-extra-sounds-en-wav-current.tar.gz on downloads.asterisk.org/pub/telephony/sounds) are available in multiple audio formats and languages.

Commercially-recorded announcements are usually created using high-end equipment in sound studios and recorded at higher quality than is needed (or usable) by the system, on the assumption that the audio can be downconverted using sound editing software to the required format (uncompressed .wav, PCM, 8kHz, 16 bits and Mono) before upload to the server.

The procedure is the same as that for importing your own recordings to the system.

=== How to upload a recording ===

To upload a recording, you need to login on the customer portal, then you have to go to DID numbers → Recordings. Once in there, click the "Upload New Recording" button and You will see a new window like this:
<div><ul>
<li style="display: inline-block;"> [[File:AddRecording.png|thumb|none|600px]] </li>
<li style="display: inline-block;"> [[File:RecordingFields.png|thumb|none|600px]] </li>
</ul></div>

* '''Name:''' you can set the name you want to identify this recording
* '''File:''' here you need to select the recording you want to upload to your account (Max Size: 25 MB).
* '''Upload:''' click on this to start uploading procedure.

: IMPORTANT NOTE: When you upload a new recording, the system can take '''up to 60 seconds''' to propagate this recording to all VoIP servers. It will not be playable until this process is complete.

Once you upload a recording you will see:

[[File:ManageRecordings.png|thumb|none|600px]]

=== Recording options ===
Once a recording is in your account you can execute the following actions:

* '''Test Dial Code''': Dial this code to hear the uploaded recording and confirm everything is correct.
: IMPORTANT NOTE : The system will attempt to convert your uncompressed (non wav49) .wav file into the required format once the upload is complete, however, the result can not be guaranteed if the file is not in the proper format.

* '''Download''': Use this option if you want to download this recording from your account.
* '''Re-upload''': Use this option if you want to upload again this recording in your account.
* '''Delete''': Use this option if you want to delete this recording from your account when you no longer need it.

=== How to use a recording ===

From our system, recordings can be used in different features, such as:
* '''[[Digital Receptionist (IVR)|Digital receptionist]]''' in your Customer Portal → DID Numbers.
* '''[[Calling Queues]]''' in your Customer Portal → DID Numbers.
* '''To [[Manage DID|Route a DID Number]]''' in your Customer Portal → DID Numbers → Manage DIDs → Select DID → Edit DID → Routing
* '''To Route [[DID Troubleshooting#Failover options|Failover Options]]''' in your Customer Portal → DID Numbers → Manage DIDs → Select DID → Edit DID → Failover

See the individual pages for these features for details.

== How to confirm a recording is proper for use ==

There's a dial code that you may use from a registered account on a device to confirm your recording was uploaded properly and in the correct format.

You can confirm an uploaded recording is good for use, by dialing the prefix dial code: "068" + the 5 unique digits of the recording, which you can check at the recordings section beside's the recording name set. This appears under the "Test dial code" column, i.e. 06856783.

Please note the dial plan within your device or system must allow 8 digit dialing for it to work.

If your recording is played back it's good for use. If the call is hanged up your recording was either corrupted, or it wasn't recorded according to our required format.

If the case, please confirm the format, delete and re-upload your recording.

== Recording (WAV File) 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.

Go under the navigation bar on '''[Reseller]''' then click on '''[Manage Rates & Packages]'''
: [[File:recording_Reseller_1.png|thumb|none|300px]]

Click on the Edit button to edit your package, or click on '''[Create a new package]''' to create a new one.

: [[File:recording_Reseller_2.png|thumb|none|700px]]

Go under the '''[Reseller System Configuration]''' Tab, and on the section "Type of configuration" select: '''[Package Configuration]''',

: [[File:recording_Reseller_3.png|thumb|none|700px]]

Then scroll down and find the feature "Recording", and enable it.

: [[File:recording_Reseller_4.png|thumb|none|500px]]

1) To add a new recording (Audio file) for your client, or to help your client adding one. Go under the [Services] at the left navigation bar, then on [Recording].

[[File:recording_Add.png|thumb|none|600px]]

2) Once on the page, click on [Add recording] tab. You will need to enter a description, and upload your audio file by clicking the [Select] button.

[[File:recording_Add_2.png|thumb|none|600px]]

3) Click [Save Recording]

Your Recording has been created successfully. It will be available to be selected in an IVR, Voicemail, Ring Group, Queue, DID Routing etc...

[[category:guides]]

Poly VVX-D230

2023-12-07T01:33:49Z

Carlb: /* Configuration */

[[Image:Poly VVX-D230 base.png|thumb|The VVX-D230's base can control ten DECT handsets]]
Aimed at small business users, the Poly (Polycom, Obahai) '''VVX-D230''' is a DECT [[IP Phones|SIP phone]] with support for up to ten cordless handsets — including the one handset which is packaged with the base.

This system (OBAHAI/VVXD230) is intended to complement the [[Polycom VVX 300, 400, etc|VVX-series]] desk phones and the [[Poly Trio 8800]] conference room speakerphone.

==Configuration==
# If the cordless handsets are not already paired to the base, press and hold the 'find' button on the base for at least five seconds. Then, from the handset, go to Menu→Settings→Registration→Register.
# Configuration of the system is done via a web interface. From the first paired handset, go to Menu→Settings→Basestation Info to obtain the IPv4 address of the base.
# Open a web browser to this address and log in with username 'admin' and default password 'admin'.

<br clear="all">
[[Image:Poly VVX-D230 Wizard.png]]

'''From the "Setup Wizard" page''', change the following parameters (changes will not be applied until you save them and reboot the device). Be sure to uncheck the "default" check box for each item you intend to modify:
*'''LocalTimeZone''' - set to your local time zone, for instance GMT-5:00 for (Eastern Time). Factory default is GMT-8:00 (Pacific Time, California).
*'''ITSP A SIPProxyServer''' - atlanta.voip.ms (or one of the other multiple [[servers]] - this must match the same server as your [[DID Troubleshooting|DID]] configuration).
*'''ITSP A DigitMap''' - (optional) (1xxxxxxxxxx|<1555>[2-9]xxxxxxS3|<1>[2-9]xxxxxxxxx|011xx.|xx.|(Mipd)|[^*#]@@.) (as an example, replacing 1-555 with your local area code) will enable both 7 and 10-digit dial for North American numbers. See the manufacturer's technical reference manual for details of the syntax.
*'''Phone1 PrimaryLine''' - "SP1 service"
*'''SP1 ITSP Profile''' - A
*'''SP1 AuthUserName''' - 123456_1 (your voip.ms user number and a [[Sub Accounts|subaccount]] number, in this example user #123456 subaccount #1
*'''SP1 AuthPassword''' - the password associated with your voip.ms subsccount
If you have additional virtual lines, repeat the same configuration steps for each subaccount using SP2, SP3... up to a maximum of eight lines and ten handsets.

<br clear="all">
[[Image:Poly VVX-D230 SP1.png]]

'''From the "SP1 Service" page''', change the following settings for the first virtual "line" (the others will be similar):
*'''X_DisplayNumber''' - A short text string identifying this line on the local handset screen. Typically the seven-digit NXX-XXXX local number, although any value will do.
*'''X_InboundCallRoute''' - DT1,DT2,DT3 (if this number is to ring to the first three extensions, for example). The default (DT1) rings only to the first extension on inbound calls.
*'''X_AcceptSipFromRegistrarOnly''' - YES (to keep [[Sip Scanner Ghost Calls|spurious]] calls out)
*'''X_KeepAliveEnable''' - YES (if you are behind NAT)
*'''X_KeepAliveExpires''' - 180 (three minutes) is reasonable
*'''DirectoryNumber''' - NPA-NXX-XXXX, the directory number associated with this line
*'''AuthUserName''', '''AuthPassword''' - should already be set from the "Setup Wizard" above; if not, set them here.
*'''CallerIDName''' - your name (15 alphanumeric chars max, no punctuation but spaces are allowed) as you want it to appear on [[Caller ID]] for outbound calls.
*'''X_CheckVoiceMailNumber''' - *97
*'''MessageWaiting''' - YES

Save everything before leaving each page of the setup. When you are finished making changes, reboot the device.

Try a test call, such as the [[Dialing Codes|echo test]] (press [4] [4] [4] [3] and the [green] button). You should hear your own voice played back just as it was received.

==See also==
* [[Polycom VVX 300, 400, etc|VVX-series]]
* [[Poly Trio 8800]]

==Documentation==
This page is merely a brief overview with just enough information to connect the handsets to VoIP.ms; see the manufacturer's website (poly.com) for more extensive documentation:

* Quick start: [https://www.poly.com/content/dam/www/products/support/voice/vvx-dect-ip-phones/setup/vvx-d230-hs-bs-class-a-qsg-access.pdf handset + base], [https://www.poly.com/content/dam/www/products/support/voice/vvx-dect-ip-phones/setup/vvx-d230-hs-chg-class-a-qsg-access.pdf expansion handset + charger]
* [https://www.poly.com/content/dam/www/products/support/voice/vvx-dect-ip-phones/user/vvxd230-adminguide-710.pdf Admin guide] and [https://www.poly.com/content/dam/www/products/support/voice/vvx-dect-ip-phones/user/vvxd230-tech-ref-710.pdf technical reference] manual from Poly.com

[[Category:SIP phones]]

Codes de Composition

2023-12-07T01:31:32Z

Carlb: /* Codes de service au Canada */

{| class="wikitable"
|+
|-
! Article in English !! Artículo en Español
|-
| [https://wiki.voip.ms/article/Dialing_Codes English] || [https://wiki.voip.ms/article/C%C3%B3digos_de_Marcado Español]
|}

VoIP.ms a des codes de composition différents en fonction de la destination que vous voulez rejoindre, et vous pouvez utiliser ces codes à partir de votre compte principal ou sous-comptes, selon le besoin. Il y a deux principales différences entre les routes lors de la numérotation, les appels locaux et internationaux, il est important de le mentionner, les appels aux États-Unis 48 (sauf Hawaï et Alaska) et au Canada sont considérés comme des appels locaux. Toutes les autres destinations seront considérées comme internationales et utiliseront la route correspondante.

== Appels locaux (USA48/Canada) ==

Pour les appels vers USA48 & Canada, il vous suffit de composer le numéro à 10 chiffres, vous pouvez le composer optionnellement en utilisant 11 chiffres, en ajoutant le préfixe 1 avant le numéro.

Exemple: 1514-316-xxxx or 514-316-xxxx (N'utilisez pas les tirets lors de la composition)

Dans la page '''Paramètres du compte''' > '''Routage de compte''', il est possible de configurer une route pour ces appels, et choisir entre la route "Value" ou "Premium".

* Value: Le Canada à partir de $0.0052

* Premium: USA 48/Canada $0.01. Remarquez que le Yukon, les Territoires du Nord-Ouest et le Nunavut sont considérés dans cette section, mais ont un taux différent (valeur à $0.0102 et $0.0260). ''

'''USA48: Ce sont les 48 états des États-Unis du continent de l'Amérique du Nord, situés au sud du Canada, en plus du District de Columbia. Ce terme exclut les États de l'Alaska et d'Hawaï, ainsi que tous les territoires en dehors de la zone des États-Unis, tel que Puerto Rico.'''

== Appels internationaux ==

Il existe des codes différents qui peuvent être utilisés pour composer des numéros internationaux (à l'extérieur de US48 & Canada). Hawaï et Alaska sont considérés dans cette section, puisque toute modification apportée à '''Paramètres du compte''' > '''Routage international''', aura une incidence sur ces destinations, cela signifie que si vous utilisez la route premium, les appels vers ces destinations utiliseront également la route premium, mais peuvent être composés comme des numéros locaux américains (10 ou 11 chiffres).

Remarque: Certaines destinations internationales peuvent être composées en utilisant uniquement le préfixe 1, cela s'applique pour une partie des pays de la NANPA. 011 & 00 sont pour le reste du pays.

Par exemple: Quand on appelle à Hawaï, Alaska, les pays des Caraïbes et les territoires américains.

Pour les appels vers des pays hors USA et Canada, nous pouvons utiliser ces codes:

* 011 + Code du pays + numéro: International
* 00 + Code du pays + numéro: International
* 033 + Code du pays + numéro: International Value
* 044 + Code du pays + numéro: International Premium

(Les signes + sont utilisés uniquement comme référence. Ne pas les inclure lors de la numérotation)

Exemple de composition pour le Mexique: 011 + le code du pays + numéro = 011-52-9999xxxxxx (ne pas utiliser les tirets lors de la numérotation).

== Codes spéciaux ==

Certains codes spéciaux que vous pouvez utiliser avec le service:

=== Les codes de [[Messagerie vocale]] ===

*97 pour accéder directement à la Messagerie vocale associée au compte. (Vous demandera un mot de passe)

*98 pour accéder à votre messagerie vocale et choisir un de vos comptes de boîte vocale. (Vous demandera code de Messagerie vocale et mot de passe)

Si vous n'avez pas accès à notre réseau VoIP et que vous souhaitiez consulter votre messagerie vocale, il vous suffit de composer votre numéro. Une fois que le système de messagerie vocale répond à votre appel, appuyez sur la touche astérisque (*).

=== Tests de qualite sonore et Test DTMF ===
4443 (Tests de qualite sonore): Ce code est utilisé pour accéder au Tests de qualite sonore, avec un nouveau compte ce code peut être composé même sans fonds, il est utile pour vérifier la qualité de votre ligne.

4747 (Test DTMF): Ce code est utilisé pour accéder au test DTMF, avec un nouveau compte ce code peut être composé même sans fonds, il est utile de vérifier si le DTMF (tonalité) est configuré correctement.

== Codes de service au États-Unis==

'''Note:''' Veuillez noter que les fonctions d'enregistrement et de transcription des appels ne fonctionneront pas pour les services 988, 911, 411.

'''1-555-555-0911''': Pour tester "[[ID de l'appelant (Caller ID) | ID de l'appelant]]" et "E911", pour tester si votre "[[ID de l'appelant (Caller ID) | ID de l'appelant]]" fonctionne correctement, vous pouvez composer ce numéro, cet enregistrement vous permettra également de savoir si votre numéro est activé avec le [[Service E911]].

'''988:''' 'Ligne d'assistance en santé mentale
'''Note:''' Veuillez noter que composer le 988 avec un numéro d'appelant sans frais vous redirigera vers le 411, puisqu'il ne s'agit pas d'un NPANXX nord-américain valide.

'''411:''' Pages Blanches (Doit être activé dans les paramètres du compte), quand il est activé les utilisateurs peuvent appeler l'annuaire de renseignements à un coût de 0,99 $ par appel.

'''911:''' Service d'urgence - Doit être activé sur votre DID au coût de 1,50 $ par mois. Pour plus d'informations, consultez [[Service E911 | Cet article]]

== Codes de service au Canada ==

'''Note:''' Veuillez noter que les fonctions d'enregistrement et de transcription des appels ne fonctionneront pas pour les services 988, 911, 411.

'''1-555-555-0911''': Pour tester "[[ID de l'appelant (Caller ID) | ID de l'appelant]]" et "E911", pour tester si votre "[[ID de l'appelant (Caller ID) | ID de l'appelant]]" fonctionne correctement, vous pouvez composer ce numéro, cet enregistrement vous permettra également de savoir si votre numéro est activé avec le [[Service E911]].

'''411''': Pages Blanches (Doit être activé dans les paramètres du compte), quand il est activé les utilisateurs peuvent appeler l'annuaire de renseignements à un coût de 0,99 $ par appel.

'''911:''' Service d'urgence - Doit être activé sur votre DID au coût de 1,50 $ par mois. Pour plus d'informations, consultez [[Service E911 | Cet article]]

'''311''': Un non-numéro d'urgence de la police, ou des services municipaux et gouvernementaux (serveurs canadiens)

Le service est disponible dans les villes suivantes (laissez-nous savoir si vous avez des problèmes):

Calgary, Alberta (18 mai 2005)
Edmonton, Alberta (16 décembre 2008)
Fort St. John, British Columbia (14 novembre 2006)
Gatineau, Quebec (22 juin 2005)
Greater Sudbury, Ontario (12 février 2007)
Halifax Regional Municipality, Nova Scotia (15 novembre 2012)[8]
''Halton Region, Ontario (18 mars 2008). Les appels vers 311 à partir de la région d'Halton ne sont actuellement pas portés par VoIP.ms. Veuillez utiliser 1-866-4HALTON
(1-866-442-5866) or 905-825-6000''
Laval, Québec (3 octobre 2007)
Montreal, Québec (mi-décembre 2007)
Ottawa, Ontario (19 septembre 2005)
Regional Municipality of Peel, Ontario (5 octobre 2009)
St. John's, Newfoundland and Labrador (27 juin 2006)
Toronto, Ontario (24 septembre 2009)
Vancouver, British Columbia
Windsor, Ontario (22 août 2005)
Winnipeg, Manitoba (16 janvier 2009)[9]

'''511''': Informations sur les prévisions météorologiques et les services aux voyageurs (serveurs canadiens)

'''811''': Services de santé (non urgents) / services de télésanté (serveurs canadiens)

'''988''': Ligne d'assistance en santé mentale
'''Note:''' Veuillez noter que composer le 988 avec un numéro d'appelant sans frais vous redirigera vers le 411, puisqu'il ne s'agit pas d'un NPANXX nord-américain valide.

'''Remarques importantes''':

* Ces services ne peuvent être appelés qu'à partir que de nos serveurs canadiens. Nous prévoyons ajouter des codes de service à nos serveurs US dès que possible.
* Ces services comptent sur votre CallerID pour acheminer votre appel vers le service en ligne approprié, par conséquent, certains services peuvent ne pas être pris en charge dans votre province.
* Il n'y a aucun frais appliqué à votre compte lorsque vous appelez le 311, 511 ou 811.

== Le service Canada 310 ==

310 est un code spécial, ces numéros ne sont accessibles qu'à partir de la région où ils appartiennent, depuis notre service, et vous devrez composer seulement les 7 chiffres, pour bien atteindre ces numéros, passer un "Caller ID" canadien valide et être enregistré à Chicago, Seattle ou l'un des serveurs canadiens.

Exemple: Composez 310-xxxx (ne pas utiliser les tirets lors de la numérotation), si vous passez par un "Caller ID" 514, c'est comme si vous
composiez le 514-310-xxxx.
[[category:Guides en français]]

Códigos de Marcado

2023-12-07T01:30:49Z

Carlb: /* Códigos de servicio para Canadá */

{| class="wikitable"
|+
|-
! Article in English !! Article en Français
|-
| [https://wiki.voip.ms/article/Dialing_Codes English] ||
[https://wiki.voip.ms/article/Codes_de_Composition Français]
|}

VoIP.ms cuenta con diferentes códigos de marcado dependiendo del destino que usted desee alcanzar, y se pueden utilizar desde la cuenta principal o una subcuenta dependiendo de sus necesidades. Es importante resaltar que existen dos tipos de rutas que determinaran la forma en que usted realice su marcación, ruta para llamadas locales y ruta para llamadas internacionales. Llamadas a Estados Unidos (excepto Hawái y Alaska) y Canadá son consideradas como llamadas locales. Todas los demás destinos serán considerados como llamadas Internacionales y usaran la ruta correspondiente.

== Llamadas locales ==

Para llamadas a Estados Unidos y Canada, usted sólo necesita marcar el número a 10 digitos, aunque de igual forma la marcacion con 11 digitos está permitida, para ello solamente agregue el prefijo 1 antes del número que desea marcar.

Ejemplo: 1514-316-xxxx o 514-316-xxxx (No use guiones al marcar)

En el caso de Canada, desde Configuración de la Cuenta (Account Settings)>> Enrutamiento de la cuenta (Account Routing), se puede establecer una ruta que se usará para estas llamadas, hay una opción para usar la ruta de valor o premium ([[Value vs. Premium|Value y Premium]]), y también, es posible cambiar el enrutamiento que usará la llamada marcando el prefijo 033 (Value) y 044 (Premium) seguido del número de 11 dígitos. Para EE. UU. solo está disponible en la ruta Premium.

*033 + 1 + Código de área + número: Canada Value (Anular la configuración de la cuenta)

*044 + 1 + Código de área + número: Canada Premium (Anular la configuración de la cuenta)
(El símbolo + solo se esta usando como referencia. No incluya como parte de su marcación)

Costo con la ruta '''Canada''' '''Value:''' Empenzando en $0.0052
Costo con la ruta '''Canada''' '''Premium:''' $0.0090
Costo con la ruta '''USA48''' '''(Premium)''': $0.0100

''Nota: excepto Yukón, Territorios del Noroeste y Nunavut los cuales son considerados en esta sección, pero tienen diferente tarifa ($0.1900 por minuto).''

'''USA48: Los Estados Unidos contiguos o Estados Unidos continentales (en inglés, contiguous United States o Mainland United States) son los 48 estados de EE. UU. localizados al sur de Canadá, además del Distrito de Columbia. La expresión excluye los estados de Alaska y Hawai, y todos los territorios insulares y protectorados de EE. UU., como Puerto Rico.'''

== Llamadas Internacionales ==

Existen diferentes códigos que se pueden utilizar para marcar números Internacionales (afuera de US48 y Canadá). Hawai y Alaska son considerados en esta sección, ya que cualquier cambio que realice para las Llamadas Internacionales en las [[Configuraciones de la Cuenta (Account Settings)|configuraciones de su cuenta]], afectará las llamadas a estos destinos. Esto significa que si utiliza la ruta premium en sus llamadas internacionales, al llamar a Hawai o Alaska la llamada será realizada usando la ruta Premium, incluso si usted marca como una llamada local (a 10 u 11 dígitos)

'''Nota''': Algunos destinos internacionales puede ser marcados usando solamente el prefijo 1, esto aplica para países que forman parte de la NANPA. 011 y 00 son utilizados para el restos de los países.
Por ej. cuando llame a Hawai, Alaska, países del Caribe y territorios de Estados Unidos.

Para llamar a países fuera de los Estados Unidos y Canadá, puede utilizar los siguiente códigos:

*011+Código del País+Número: Internacional
*00+Código del País+Número: Internacional
*033+Código del País+Número: Internacional Value (asegura el uso de la ruta Value, sin importar que no la haya elegido en las [[Configuraciones de la Cuenta (Account Settings)|configuraciones de su cuenta]])
*044+Código del País+Número: Internacional Premium (asegura el uso de la ruta Premium, sin importar que no la haya elegido en las [[Configuraciones de la Cuenta (Account Settings)|configuraciones de su cuenta]])

(El símbolo + es usado solo como referencia. No lo incluya como parte de su marcación)

Ejemplo de Marcado a México: 011+Código del País (52) +Número incluyendo código de la ciudad. Ejemplo: 011-52-999xxxxxxx (no utilice guiones al marcar)

== Códigos Especiales ==

Estos son algunos códigos especiales que pueden ser usados junto con VoIP.ms

=== Código de Acceso a su Buzón de Voz ===
*Marque <nowiki>*</nowiki>97 para acceder directamente al [[Buzón de voz (Voicemail)|buzón]] que se encuentra asociado con la cuenta desde la cual está marcando. (solamente se le solicitará la contraseña del buzón)

*Marque <nowiki>*</nowiki>98 para acceder a cualquier [[Buzón de voz (Voicemail)|buzón]] en su cuenta, sin importar desde qué cuenta marque.(Se le solicitará el número del buzón y la contraseña).

Si no tiene acceso a la red de VoIP.ms y desea consultar su [[Buzón de voz (Voicemail)|buzón]], usted puede simplemente marcar su número y una vez que el [[Buzón de voz (Voicemail)|buzón]] tome la llamada presiona la tecla asterisco (*).

=== Balance de Cuenta ===

Marque * 225 (* bal): este código le permite acceder al saldo de su cuenta VoIP.ms. Puede habilitarse o deshabilitarse en [[Subcuentas]].

=== Pruebas de Eco y Tonos de Teclado (Echo & DTMF test) ===

4443 (Prueba de Eco - Echo Test): Este código le permite realizar una prueba eco, con una cuenta nueva puede utilizar este código incluso si no tiene fondos, esta prueba es útil para validar la calidad del audio.

4747 (Tonos de Teclado - DTMF Test): Este código es utilizado para probar los tonos de su teclado, de igual forma puede ser utilizado incluso sin fondos, este código es útil para validar que la configuración de DTMF es la correcta.

== Códigos de servicio para Estados Unidos ==

'''Nota:''' Tenga en cuenta que las funciones de grabación y transcripción de llamadas no funcionarán para los servicios 988, 911, 411.

'''1-555-555-0911''' (Prueba para el [[Numero Identificador (Caller ID)|CallerID]] y servicio de e911): Puede usar este código para probar si funciona el [[Numero Identificador (Caller ID)|CallerID]] que usted está enviando y de igual forma le indica si el número está activado con el servicio de e911.

'''988:''' Línea Directa de Salud Mental
'''Nota:''' Tenga en cuenta que marcar 988 con un identificador de llamadas gratuito lo redirigirá al 411, ya que este no es un NPANXX norteamericano válido.

'''411:''' Asistencia de Directorio (debe ser habilitado en las configuraciones de su cuenta), cuando lo habilite puede marcar para obtener asistencia de directorio a un costo de $0.99 por llamada.

'''911:''' Servicio de emergencia - Debe estar habilitado en su DID con un costo de $1.50 por mes. Para más información consultar [[Servicio_E911 | este artículo]]

== Códigos de servicio para Canadá ==

'''Nota:''' Tenga en cuenta que las funciones de grabación y transcripción de llamadas no funcionarán para los servicios 988, 911, 411.

'''1-555-555-0911''' (Prueba para el [[Numero Identificador (Caller ID)|CallerID]] y servicio de e911): Puede usar este código para probar si funciona el [[Numero Identificador (Caller ID)|CallerID]] que usted está enviando y de igual forma le indica si el número está activado con el servicio de e911.

'''411:''' Asistencia de Directorio (debe ser habilitado en las configuraciones de su cuenta), cuando lo habilite puede marcar para obtener asistencia de directorio a un costo de $0.99 por llamada.

'''911:''' Servicio de emergencia - Debe estar habilitado en su DID con un costo de $1.50 por mes. Para más información consultar [[Servicio_E911 | este artículo]]

'''311:''' Servicio de Policía (no emergencias), Municipales y otros servicios gubernamentales (Solamente se puede acceder usando servidores Canadienses y pasando un [[Numero Identificador (Caller ID)|CallerID]] válido de una ciudad donde el servicio esté disponible.)

El servicio se ha puesto a disposición en las siguientes comunidades (con fecha de inicio), háganos saber si tiene algún problema:

Calgary, Alberta (18 Mayo 2005)
Edmonton, Alberta (16 Diciembre 2008)
Fort St. John, British Columbia (14 Noviembre 2006)
Gatineau, Quebec (22 Junio 2005)
Greater Sudbury, Ontario (12 Febrero 2007)
Halifax Regional Municipality, Nova Scotia (15 Noviembre 2012)[8]
''Halton Region, Ontario (18 Marzo 2008). Actualmente, las llamadas al 311 desde la región de Halton no son compatibles con VoIP.ms. Por favor use 1-866-4HALTON
(1-866-442-5866) o 905-825-6000''
Laval, Quebec (3 Octubre 2007)
Montreal, Quebec (mid-Diciembre 2007)
Ottawa, Ontario (19 Septiembre 2005)
Regional Municipality of Peel, Ontario (5 Octubre 2009)
St. John's, Newfoundland and Labrador (27 Junio 2006)
Toronto, Ontario (24 Septiembre 2009)
Vancouver, British Columbia
Windsor, Ontario (22 Agosto 2005)
Winnipeg, Manitoba (16 Enero 2009)[9]

'''511:''' Servicios de clima e información al viajero (Solamente se puede acceder usando servidores Canadienses y pasando un [[Numero Identificador (Caller ID)|CallerID]] válido de una ciudad donde el servicio esté disponible.)

'''811:''' Servicios no urgentes de salud (Health Teletriage/Telehealth). (Solamente se puede acceder usando servidores Canadienses y pasando un [[Numero Identificador (Caller ID)|CallerID]] válido de una ciudad donde el servicio esté disponible.)

'''988:''' Línea Directa de Salud Mental
'''Nota:''' Tenga en cuenta que marcar 988 con un identificador de llamadas gratuito lo redirigirá al 411, ya que este no es un NPANXX norteamericano válido.

'''Notas Importantes:'''

* Los servicios solo se pueden llamar desde servidores canadienses. Hay planes para agregar códigos de servicio a los servidores de EE. UU. Lo antes posible. Esto no se aplica al servicio 411.
* Los servicios dependen en el [[Numero Identificador (Caller ID)|CallerID]] para enrutar su llamada con la linea de servicio adecuada, por lo tanto algunos servicios pueden no estar soportados en su provincia o área especifica.
* No se aplican cargos en su cuenta cuando marca al 311, 511 o 811.

== Servicio 310 para Canadá ==

''Exchange'' 310 es un código especial, este tipo de números solamente sólo pueden alcanzarse desde el area al cual pertenecen. Desde VoIP.ms usted solamente necesita marcar usando 7 digitos para alcanzar estos números, de igual forma es importante pasar un [[Numero Identificador (Caller ID)|CallerID]] canadiense valido y estar registrado en el servidor de Chicago, Seattle o cualquier de los servidores canadienses.

'''Ejemplo''': Al marcar 310-xxxx (recuerde no utilizar guiones al marcar), si usted esta enviando un [[Numero Identificador (Caller ID)|CallerID]] con el código de area 514, es como si estuviera marcando 514-310-xxxx.

== Casos Especiales ==

La terminación de llamadas no esta soportada a los números 1900, números premium y números gratuitos internacionales (fuera de EE. UU. Y Canadá). Si tiene preguntas sobre esto, no dude en ponerse en contacto con nuestro personal de soporte a través de [mailto: support@voip.ms | soporte de tickets] o chat en vivo.

[[category:Guías]]

Using ring groups with a third-party spamfilter service

2023-12-07T00:59:17Z

Carlb: /* See also */

The voip.ms service has a built-in filter which can be used to block unwanted calls from telemarketers, debt buyers, fraudsters and other unwanted callers. [[CallerID Filtering]] allows the user to supply a list of numbers; any number which matches an entry on the list may be blocked, redirected or handled in various ways.

As telemarketers will try to circumvent the blocks by calling from some other number, various third-party services have been created to build blacklists of thousands of numbers of known unwanted callers. These are packaged and deployed in various forms; one form of third-party spamfilter requires that the user direct all inbound calls to one or more [[Ring Groups]].

In this configuration, inbound calls ring both at the subscriber's handset and at a second location - in this case, the unwanted call filtering service. If the call is legit, the third-party server ignores it; if the call is unwanted, the third-party server answers immediately so that the subscriber's phone gives only a brief single ring or doesn't ring at all.

One such server is the Jolly Roger Telephone Company (jollyrogertelco.com) which, despite the name, is not a replacement for voip.ms or any other telephone company; it is a third-party call filter service to block unwanted telemarketing calls by directing them to one of several automated robots instead of to a real human being.

The voip.ms service offers multiple features which make it well suited for use in conjunction with such third-party services:
* [[Ring Groups]] allow the same inbound call to ring at two or more locations at once
* [[SIP_URI#Creating a new SIP URI|Creating a SIP URI]] allows calls to be forwarded back out of the system to other Internet services at no cost
* If a third-party service doesn't support SIP URI to accept calls, [[Call Forwarding]] entries can be created to send calls back out to the public switched telephone network

==Configuration==
In this example, the third-party spam blocker is shown as jollyrogertelco.com; the setup will be similar for other providers which rely on the subscriber's access to [[Ring Groups]] to send all incoming calls to both the local handset and an outside service.

1) The first step is to obtain a number for inbound calls on the voip.ms service, if you don't already have one. See: [[Order a DID Number]]

2) The next step is to create an account on the third-party service, for example https://www.jollyrogertelco.com calls this subscription their "landlubber" service (offering "landline" or "VoIP" as options). When the system asks for your telephone number, provide the number which you're sending on call display on your voip.ms outbound calls.

3) The third-party server will send an e-mail once the service has been set up on their end. This looks something like:

: ''This is a note to confirm your service with the Jolly Roger Telephone Company!''
: ''You have one telephone number active with us!''
: ''Your first telephone number is 1NXXNXXXXXX.''
: ''See the 'Pick A Robot' link from www.jollyrogertelephone.com to find the numbers to our bots. If you don't have time to look them up now, just dial a random bot at 206-259-4999 for the US, 020-3813-1739 for the UK''

4) It should be possible to get the SIP code from the "settings" tab on https://jollyrogertelephone.com/amember/personal-options (it's at the bottom, next to a red-button link "How to use SIP codes").

5) Another alternative is to call the supplied number, +1 206-259-4999. A seemingly human-sounding voice will ask "Hello? Hello?". This is the robot. Hang up. The third-party server will send an automated e-mail indicating that the robot took the call and will provide the SIP configuration information:

:''A caller from NXX-NXX-XXXX dialled 206-259-4999 and was marooned for 10 seconds with a Jolly Roger bot named 'Kim the Kraken' The recording of the call is attached.''
:''Want SIP? You got it! Your SIP Code is 3jrt66r. For instructions on how to use SIP to integrate with Jolly Roger Telephone, click here: [https://jollyrogertelephone.com/faqs/#FAQ318 jollyrogertelephone.com/how-to-integrate-with-sip/].''
:(The actual SIP Code will vary for each subscriber. In this case, 3jrt66r is used as a placeholder as an example.)

6) Create the SIP URI from the supplied code by opening https://www.voip.ms/m/sipuri.php on the voip.ms control panel. See [[SIP URI#Creating a new SIP URI]]. For this third-party server, the SIP URI is the SIP Code supplied in the e-mail, plus a suffix, so '''3jrt66r''' could become '''3jrt66r'''-4949@jrt.bz for example:

[[Image:Forward.jpg]]
:'''Create new SIP URI'''
:'''SIP URI''' sip: 3jrt66r-4949@jrt.bz
:'''Description''': ...whatever...

:(Apparently the SIP URI with the -4999@jrt.bz suffix answers everything as the robot, the -4949@jrt.bz suffix answers just the unwanted calls as the bot.)

7) Once the SIP URI for the third-party server exists, it needs to be added to a [[Ring Group]]. On the voip.ms control panel https://www.voip.ms/m/ringgroup.php click the "Create a new ring group" button.

[[Image:RingGroups.png]]

:A dialogue box will pop up asking which extensions should ring when an incoming call arrives. Check the box for the SIP URI you just created (above) and check the box for whichever handset you intend to use to answer calls. This will cause incoming calls to ring in both places at once.

:The "ring time" for the third-party server should be set to something relatively short so that, if the call is not answered, the various fallback settings for the DID may be used.

8) The final step is to direct one or more of your direct inbound dial (DID) numbers into the ring group using [[Manage DID]] on the control panel, https://www.voip.ms/m/managedid.php

:Select an inbound number:

[[Image:EditDID.png|thumb|none|600px]]

:Change the routing of that inbound number to "ring group" and pick the ring group which you just created (above):

[[File:Routing.png|thumb|none|600px]]

9) Optionally, you may use [[CallerID Filtering]] to take the numbers which call you most often (or everyone in your [[Phone_book#Manage_your_groups|phone book]]) and send them directly to one of your extensions - without going through any of the filters. There is no requirement to do this, but it may cut down on the number of notification e-mails you receive.

:''Done! Goodbye, telemarketers.''

== See also ==
* [[Call Hunting]] differs from [[Ring Groups|using ring groups]] in that up to eight extensions are tried sequentially, instead of simultaneously/in parallel. By trying the spam-filter bots first, then failing over to a SIP handset a few seconds later (and, if that fails, a second line on the same handset or voicemail), a Call Hunting configuration would avoid the handset ringing once before the 'bot blocks the unwanted call.
* [[DigitalReceptionist IVR|Interactive Voice Response]] can be a useful tool to block spurious ADAD/robocall and misdial traffic from reaching live persons at your site. When the IVR answers, the caller will hear a customizable message like "Thank you for calling XYZ Inc, for Sales press 1, for Service press 2..." before the call is passed to the selected extension.
* [[Call Transcription]] can be useful if you are receiving many problem calls; it is activated for all inbound calls on a per-DID basis and will automatically transform the call audio to e-mailed text.
* [[Nomorobo]] and [[How to Prevent Robocalls with Nomorobo?]] introduce another, similar alternative, but for a provider that requires the calls be sent back out to a regular telephone number (instead of a SIP URI). See that provider's documentation on https://nomorobo.zendesk.com/hc/en-us/articles/205065739-Voip-ms for more detail.

==External links==
* [https://jollyrogertelephone.com/faqs/#FAQ318 Jolly Roger FAQ: How to integrate with SIP] is a general overview on how to configure SIP addresses and ring groups to block unwanted calls.
* [https://www.npr.org/2016/02/25/468149405/jolly-roger-telephone-company-uses-software-to-entrap-telemarketers NPR], [https://www.cbsnews.com/news/jolly-roger-telephone-company-robot-annoy-telemarketers/ CBS], [http://fortune.com/2016/02/24/robot-telemarketers-jolly-roger/ Fortune], [https://www.pcworld.com/article/3028541/privacy/tired-of-telemarketers-now-you-can-turn-the-tables-on-them-with-this-clever-bot.html PC World], [https://www.nytimes.com/2016/02/25/fashion/a-robot-that-has-fun-at-telemarketers-expense.html NY Times], [http://consumersunion.org/campaign-updates/end-robocalls-speaks-to-the-jolly-roger-telephone-company/ Consumers Union], [http://mashable.com/2016/02/03/robot-annoys-telemarketers/ Mashable], [http://www.smh.com.au/technology/technology-news/phone-robot-keeps-annoying-telemarketers-talking-for-as-long-as-possible-20160201-gmj83u.html SMH] and [http://www.ibtimes.co.in/why-t-apple-google-other-companies-could-enlist-help-jolly-roger-telephone-company-crack-690816 IBT] give the background on why subscribers would want to use VoIP features to put an end to unwanted marketing calls.

[[Category:Guides]]

Dialing Codes

2023-12-07T00:54:30Z

Carlb: /* Local calls (USA48/Canada) */

{| class="wikitable"
|+
|-
! Article en Français !! Artículo en Español
|-
| [https://wiki.voip.ms/article/Codes_de_Composition Français] ||
[https://wiki.voip.ms/article/C%C3%B3digos_de_Marcado Español]
|}

[http://voip.ms/ VoIP.ms] has different dialing codes depending on the destination you want to reach, and you can use these codes from your main or sub account depending on your needs. It is important to mention that there are two main distinctions between the routes when dialing local and international calls. Calls to United States 48 (except Hawaii & Alaska) and Canada are considered local calls. All the other destinations will be considered International and will use the corresponding route.

== Local calls (USA48/Canada) ==

For calls to USA48 & Canada, you only need to dial the complete 10 digits number, optionally you can dial using 11 digits, adding the prefix 1 before the number.

Example: 1514-316-xxxx or 514-316-xxxx (do not use the dashes when dialing)

On the case of Canada, from the Account Settings >> Account Routing, it can be set a route for these calls, there is an option to use value or premium route, and also, it's possible to switch the routing the call will use by dialing the prefix 033 (Value) and 044 (Premium) followed by the 11 digits number. The US is only available on the Premium route.

*033 + 1 + Area Code + number: Canada Value (override account setting)

*044 + 1 + Area Code + number: Canada Premium (override account setting)
(+ signs used as reference only. Do not include + signs when dialing)

Cost with the '''Canada''' '''Value''' route: Starting at $0.0052
Cost with the '''Canada''' '''Premium''' route: $0.0090
Cost with the '''USA48''' route ('''Premium'''): $0.0100

''Note: Yukon, Northwest Territories & Nunavut are considered in this section but have a different rate ($0.1900 per minute).''

'''USA48: The contiguous United States are the 48 U.S. states on the continent of North America that are south of Canada, plus the District of Columbia. The term excludes the states of Alaska and Hawaii, and all off-shore U.S. territories and possessions, such as Puerto Rico.
'''

== International calls ==

There are different codes that can be used to dial International numbers (Outside US48 & Canada). Hawaii and Alaska are considered in this section, since any change made on the [[Account Settings]] >> Account Routing for International calls, will affect also these destinations, this means if we set to use premium route, calls to these destinations will use also premium route, but they can be dialed as local US numbers (10 or 11 digits).

Note: Some International destinations can be dialed using only the prefix 1, this applies for countries part of the NANPA. 011 & 00 are for the rest of countries.
E.g. when dialing Hawaii, Alaska, Caribbean countries an U.S. Territories.

For dialing to countries outside US & Canada, we can follow these codes:

*011 + Country Code + number: International
*00 + Country Code + number: International
*033 + Country Code + number: International Value (override account setting)
*044 + Country Code + number: International Premium (override account setting)

(+ signs used as reference only. Do not include + signs when dialing)

Example Dialing to Mexico: 011+country code+number 011-52-9999xxxxxx (do not use the dashes when dialing)

== Special codes ==

Some special codes that you can use with the service:

=== [[Voicemail]] Access Codes ===
*97 to access directly the Mailbox associated to the account you are dialing from. (Will prompt for Password only)

*98 to access your Voicemail and choose one of your Mailbox accounts. (Will prompt for Mailbox ID and Password)

If you don't have access to our VoIP network and would like to check your Voicemail, you can simply dial your number. Once the Voicemail system answers your call, press the asterisk key (*).

=== Account Balance ===

Dial *225 (*bal): This code allow you to access your VoIP.ms Account Balance. It can be Enable or Disable on [[Sub Accounts]].

=== Echo & DTMF test ===

4443 (Echo Test): This code is used to access the echo test, with a new account this code can be dialed even without funds, it is useful to verify the quality on your line.

4747 (DTMF Test): This code is used to access the dtmf test, with a new account this code can be dialed even without funds, it is useful to verify if the dtmf is configured properly.

== Service codes for US ==

'''Note:''' Please note that call recording and call transcription features will not work for 988, 911, 411 services.

'''1-555-555-0911:''' Test CallerID and [[e911]] Test: to test if your caller Id is working properly you can dial this number, also this recording will let you know if your number is activated with the e911 service.

'''988:''' 'Mental Health Hotline
'''Note:''' Please note that dialing 988 with a toll free callerID will redirect you to 411, since this is not a valid North American NPANXX

'''411:''' Directory Assistance (Must be enabled in [[Account Settings]]), when enabled users can dial the directory assistance at a cost of $0.99 per call.

'''911:''' Emergency service - Must be enabled on your DID at a cost of $1.50 per month. For more information, consult [[E911 | this article]]

== Service codes for Canada ==

'''Note:''' Please note that call recording and call transcription features will not work for 988, 911, 411 services.

'''1-555-555-0911:''' Test CallerID and [[e911]] Test: to test if your caller Id is working properly you can dial this number, also this recording will let you know if your number is activated with the e911 service.

'''411:''' Directory Assistance (Must be enabled in [[Account Settings]]), when enabled users can dial the directory assistance at a cost of $0.99 per call.

'''911:''' Emergency service - Must be enabled on your DID at a cost of $1.50 per month. For more information, consult [[E911 | this article]]

'''311:''' Non-Emergency Police, Municipal and Other Governmental Services (Canadian Servers)

The service has been made available in the following communities (with starting date), please let us know if you experience an issue:

Calgary, Alberta (18 May 2005)
Edmonton, Alberta (16 December 2008)
Fort St. John, British Columbia (14 November 2006)
Gatineau, Quebec (22 June 2005)
Greater Sudbury, Ontario (12 February 2007)
Halifax Regional Municipality, Nova Scotia (15 November 2012)[8]
''Halton Region, Ontario (18 March 2008). The calls to 311 from the Halton Region are not currently supported with VoIP.ms. Please use 1-866-4HALTON
(1-866-442-5866) or 905-825-6000''
Laval, Quebec (3 October 2007)
Montreal, Quebec (mid-December 2007)
Ottawa, Ontario (19 September 2005)
Regional Municipality of Peel, Ontario (5 October 2009)
St. John's, Newfoundland and Labrador (27 June 2006)
Toronto, Ontario (24 September 2009)
Vancouver, British Columbia
Windsor, Ontario (22 August 2005)
Winnipeg, Manitoba (16 January 2009)[9]

'''511:''' Provision of Weather and Traveler Information Services (Canadian Servers)

'''811:''' Non-Urgent Health Teletriage / telehealth Services (Canadian Servers)

'''988:''' Mental Health Hotline
'''Note:''' Please note that dialing 988 with a toll free callerID will redirect you to 411, since this is not a valid North American NPANXX

'''Important Notes:'''

* The services can only be Called from Canadian servers. There are plans on adding service codes to US servers as soon as possible. This doesn't apply to the 411 service.
* The services rely on your [[Caller ID]] to route your call to the proper service line, therefore, some services may not be supported in your province or for your specific area.
* The are no charges applied to your account when you call 311, 511 or 811.

== Canada 310 service ==

Exchange 310 is a special code, these numbers are only reachable from the area where they belong, from the service, you will need to dial only the 7 digits to properly reach these numbers, pass a valid Canadian CallerID and be registered in Chicago, Seattle or any of the Canadian servers.

Example: Dial 310-xxxx (do not use the dashes when dialing), if you pass a caller id in area code 514, dialing seven digits 310-xxxx is like dialing 514-310-xxxx.

== Special cases ==

Termination is not supported to 1900 numbers, premium numbers and International Toll Free numbers (out of USA and Canada). If you have questions about please fell free to contact our support staff via [mailto:support@voip.ms | ticket support ] or live chat.

Dialing Codes

2023-12-07T00:47:13Z

Carlb: /* Service codes for Canada */ - the limited per-municipality deployment is for 311, not 9-1-1

{| class="wikitable"
|+
|-
! Article en Français !! Artículo en Español
|-
| [https://wiki.voip.ms/article/Codes_de_Composition Français] ||
[https://wiki.voip.ms/article/C%C3%B3digos_de_Marcado Español]
|}

[http://voip.ms/ VoIP.ms] has different dialing codes depending on the destination you want to reach, and you can use these codes from your main or sub account depending on your needs. It is important to mention that there are two main distinctions between the routes when dialing local and international calls. Calls to United States 48 (except Hawaii & Alaska) and Canada are considered local calls. All the other destinations will be considered International and will use the corresponding route.

== Local calls (USA48/Canada) ==

For calls to USA48 & Canada, you only need to dial the complete 10 digits number, optionally you can dial using 11 digits, adding the prefix 1 before the number.

Example: 1514-316-xxxx or 514-316-xxxx (do not use the dashes when dialing)

On the case of Canada, from the Account Settings >> Account Routing, it can be set a route for these calls, there is an option to use value or premium route, and also, it's possible to switch the routing the call will use by dialing the prefix 033 (Value) and 044 (Premium) followed by the 11 digits number. The US is only available on the Premium route.

*033 + 1 + Area Code + number: Canada Value (override account setting)

*044 + 1 + Area Code + number: Canada Premium (override account setting)
(+ signs used as reference only. Do not include + signs when dialing)

Cost with the '''Canada''' '''Value''' route: Starting at $0.0052
Cost with the '''Canada''' '''Premium''' route: $0.0090
Cost with the '''USA48''' route ('''Premium'''): $0.0100

''Note (except Yukon, North West Territories & Nunavut are considered in this section but have a different rate ($0.1900 per minute).''

'''USA48: The contiguous United States are the 48 U.S. states on the continent of North America that are south of Canada, plus the District of Columbia. The term excludes the states of Alaska and Hawaii, and all off-shore U.S. territories and possessions, such as Puerto Rico.
'''

== International calls ==

There are different codes that can be used to dial International numbers (Outside US48 & Canada). Hawaii and Alaska are considered in this section, since any change made on the [[Account Settings]] >> Account Routing for International calls, will affect also these destinations, this means if we set to use premium route, calls to these destinations will use also premium route, but they can be dialed as local US numbers (10 or 11 digits).

Note: Some International destinations can be dialed using only the prefix 1, this applies for countries part of the NANPA. 011 & 00 are for the rest of countries.
E.g. when dialing Hawaii, Alaska, Caribbean countries an U.S. Territories.

For dialing to countries outside US & Canada, we can follow these codes:

*011 + Country Code + number: International
*00 + Country Code + number: International
*033 + Country Code + number: International Value (override account setting)
*044 + Country Code + number: International Premium (override account setting)

(+ signs used as reference only. Do not include + signs when dialing)

Example Dialing to Mexico: 011+country code+number 011-52-9999xxxxxx (do not use the dashes when dialing)

== Special codes ==

Some special codes that you can use with the service:

=== [[Voicemail]] Access Codes ===
*97 to access directly the Mailbox associated to the account you are dialing from. (Will prompt for Password only)

*98 to access your Voicemail and choose one of your Mailbox accounts. (Will prompt for Mailbox ID and Password)

If you don't have access to our VoIP network and would like to check your Voicemail, you can simply dial your number. Once the Voicemail system answers your call, press the asterisk key (*).

=== Account Balance ===

Dial *225 (*bal): This code allow you to access your VoIP.ms Account Balance. It can be Enable or Disable on [[Sub Accounts]].

=== Echo & DTMF test ===

4443 (Echo Test): This code is used to access the echo test, with a new account this code can be dialed even without funds, it is useful to verify the quality on your line.

4747 (DTMF Test): This code is used to access the dtmf test, with a new account this code can be dialed even without funds, it is useful to verify if the dtmf is configured properly.

== Service codes for US ==

'''Note:''' Please note that call recording and call transcription features will not work for 988, 911, 411 services.

'''1-555-555-0911:''' Test CallerID and [[e911]] Test: to test if your caller Id is working properly you can dial this number, also this recording will let you know if your number is activated with the e911 service.

'''988:''' 'Mental Health Hotline
'''Note:''' Please note that dialing 988 with a toll free callerID will redirect you to 411, since this is not a valid North American NPANXX

'''411:''' Directory Assistance (Must be enabled in [[Account Settings]]), when enabled users can dial the directory assistance at a cost of $0.99 per call.

'''911:''' Emergency service - Must be enabled on your DID at a cost of $1.50 per month. For more information, consult [[E911 | this article]]

== Service codes for Canada ==

'''Note:''' Please note that call recording and call transcription features will not work for 988, 911, 411 services.

'''1-555-555-0911:''' Test CallerID and [[e911]] Test: to test if your caller Id is working properly you can dial this number, also this recording will let you know if your number is activated with the e911 service.

'''411:''' Directory Assistance (Must be enabled in [[Account Settings]]), when enabled users can dial the directory assistance at a cost of $0.99 per call.

'''911:''' Emergency service - Must be enabled on your DID at a cost of $1.50 per month. For more information, consult [[E911 | this article]]

'''311:''' Non-Emergency Police, Municipal and Other Governmental Services (Canadian Servers)

The service has been made available in the following communities (with starting date), please let us know if you experience an issue:

Calgary, Alberta (18 May 2005)
Edmonton, Alberta (16 December 2008)
Fort St. John, British Columbia (14 November 2006)
Gatineau, Quebec (22 June 2005)
Greater Sudbury, Ontario (12 February 2007)
Halifax Regional Municipality, Nova Scotia (15 November 2012)[8]
''Halton Region, Ontario (18 March 2008). The calls to 311 from the Halton Region are not currently supported with VoIP.ms. Please use 1-866-4HALTON
(1-866-442-5866) or 905-825-6000''
Laval, Quebec (3 October 2007)
Montreal, Quebec (mid-December 2007)
Ottawa, Ontario (19 September 2005)
Regional Municipality of Peel, Ontario (5 October 2009)
St. John's, Newfoundland and Labrador (27 June 2006)
Toronto, Ontario (24 September 2009)
Vancouver, British Columbia
Windsor, Ontario (22 August 2005)
Winnipeg, Manitoba (16 January 2009)[9]

'''511:''' Provision of Weather and Traveler Information Services (Canadian Servers)

'''811:''' Non-Urgent Health Teletriage / telehealth Services (Canadian Servers)

'''988:''' Mental Health Hotline
'''Note:''' Please note that dialing 988 with a toll free callerID will redirect you to 411, since this is not a valid North American NPANXX

'''Important Notes:'''

* The services can only be Called from Canadian servers. There are plans on adding service codes to US servers as soon as possible. This doesn't apply to the 411 service.
* The services rely on your [[Caller ID]] to route your call to the proper service line, therefore, some services may not be supported in your province or for your specific area.
* The are no charges applied to your account when you call 311, 511 or 811.

== Canada 310 service ==

Exchange 310 is a special code, these numbers are only reachable from the area where they belong, from the service, you will need to dial only the 7 digits to properly reach these numbers, pass a valid Canadian CallerID and be registered in Chicago, Seattle or any of the Canadian servers.

Example: Dial 310-xxxx (do not use the dashes when dialing), if you pass a caller id in area code 514, dialing seven digits 310-xxxx is like dialing 514-310-xxxx.

== Special cases ==

Termination is not supported to 1900 numbers, premium numbers and International Toll Free numbers (out of USA and Canada). If you have questions about please fell free to contact our support staff via [mailto:support@voip.ms | ticket support ] or live chat.

Using ring groups with a third-party spamfilter service

2023-12-07T00:40:42Z

Carlb: /* See also */

The voip.ms service has a built-in filter which can be used to block unwanted calls from telemarketers, debt buyers, fraudsters and other unwanted callers. [[CallerID Filtering]] allows the user to supply a list of numbers; any number which matches an entry on the list may be blocked, redirected or handled in various ways.

As telemarketers will try to circumvent the blocks by calling from some other number, various third-party services have been created to build blacklists of thousands of numbers of known unwanted callers. These are packaged and deployed in various forms; one form of third-party spamfilter requires that the user direct all inbound calls to one or more [[Ring Groups]].

In this configuration, inbound calls ring both at the subscriber's handset and at a second location - in this case, the unwanted call filtering service. If the call is legit, the third-party server ignores it; if the call is unwanted, the third-party server answers immediately so that the subscriber's phone gives only a brief single ring or doesn't ring at all.

One such server is the Jolly Roger Telephone Company (jollyrogertelco.com) which, despite the name, is not a replacement for voip.ms or any other telephone company; it is a third-party call filter service to block unwanted telemarketing calls by directing them to one of several automated robots instead of to a real human being.

The voip.ms service offers multiple features which make it well suited for use in conjunction with such third-party services:
* [[Ring Groups]] allow the same inbound call to ring at two or more locations at once
* [[SIP_URI#Creating a new SIP URI|Creating a SIP URI]] allows calls to be forwarded back out of the system to other Internet services at no cost
* If a third-party service doesn't support SIP URI to accept calls, [[Call Forwarding]] entries can be created to send calls back out to the public switched telephone network

==Configuration==
In this example, the third-party spam blocker is shown as jollyrogertelco.com; the setup will be similar for other providers which rely on the subscriber's access to [[Ring Groups]] to send all incoming calls to both the local handset and an outside service.

1) The first step is to obtain a number for inbound calls on the voip.ms service, if you don't already have one. See: [[Order a DID Number]]

2) The next step is to create an account on the third-party service, for example https://www.jollyrogertelco.com calls this subscription their "landlubber" service (offering "landline" or "VoIP" as options). When the system asks for your telephone number, provide the number which you're sending on call display on your voip.ms outbound calls.

3) The third-party server will send an e-mail once the service has been set up on their end. This looks something like:

: ''This is a note to confirm your service with the Jolly Roger Telephone Company!''
: ''You have one telephone number active with us!''
: ''Your first telephone number is 1NXXNXXXXXX.''
: ''See the 'Pick A Robot' link from www.jollyrogertelephone.com to find the numbers to our bots. If you don't have time to look them up now, just dial a random bot at 206-259-4999 for the US, 020-3813-1739 for the UK''

4) It should be possible to get the SIP code from the "settings" tab on https://jollyrogertelephone.com/amember/personal-options (it's at the bottom, next to a red-button link "How to use SIP codes").

5) Another alternative is to call the supplied number, +1 206-259-4999. A seemingly human-sounding voice will ask "Hello? Hello?". This is the robot. Hang up. The third-party server will send an automated e-mail indicating that the robot took the call and will provide the SIP configuration information:

:''A caller from NXX-NXX-XXXX dialled 206-259-4999 and was marooned for 10 seconds with a Jolly Roger bot named 'Kim the Kraken' The recording of the call is attached.''
:''Want SIP? You got it! Your SIP Code is 3jrt66r. For instructions on how to use SIP to integrate with Jolly Roger Telephone, click here: [https://jollyrogertelephone.com/faqs/#FAQ318 jollyrogertelephone.com/how-to-integrate-with-sip/].''
:(The actual SIP Code will vary for each subscriber. In this case, 3jrt66r is used as a placeholder as an example.)

6) Create the SIP URI from the supplied code by opening https://www.voip.ms/m/sipuri.php on the voip.ms control panel. See [[SIP URI#Creating a new SIP URI]]. For this third-party server, the SIP URI is the SIP Code supplied in the e-mail, plus a suffix, so '''3jrt66r''' could become '''3jrt66r'''-4949@jrt.bz for example:

[[Image:Forward.jpg]]
:'''Create new SIP URI'''
:'''SIP URI''' sip: 3jrt66r-4949@jrt.bz
:'''Description''': ...whatever...

:(Apparently the SIP URI with the -4999@jrt.bz suffix answers everything as the robot, the -4949@jrt.bz suffix answers just the unwanted calls as the bot.)

7) Once the SIP URI for the third-party server exists, it needs to be added to a [[Ring Group]]. On the voip.ms control panel https://www.voip.ms/m/ringgroup.php click the "Create a new ring group" button.

[[Image:RingGroups.png]]

:A dialogue box will pop up asking which extensions should ring when an incoming call arrives. Check the box for the SIP URI you just created (above) and check the box for whichever handset you intend to use to answer calls. This will cause incoming calls to ring in both places at once.

:The "ring time" for the third-party server should be set to something relatively short so that, if the call is not answered, the various fallback settings for the DID may be used.

8) The final step is to direct one or more of your direct inbound dial (DID) numbers into the ring group using [[Manage DID]] on the control panel, https://www.voip.ms/m/managedid.php

:Select an inbound number:

[[Image:EditDID.png|thumb|none|600px]]

:Change the routing of that inbound number to "ring group" and pick the ring group which you just created (above):

[[File:Routing.png|thumb|none|600px]]

9) Optionally, you may use [[CallerID Filtering]] to take the numbers which call you most often (or everyone in your [[Phone_book#Manage_your_groups|phone book]]) and send them directly to one of your extensions - without going through any of the filters. There is no requirement to do this, but it may cut down on the number of notification e-mails you receive.

:''Done! Goodbye, telemarketers.''

== See also ==
* [[Call Hunting]] differs from [[Ring Groups|using ring groups]] in that up to eight extensions are tried sequentially, instead of simultaneously/in parallel. By trying the spam-filter bots first, then failing over to a SIP handset a few seconds later (and, if that fails, a second line on the same handset or voicemail), a Call Hunting configuration would avoid the handset ringing once before the 'bot blocks the unwanted call.
* [[DigitalReceptionist IVR|Interactive Voice Response]] can be a useful tool to block spurious ADAD/robocall and misdial traffic from reaching live persons at your site. When the IVR answers, the caller will hear "Thank you for calling XYZ Inc, for Sales press 1, for Service press 2..." before the call is passed to the selected extension.
* [[Call Transcription]] can be useful if you are receiving many problem calls; it is activated for all inbound calls on a per-DID basis and will automatically transform the call audio to e-mailed text.
* [[Nomorobo]] and [[How to Prevent Robocalls with Nomorobo?]] introduce another, similar alternative, but for a provider that requires the calls be sent back out to a regular telephone number (instead of a SIP URI). See that provider's documentation on https://nomorobo.zendesk.com/hc/en-us/articles/205065739-Voip-ms for more detail.

==External links==
* [https://jollyrogertelephone.com/faqs/#FAQ318 Jolly Roger FAQ: How to integrate with SIP] is a general overview on how to configure SIP addresses and ring groups to block unwanted calls.
* [https://www.npr.org/2016/02/25/468149405/jolly-roger-telephone-company-uses-software-to-entrap-telemarketers NPR], [https://www.cbsnews.com/news/jolly-roger-telephone-company-robot-annoy-telemarketers/ CBS], [http://fortune.com/2016/02/24/robot-telemarketers-jolly-roger/ Fortune], [https://www.pcworld.com/article/3028541/privacy/tired-of-telemarketers-now-you-can-turn-the-tables-on-them-with-this-clever-bot.html PC World], [https://www.nytimes.com/2016/02/25/fashion/a-robot-that-has-fun-at-telemarketers-expense.html NY Times], [http://consumersunion.org/campaign-updates/end-robocalls-speaks-to-the-jolly-roger-telephone-company/ Consumers Union], [http://mashable.com/2016/02/03/robot-annoys-telemarketers/ Mashable], [http://www.smh.com.au/technology/technology-news/phone-robot-keeps-annoying-telemarketers-talking-for-as-long-as-possible-20160201-gmj83u.html SMH] and [http://www.ibtimes.co.in/why-t-apple-google-other-companies-could-enlist-help-jolly-roger-telephone-company-crack-690816 IBT] give the background on why subscribers would want to use VoIP features to put an end to unwanted marketing calls.

[[Category:Guides]]

Other Services

2023-12-07T00:26:24Z

Carlb: /* Jolly Roger */

One of the advantages of VoIP.ms is that is not limited to SIP devices, systems, or softphones. Our service can be integrated with other services that you may find useful for your daily tasks.

Here you can check some of the services that you can use with your VoIP.ms account.

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

'''Looking for a Analog Telephone Adapter (ATA)? [[ATA_Devices | Click here]] to see all models.'''

__TOC__

== Web dialers ==

=== Dialfire ===

[[File:Dialfire1.png|300px|thumb|left|Dialfire]]

'''Product:''' Cloud dialer that instantly turns your browser into a complete outbound call center.

'''Company:''' cloud IT Services GmbH

'''Overview:''' While it's very easy to get started with Dialfire, some campaigns are more complex than others and this is why Dialfire has a wide range of advanced features that put you in control.

'''Main features:'''
: Outbound Calling | Inbound Call Handling | Customizable | Multi-Step Campaigns
: Real-Time Analytics | Integration | Call Monitoring & Recording | Reliability & Security | Multi-Tenancy

[[Dialfire|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

== Call Routing & Managing ==

=== Callroute ===

[[File:Callroute logo1.png|400px|thumb|left|Callroute]]

'''Product:''' Callroute is an advanced call routing, number management and user provisioning solution.

'''Company:''' Sipsinergy

'''Overview:''' It allows you to connect any phone system together and to any service provider without the need for any infrastructure or deep technical knowledge.

'''Main features:'''
: Connect your existing VoIP provider | MS Teams Provisioning | All in one PSTN solution | SIP phone support
: Call Recording | Multiple inbound numbers | Complete number management | Reporting

[[Callroute|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

== SMS/MMS Services ==

=== Textable ===

[[File:Textable_logo2.png|400px|thumb|left|Textable]]

'''Product:''' Messaging platform for VoIP phone numbers used by individuals, businesses and Managed Service Providers.

'''Company:''' Taurix LLC

'''Overview:''' Keep in touch with your customers – on the go.

Real-time messaging application in your browser window with push notifications and always in sync with your mobile app.

[[Textable|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

== Spam calls blocking services ==

=== Nomorobo ===

[[File:Nomorobo.png|300px|thumb|left|Nomorobo]]

'''Product:''' Nomorobo is a cloud-based Internet service that helps to block and hangs up illegal calls from telemarketers. .

'''Company:''' Telephone Science Corp

'''Overview:''' VoIP.ms works with Nomorobo to block unsolicited robocalls.

[[Nomorobo|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

=== Jolly Roger ===

[[File:Jollyroger-logo.png|300px|thumb|left|jollyrogertelephone.com]]

'''Product:''' Jolly Roger is a cloud-based Internet service that helps to block spam calls from telemarketers and robodialers.

'''Company:''' jollyrogertelephone.com

'''Overview:''' VoIP.ms works well with Jolly Roger, a spam filter which can be enabled by using [[Ring Groups]] to forward calls to a [[SIP address]]. As an effective block to unsolicited robocalls, this is a spam filter with attitude. It uses TRUECNAM to identify spam, then answers as if it were human - just to waste the caller's time until they take the hint and stop calling you.

[[Using ring groups with a third-party spamfilter service|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

Other Services

2023-12-07T00:25:17Z

Carlb: /* Spam calls blocking services */

One of the advantages of VoIP.ms is that is not limited to SIP devices, systems, or softphones. Our service can be integrated with other services that you may find useful for your daily tasks.

Here you can check some of the services that you can use with your VoIP.ms account.

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

'''Looking for a Analog Telephone Adapter (ATA)? [[ATA_Devices | Click here]] to see all models.'''

__TOC__

== Web dialers ==

=== Dialfire ===

[[File:Dialfire1.png|300px|thumb|left|Dialfire]]

'''Product:''' Cloud dialer that instantly turns your browser into a complete outbound call center.

'''Company:''' cloud IT Services GmbH

'''Overview:''' While it's very easy to get started with Dialfire, some campaigns are more complex than others and this is why Dialfire has a wide range of advanced features that put you in control.

'''Main features:'''
: Outbound Calling | Inbound Call Handling | Customizable | Multi-Step Campaigns
: Real-Time Analytics | Integration | Call Monitoring & Recording | Reliability & Security | Multi-Tenancy

[[Dialfire|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

== Call Routing & Managing ==

=== Callroute ===

[[File:Callroute logo1.png|400px|thumb|left|Callroute]]

'''Product:''' Callroute is an advanced call routing, number management and user provisioning solution.

'''Company:''' Sipsinergy

'''Overview:''' It allows you to connect any phone system together and to any service provider without the need for any infrastructure or deep technical knowledge.

'''Main features:'''
: Connect your existing VoIP provider | MS Teams Provisioning | All in one PSTN solution | SIP phone support
: Call Recording | Multiple inbound numbers | Complete number management | Reporting

[[Callroute|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

== SMS/MMS Services ==

=== Textable ===

[[File:Textable_logo2.png|400px|thumb|left|Textable]]

'''Product:''' Messaging platform for VoIP phone numbers used by individuals, businesses and Managed Service Providers.

'''Company:''' Taurix LLC

'''Overview:''' Keep in touch with your customers – on the go.

Real-time messaging application in your browser window with push notifications and always in sync with your mobile app.

[[Textable|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

== Spam calls blocking services ==

=== Nomorobo ===

[[File:Nomorobo.png|300px|thumb|left|Nomorobo]]

'''Product:''' Nomorobo is a cloud-based Internet service that helps to block and hangs up illegal calls from telemarketers. .

'''Company:''' Telephone Science Corp

'''Overview:''' VoIP.ms works with Nomorobo to block unsolicited robocalls.

[[Nomorobo|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

=== Jolly Roger ===

[[File:Jollyroger-logo.png|300px|thumb|left|jollyrogertelephone.com]]

'''Product:''' Jolly Roger is a cloud-based Internet service that helps to block spam calls from telemarketers and robodialers.

'''Company:''' jollyrogertelephone.com

'''Overview:''' VoIP.ms works well with Jolly Roger, a spam filter which can be enabled using [[Ring Groups]] to forward calls to a [[SIP address]]. As an effective block to unsolicited robocalls, this is a spam filter with attitude. It uses TRUECNAM to identify spam, then answers as if it were human - just to waste the caller's time until they take the hint and stop calling you.

[[Using ring groups with a third-party spamfilter service|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

File:Jollyroger-logo.png

2023-12-07T00:13:09Z

Carlb: uploaded a new version of "File:Jollyroger-logo.png": fix white-on-transparent colour scheme, which is making this invisible

Proprietary logo for jollyrogertelephone.com - which is described in [[Using ring groups with a third-party spamfilter service]] as an example of a spam filter which may be reached directly using a [[SIP address]]. An alternative to sending calls back out to the PSTN, [[Nomorobo]]-style, for spam blocking.

File:Jollyroger-logo.png

2023-12-07T00:11:46Z

Carlb: Proprietary logo for jollyrogertelephone.com - which is described in Using ring groups with a third-party spamfilter service as an example of a spam filter which may be reached directly using a SIP address. An alternative to sending calls back out

Poly VVX-D230

2023-09-28T05:51:11Z

Carlb: /* See alos */

[[Image:Poly VVX-D230 base.png|thumb|The VVX-D230's base can control ten DECT handsets]]
Aimed at small business users, the Poly (Polycom, Obahai) '''VVX-D230''' is a DECT [[IP Phones|SIP phone]] with support for up to ten cordless handsets — including the one handset which is packaged with the base.

This system (OBAHAI/VVXD230) is intended to complement the [[Polycom VVX 300, 400, etc|VVX-series]] desk phones and the [[Poly Trio 8800]] conference room speakerphone.

==Configuration==
# If the cordless handsets are not already paired to the base, press and hold the 'find' button on the base for at least five seconds. Then, from the handset, go to Menu→Settings→Registration→Register.
# Configuration of the system is done via a web interface. From any paired handset, go to Menu→Settings→Basestation Info to obtain the IPv4 address of the base.
# Open a web browser to this address and log in with username 'admin' and default password 'admin'.

<br clear="all">
[[Image:Poly VVX-D230 Wizard.png]]

'''From the "Setup Wizard" page''', change the following parameters (changes will not be applied until you save them and reboot the device). Be sure to uncheck the "default" check box for each item you intend to modify:
*'''LocalTimeZone''' - set to your local time zone, for instance GMT-5:00 for (Eastern Time). Factory default is GMT-8:00 (Pacific Time, California).
*'''ITSP A SIPProxyServer''' - atlanta.voip.ms (or one of the other multiple [[servers]] - this must match the same server as your [[DID Troubleshooting|DID]] configuration).
*'''ITSP A DigitMap''' - (optional) (1xxxxxxxxxx|<1555>[2-9]xxxxxxS3|<1>[2-9]xxxxxxxxx|011xx.|xx.|(Mipd)|[^*#]@@.) (as an example, replacing 1-555 with your local area code) will enable both 7 and 10-digit dial for North American numbers. See the manufacturer's technical reference manual for details of the syntax.
*'''Phone1 PrimaryLine''' - "SP1 service"
*'''SP1 ITSP Profile''' - A
*'''SP1 AuthUserName''' - 123456_1 (your voip.ms user number and a [[Sub Accounts|subaccount]] number, in this example user #123456 subaccount #1
*'''SP1 AuthPassword''' - the password associated with your voip.ms subsccount
If you have additional virtual lines, repeat the same configuration steps for each subaccount using SP2, SP3... up to a maximum of eight lines and ten handsets.

<br clear="all">
[[Image:Poly VVX-D230 SP1.png]]

'''From the "SP1 Service" page''', change the following settings for the first virtual "line" (the others will be similar):
*'''X_DisplayNumber''' - A short text string identifying this line on the local handset screen. Typically the seven-digit NXX-XXXX local number, although any value will do.
*'''X_InboundCallRoute''' - DT1,DT2,DT3 (if this number is to ring to the first three extensions, for example). The default (DT1) rings only to the first extension on inbound calls.
*'''X_AcceptSipFromRegistrarOnly''' - YES (to keep [[Sip Scanner Ghost Calls|spurious]] calls out)
*'''X_KeepAliveEnable''' - YES (if you are behind NAT)
*'''X_KeepAliveExpires''' - 180 (three minutes) is reasonable
*'''DirectoryNumber''' - NPA-NXX-XXXX, the directory number associated with this line
*'''AuthUserName''', '''AuthPassword''' - should already be set from the "Setup Wizard" above; if not, set them here.
*'''CallerIDName''' - your name (15 alphanumeric chars max, no punctuation but spaces are allowed) as you want it to appear on [[Caller ID]] for outbound calls.
*'''X_CheckVoiceMailNumber''' - *97
*'''MessageWaiting''' - YES

Save everything before leaving each page of the setup. When you are finished making changes, reboot the device.

Try a test call, such as the [[Dialing Codes|echo test]] (press [4] [4] [4] [3] and the [green] button). You should hear your own voice played back just as it was received.

==See also==
* [[Polycom VVX 300, 400, etc|VVX-series]]
* [[Poly Trio 8800]]

==Documentation==
This page is merely a brief overview with just enough information to connect the handsets to VoIP.ms; see the manufacturer's website (poly.com) for more extensive documentation:

* Quick start: [https://www.poly.com/content/dam/www/products/support/voice/vvx-dect-ip-phones/setup/vvx-d230-hs-bs-class-a-qsg-access.pdf handset + base], [https://www.poly.com/content/dam/www/products/support/voice/vvx-dect-ip-phones/setup/vvx-d230-hs-chg-class-a-qsg-access.pdf expansion handset + charger]
* [https://www.poly.com/content/dam/www/products/support/voice/vvx-dect-ip-phones/user/vvxd230-adminguide-710.pdf Admin guide] and [https://www.poly.com/content/dam/www/products/support/voice/vvx-dect-ip-phones/user/vvxd230-tech-ref-710.pdf technical reference] manual from Poly.com

[[Category:SIP phones]]

Poly VVX-D230

2023-09-28T05:50:21Z

Carlb: /* Configuration */

[[Image:Poly VVX-D230 base.png|thumb|The VVX-D230's base can control ten DECT handsets]]
Aimed at small business users, the Poly (Polycom, Obahai) '''VVX-D230''' is a DECT [[IP Phones|SIP phone]] with support for up to ten cordless handsets — including the one handset which is packaged with the base.

This system (OBAHAI/VVXD230) is intended to complement the [[Polycom VVX 300, 400, etc|VVX-series]] desk phones and the [[Poly Trio 8800]] conference room speakerphone.

==Configuration==
# If the cordless handsets are not already paired to the base, press and hold the 'find' button on the base for at least five seconds. Then, from the handset, go to Menu→Settings→Registration→Register.
# Configuration of the system is done via a web interface. From any paired handset, go to Menu→Settings→Basestation Info to obtain the IPv4 address of the base.
# Open a web browser to this address and log in with username 'admin' and default password 'admin'.

<br clear="all">
[[Image:Poly VVX-D230 Wizard.png]]

'''From the "Setup Wizard" page''', change the following parameters (changes will not be applied until you save them and reboot the device). Be sure to uncheck the "default" check box for each item you intend to modify:
*'''LocalTimeZone''' - set to your local time zone, for instance GMT-5:00 for (Eastern Time). Factory default is GMT-8:00 (Pacific Time, California).
*'''ITSP A SIPProxyServer''' - atlanta.voip.ms (or one of the other multiple [[servers]] - this must match the same server as your [[DID Troubleshooting|DID]] configuration).
*'''ITSP A DigitMap''' - (optional) (1xxxxxxxxxx|<1555>[2-9]xxxxxxS3|<1>[2-9]xxxxxxxxx|011xx.|xx.|(Mipd)|[^*#]@@.) (as an example, replacing 1-555 with your local area code) will enable both 7 and 10-digit dial for North American numbers. See the manufacturer's technical reference manual for details of the syntax.
*'''Phone1 PrimaryLine''' - "SP1 service"
*'''SP1 ITSP Profile''' - A
*'''SP1 AuthUserName''' - 123456_1 (your voip.ms user number and a [[Sub Accounts|subaccount]] number, in this example user #123456 subaccount #1
*'''SP1 AuthPassword''' - the password associated with your voip.ms subsccount
If you have additional virtual lines, repeat the same configuration steps for each subaccount using SP2, SP3... up to a maximum of eight lines and ten handsets.

<br clear="all">
[[Image:Poly VVX-D230 SP1.png]]

'''From the "SP1 Service" page''', change the following settings for the first virtual "line" (the others will be similar):
*'''X_DisplayNumber''' - A short text string identifying this line on the local handset screen. Typically the seven-digit NXX-XXXX local number, although any value will do.
*'''X_InboundCallRoute''' - DT1,DT2,DT3 (if this number is to ring to the first three extensions, for example). The default (DT1) rings only to the first extension on inbound calls.
*'''X_AcceptSipFromRegistrarOnly''' - YES (to keep [[Sip Scanner Ghost Calls|spurious]] calls out)
*'''X_KeepAliveEnable''' - YES (if you are behind NAT)
*'''X_KeepAliveExpires''' - 180 (three minutes) is reasonable
*'''DirectoryNumber''' - NPA-NXX-XXXX, the directory number associated with this line
*'''AuthUserName''', '''AuthPassword''' - should already be set from the "Setup Wizard" above; if not, set them here.
*'''CallerIDName''' - your name (15 alphanumeric chars max, no punctuation but spaces are allowed) as you want it to appear on [[Caller ID]] for outbound calls.
*'''X_CheckVoiceMailNumber''' - *97
*'''MessageWaiting''' - YES

Save everything before leaving each page of the setup. When you are finished making changes, reboot the device.

Try a test call, such as the [[Dialing Codes|echo test]] (press [4] [4] [4] [3] and the [green] button). You should hear your own voice played back just as it was received.

==See alos==
* [[Polycom VVX 300, 400, etc|VVX-series]]
* [[Poly Trio 8800]]

==Documentation==
This page is merely a brief overview with just enough information to connect the handsets to VoIP.ms; see the manufacturer's website (poly.com) for more extensive documentation:

* Quick start: [https://www.poly.com/content/dam/www/products/support/voice/vvx-dect-ip-phones/setup/vvx-d230-hs-bs-class-a-qsg-access.pdf handset + base], [https://www.poly.com/content/dam/www/products/support/voice/vvx-dect-ip-phones/setup/vvx-d230-hs-chg-class-a-qsg-access.pdf expansion handset + charger]
* [https://www.poly.com/content/dam/www/products/support/voice/vvx-dect-ip-phones/user/vvxd230-adminguide-710.pdf Admin guide] and [https://www.poly.com/content/dam/www/products/support/voice/vvx-dect-ip-phones/user/vvxd230-tech-ref-710.pdf technical reference] manual from Poly.com

[[Category:SIP phones]]

IP Phones

2023-09-26T17:15:17Z

Carlb: Replace .jpg with a .png duplicate. Original is failing to resize due to server error. "Error creating thumbnail: Incomplete GD library configuration: missing function imagecreatefromjpeg"

== Blog Articles ==

* '''VoIP – Bring Your Own Device (BYOD)''': To be able to place or receive calls using VoIP, you may need a hardware setup that will allow you to use your regular phone (ATA) or a special phone that connects directly to VoIP networks (IP Phone). For more information, Take a peek at our blog article [https://wiki.voip.ms/article/VoIP_%E2%80%93_Bring_Your_Own_Device_(BYOD) VoIP – Bring Your Own Device (BYOD)]

* '''IP Phone:''' An IP Phone uses voice over IP (VoIP) technologies allowing telephone calls to be made over an IP network such as the Internet instead of the ordinary PSTN system. Still not clear? Take a peek at our blog article : [https://wiki.voip.ms/article/Back_to_Basics:_What_is_an_IP_Phone%3F Back to Basics - What is an IP Phone?]

'''Looking for a Analog Telephone Adapter (ATA)? [[ATA_Devices | Click here]] to see all models.'''

__TOC__

==IP Phones ==

===3COM 3108 Wireless Phone===

[[File:3com_3108_1.jpg|300px|thumb|left|3COM 3108 Wireless Phone]]

'''Product:''' 3COM 3108 Wireless Phone

'''Company:''' 3COM

'''Overview:''' The 3Com 3108 Wireless Telephone is a Session Initiation Protocol (SIP)-based wireless Voice over Internet Protocol (VoIP) telephone. SIP is an internationally recognized standard <nowiki>(IETF RFC 3261)</nowiki> for implementing VoIP. You can make and receive VoIP calls as long as your Wireless Telephone is registered with a SIP proxy server and you are operating it within range of an IEEE 802.11b/g enabled wireless network (WLAN). The SIP proxy server can belong to a wireless Internet Telephony Service Provider (ITSP) or corporate VoIP PBX system, such as the 3Com NBX® System.

[[3COM_3108_Wireless_Phone|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

===Aastra 6730i/6731i VoIP Phone===

[[File:Aastra6730i.jpg|300px|frame|left|Aastra 6730i VoIP Phone]]

'''Product:''' Aastra 6730i/6731i VoIP Phone

'''Company:''' Aastra

'''Overview:'''The Aastra 6730i, a new member of the carrier-grade, open-standards-based 67xi SIP portfolio, offers exceptional features and flexibility in an enterprise grade IP telephone. With a sleek, elegant design and a compact footprint, this multi-line SIP telephone delivers the advanced features and performance traditionally found only in higher priced products. Featuring a 3 line LCD display, the 6730i supports up to 6 lines with call appearances, offers advanced XML capability to access custom applications and is fully interoperable with leading IP-PBX platforms.

Supported by a host of Aastra configuration options, XML development tools, and continuous product enhancements via software releases, the value offered by the 6730i makes it is ideally suited for light telephone use for the small and home-based business applications.

[[Aastra_6730i_VoIP_Phone|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

===Audiocodes===

====400HD Series====

[[File:Audiocodes 420HD.jpg|300px|thumb|left|Audiocodes 420HD IP Phone]]

'''Product:''' Audiocodes 400HD Series

'''Company:''' Audiocodes

'''Overview:''' The [https://www.audiocodes.com/solutions-products/products/ip-phones AudioCodes 400HD series] of IP phones is a range of easy-to-use, feature-rich desktop devices for the service provider hosted services, enterprise IP telephony and contact center markets. Based on the same advanced, field-proven underlying technology as our other VoIP products, AudioCodes high quality IP phones enable systems integrators and end customers to build end-to-end VoIP solutions.

[[Audiocodes 400HD|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

===Cisco===

====Cisco 88XX & 68XX series====

[[File:8800_Series.png|300px|thumb|left|Cisco 8800 Series]]

'''Product:''' 88XX & 68XX series

'''Company:''' Cisco

'''Overview:''' The ''Cisco IP Phone 6800'' Series multiplatform phones are designed for affordability. They deliver reliable, business-grade audio, with Gigabit Ethernet integration and low power usage.

Ideal for customers with moderate to active VoIP needs, the 6800 Series phones are supported on Cisco-approved third-party unified communications as a service (UCaaS) providers.

The ''Cisco IP Phone 8800'' Series is a great fit for businesses of all sizes seeking secure, high-quality, full-featured VoIP. Select models provide affordable entry to HD video and support for highly-active, in-campus mobile workers. This advanced series provides flexible deployment options: on-premises, cloud and Cisco pre-approved third-party UCaaS providers.

[[Cisco IP Phone 68XX and 88XX|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====Cisco Linksys SPA942 NA====

[[File:SPA942-0.jpg|300px|thumb|left|Cisco Linksys SPA942 NA]]

'''Product:''' Linksys SPA942 NA

'''Company:''' Cisco

'''Overview:''' Four active lines and dual switched Ethernet ports offer incredible flexibility to the home or small-business Internet telephony user. Part of Cisco Small Business IP Phone Series, the SPA942 IP Phone can be configured to carry a unique phone number or extension for each of its lines or can be configured to use a shared number over multiple phones.

Based on the SIP standard, the SPA942 is able to easily integrate new features and services offered by providers. A high-resolution display makes for an easy menu- and web-based configuration.

[[Cisco_Linksys_SPA942_NA|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====Cisco SPA525G====

[[File:525g.jpg|300px|thumb|left|Cisco SPA525g Phone Adapter with Router]]

'''Product:''' Cisco SPA525G

'''Company:''' Cisco

'''Overview:''' The Cisco SPA525G 5-line IP Phone with Color Display is a full-featured VoIP (Voice over Internet Protocol) phone that provides voice communication over an IP network. It provides traditional features, such as call forwarding, redialing, speed dialing, transferring calls, conference calling, and accessing voice mail. Calls can be made or received with a handset, headset or speaker.
Your Cisco IP Phone provides a web interface for the phone user that allows you to configure some features of your phone by using a web browser.
This article will guide you through the steps for basic configuration to make it work with VoIP.ms

[[Cisco SPA525G|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====Cisco IP Phone 7940/7960====

[[File:Cisco_7960g.png|300px|thumb|left|Cisco IP Phone 7940/7960]]

'''Product:''' Cisco IP Phone 7940/7960

'''Company:''' Cisco

'''Overview:''' The Cisco IP Phone 7940/7960 is a full-featured telephone that provides voice communication over an IP network. This phone functions as a traditional analog phone, allowing you to place and receive telephone calls. This phone also supports features, such as network call forwarding, redialing, speed dialing, transferring calls, placing conference calls, and accessing voice mail. Phones require Power Over Ethernet (PoE) or a 48V AC Adapter to power up.

[[Cisco_IP_Phone_7940/7960|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====Cisco SPA30x and SPA50x series IP phones====

[[File:Cisco_spa504g.jpg|300px|thumb|left|Cisco SPA504G Phone]]

'''Product:''' Cisco SPA504G Phone

'''Company:''' Cisco

'''Overview:'''
The Cisco SPA504G is an office-style desk telephone with built-in voice over the Internet.

It is one in a series of similar models (SPA30x and SPA50x) which vary primarily in the number of lines (extensions) on the 'phone, power source (some models use power-over-Ethernet) and the availability of a second Ethernet connector. These devices are well-suited to offices and IP PBX applications. These do not provide a virtual line for connecting analog devices such as standard telephone handsets; they are instead self-contained to connect directly to VoIP.

[[Cisco_SPA504G_Phone|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

=== Dinstar===

==== Dinstar C60 Series ====

[[File:C60U_F.png|300px|thumb|left|Dinstar C60UP]]

C60 series are based on high innovative SIP technology, which is ideal for all kinds of business communication. It integrates with 132x64-pixel graphical LCD with back-light, elegant and intuitive user interface, which indicate you can enjoy good user experience.

[[Dinstar_C60_series|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

===Fanvil ===

====Fanvil X4G====

[[File:FanfillX4g.jpg|300px|thumb|left|Fanvil X4G]]

'''Product:''' Fanvil X4G

'''Company:''' Fanvil

'''Overview:''' The Fanvil X4G has a 2.8" main color screen and a secondary 2.4" DSS color screen. The user interface is sleek, colorful and easy to navigate. It has a one button call function and a call log and the ability to store 500 phonebook entries. The X4G's high compatibility supports various systems including 3CX, Avaya, OpenVox, NEC, Elastix, Asterisk, Matrix, Broadsoft, Epygi and more.

[[Fanvill X4G|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====Fanvil V62====

[[File:Fanvil-v62-VoIPms.png|300px|thumb|left|Fanvil V62]]

'''Product:''' Fanvil V62

'''Company:''' Fanvil

'''Overview:''' V62 is more than an efficient telephone but a delicate work of art, providing a smart and smooth business communication experience for enterprises. As the essential business phone featuring a graphical Dot-matrix screen with backlight and necessary VoIP features and other extended features, V62 is a great combination of elegant outside and powerful inside.

[[Fanvil V62|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====Fanvil V63====

[[File:Fanvil-v63-VoIPms.png|300px|thumb|left|Fanvil V63]]

'''Product:''' Fanvil V63

'''Company:''' Fanvil

'''Overview:''' V63 is more than an efficient telephone but a delicate work of art, providing a smart and smooth business communication experience for enterprises. As the essential business phone featuring 2.8 color screen and necessary VoIP features and other extended features, V63 is a perfect combination of elegant outside and powerful inside.

[[Fanvil V63|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====Fanvil V65====

[[File:Fanvil-v65-VoIPms.jpg|300px|thumb|left|Fanvil V65]]

'''Product:''' Fanvil V65

'''Company:''' Fanvil

'''Overview:''' V65 is more than an efficient telephone but a delicate work of art, providing a smart and smooth business communication experience for executives and managers. As the prime business phone featuring an adjustable screen and built-in Bluetooth 4.2 and 2.4G/5G Wi-Fi, V65 is a perfect combination of elegant outside and powerful inside.

[[Fanvil V62|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====Fanvil V67====

[[File:Fanvil-V67-VoIPms.jpg|300px|thumb|left|Fanvil V67]]

'''Product:''' Fanvil V67

'''Company:''' Fanvil

'''Overview:''' V67 is more than an efficient telephone but also a delicate work of art, which provides a more intelligent and elegant office operation experience for executives, managers and teleworkers. With brand new design, V67 features an adjustable touch screen and a keypad with colorful light effect that improve the beauty and comfort of office desktop.

[[Fanvil V67|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====Fanvil X4U-V2====

[[File:Fanvil-X4U-V2-VoIPms.jpg|300px|thumb|left|Fanvil X4U-V2]]

'''Product:''' Fanvil X4U-V2

'''Company:''' Fanvil

'''Overview:''' The X4/X4G is a feature-rich sip phone for business. The 4-Line IP Phone has been designed by pursuing ease of use in even the tiniest details. Dual 10/100 Mbps(X4G: 10/100/1000 Mbps) network ports with integrated PoE are ideal for extended network use. Delivering a superb sound quality as well as rich visual experience. With second DSS color screen, the IP Phone supports up to 30 DSS keys which improve work efficiency. Using standard encryption protocols to perform highly secure remote provisioning and software upgrades.

[[Fanvil X4U-V2|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====Fanvil X6U-V2====

[[File:Fanvil-X6U-V2-VoIPms.jpg|300px|thumb|left|Fanvil X6U-V2]]

'''Product:''' Fanvil X6U-V2

'''Company:''' Fanvil

'''Overview:''' Featuring three color displays, newly added line keys with LED light and built-in Bluetooth, Fanvil X6U provides the direct access to instructions, aiming to offering greater flexibility, productivity, to exceed the different demands of businesses. Wideband codec of G.722 and Opus in this device delivers you an immersive HD audio experience in both high band and low band with the network.

[[Fanvil X6U-V2|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

===Linkvil by Fanvil===

====Linkvil W611W====

[[File:Linkvil_W611W_by_Fanvil.jpg|300px|thumb|left|Linkvil W611W by Fanvil]]

'''Product:''' Linkvil W611W by Fanvil

'''Company:''' Fanvil

'''Overview:''' LINKVIL W611W is a portable, elegant Wi-Fi phone designed for mobile communication applications. Certified to IP67 standard, W611W is highly waterproof, dustproof, and drop-safe from 1.8-meter height. It has an excellent performance in different environments with humidity and dust. W611W integrates Wi-Fi 6, bringing a superb wireless communication experience. Moreover, it integrates Bluetooth 5.0 for pairing with headsets and mobile devices. Installed with a rechargeable 1900mAh battery, W611W is ready for 9 hours’ talk time or 200 hours standby time. W611W is widely used in various wireless scenarios such as enterprises, shopping malls, residential areas, hotels and warehouses, providing users with a high-quality mobile communication experience.

[[Linkvil_W611W_by_Fanvil | See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

[[Category:Devices]]

=== Flyingvoice ===

====Flyingvoice====

[[File:Flyingvoice_Phones.jpg|300px|thumb|left|Flyingvoice]]

'''Product:''' '''FIP11C / FIP11CP''', '''FIP13G''', '''FIP14G''', '''FIP15G''', '''FIP10/FIP10P''', '''FIP12WP''', '''FIP16'''

'''Company:''' Flyingvoice

'''Overview:''' Flyingvoice is a provider of communication terminal equipment and one-stop VoIP CPE solutions. They offer a full range of VoIP products, such as VoIP phones, ATAs, gateways and routers for businesses and consumers. Their WiFI IP phones offer a wireless option, so you do not need a wired internet connection to either make or receive VoIP calls.

[[Flyingvoice|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

[[Category:Devices]]

===Fortinet===

====FortiFone FON-570====

[[File:FON-570_Phone.jpg|300px|thumb|left|Fortinet FortiFone FON-570]]

'''Product:''' FortiFone FON-570

'''Company:''' Fortinet

'''Overview:''' Featuring a large 7” color touchscreen and premium HD call quality, this IP phone is great for efficient communications. Combined dedicated feature keys and programable keys expandable to 109, you have the flexibility to control your calls within your fingertips.

*7" color screen
*7 dedicated feature keys
*109 programable phone keys
*Full duplex speakerphone
*2 x 10/100/1000 Ethernet ports
*Integrated Power over Ethernet (PoE) support

[[Fortinet_FortiFone_FON-570|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====FortiFone FON-375====

[[File:FON-375_Phone.jpg|300px|thumb|left|Fortinet FortiFone FON-375]]

'''Product:''' FortiFone FON-375

'''Company:''' Fortinet

'''Overview:''' A reliable IP phone delivers HD sound quality, ideal for office workers who need efficient communications. An easy-to-read color screen and a programable second screen make it easy to display which lines are in use and who is on a call.

:*Dual color screens: 2.8" + 2.4”
:*8 dedicated feature keys
:*30 programable phone keys
:*Full duplex speakerphone
:*2 x 10/100/1000 Ethernet ports
:*Integrated Power over Ethernet (PoE) support

[[Fortinet_FortiFone_FON-375|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====FortiFone FON-175====

[[File:FON-175_Phone.jpg|300px|thumb|left|Fortinet FortiFone FON-175]]

'''Product:''' FortiFone FON-175

'''Company:''' Fortinet

'''Overview:''' A quality, two-line IP phone delivers reliable communications with HD audio quality. This entry-level business phone is easy to use that works in any office.

*2.4" color screen
*5 dedicated feature keys
*Full duplex speakerphone
*2 x 10/100 Ethernet ports
*Integrated Power over Ethernet (PoE) support

[[Fortinet_FortiFone_FON-175|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

===Gigaset A510 IP===

[[File:Gigaset_a510_IP.jpg#file|300px|thumb|left]]

'''Product:'''Gigaset A510 IP

'''Company:'''Gigaset

'''Overview:'''The Gigaset A510 and C610 IP phones are fitting solutions if you are looking for the flexibility of VoIP and the convenience of using a cordless handset.

[[Gigaset_A510_IP| See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

===Grandstream===

====DP Series====

=====Grandstream DP715/DP710=====

[[File:Grandstream 715-710.png|300px|thumb|left]]

'''Product:''' Grandstream DP715/DP710

'''Company:''' Grandstream

'''Overview:''' The DP715/710 is the next generation of powerful, affordable, high quality and simple to configure DECT Cordless IPPhone for small business and residential users. Their compact size, superb voice quality, rich feature set, market leading price-performance and wide range radio coverage enable consumers to maximize the power of IP voice application and mobility for a minimum investment.

[[Grandstream_DP715/DP710| See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

=====Grandstream DP750/DP720=====

[[File:DP750-720.png|300px|thumb|left]]

'''Product:''' Grandstream DP750/DP720

'''Company:''' Grandstream

'''Overview:''' The DP750/720 base station and handsets allows you to deploy an immersive DECT environment that allows users to communicate free from their desktop using Grandstream’s DP720 DECT handsets. The DP750 pairs with up to 5 DP720s to create a powerful and mobile network solution with up to 10 lines per handset, and 5 concurrent calls per DECT system.

[[Grandstream_DP750/DP720| See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====GXP Series====

=====Grandstream GXP1630 IP Phone=====

[[File:Gxp1630.jpg|300px|thumb|left|Grandstream GXP2120 IP Phone]]

'''Product:''' Grandstream GXP1630 IP Phone

'''Company:''' Grandstream

'''Overview:''' The GXP1630 comes equipped with a suite of VoIP features that are deployed in a clear and easy-to-use fashion. Focused primarily for low to medium call volumes and efficient call handling, its 3 line/SIP account design and 8 dual-colored BLF/speed dial keys gives this versatility. The GXP1630 also supports the best possible connection speeds and call quality with its dual Gigabit ports and HD audio on both speakerphone and handset. With other features such as its integrated PoE, 3 XML programmable soft keys and 4-way conferencing support, the GXP1630 is a high-quality and versatile Basic IP phone.

[[Grandstream_GXP1630|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

=====Grandstream GXP2120 IP Phone=====

[[File:Gxp2110.png|300px|thumb|left|Grandstream GXP2120 IP Phone]]

'''Product:''' Grandstream GXP2120 IP Phone

'''Company:''' Grandstream

'''Overview:''' The Grandstream GXP2120 is a 6 line SIP Phone which features HD Voice hardware and software support and a large 320 x 160 backlit graphical LCD. The GXP2120 can handle 6 SIP accounts represented by 6 dual-color line keys and 4 XML programmable context-sensitive soft keys. In addition, the GXP2120 has 7 dual-color BLF extension keys for the most common calls and transfers making it an ideal phone for an office user with moderate to heavy interoffice calling needs.

[[Grandstream_GXP2120_IP_Phone|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

=====Grandstream GXP2135 IP Phone=====

[[File:GXP2135-device.jpg|300px|thumb|left|Grandstream GXP2135 IP Phone]]

'''Product:''' Grandstream GXP2135 IP Phone

'''Company:''' Grandstream

'''Overview:''' The GXP2135 is the ideal selection for busy users who value call control, productivity and usability, and manage medium to heavy call volumes. Equipped with 8 lines and 4 SIP accounts, a 2.8-inch color LCD display, and 32 digital speed dial/BLF keys, the GXP2135 enables quick and powerful usability.

As all Grandstream IP phones do, the GXP2135 features state-of-the-art security encryption technology (SRTP and TLS). The GXP2135 supports a variety of automated provisioning options, including zero-configuration with Grandstream’s UCM series IP PBXs, encrypted XML files and TR-069, to make mass deployment extremely easy.

[[Grandstream GXP2135|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

=====Grandstream GXP2170=====

[[File:Grandstream_GXP2170.png|300px|thumb|left|Grandstream GXP2170]]

'''Product:''' Grandstream GXP2170

'''Company:''' Grandstream

'''Overview:''' The GXP2170 is a powerful High-End IP phone that is ideal for busy users who handle high call volumes. Receptionists, administrators, sales staff and other call-intensive rolls can enjoy efficiency by utilizing the GXP2170’s 12 line keys, 4.3 inch color display LCD and 48 digital, on-screen speed dial/BLF keys.

[[Grandstream_GXP2170|See Configuration Details]]
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=====Grandstream GXP2200=====

[[File:Grandstream_GXP2200.png|300px|thumb|left|Grandstream GXP2200]]

'''Product:''' Grandstream GXP2200

'''Company:''' Grandstream

'''Overview:''' The GXP2200 is one of the most advanced AndroidTM desktop IP phones available on the market today. The innovative phone includes the AndroidTM version 2.3 operating system with a 4.3 inch capacitive touchscreen LCD and the ability to host 6 SIP accounts. Web applications such as news, social media sites, and games can be downloaded directly via Google Play Store, and applications can be created to fit any need and downloaded directly to the phone for customized use.

[[Grandstream_GXP2200|See Configuration Details]]
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====GRP Series====

: '''Note that the GRP2615 configurations are the same to all GRP series.'''

=====Grandstream GRP2615=====

[[File:Grandstream_GRP2615.png|300px|thumb|left|Grandstream GRP2615]]

'''Product:''' Grandstream GRP2615

'''Company:''' Grandstream

'''Overview:''' The GRP2615 is a high-end carrier-grade IP phone featuring a sleek design and a suite of next-generation features including integrated Wi-Fi, Bluetooth support, 40 multipurpose keys (MPKs), an available extension module, dual Gigabit ports and more. This device features a large 4.3 inch color LCD with swappable face plates to allow for easy logo customization. For cloud provisioning and centralized management, the GRP2615 is supported by Grandstream’s Device Management System (GDMS), which provides a centralized interface to configure, provision, manage and monitor deployments of Grandstream endpoints.

[[Grandstream_GRP2615|See Configuration Details]]
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====WP Series====
:''There are multiple IP phones in this series, which are all very similar in configuration. All are self-contained IP handsets which connect directly to Wi-Fi and register directly with any of the VOiP.ms servers.''

=====Grandstream WP810=====

[[File:Grandstream WP810.png|150px|thumb|left|Grandstream WP810]]

'''Product:''' Grandstream WP810

'''Company:''' Grandstream

'''Overview:''' The WP810, WP820, WP822 and WP825 are cordless SIP IP phones which connect directly to the local network using dual-band Wi-Fi, eliminating the need for a cordless base station or other hardware. Effectively, the WP810 (with its charging cradle) is a self-contained two-line SIP solution.

[[Grandstream WP810, WP820, WP822 and WP825|See Configuration Details]]
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===Innomedia===

====BuddyTalk BT110====

[[File:Buddy_Talk_BT110_Innomedia.jpg|250px|thumb|left|Innomedia BuddyTalk BT110]]

'''Product:''' BuddyTalk BT110

'''Company:''' Innomedia

'''Overview:''' Powered by the Amazon Alexa Voice Service (AVS), the BuddyTalk series of products supports a broad suite of AVS enabled smart speaker features.
Equipped with advanced audio processing and VoIP technologies, BuddyTalk devices are intelligent speakerphones that deliver unprecedented flexibility in calling, superior voice quality, and high levels of security.

[[BuddyTalk_-_BT110|See Configuration Details]]
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===Konftel===

==== Konftel 300Wx IP ====
[[File:Konftel-300Wx-IP.png|300px|thumb|left|Konftel 300Wx IP]]

'''Product:''' Konftel 300Wx IP

'''Company:''' Konftel

'''Overview:''' The Konftel 300Wx IP wireless conference phone allows you to hold conference calls in HD quality wherever is convenient for you – without worrying about network and power outlets. Reliable and secure DECT technology. The accompanying IP DECT base can handle up to 20 registered Konftel 300Wx devices and five ongoing calls.

The rechargeable battery ensures more than 60 hours of call time, so you can talk for a full working week without recharging! A USB port makes the Konftel 300Wx ready for all the apps and services we use to communicate and collaborate via computers. Combine meeting apps and regular phone calls. OmniSound® delivers superb sound quality

[[Konftel 300Wx|See Configuration Details]]
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==== Konftel 300IPx ====

[[File:Konftel300ipx-conference-phone.jpg|300px|thumb|left|Konftel 300IPx]]

'''Product:''' Konftel 300IPx

'''Company:''' Konftel

'''Overview:''' The [https://www.konftel.com/en/products/konftel-300ipx KONFTEL 300IPx] together with the Konftel Unite app brings a whole new easiness to conference calls. It is highly intuitive and based on our natural mobile behavior. The new generation of IP conference phone is – The Art of Easiness.

[[Konftel 300IPx|See Configuration Details]]
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===Panasonic===

====Panasonic KX-TGP 550====

[[File:Pana550b.jpg|300px|thumb|left|Panasonic KX-TGP 550]]

'''Product:''' Panasonic KX-TGP 550

'''Company:''' Panasonic

'''Overview:''' The Panasonic KX-TGP550 responses to the needs of SIP IP-Centrix/Hosted PBX systems and Asterisk users. Conveniently, no need to set up a system telephone at every base. This system also enables you to use a range of convenient services provided by the carrier such as Call Forward, Voice Mail, etc.

*Support for 3 simultaneous network conversations
*Up to 6 DECT cordless handsets
*Support for up to 8 SIP registrations (e.g. up to 8 DID lines or extensions)
*Hands-free speakerphone on a cordless handset

[[Panasonic KX-TGP 550|See Configuration Details]]
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====Panasonic KX-HDV130C====

[[File:Panasonic_KXHDV130_01.jpg|300px|thumb|left|Panasonic KX-HDV130C]]

'''Product:''' Panasonic KX-HDV130C

'''Company:''' Panasonic

'''Overview:''' The Panasonic KX-HDV130 SIP desk phone delivers the ideal balance of low cost and high quality, along a range of value added features.

*Support for up to 2 SIP registrations (e.g. up to 2 DID lines or extensions)
*Support for 3 simultaneous network conversations (3-way conferencing)*
*2 Programmable keys / Line keys
*Hands-free speakerphone on a cordless handset

[[Panasonic KX-HDV130C|See Configuration Details]]
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====Panasonic KX-HDV230====

[[File:Panasonic_KXHDV230_01.jpg|300px|thumb|left|Panasonic KX-HDV230]]

'''Product:''' Panasonic KX-HDV230

'''Company:''' Panasonic

'''Overview:''' The Panasonic KX-HDV230 IP phone offers streamlined functions and the high definition voice quality that's essential for effective communication.

*Support for up to 6 SIP registrations (e.g. up to 6 DID lines or extensions)
*24 Flexible Function Keys
*2 ethernet ports 10/100/1000 Base -T
*Large LCD with Backlight

[[Panasonic KX-HDV130C|See Configuration Details]]
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====Panasonic KX-HDV330====

[[File:Panasonic_KXHDV330_01.jpg|300px|thumb|left|Panasonic KX-HDV330]]

'''Product:''' Panasonic KX-HDV330

'''Company:''' Panasonic

'''Overview:''' The Panasonic KX-HDV330 is a multi-functional business SIP phone equipped with a colour touch panel for intuitive operation.

*Large LCD with Backlight
*Built-in Bluetooth®
*Support for up to 12 SIP registrations (e.g. up to 12 DID lines or extensions)
*24 Flexible Function Keys
*Hands-free speakerphone on a cordless handset

[[Panasonic KX-HDV130C|See Configuration Details]]
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===Pirelli DP-L10===

[[File:Pirelli1.jpg|300px|thumb|left|Pirelli DP-L10]]

'''Product:''' Pirelli DP-L10

'''Company:''' Pirelli

'''Overview:''' Pirelli's DualPhone DP-L10 phone combines GSM tri-band capabilities with SIP-based Wireless VoIP via an integrated WLAN 802.11b/g interface, combining cell phone technology with the added advantage of transporting phone calls over the Internet.

The Dual Phone enables mobile phone features such as Internet browsing, e-mail, SMS, and MMS, which can also be used through a fixed-broadband connection, providing the same mobile services with the added benefits of greater bandwidth and cost savings.

[[Pirelli DP-L10|See Configuration Details]]
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===Poly (Polycom, Obahai, Plantronics)===
Poly.com has been acquired by Hewlett-Packard.

====Poly Edge B10====

[[File:PolyEdgeB10.png|300px|thumb|left|Poly Edge B10]]

'''Product:''' Poly Edge B10

'''Company:''' Poly

'''Overview:''' Sleek design meets Poly pro-grade audio at a shockingly affordable price: That’s what makes the Poly Edge B Series IP Phones the genius choice for any growing business. Easy to use with illuminated keys where you need them. Plug-and-play provisioning and hardcore reliability make it pure value in a low-cost business phone.

[[Poly_Edge_B10|See Configuration Details]]

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====Polycom SoundStation IP 4000 Conference Phone====

[[File:Polycom-4000-0.gif|300px|thumb|left|Polycom SoundStation IP 4000 Conference Phone]]

'''Product:''' Polycom SoundStation IP 4000 Conference Phone

'''Company:''' Poly

'''Overview:''' the SoundStation IP 4000 SIP voice conferencing unit is the answer for organizations that are ready for the
benefits and versatility of a SIP enabled business. Designed for offices or small to medium-sized conference rooms, the SoundStation IP 4000 SIP provides remarkable room coverage. You can speak naturally from up to 10-feet away from a microphone and still be heard clearly on the far end of the call. Need coverage for a larger room? The optional extension microphones offer an increased pickup for larger rooms. Plus, with gated microphone technology, echo and background noise is almost entirely eliminated.

The SoundStation IP 4000 SIP also provides familiar features that are easy to use. The menu-driven user interface, viewable on a high-resolution backlit LCD, offers convenient access to business telephony functions you use most including transfer, hold, redial, and conference.

[[Polycom_SoundStation_IP_4000_Conference_Phone|See Configuration Details]]
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====Polycom SoundPoint IP 501, 550, 650, etc.====

[[File:Polycom501.jpg|258px|thumb|left|Polycom SoundPoint IP 501]]

'''Product:''' Polycom SoundPoint IP 501, 550, 650, etc.

'''Company:''' Poly

'''Overview:''' The SoundPoint IP series are multi-line (3-6 lines depending on model) Voice over IP telephones that seamlessly integrate with IP PBX and softswitch vendors’ IP solutions.

As protocols develop and standards evolve, it’s easy to update the phone in the field via a software download, thus enabling new features and functionality for the phone and protecting your investment.

Whether you deploy MGCP or SIP standards, Polycom offers a solution that fits all your business communication needs.

[[Polycom_SoundPoint_IP_501|See Configuration Details]]
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====Polycom SoundPoint IP 331====

[[File:Polycom_Soundpoint_IP_331.png|258px|thumb|left|Polycom SoundPoint IP 331]]

'''Product:''' Polycom SoundPoint IP 331

'''Company:''' Poly

'''Overview:''' The Polycom IP 331 is engineered to make installation, configuration, and upgrades as simple and efficient as possible. The phones' standard base stand can be reversed to become a wall mount, eliminating the need for a separate accessory. Built-in IEEE 802.3af PoE circuitry and a dual-port Ethernet switch enables flexible deployment options and savings on cabling expenses.

[[Polycom_Soundpoint_IP_331|See Configuration Details]]
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====Polycom SoundPoint IP 601====

[[File:Voipms-polycom601.jpg|258px|thumb|left|Polycom SoundPoint IP 601]]

'''Product:''' Polycom SoundPoint IP 601

'''Company:''' Poly

'''Overview:''' The 6-line Polycom® SoundPoint IP™ 601 offers industry-leading functionality and call handling unmatched voice quality an intuitive user interface & expandability to 12 lines!

[[Polycom_SoundPoint_IP_601|See Configuration Details]]
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====Poly VVX-D230====

[[File:Poly VVX-D230 base.png|250px|thumb|left|Poly VVX-D230 cordless base]]

'''Product:''' Poly VVX-D230 cordless DECT phone

'''Company:''' Poly (Obahai, Polycom)

'''Overview:''' The VVX series are SIP desk phones aimed at small to mid-sized business; the VVX-D230 is a cordless system on which one base may control up to ten handsets.

[[Poly VVX-D230|See Configuration Details]]
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====Poly VVX 300, 400, etc====

[[File:Vvx300.png|250px|thumb|left|Polycom VVX 300 Series]]

'''Product:''' Polycom VVX Series

'''Company:''' Poly

'''Overview:''' The VVX series provides high-quality audio (HD Voice) and video communications from 6 lines and up.

[[Polycom VVX 300, 400, etc|See Configuration Details]]
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=== Positron IP phones ===

[[File:PositronLogo.jpeg|250px|thumb|left|Positron IP phones]]

'''Product:''' Positron

'''Company:''' [http://www.positrontelecom.com/products.html Positron]

'''Overview:''' Positron IP Phones is an affordable next-generation SIP phone including wideband audio support, ethernet ports and integrated Power over Ethernet.

All the IP Phones are optimized for executives, administrative assistants and those working with bandwidth-intensive applications on collocated PCs. Line keys can also be configured as IP PBX features such as BLF, SCA, Intercom, Call Pickup, Call Park, and many others. The high-resolution screen and the HD voice features provide a high quality visual and authentic audio experience.

==== Positron IP304 ====

[[File:IP304.png |250px|thumb|left|Positron IP304]]

'''Product:''' Positron IP304

'''Company:''' [http://www.positrontelecom.com/products.html Positron]

'''Overview:''' Positron IP304 is an affordable next-generation SIP phone with wideband audio support, dual Ethernet port and integrated Power over Ethernet.

The IP304 enterprise VoIP phone is Positron’s entry-level phone with 3 VoIP accounts. Its functionalities make it easy to place calls on hold, to transfer calls and to initiate a conference call.

With HD audio for both headset and speakerphone, IP304 provides a clear authentic audio experience

[[Positron IP304 | View configuration for Positron IP304]]
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==== Positron IP304C ====

[[File:IP304C.png |250px|thumb|left|Positron IP304C]]

'''Product:''' Positron IP304C

'''Company:''' [http://www.positrontelecom.com/products.html Positron]

'''Overview:''' Positron IP304C is an innovative enterprise-level IP Phone that features 4 line keys, color display, 3.5” TFT-LCD with 480 x 320 pixel. It supports up to a 5-way conference.

IP304C is optimized for executives, administrative assistants and those working with bandwidth-intensive applications on collocated PCs. Line keys can also be configured as IP PBX features such as BLF, SCA, Intercom, Call Pickup, Call Park, and many others.

The high-resolution TFT-LCD screen and the HD voice features provide a high quality visual and authentic audio experience.

[[Positron IP304C | View configuration for Positron IP304C]]
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==== Positron IP408 ====

[[File:IP408.png |250px|thumb|left|Positron IP408]]

'''Product:''' Positron IP408

'''Company:''' [http://www.positrontelecom.com/products.html Positron]

'''Overview:''' Positron IP408 is an affordable next-generation SIP 2.0 phone including wideband audio support and WAN/LAN Ethernet ports with route and bridge mode.

The IP408 enterprise VoIP phone supports 4 SIP lines. Its functionalities make it easy to place calls on hold, to transfer calls and to initiate a conference call.

With HD audio for both headset and speakerphone, IP408 provides a clear authentic audio experience

[[Positron IP408 | View configuration for Positron IP408]]
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==== Positron IP410C ====

[[File:IP410C.png |250px|thumb|left|Positron IP410C]]

'''Product:''' Positron IP410C

'''Company:''' [http://www.positrontelecom.com/products.html Positron]

'''Overview:''' Positron IP410C is an affordable next-generation SIP Phone that features 4 line keys, 10 programmable extension keys, color display, wideband audio support and dual Ethernet port with integrated Power over Ethernet.

IP410C is optimized for executives, administrative assistants and those working with bandwidth-intensive applications on collocated PCs. Line keys can also be configured as IP PBX features such as BLF, SCA, Intercom, Call Pickup, Call Park, and many others.

The high-resolution TFT-LCD screen and the HD voice features provide a high quality visual and authentic audio experience.

[[Positron IP410C | View configuration for Positron IP410C]]
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==== Positron IP410G ====

[[File:IP410G.png |250px|thumb|left|Positron IP410G]]

'''Product:''' Positron IP410G

'''Company:''' [http://www.positrontelecom.com/products.html Positron]

'''Overview:''' Positron IP410G is an innovative enterprise-level color IP Phone that features 4 line keys, 10 programmable extension keys, color display, 3.5” TFT-LCD with 480*320 pixel, wideband audio support and dual Gigabit Ethernet port with integrated Power over Ethernet.

IP410G is optimized for executives, administrative assistants and those working with bandwidth-intensive applications on collocated PCs. Ten programmable keys can also be configured as IP PBX features such as BLF, SCA, Intercom, Call Pickup, Call Park, and many others.

The high-resolution TFT-LCD screen and the HD voice features provide a high quality visual and authentic audio experience.

[[Positron IP410G | View configuration for Positron IP410G]]
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===Siemens Gigaset C450-Ip===

[[File:Siemens_gigaset_c450_IP.jpg|300px|thumb|left|Siemens Gigaset C450-Ip]]

'''Product:''' Siemens Gigaset C450-Ip

'''Company:''' Siemens

'''Overview:''' The Gigaset C450 is an entry phone in the Siemens Gigaset product line, the addition of IP connectivity via a standard Ethernet 10/100 BaseT interface and SIP protocol within a reasonable price bracket, opens large applications where wireless operation and SIP are required.
The Ethernet connectivity allows the unit to work without a PC.

The dual mode (SIP/PSTN) enables incoming call on a legacy interface with existing directory number and safe emergency outgoing calls.

The presentation of the unit with an independent base and a dedicated phone charger/support will simplify cabling.

[[Siemens Gigaset C450-Ip|See Configuration Details]]
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===Snom===

====Snom 320 VoIP Phone====

[[File:Snom320.png|300px|frame|left|Snom 320 VoIP Phone]]

'''Product:''' Snom 320 VoIP Phone

'''Company:''' Snom

'''Overview:'''Ideal for the office and everyone who spends a lot of time on the phone, the snom 320 is an affordable, yet powerful SIP business phone with a built-in, full-duplex speakerphone and three-party conference bridging.

[[SNOM 320|See Configuration Details]]
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====Snom m3 VoIP Phone====

[[File:Snom_m3.png|300px|thumb|left|Snom m3 VoIP Phone]]

'''Product:''' Snom m3 VoIP Phone

'''Company:''' Snom

'''Overview:''' The snom m9 is the next generation DECT handheld (CAT-iq) that empowers users with the convenience of wireless communication along with the widely accepted benefits and feature richness of Voice-over-IP telephony.

The snom m9 promises to deliver excellent speech quality through digital and wideband audio (klarVOICE).

By combining professional functions of versatile business communication with the intuitive features of the mobile-carrier world, the snom m9 is ideally suited for professional and private use alike. With features such as hands-free mode, calling line identification (CLI) by displaying name, number, and image of the caller as well as typical mobile-phone features such as Address Book, Calendar, Calculator and Alarm function, the snom m9 provides the perfect blend of mobility and accessibility.

[[Snom_m3_VoIP_Phone|See Configuration Details]]
<br clear="all" />

====SNOM C520====

[[File:snom_c520.png|300px|thumb|left|C520]]

'''Product:''' SNOM C520 Conferencing

'''Company:''' Snom

'''Overview:''' With its modern and sleek design, the C520 fits seamlessly into your working day. Two detachable DECT microphones can be positioned freely or carried in the room as required to ensure the best sound and voice quality.

Built-in charging stations with magnetic bays directly on the base station mean both microphones are always charged and ready for use in the next meeting. The conference phone also features automatic volume control and digital noise reduction so that all call participants can be understood in best sound quality.

[[Snom C520|See Configuration Details]]
<br clear="all" />

====SNOM professional D7XX====

[[File:Snom.jpg|300px|thumb|left|Snom D7XX]]

'''Products:''' D120, D717, D735, D785

'''Company:''' Snom

'''Overview:''' The [https://www.snom.com/en/ip-phones/desk-phones/d7xx-series/ professional D7XX] Series telephones are both aesthetically appealing and highly practical, meeting business requirements when a telephone is a key tool in daily work.

These high-performance devices are future-proofed and provide the best in Wideband HD audio, ensuring crystal clear sound quality. They are Bluetooth compatible to meet the connectivity requirements of today’s offices.

[[Snom IP Phones|See Configuration Details]]
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====SNOM M100 KLE====

[[File:M100.jpg|300px|thumb|left|M100 KLE]]

'''Products:''' M100 KLE SIP DECT 4-Line Base Station

'''Company:''' Snom

'''Overview:''' The [https://www.snomamericas.com/en/pd/ip-phones/m-series/m-kle-series/m100-kle/ M100 KLE SIP DECT 4-Line Base Station] supports up to 10 phones in the Snom KLE DECT 4-Line Series, including the M10 and M10R SIP DECT 4-Line handsets and the M18 KLE SIP DECT 4-Line deskset. This cordless family of phones features four programmable LED backlit line keys on the handsets and desk sets.

With key system emulation, the M100 KLE Series handles shared line appearances locally without the need for SCA (shared called appearances) support from your provider. This allows an easy and intuitive method for your customers to see incoming calls, hold calls, and resume calls from any handset or deskset with a simple press of a button.

[[M100_KLE_SIP_DECT_4-Line_Base_Station|See Configuration Details]]

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=== Vtech ===

==== Vtech Conference Station ====

[[File:VCS754-thumb.PNG|300px|thumb|left|Vtech VCS Series]]

'''Product:''' VCSV752 & CS754

'''Company:''' Vtech

'''Overview:''' The [https://businessphones.vtech.com/pd/3439/VCS754-ErisStation-SIP-Conference-Phone-with-Four-Wireless-Mics Vtech VCS754 ErisStation] conference phone features a compact, all-in-one design makes it easy to keep everything together—no clutter, no hassle. Built-in charging stations with magnetic bays ensure the microphones are charged and available for the next meeting.

[[Vtech Conference Station|See Configuration Details]]
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==== Vtech VSP Series ====

[[File:VSP736 ErisTerminal.jpg|300px|thumb|left|VSP Series]]

'''Products:''' VSP600 - VSP715 - VSP725 - VSP726 - VSP736

'''Company:''' Vtech

'''Overview:''' The [https://businessphones.vtech.com/products/sip-phones/vsp700 Vtech VSP700 Series] comes with all the essential features you need to keep pace with your business and your budget. Depending on the model, support two to six SIP accounts with these easy-to-use phones.

[[Vtech VSP Series|See Configuration Details]]
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===Yealink===

====Yealink Voice Solutions====

[[File:Yealink_easyVoip.png|300px|thumb|left|Yealink SIP-T28P]]

'''Product:''' Yealink W60B, Yealink T21, Yealink T42S

'''Company:''' Yealink

'''Overview:''' Yealink offers solutions for each customer's needs, starting from basic to more complex ones.

[[Yealink Voice Solutions|See Configuration Details]]
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====Yealink SIP-T28P (VSRF)====

[[File:Yealink-sip-t28p.jpg|300px|thumb|left|Yealink SIP-T28P]]

'''Product:''' Yealink SIP-T28P

'''Company:''' Yealink

'''Overview:''' Yealink SIP-T28P represents the next generation VoIP phone which designed for the business user who needs rich telephony features, friendly UI and super voice quality. It is equipped with the TI TITAN chipset, offers high definition voice quality through TI voice engine, HD handset, HD speaker and HD codec (G.722).

By the large, high-resolution graphical display, and together with all the 48 keys, SIP-T28P offers an excellent user experience to configure, make calls, express XML browser, etc. Moreover, to avoid problems with unwanted violations of your audio data, Yealink SIP-T28P supports the security standards TLS, SRTP, HTTPS, 802.1x, Open VPN and AES encryption which are necessary to protect against electronic eavesdropping and data theft.

[[Yealink_SIP-T28P|See Configuration Details]]
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====Yealink SIP-W73P (W70B) DECT====

[[File:Yealink_SIP-W73P.png|300px|thumb|left|Yealink SIP-W73P DECT IP Phone]]

'''Product:''' Yealink SIP-W73P DECT IP PHONE

'''Company:''' Yealink

'''Overview:''' The Yealink W70B is the DECT IP base station for small and medium-sized businesses. Paring with up to a total of 10 Yealink W73H/W56H/W59R/CP930W/color screen DDPhone (T54W+DD10K) DECT handsets, W70B allows you to enjoy superb mobility and efficient flexibility immediately as well as significantly eliminates additional wiring troubles and charges. A powerful chip ensuring a better and higher performance, this DECT IP base station not only supports up to 10 VoIP accounts and 20 simultaneous calls, but also speeds up its startup and signal connection, provides no perception upgrade, slashes its upgrade downtime as well.

[[Yealink_SIP-W73P|See Configuration Details]]
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====Yealink SIP-T54W====

[[File:Yealink_SIP-T54W.png|300px|thumb|left|Yealink SIP-T54W Prime Business Phone]]

'''Product:''' Yealink SIP-T54W

'''Company:''' Yealink

'''Overview:''' Yealink SIP-T54W is an easy-to-use Prime Business Phone with an adjustable 4.3-inch color LCD screen that you can easily and flexibly find the comfortable viewing angle according to the personal and environmental needs. With the built-in Bluetooth 4.2 and the built-in dual band 2.4G/5G Wi-Fi, the SIP-T54W IP Phone ensures you to keep up with the modern wireless technology and take the first chance in the future wireless age. Its built-in USB 2.0 port allows for USB recording or a direct wired/wireless USB headset or up to three Yealink EXP50 expansion modules connection. Benefitting from these features, the Yealink SIP-T54W is a powerful and expandable office phone that delivers optimum desktop efficient and productivity

[[Yealink_SIP-T54W|See Configuration Details]]
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====Yealink SIP-T46U====

[[File:Yealink_SIP-T46U.png|300px|thumb|left|Yealink SIP-T46U Ultra-elegant Gigabit IP Phone]]

'''Product:''' Yealink SIP-T46U

'''Company:''' Yealink

'''Overview:''' The SIP-T46U IP phone is an ultimate communication tool that has the better overall performance. The phone employs an appealing high-resolution TFT color display that looks brighter and more vibrant. United Yealink Optima HD Voice technology and wideband codec of Opus, the T46U awards you the superb audio quality and crystal-clear voice communications. Moreover, the T46U puts dual USB ports in a phone that makes Bluetooth, Wi-Fi, USB headset and USB recording come true, and you can use any two of them freely according to your needs.

[[Yealink_SIP-T46U|See Configuration Details]]
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====Yealink SIP-T33G====

[[File:Yealink_SIP-T33G.png|300px|thumb|left|Yealink SIP-T33G Classic Business IP Phone]]

'''Product:''' Yealink SIP-T33G

'''Company:''' Yealink

'''Overview:''' Yealink SIP-T33G offers support for 4 lines and includes local 5-way
conferencing. For its fashionable appearance as well as an extra-large 320x240-pixel color display with backlight, it brings comfortable operation experience and clear visual experience for users. Designed with a new powerful chip, it helps greatly improved work efficiency. Additional features include a dual-port Gigabit Ethernet with integrated PoE, EHS35 support for Yealink wireless headset, and adjustable multi-angle stand support. These features allow the SIP-T33G to be a high-quality but cost-effective classic IP phone that maximizes productivity in both small and large office environments.

[[Yealink_SIP-T33G|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

[[Category:Devices]]

===Zycoo ZP502===

[[File:ZP502.jpg|left|Zycoo ZP502]]

'''Product:''' Zycoo ZP502

'''Company:''' Zycoo ZP502

'''Overview:''' ZP502 VoIP Phone is ZYCOO business IP phone terminal which adopts SIP and IAX2 protocols.

With Broadcom solution, compatible with various Platforms such as Asterisk, FreePBX, Broadsoft, Cisco call manager, etc.

[[Zycoo ZP502|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

Poly VVX-D230

2023-09-25T16:24:33Z

Carlb:

[[Image:Poly VVX-D230 base.png|thumb|The VVX-D230's base can control ten DECT handsets]]
Aimed at small business users, the Poly (Polycom, Obahai) '''VVX-D230''' is a DECT [[IP Phones|SIP phone]] with support for up to ten cordless handsets — including the one handset which is packaged with the base.

This system (OBAHAI/VVXD230) is intended to complement the [[Polycom VVX 300, 400, etc|VVX-series]] desk phones and the [[Poly Trio 8800]] conference room speakerphone.

==Configuration==
# If the cordless handsets are not already paired to the base, press and hold the 'find' button on the base for at least five seconds. Then, from the handset, go to Menu→Settings→Registration→Register.
# Configuration of the system is done via a web interface. From any paired handset, go to Menu→Settings→Basestation Info to obtain the IPv4 address of the base.
# Open a web browser to this address and log in with username 'admin' and default password 'admin'.

<br clear="all">
[[Image:Poly VVX-D230 Wizard.png]]

'''From the "Setup Wizard" page''', change the following parameters (changes will not be applied until you save them and reboot the device). Be sure to uncheck the "default" check box for each item you intend to modify:
*'''LocalTimeZone''' - set to your local time zone, for instance GMT-5:00 for (Eastern Time). Factory default is GMT-8:00 (Pacific Time, California).
*'''ITSP A SIPProxyServer''' - atlanta.voip.ms (or one of the other multiple [[servers]] - this must match the same server as your [[DID Troubleshooting|DID]] configuration).
*'''ITSP A DigitMap''' - (optional) (1xxxxxxxxxx|<1555>[2-9]xxxxxxS3|<1>[2-9]xxxxxxxxx|011xx.|xx.|(Mipd)|[^*#]@@.) (as an example, replacing 1-555 with your local area code) will enable both 7 and 10-digit dial for North American numbers. See the manufacturer's technical reference manual for details of the syntax.
*'''Phone1 PrimaryLine''' - "SP1 service"
*'''SP1 ITSP Profile''' - A
*'''SP1 AuthUserName''' - 123456_1 (your voip.ms user number and a [[Sub Accounts|subaccount]] number, in this example user #123456 subaccount #1
*'''SP1 AuthPassword''' - the password associated with your voip.ms subsccount
If you have additional virtual lines, repeat the same configuration steps for each subaccount using SP2, SP3... up to a maximum of eight lines and ten handsets.

<br clear="all">
[[Image:Poly VVX-D230 SP1.png]]

'''From the "SP1 Service" page''', change the following settings for the first virtual "line" (the others will be similar):
*'''X_DisplayNumber''' - A short text string identifying this line on the local handset screen. Typically the seven-digit NXX-XXXX local number, although any value will do.
*'''X_AcceptSipFromRegistrarOnly''' - YES (to keep [[Sip Scanner Ghost Calls|spurious]] calls out)
*'''X_KeepAliveEnable''' - YES (if you are behind NAT)
*'''X_KeepAliveExpires''' - 180 (three minutes) is reasonable
*'''DirectoryNumber''' - NPA-NXX-XXXX, the directory number associated with this line
*'''AuthUserName''', '''AuthPassword''' - should already be set from the "Setup Wizard" above; if not, set them here.
*'''CallerIDName''' - your name (15 alphanumeric chars max, no punctuation but spaces are allowed) as you want it to appear on [[Caller ID]] for outbound calls.
*'''X_CheckVoiceMailNumber''' - *97
*'''MessageWaiting''' - YES

Save everything before leaving each page of the setup. When you are finished making changes, reboot the device.

Try a test call, such as the [[Dialing Codes|echo test]] (press [4] [4] [4] [3] and the [green] button). You should hear your own voice played back just as it was received.

==See alos==
* [[Polycom VVX 300, 400, etc|VVX-series]]
* [[Poly Trio 8800]]

==Documentation==
This page is merely a brief overview with just enough information to connect the handsets to VoIP.ms; see the manufacturer's website (poly.com) for more extensive documentation:

* Quick start: [https://www.poly.com/content/dam/www/products/support/voice/vvx-dect-ip-phones/setup/vvx-d230-hs-bs-class-a-qsg-access.pdf handset + base], [https://www.poly.com/content/dam/www/products/support/voice/vvx-dect-ip-phones/setup/vvx-d230-hs-chg-class-a-qsg-access.pdf expansion handset + charger]
* [https://www.poly.com/content/dam/www/products/support/voice/vvx-dect-ip-phones/user/vvxd230-adminguide-710.pdf Admin guide] and [https://www.poly.com/content/dam/www/products/support/voice/vvx-dect-ip-phones/user/vvxd230-tech-ref-710.pdf technical reference] manual from Poly.com

[[Category:SIP phones]]

ReadyNet AC1000MS and AC1300MS

2023-09-25T16:09:51Z

Carlb:

[[Image:Readynet-ac1000ms.png|thumb|ReadyNet AC1000MS]]
The '''ReadyNet AC1000MS and AC1300MS''' are dual-band wi-fi routers which include a built-in [[ATA Device|analogue telephone adapter]]. The AC1300MS provides one virtual line (1x FxS) while the AC1000MS provides two lines (2x FxS) plus a port to allow an external USB hard drive to be connected to build a file server.

The advantage of these units is that they combine a wi-fi router and an ATA in one box, much as the Cisco [[Cisco WRP400|WRP400 and WRP500]] did before their discontinuation in 2017.

== Configuration ==
First, connect a PC to the router; this may be done using either a hard-wired Ethernet connection or by Wi-Fi. The default network names (SSID) and password for Wi-Fi are marked on a sticker on the underside of the router.

The configuration screen, by default, is at <nowiki>http://192.168.11.1</nowiki> - the username is "user" and the password is marked on the sticker on the underside of the unit.

First, go to the "Administration" tab, section "Time/Date setting". Select your local time zone, click "sync with host", then click "save and apply".

[[Image:Readynet-timedate.png]]

Next, go to the "SIP" tab. Most of these settings may be left as-is, but "SIP T1" may need to be set to 1000ms to prevent outbound audio from breaking up on high-latency connections.

"NAT transversal" can usually be left as "Disabled", unless the ReadyNet box is behind another router. Optionally, you may also wish to set a [[Dial Plan for Linksys ATAs|dial plan]]; the syntax is fairly similar to the Cisco/Linksys ATA's, except that each rule is placed on a separate line instead of being separated by pipe (|) characters on a single line.

Click "save and apply".

[[Image:Readynet-siptab.png]]

Lastly, go to the "FXS1" or "FXS2" tab (there's one for each line on the AC1000MS, both look similar). FXS1 is shown in this example, although both lines may be configured separately for different [[Sub Accounts|subaccounts]] (the most common option) or indepedently for different users or even different providers.

There are multiple values to be set on this page:
* Line Enable: '''Enable'''
* Proxy Server and Outbound Server: Any of the VoIP.ms [[servers]], for instance '''toronto.voip.ms'''. Be sure to point your [[DID Troubleshooting|DID]] to this same server.
* Proxy Port and Outbound Port: This can usually be left as '''5060''', although a few other options (such as '''5080''') are supported by the VoIP.ms servers.
* Display Name: Your name, as you intend it to appear on call display - preferably 15 characters or less.
* Account and Phone Number: Set both to your VoIP.ms account or subaccount number. For instance, user #123456 subaccount #1 would be written as '''123456_1'''.
* Password: The password associated with your VoIP.ms subaccount (above).

[[Image:Readynet-fxstab.png]]

and lastly, in the "advanced" section:

* Register Refresh Interval (sec): should be set to something low like '''120''' (two minutes) or '''180''' (three minutes) if you need to periodically re-register to keep the connection open on your network.

[[Image:Readynet-fxs-tab.png]]

Once configuration is complete, the "PHONE 1" or "PHONE 2" LED should come on solid green on the router, the "Status" tab should indicate "Registered" and the VoIP.ms web interface should show the subaccount as having successfully registered.

Configuration of the second FXS line tab is the same or similar to the first.

Try a test call, such as '''4443#''' to invoke the echo test. Try placing an ordinary call to or from the ReadyNet router. The system should be ready for use.

==External links==
* [https://www.readynetsolutions.com/ac1000ms-wireless-ac-voip-router AC1000MS] [https://www.readynetsolutions.com/ac1000ms-support-and-downloads support and downloads]
* [https://www.readynetsolutions.com/ac1300ms-wireless-ac-voip-router AC1300MS] [https://www.readynetsolutions.com/ac1300ms-support-and-downloads support and downloads] on ReadyNetSolutions.com

[[Category:Analog Telephone Adapters]]
[[Category:Routers]]

ReadyNet AC1000MS and AC1300MS

2023-09-25T16:09:31Z

Carlb: Duplicate of Image:Readynet-ac1000ms.jpg. Original is failing to resize due to server error. "Error creating thumbnail: Incomplete GD library configuration: missing function imagecreatefromjpeg"

[[Image:Readynet-ac1000ms.pmg|thumb|ReadyNet AC1000MS]]
The '''ReadyNet AC1000MS and AC1300MS''' are dual-band wi-fi routers which include a built-in [[ATA Device|analogue telephone adapter]]. The AC1300MS provides one virtual line (1x FxS) while the AC1000MS provides two lines (2x FxS) plus a port to allow an external USB hard drive to be connected to build a file server.

The advantage of these units is that they combine a wi-fi router and an ATA in one box, much as the Cisco [[Cisco WRP400|WRP400 and WRP500]] did before their discontinuation in 2017.

== Configuration ==
First, connect a PC to the router; this may be done using either a hard-wired Ethernet connection or by Wi-Fi. The default network names (SSID) and password for Wi-Fi are marked on a sticker on the underside of the router.

The configuration screen, by default, is at <nowiki>http://192.168.11.1</nowiki> - the username is "user" and the password is marked on the sticker on the underside of the unit.

First, go to the "Administration" tab, section "Time/Date setting". Select your local time zone, click "sync with host", then click "save and apply".

[[Image:Readynet-timedate.png]]

Next, go to the "SIP" tab. Most of these settings may be left as-is, but "SIP T1" may need to be set to 1000ms to prevent outbound audio from breaking up on high-latency connections.

"NAT transversal" can usually be left as "Disabled", unless the ReadyNet box is behind another router. Optionally, you may also wish to set a [[Dial Plan for Linksys ATAs|dial plan]]; the syntax is fairly similar to the Cisco/Linksys ATA's, except that each rule is placed on a separate line instead of being separated by pipe (|) characters on a single line.

Click "save and apply".

[[Image:Readynet-siptab.png]]

Lastly, go to the "FXS1" or "FXS2" tab (there's one for each line on the AC1000MS, both look similar). FXS1 is shown in this example, although both lines may be configured separately for different [[Sub Accounts|subaccounts]] (the most common option) or indepedently for different users or even different providers.

There are multiple values to be set on this page:
* Line Enable: '''Enable'''
* Proxy Server and Outbound Server: Any of the VoIP.ms [[servers]], for instance '''toronto.voip.ms'''. Be sure to point your [[DID Troubleshooting|DID]] to this same server.
* Proxy Port and Outbound Port: This can usually be left as '''5060''', although a few other options (such as '''5080''') are supported by the VoIP.ms servers.
* Display Name: Your name, as you intend it to appear on call display - preferably 15 characters or less.
* Account and Phone Number: Set both to your VoIP.ms account or subaccount number. For instance, user #123456 subaccount #1 would be written as '''123456_1'''.
* Password: The password associated with your VoIP.ms subaccount (above).

[[Image:Readynet-fxstab.png]]

and lastly, in the "advanced" section:

* Register Refresh Interval (sec): should be set to something low like '''120''' (two minutes) or '''180''' (three minutes) if you need to periodically re-register to keep the connection open on your network.

[[Image:Readynet-fxs-tab.png]]

Once configuration is complete, the "PHONE 1" or "PHONE 2" LED should come on solid green on the router, the "Status" tab should indicate "Registered" and the VoIP.ms web interface should show the subaccount as having successfully registered.

Configuration of the second FXS line tab is the same or similar to the first.

Try a test call, such as '''4443#''' to invoke the echo test. Try placing an ordinary call to or from the ReadyNet router. The system should be ready for use.

==External links==
* [https://www.readynetsolutions.com/ac1000ms-wireless-ac-voip-router AC1000MS] [https://www.readynetsolutions.com/ac1000ms-support-and-downloads support and downloads]
* [https://www.readynetsolutions.com/ac1300ms-wireless-ac-voip-router AC1300MS] [https://www.readynetsolutions.com/ac1300ms-support-and-downloads support and downloads] on ReadyNetSolutions.com

[[Category:Analog Telephone Adapters]]
[[Category:Routers]]

File:Readynet-ac1000ms.png

2023-09-25T16:08:25Z

Carlb: Duplicate of :Image:Readynet-ac1000ms.jpg. Original is failing to resize due to server error. "Error creating thumbnail: Incomplete GD library configuration: missing function imagecreatefromjpeg" ReadyNet's AC1000MS

Duplicate of [[:Image:Readynet-ac1000ms.jpg]]. Original is failing to resize due to server error. "Error creating thumbnail: Incomplete GD library configuration: missing function imagecreatefromjpeg"

[[ReadyNet AC1000MS and AC1300MS|ReadyNet's AC1000MS]] is a dual-band wi-fi router with two built-in ATA FxS ports for VoIP. (Manufacturer's photo)

Poly VVX-D230

2023-09-25T15:46:18Z

Carlb:

[[Image:Poly VVX-D230 base.png|thumb|The VVX-D230's base can control ten DECT handsets]]
Aimed at small business users, the Poly (Polycom, Obahai) VVX-D230 is a DECT [[IP Phones|SIP phone]] with support for up to ten cordless handsets — including the one handset which is packaged with the base.

This system (OBAHAI/VVXD230) is intended to complement the [[Polycom VVX 300, 400, etc|VVX-series]] desk phones and the [[Poly Trio 8800]] conference room speakerphone.

==Configuration==
# If the cordless handsets are not already paired to the base, press and hold the 'find' button on the base for at least five seconds. Then, from the handset, go to Menu→Settings→Registration→Register.
# Configuration of the system is done via a web interface. From any paired handset, go to Menu→Settings→Basestation Info to obtain the IPv4 address of the base.
# Open a web browser to this address and log in with username 'admin' and default password 'admin'.

<br clear="all">
[[Image:Poly VVX-D230 Wizard.png]]

'''From the "Setup Wizard" page''', change the following parameters (changes will not be applied until you save them and reboot the device). Be sure to uncheck the "default" check box for each item you intend to modify:
*'''LocalTimeZone''' - set to your local time zone, for instance GMT-5:00 for (Eastern Time). Factory default is GMT-8:00 (Pacific Time, California).
*'''ITSP A SIPProxyServer''' - atlanta.voip.ms (or one of the other multiple [[servers]] - this must match the same server as your [[DID Troubleshooting|DID]] configuration).
*'''ITSP A DigitMap''' - (optional) (1xxxxxxxxxx|<1555>[2-9]xxxxxxS3|<1>[2-9]xxxxxxxxx|011xx.|xx.|(Mipd)|[^*#]@@.) (as an example, replacing 1-555 with your local area code) will enable both 7 and 10-digit dial for North American numbers. See the manufacturer's technical reference manual for details of the syntax.
*'''Phone1 PrimaryLine''' - "SP1 service"
*'''SP1 ITSP Profile''' - A
*'''SP1 AuthUserName''' - 123456_1 (your voip.ms user number and a [[Sub Accounts|subaccount]] number, in this example user #123456 subaccount #1
*'''SP1 AuthPassword''' - the password associated with your voip.ms subsccount
If you have additional virtual lines, repeat the same configuration steps for each subaccount using SP2, SP3... up to a maximum of eight lines and ten handsets.

<br clear="all">
[[Image:Poly VVX-D230 SP1.png]]

'''From the "SP1 Service" page''', change the following settings for the first virtual "line" (the others will be similar):
*'''X_DisplayNumber''' - A short text string identifying this line on the local handset screen. Typically the seven-digit NXX-XXXX local number, although any value will do.
*'''X_AcceptSipFromRegistrarOnly''' - YES (to keep [[Sip Scanner Ghost Calls|spurious]] calls out)
*'''X_KeepAliveEnable''' - YES (if you are behind NAT)
*'''X_KeepAliveExpires''' - 180 (three minutes) is reasonable
*'''DirectoryNumber''' - NPA-NXX-XXXX, the directory number associated with this line
*'''AuthUserName''', '''AuthPassword''' - should already be set from the "Setup Wizard" above; if not, set them here.
*'''CallerIDName''' - your name (15 alphanumeric chars max, no punctuation but spaces are allowed) as you want it to appear on [[Caller ID]] for outbound calls.
*'''X_CheckVoiceMailNumber''' - *97
*'''MessageWaiting''' - YES

Save everything before leaving each page of the setup. When you are finished making changes, reboot the device.

Try a test call, such as the [[Dialing Codes|echo test]] (press [4] [4] [4] [3] and the [green] button). You should hear your own voice played back just as it was received.

==See alos==
* [[Polycom VVX 300, 400, etc|VVX-series]]
* [[Poly Trio 8800]]

==Documentation==
This page is merely a brief overview with just enough information to connect the handsets to VoIP.ms; see the manufacturer's website (poly.com) for more extensive documentation:

* Quick start: [https://www.poly.com/content/dam/www/products/support/voice/vvx-dect-ip-phones/setup/vvx-d230-hs-bs-class-a-qsg-access.pdf handset + base], [https://www.poly.com/content/dam/www/products/support/voice/vvx-dect-ip-phones/setup/vvx-d230-hs-chg-class-a-qsg-access.pdf expansion handset + charger]
* [https://www.poly.com/content/dam/www/products/support/voice/vvx-dect-ip-phones/user/vvxd230-adminguide-710.pdf Admin guide] and [https://www.poly.com/content/dam/www/products/support/voice/vvx-dect-ip-phones/user/vvxd230-tech-ref-710.pdf technical reference] manual from Poly.com

[[Category:SIP phones]]

File:Poly VVX-D230 base.png

2023-09-25T15:39:41Z

Carlb: This image is a duplicate of :Image:Poly-vvx-d230-base.jpg, uploaded because JPG thumbnails are failing on this server with error "Error creating thumbnail: Incomplete GD library configuration: missing function imagecreatefromjpeg". Polycom (Obahai)

This image is a duplicate of [[:Image:Poly-vvx-d230-base.jpg]], uploaded because JPG thumbnails are failing on this server with error "Error creating thumbnail: Incomplete GD library configuration: missing function imagecreatefromjpeg".

Polycom (Obahai) [[Poly VVX-D230]] cordless DECT handset with base. As one base can control ten DECT cordless handsets, the other handsets in the system come with just a charging cradle, not the full base. Poly is a division of Hewlett-Packard. image is copyright to the manufacturer.

File:Poly VVX-D230 SP1.png

2023-09-25T15:33:09Z

Carlb: Configuration screen for Poly VVX-D230 DECT cordless SIP phone.

Configuration screen for [[Poly VVX-D230]] DECT cordless SIP phone.

File:Poly VVX-D230 Wizard.png

2023-09-25T15:32:45Z

Carlb: Configuration screen shot for Poly VVX-D230 DECT cordless SIP phone.

Configuration screen shot for [[Poly VVX-D230]] DECT cordless SIP phone.

Devices

2023-09-25T07:00:22Z

Carlb: /* Polycom VVX 300, 400, etc */

== Blog Articles ==

* '''VoIP – Bring Your Own Device (BYOD)''': To be able to place or receive calls using VoIP, you may need a hardware setup that will allow you to use your regular phone (ATA) or a special phone that connects directly to VoIP networks (IP Phone). For more information, Take a peek at our blog article [https://wiki.voip.ms/article/VoIP_%E2%80%93_Bring_Your_Own_Device_(BYOD) VoIP – Bring Your Own Device (BYOD)]

* '''ATA:''' An '''Analog Telephone Adapter''' 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. Still not clear? Take a peek at our blog article : [https://wiki.voip.ms/article/Back_to_Basics_-_What_is_an_ATA%3F Back to Basics - What is an ATA?]

* '''IP Phone:''' An IP Phone uses voice over IP (VoIP) technologies allowing telephone calls to be made over an IP network such as the Internet instead of the ordinary PSTN system. Still not clear? Take a peek at our blog article : [https://wiki.voip.ms/article/Back_to_Basics:_What_is_an_IP_Phone%3F Back to Basics - What is an IP Phone?]

__TOC__

==ATAs==

===Atcom AG188N===

[[File:Atcom-ag188n.jpg|300px|thumb|left|Atcom AG188N]]

'''Product:''' Atcom AG188N

'''Company:''' Atcom

'''Overview:''' AG188N voice gateway is an Internet-based one port voice gateway. AG188N ATA adapts multi-voice control protocols and voice compression codec to directly convert analog voice into IP packet for internet transport, thus effectively using the existing bandwidth to provide PSTN quality voice service.

AG188N ATA one FXS voice over IP gateway supports SIP, and IAX2 protocol, offering one 10/100Mbps Ethernet interface, one RJ11 telephone interface, one built-in router, and lifeline port. AG-188N ATA is small and can be widely used in SOHO, small office and enterprise branches.

[[Atcom AG188N|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

===Auerswald COMpact 5010===

[[File:Auerswald_5010.jpg|300px|thumb|left|Auerswald COMpact 5010]]

'''Product:''' Auerswald COMpact 5010

'''Company:''' Auerswald

'''Overview:''' Communication center for small companies and ambitious private users with ample expansion possibilities

The COMpact 5010 VoIP makes an ideal communication center for agencies, law firms and small craft businesses as well as the technology-conscious. With the according modules, it gives you 8 trunk lines (analog, ISDN, VoIP) and connections for 10 internal participants (analog, ISDN, VoIP). And if you want, our Automatic Update will always keep the operating software of your COMpact system up to speed!

[[Auerswald COMpact 5010|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

===Cisco===

====Cisco Linksys PAP2====

[[File:Pap2.jpg|300px|thumb|left|Cisco Linksys PAP2]]

'''Product:''' Linksys PAP2

'''Company:''' Cisco

'''Overview:''' The Linksys PAP2 is an analog telephony adapter (ATA), which allows for the connection of up to two analog telephones to a soft switch or VoIP carrier using Session Initiation Protocol (SIP). The physical connections on the device are two RJ11 jacks for the telephones and one RJ45 jack for ethernet. The ethernet connection is 10 Mbit/s, half-duplex. The device is configured through a web interface.
Each analog telephone is presented as a distinct SIP user.

[[Cisco Linksys PAP2|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====Cisco Linksys PAP2T====

[[File:Pap2t.jpg|300px|thumb|left|Cisco Linksys PAP2T]]

'''Product:''' Linksys PAP2T

'''Company:''' Cisco

'''Overview:''' The Cisco PAP2T Internet Phone Adapter with 2 VoIP Ports enables high-quality feature-rich VoIP service through your broadband Internet connection. Just plug it into your home router or gateway and use the two standard telephone ports to connect analog phones or fax machines. Each phone port operates independently, with separate phone service and phone numbers, like having two telephone lines. You'll get clear reception and reliable fax connections, even while using the Internet.

With Internet telephony, along with low domestic and international phone rates, impressive arrays of special telephone features are available. Choose your preferred local dialing number, regardless of where you live. Or add a virtual telephone number to have forwarded to your Internet phone. You can even add a toll-free number. The Cisco PAP2T Internet Phone Adapter is compatible with these and all of the other special telephone features that are available from your Internet telephone service provider, such as caller ID, call waiting, voicemail, call forwarding, distinctive ring, and much more.

[[Cisco Linksys PAP2T|See Configuration Details]]<br>
[[PAP2T With Static IP|See Configuration Details with Static IP]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====Cisco SPA112 and SPA122====

[[File:SPA112.jpg|300px|thumb|left|Cisco SPA112]]

'''Product:''' SPA112, SPA122

'''Company:''' Cisco

'''Overview:''' The Cisco SPA112 2 Port Adapter connects to VoIP service through a wired broadband Internet connection and provides two virtual telephone's (FxS) lines to which standard touch-tone 'phones may be connected. The SPA122 is very similar to the SPA112 but includes a second network connection, allowing it to be installed as a bridge or router.

Both included two standard telephone ports to easily connect existing analog phones or fax machines. Each phone line could be configured independently.

Introduced in late 2011 and discontinued in mid-2020, these popular boxes represented an inexpensive means to continue using existing analog hardware while migrating to voice over IP. Their successors are the (very similar) [[Cisco ATA191 and ATA192]].

[[Cisco SPA112|See Configuration Details]]

<div style="width:100%;overflow:hidden;clear:both"></div>

====Cisco WRP400 and WRP500====

[[File:WRP400.jpg|300px|thumb|left|Cisco WRP400]]

'''Product:''' WRP400, WRP500

'''Company:''' Cisco

'''Overview:''' The Cisco WRP400/WRP500 wi-fi routers connect to VoIP service through a wired broadband Internet connection and provide two virtual telephone's (FxS) lines to which standard touch-tone 'phones may be connected. These devices effectively deliver a complete local area network and a two-line analog telephone adapter in one box.

The WRP400 provided a wired LAN (four 100baseT downlink ports), plus full-featured wireless B/G networking. Its faster but more expensive successor, the WRP500, provides quicker wi-fi (802.11ac, dual-band) and wired LAN (1000baseT) connections. If wired Internet is not available, uplink for these routers can fall back to a mobile Internet connection if appropriate hardware is connected to the USB port.

The telephony capabilities of the WRP400/WRP500 are very similar to the SPA122. Both include two standard telephone ports to easily connect existing analog phones or fax machines. Each phone line can be configured independently. T.38 faxing is supported if a provider provides this capability.

[[Cisco WRP400|See Configuration Details]]

<div style="width:100%;overflow:hidden;clear:both"></div>

====Cisco SPA2100 Phone Adapter====

[[File:Sipura2100.jpg|300px|thumb|left|Cisco SPA2100 Phone Adapter]]

'''Product:''' Cisco SPA2100 Phone Adapter

'''Company:''' Cisco

'''Overview:''' Inexpensive, easy to install, and simple to use, the Cisco® SPA2100 Phone Adapter with Router (Figure 1) connects a standard telephone or fax machine to an IP-based data network. Voice over IP (VoIP) service providers can offer residential and business users traditional and enhanced communication services via the customer's broadband connection to the Internet.

The Cisco SPA2100 features two basic telephone ports to connect existing analog phones or fax machines to a private branch exchange (PBX) system. It also includes two RJ-45 Ethernet interfaces to connect to a home or office LAN, as well as an Ethernet connection to connect a broadband modem or router (WAN). Each phone line can be configured independently via software controlled by the service provider or the end user. With the SPA2100, users are able to protect and extend their investment in their existing analog telephones, conference speakerphones, and fax machines, as well as control their migration to IP with an extremely affordable, reliable solution.

[[Cisco SPA2100 Phone Adapter|See Configuration Details]]
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====Cisco SPA2102 Phone Adapter with Router====

[[File:Spa2102b.jpg|300px|thumb|left|Cisco SPA2102 Phone Adapter with Router]]

'''Product:''' Cisco SPA2102 Phone Adapter with Router

'''Company:''' Cisco

'''Overview:''' Inexpensive, easy to install, and simple to use, the Cisco SPA2102 Phone Adapter with Router connects a standard telephone or fax machine to an IP-based data network. Voice over IP (VoIP) service providers can offer residential and business users traditional and enhanced communication services via the customer's broadband connection to the Internet.

The Cisco SPA2102 features two basic telephone ports to connect existing analog phones or fax machines to a private branch exchange (PBX) system. It also includes two 100BASE-T RJ-45 Ethernet interfaces to connect to a home or office LAN, as well as an Ethernet connection to connect a broadband modem or router (WAN). Each phone line can be configured independently via software controlled by the service provider or the end user. With the SPA2102, users are able to protect and extend their investment in their existing analog telephones, conference speakerphones, and fax machines, as well as control their migration to IP with an extremely affordable, reliable solution.

[[Cisco_SPA2102_Phone_Adapter_with_Router|See Configuration Details]]
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===Grandstream===

====Grandstream HandyTone 486====

[[File:HT486.jpg|300px|thumb|left|Grandstream HandyTone 486]]

'''Product:''' Grandstream HandyTone 486

'''Company:''' Grandstream

'''Overview:''' Grandstream's award-winning HandyTone-486 is an all-in-one VoIP integrated access device that features superb audio quality, rich functionalities, high level of integration, compactness, and ultra-affordability. The HandyTone-486 is fully compatible with SIP industry standard and can interoperate with many other SIP compliant devices and software on the market.

Grandstream HandyTone-486 has been awarded the Best of Show product in 2004 Internet Telephony Conference and Expo.

[[Grandstream_HandyTone_486|See Configuration Details]]
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====Grandstream HandyTone 502 - HT502====

[[File:Ht502.jpg|300px|thumb|left|Grandstream HandyTone 502 - HT502]]

'''Product:''' Grandstream HandyTone 502 - HT502

'''Company:''' Grandstream

'''Overview:''' The HT502 is a powerful VoIP router. The product's inclusion of an integrated high-performance NAT router and 10Mbps Ethernet WAN and LAN ports enables a shared broadband connection between multiple ethernet devices.

In addition to being SIP 2.0 standard compliant, the product supports Universal Plug-in-Play (UPnP), up to 2 SIP account profiles, and advanced telephony features. It provides two lines (FxS), supporting both tone and [[pulse dial]] for compatibility with any standard telephone.

Enhanced security

Automated provisioning using symmetric and asymmetric voice

Support for a broad range of popular voice codec

[[Grandstream_HandyTone_502_-_HT502|See Configuration Details]]
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====Grandstream HandyTone 702 - HT702====

[[File:Ht702.jpg|300px|thumb|left|Grandstream HandyTone 702 - HT702]]

'''Product:''' Grandstream HandyTone 702 - HT702

'''Company:''' Grandstream

'''Overview:''' The HT702 is Grandstream´s Newest powerful VoIP router. The product's inclusion of an integrated high-performance NAT router and 10Mbps Ethernet WAN and LAN ports enables a shared broadband connection between multiple ethernet devices.

In addition to being SIP 2.0 standard compliant, the product supports Universal Plug-in-Play (UPnP), up to 2 SIP account profiles/Phones and advanced telephony features.

Enhanced Security

Automated provisioning using symmetric and asymmetric voice

Support for a broad range of popular voice codec

[[Grandstream_HandyTone_702_-_HT702|See Configuration Details]]
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====Grandstream HandyTone 802 - HT802====

[[File:HT802 Device.jpg|300px|thumb|left|Grandstream HandyTone 802 - HT802]]

'''Product:''' Grandstream HandyTone 802 - HT802

'''Company:''' Grandstream

'''Overview:''' The HT802 delivers powerful VoIP technology and routing capabilities to home and office environments, and allows users to successfully connect their analog devices to a manageable and powerful VoIP network. Built upon Grandstream’s market-leading SIP ATA/gateway technology, with millions of units successfully deployed worldwide, this powerful ATA features exceptional voice quality in various applications and environments.

The HT802 comes with 2 easy-to-use FXS ports, state-of-the-art encryption with a unique security certificate per unit, automated provisioning for volume deployment and device management and outstanding network performance.

TLS and SRTP security encryption technology to protect calls and accounts

Supports 2 SIP profiles through 2 FXS ports and a single 10/100Mbps port

Support for a broad range of popular voice codec

[[Grandstream HandyTone 802 - HT802|See Configuration Details]]
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===Mediatrix C7 and Mediatrix 4100 Series===

[[File:Mediatrix_C7.jpg|300px|thumb|left|Mediatrix C7]]
[[File:Mediatrix_4100_1_media5.jpg|300px|thumb|left|Mediatrix 4102]]

'''Product:''' : Mediatrix C7 and Mediatrix 4100 Series running DGW 2.0 firmware

'''Company:''' Media5 Corporation

'''Overview:''' The Mediatrix® C7 and Mediatrix® 4100 Series feature VoIP adaptors that interconnect up to 24 analog telephones, faxes, and modems ports into SIP-based systems. These series allow operators to shorten deployment time to enable cloud telephony services into branch offices and SMBs.

The DGW 2.0 firmware also offers security features such as SIP over TLS, SRTP, certificates management, and HTTPS designed to bring enhanced security for the network management, SIP signaling, and media transmission aspects.

[[Media5 Mediatrix C7 and 4100|See Configuration Details]]
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===Netgear WGR615V===

[[File:NetgearWGR615V.jpg|300px|thumb|left|Netgear WGR615V]]

'''Product:''' WGR615V

'''Company:''' Netgear

'''Overview:''' The WGR615V was never sold directly to the public when it was new, but since the model is discontinued, lots of them are being sold.
It is a wireless (802.11 G) router that happens to have a built-in ATA.
It is possible to use just the ATA part.

[[Netgear WGR615V|See Configuration Details]]
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=== OBiHAI (Polycom) ===

====OBi 100/110 & OBi 200====

[[File:OBi110-ATA.jpg|300px|thumb|left|OBi110]]

'''Product:''' OBi 100/110 & OBi 200

'''Company:''' Polycom (OBIHAI Technology Inc)

'''Overview:''' With OBi devices you are in control of your communications life. From the OBi's on-board connections as well as via the Internet, you have the power to bridge mobile, fixed line and Internet telephone services. The OBi100 & OBi110 are stand-alone, dedicated devices, built with a high-performance "system on a chip" platform to ensure high-quality voice conversations. Every OBi device has high availability and reliability. It is 'always-on' to make or receive a call. With an OBi device, a computer is not required and a computer does not need to be on to talk on the phone or use the OBi's powerful service bridging features. To get started, all you need is a phone, power and a connection to the Internet.

The OBi110 has one phone port and one line port. Use the OBi110 when you need an analog line connection to a traditional telephone network or service. Think of the OBi110 LINE port as a gateway to traditional phone service. If you do not have traditional phone service at home, then the OBi100 is probably the right product to get.

[[OBi 100/110 & OBi 200|See Configuration Details]]
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====Polycom OBi300====

[[File:obi300.png|300px|thumb|left|Polycom OBi300]]

'''Product:''' Polycom OBi300

'''Company:''' Polycom

'''Overview:''' Polycom OBi300 expands your service portfolio by enhancing the communication possibilities of home offices with flexibility in voice, fax and modem applications as they transition to the digital communications world.
* Flexible voice and fax applications
* Help home offices maximize current analog investment by keeping one analog phone or fax machine and connect them to VoIP services
* Supports T.38 fax standard so home office users can send or receive reliable facsimile calls over the Internet

[[OBi300|See Configuration Details]]
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===ReadyNet AC1000MS and AC1300MS===

[[File:readynet-ac1000ms.jpg|300px|thumb|left|ReadyNet AC1000MS]]

'''Product:''' ReadyNet AC1000MS (two lines) or AC1300MS (one line)

'''Company:''' ReadyNet Solutions (Phonex Broadband Corporation)

'''Overview:''' The ReadyNet AC1000MS is a full-featured 1200 megabit-per-second dual-band Wi-Fi router with a built in two-line SIP ATA, eliminating the need for two separate boxes. Both lines may be configured independently.

[[ReadyNet AC1000MS and AC1300MS|See Configuration Details]]

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===Telco AC-211===

[[File:Ac211n.jpg|300px|thumb|left|Telco AC-211]]

'''Product:''' Telco AC-211

'''Company:''' Telco

'''Overview:''' Telco AC-211 is a SIP ATA supporting two lines and numerous configuration options. This ATA is most commonly found as the AC-211-SR & AC-211N from the now-defunct SunRocket service. This device works well with VoIP.ms once configured.

[[Telco_AC-211|See Configuration Details]]

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===TP-Link TD-VG3631===

[[File:Tplink vg3631 front.png|300px|thumb|left|TPLink VG3631]]

'''Product:''' TP-Link TD-VG3631

'''Company:''' TP-Link

'''Overview:''' High-speed DSL modem, NAT router, and wireless access point in one device providing a one-stop networking solution Wireless N speed up to 300Mbps makes it ideal for heavy bandwidth consuming or interruption sensitive applications like online gaming, Internet calls and even the HD video streaming Supporting both traditional landlines and VoIP network offers you a multiple choice when making phone calls. Various call features such as caller ID, call waiting, call holding, call forwarding, 3-way conference calls and voicemail USB 2.0 port supports Storage Sharing (Samba or FTP), Media Server, Print Server and multi accounts Ethernet WAN (EWAN) offers another broadband connectivity option for connecting to Cable, VDSL or Fiber modems IP based Bandwidth Control makes it easier for you to manage the bandwidth of devices that connected to the modem router Wi-Fi On/Off Button allows users to simply turn their wireless radio on or off.

[[TP-Link TD-VG3631|See Configuration Details]]
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==IP Paging, Speakers, Stobe Lights==

===Algo Technologies SIP endpoints===

[[File:AlgoTechnologies.jpg|300px|thumb|left|Algo Technologies SIP endpoints]]

'''Product:''' IP speakers, paging adapters, strobe lights, clocks, push buttons, doorphones/intercoms, and more

'''Company:''' Algo Technologies SIP endpoints

'''Overview:''' Algo is a telecommunications manufacturer of endpoints and accessories including IP speakers, paging adapters, strobe lights, clocks, push buttons, doorphones / intercoms, and specialty handsets (PTT, PTM).

[[Algo_Technologies_SIP_endpoints|See Configuration Details]]
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==IP Phones==

===3COM 3108 Wireless Phone===

[[File:3com_3108_1.jpg|300px|thumb|left|3COM 3108 Wireless Phone]]

'''Product:''' 3COM 3108 Wireless Phone

'''Company:''' 3COM

'''Overview:''' The 3Com 3108 Wireless Telephone is a Session Initiation Protocol (SIP)-based wireless Voice over Internet Protocol (VoIP) telephone. SIP is an internationally recognized standard <nowiki>(IETF RFC 3261)</nowiki> for implementing VoIP. You can make and receive VoIP calls as long as your Wireless Telephone is registered with a SIP proxy server and you are operating it within range of an IEEE 802.11b/g enabled wireless network (WLAN). The SIP proxy server can belong to a wireless Internet Telephony Service Provider (ITSP) or corporate VoIP PBX system, such as the 3Com NBX® System.

[[3COM_3108_Wireless_Phone|See Configuration Details]]
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===Aastra 6730i/6731i VoIP Phone===

[[File:Aastra6730i.jpg|300px|frame|left|Aastra 6730i VoIP Phone]]

'''Product:''' Aastra 6730i/6731i VoIP Phone

'''Company:''' Aastra

'''Overview:'''The Aastra 6730i, a new member of the carrier-grade, open-standards-based 67xi SIP portfolio, offers exceptional features and flexibility in an enterprise grade IP telephone. With a sleek, elegant design and a compact footprint, this multi-line SIP telephone delivers the advanced features and performance traditionally found only in higher priced products. Featuring a 3 line LCD display, the 6730i supports up to 6 lines with call appearances, offers advanced XML capability to access custom applications and is fully interoperable with leading IP-PBX platforms.

Supported by a host of Aastra configuration options, XML development tools, and continuous product enhancements via software releases, the value offered by the 6730i makes it is ideally suited for light telephone use for the small and home-based business applications.

[[Aastra_6730i_VoIP_Phone|See Configuration Details]]
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===Audiocodes===

====400HD Series====

[[File:Audiocodes 420HD.jpg|300px|thumb|left|Audiocodes 420HD IP Phone]]

'''Product:''' Audiocodes 400HD Series

'''Company:''' Audiocodes

'''Overview:''' The [https://www.audiocodes.com/solutions-products/products/ip-phones AudioCodes 400HD series] of IP phones is a range of easy-to-use, feature-rich desktop devices for the service provider hosted services, enterprise IP telephony and contact center markets. Based on the same advanced, field-proven underlying technology as our other VoIP products, AudioCodes high quality IP phones enable systems integrators and end customers to build end-to-end VoIP solutions.

[[Audiocodes 400HD|See Configuration Details]]
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===Cisco===

====Cisco 88XX & 68XX series====

[[File:8800_Series.png|300px|thumb|left|Cisco 8800 Series]]

'''Product:''' 88XX & 68XX series

'''Company:''' Cisco

'''Overview:''' The ''Cisco IP Phone 6800'' Series multiplatform phones are designed for affordability. They deliver reliable, business-grade audio, with Gigabit Ethernet integration and low power usage.

Ideal for customers with moderate to active VoIP needs, the 6800 Series phones are supported on Cisco-approved third-party unified communications as a service (UCaaS) providers.

The ''Cisco IP Phone 8800'' Series is a great fit for businesses of all sizes seeking secure, high-quality, full-featured VoIP. Select models provide affordable entry to HD video and support for highly-active, in-campus mobile workers. This advanced series provides flexible deployment options: on-premises, cloud and Cisco pre-approved third-party UCaaS providers.

[[Cisco IP Phone 68XX and 88XX|See Configuration Details]]
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====Cisco Linksys SPA942 NA====

[[File:SPA942-0.jpg|300px|thumb|left|Cisco Linksys SPA942 NA]]

'''Product:''' Linksys SPA942 NA

'''Company:''' Cisco

'''Overview:''' Four active lines and dual switched Ethernet ports offer incredible flexibility to the home or small-business Internet telephony user. Part of Cisco Small Business IP Phone Series, the SPA942 IP Phone can be configured to carry a unique phone number or extension for each of its lines or can be configured to use a shared number over multiple phones.

Based on the SIP standard, the SPA942 is able to easily integrate new features and services offered by providers. A high-resolution display makes for an easy menu- and web-based configuration.

[[Cisco_Linksys_SPA942_NA|See Configuration Details]]
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====Cisco SPA525G====

[[File:525g.jpg|300px|thumb|left|Cisco SPA525g Phone Adapter with Router]]

'''Product:''' Cisco SPA525G

'''Company:''' Cisco

'''Overview:''' The Cisco SPA525G 5-line IP Phone with Color Display is a full-featured VoIP (Voice over Internet Protocol) phone that provides voice communication over an IP network. It provides traditional features, such as call forwarding, redialing, speed dialing, transferring calls, conference calling, and accessing voice mail. Calls can be made or received with a handset, headset or speaker.
Your Cisco IP Phone provides a web interface for the phone user that allows you to configure some features of your phone by using a web browser.
This article will guide you through the steps for basic configuration to make it work with VoIP.ms

[[Cisco SPA525G|See Configuration Details]]
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====Cisco IP Phone 7940/7960====

[[File:Cisco_7960g.png|300px|thumb|left|Cisco IP Phone 7940/7960]]

'''Product:''' Cisco IP Phone 7940/7960

'''Company:''' Cisco

'''Overview:''' The Cisco IP Phone 7940/7960 is a full-featured telephone that provides voice communication over an IP network. This phone functions as a traditional analog phone, allowing you to place and receive telephone calls. This phone also supports features, such as network call forwarding, redialing, speed dialing, transferring calls, placing conference calls, and accessing voice mail. Phones require Power Over Ethernet (PoE) or a 48V AC Adapter to power up.

[[Cisco_IP_Phone_7940/7960|See Configuration Details]]
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====Cisco SPA30x and SPA50x series IP phones====

[[File:Cisco_spa504g.jpg|300px|thumb|left|Cisco SPA504G Phone]]

'''Product:''' Cisco SPA504G Phone

'''Company:''' Cisco

'''Overview:'''
The Cisco SPA504G is an office-style desk telephone with built-in voice over the Internet.

It is one in a series of similar models (SPA30x and SPA50x) which vary primarily in the number of lines (extensions) on the 'phone, power source (some models use power-over-Ethernet) and the availability of a second Ethernet connector. These devices are well-suited to offices and IP PBX applications. These do not provide a virtual line for connecting analog devices such as standard telephone handsets; they are instead self-contained to connect directly to VoIP.

[[Cisco_SPA504G_Phone|See Configuration Details]]
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=== Dinstar===

==== Dinstar C60 Series ====

[[File:C60U_F.png|300px|thumb|left|Dinstar C60UP]]

C60 series are based on high innovative SIP technology, which is ideal for all kinds of business communication. It integrates with 132x64-pixel graphical LCD with back-light, elegant and intuitive user interface, which indicate you can enjoy good user experience.

[[Dinstar_C60_series|See Configuration Details]]
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===Fanvil ===

====Fanvil X4G====

[[File:FanfillX4g.jpg|300px|thumb|left|Fanvil X4G]]

'''Product:''' Fanvil X4G

'''Company:''' Fanvil

'''Overview:''' The Fanvil X4G has a 2.8" main color screen and a secondary 2.4" DSS color screen. The user interface is sleek, colorful and easy to navigate. It has a one button call function and a call log and the ability to store 500 phonebook entries. The X4G's high compatibility supports various systems including 3CX, Avaya, OpenVox, NEC, Elastix, Asterisk, Matrix, Broadsoft, Epygi and more.

[[Fanvill X4G|See Configuration Details]]
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[[Category:Devices]]

=== Flyingvoice ===

====Flyingvoice ====

[[File:Flyingvoice_Phones.jpg|300px|thumb|left|Flyingvoice]]

'''Product:''' '''FIP11C / FIP11CP''', '''FIP13G''', '''FIP14G''', '''FIP15G''', '''FIP10/FIP10P''', '''FIP12WP''', '''FIP16'''

'''Company:''' Flyingvoice

'''Overview:''' Flyingvoice is a provider of communication terminal equipment and one-stop VoIP CPE solutions. They offer a full range of VoIP products, such as VoIP phones, ATAs, gateways and routers for businesses and consumers. Their WiFI IP phones offer a wireless option, so you do not need a wired internet connection to either make or receive VoIP calls.

[[Flyingvoice|See Configuration Details]]
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[[Category:Devices]]

===Fortinet===

====FortiFone FON-570====

[[File:FON-570_Phone.jpg|300px|thumb|left|Fortinet FortiFone FON-570]]

'''Product:''' FortiFone FON-570

'''Company:''' Fortinet

'''Overview:''' Featuring a large 7” color touchscreen and premium HD call quality, this IP phone is great for efficient communications. Combined dedicated feature keys and programable keys expandable to 109, you have the flexibility to control your calls within your fingertips.

*7" color screen
*7 dedicated feature keys
*109 programable phone keys
*Full duplex speakerphone
*2 x 10/100/1000 Ethernet ports
*Integrated Power over Ethernet (PoE) support

[[Fortinet_FortiFone_FON-570|See Configuration Details]]
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====FortiFone FON-375====

[[File:FON-375_Phone.jpg|300px|thumb|left|Fortinet FortiFone FON-375]]

'''Product:''' FortiFone FON-375

'''Company:''' Fortinet

'''Overview:''' A reliable IP phone delivers HD sound quality, ideal for office workers who need efficient communications. An easy-to-read color screen and a programable second screen make it easy to display which lines are in use and who is on a call.

:*Dual color screens: 2.8" + 2.4”
:*8 dedicated feature keys
:*30 programable phone keys
:*Full duplex speakerphone
:*2 x 10/100/1000 Ethernet ports
:*Integrated Power over Ethernet (PoE) support

[[Fortinet_FortiFone_FON-375|See Configuration Details]]
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====FortiFone FON-175====

[[File:FON-175_Phone.jpg|300px|thumb|left|Fortinet FortiFone FON-175]]

'''Product:''' FortiFone FON-175

'''Company:''' Fortinet

'''Overview:''' A quality, two-line IP phone delivers reliable communications with HD audio quality. This entry-level business phone is easy to use that works in any office.

*2.4" color screen
*5 dedicated feature keys
*Full duplex speakerphone
*2 x 10/100 Ethernet ports
*Integrated Power over Ethernet (PoE) support

[[Fortinet_FortiFone_FON-175|See Configuration Details]]
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===Gigaset A510 IP===

[[File:Gigaset_a510_IP.jpg#file|300px|thumb|left]]

'''Product:'''Gigaset A510 IP

'''Company:'''Gigaset

'''Overview:'''The Gigaset A510 and C610 IP phones are fitting solutions if you are looking for the flexibility of VoIP and the convenience of using a cordless handset.

[[Gigaset_A510_IP| See Configuration Details]]
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===Grandstream===

====Grandstream DP715/DP710====

[[File:Grandstream 715-710.png|300px|thumb|left]]

'''Product:''' Grandstream DP715/DP710

'''Company:''' Grandstream

'''Overview:''' The DP715/710 is the next generation of powerful, affordable, high quality and simple to configure DECT Cordless IPPhone for small business and residential users. Their compact size, superb voice quality, rich feature set, market leading price-performance and wide range radio coverage enable consumers to maximize the power of IP voice application and mobility for a minimum investment.

[[Grandstream_DP715/DP710| See Configuration Details]]
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====Grandstream DP750/DP720====

[[File:DP750-720.png|300px|thumb|left]]

'''Product:''' Grandstream DP750/DP720

'''Company:''' Grandstream

'''Overview:''' The DP750/720 base station and handsets allows you to deploy an immersive DECT environment that allows users to communicate free from their desktop using Grandstream’s DP720 DECT handsets. The DP750 pairs with up to 5 DP720s to create a powerful and mobile network solution with up to 10 lines per handset, and 5 concurrent calls per DECT system.

[[Grandstream_DP750/DP720| See Configuration Details]]
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====Grandstream WP810====
[[File:Grandstream WP810.png|150px|thumb|left]]

'''Product:''' Grandstream WP810

'''Company:''' Grandstream

'''Overview:''' The WP810, WP820, WP822 and WP825 are cordless SIP IP phones which connect directly to the local network using dual-band Wi-Fi, eliminating the need for a cordless base station or other hardware. Effectively, the WP810 (with its charging cradle) is a self-contained two-line SIP solution.

[[Grandstream WP810, WP820, WP822 and WP825|See Configuration Details]]

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====Grandstream GXP1630 IP Phone====

[[File:Gxp1630.jpg|300px|thumb|left|Grandstream GXP2120 IP Phone]]

'''Product:''' Grandstream GXP1630 IP Phone

'''Company:''' Grandstream

'''Overview:''' The GXP1630 comes equipped with a suite of VoIP features that are deployed in a clear and easy-to-use fashion. Focused primarily for low to medium call volumes and efficient call handling, its 3 line/SIP account design and 8 dual-colored BLF/speed dial keys gives this versatility. The GXP1630 also supports the best possible connection speeds and call quality with its dual Gigabit ports and HD audio on both speakerphone and handset. With other features such as its integrated PoE, 3 XML programmable soft keys and 4-way conferencing support, the GXP1630 is a high-quality and versatile Basic IP phone.

[[Grandstream_GXP1630|See Configuration Details]]
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====Grandstream GXP2120 IP Phone====

[[File:Gxp2110.png|300px|thumb|left|Grandstream GXP2120 IP Phone]]

'''Product:''' Grandstream GXP2120 IP Phone

'''Company:''' Grandstream

'''Overview:''' The Grandstream GXP2120 is a 6 line SIP Phone which features HD Voice hardware and software support and a large 320 x 160 backlit graphical LCD. The GXP2120 can handle 6 SIP accounts represented by 6 dual-color line keys and 4 XML programmable context-sensitive soft keys. In addition, the GXP2120 has 7 dual-color BLF extension keys for the most common calls and transfers making it an ideal phone for an office user with moderate to heavy interoffice calling needs.

[[Grandstream_GXP2120_IP_Phone|See Configuration Details]]
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====Grandstream GXP2135 IP Phone====

[[File:GXP2135-device.jpg|300px|thumb|left|Grandstream GXP2135 IP Phone]]

'''Product:''' Grandstream GXP2135 IP Phone

'''Company:''' Grandstream

'''Overview:''' The GXP2135 is the ideal selection for busy users who value call control, productivity and usability, and manage medium to heavy call volumes. Equipped with 8 lines and 4 SIP accounts, a 2.8-inch color LCD display, and 32 digital speed dial/BLF keys, the GXP2135 enables quick and powerful usability.

As all Grandstream IP phones do, the GXP2135 features state-of-the-art security encryption technology (SRTP and TLS). The GXP2135 supports a variety of automated provisioning options, including zero-configuration with Grandstream’s UCM series IP PBXs, encrypted XML files and TR-069, to make mass deployment extremely easy.

[[Grandstream GXP2135|See Configuration Details]]
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====Grandstream GXP2170====

[[File:Grandstream_GXP2170.png|300px|thumb|left|Grandstream GXP2170]]

'''Product:''' Grandstream GXP2170

'''Company:''' Grandstream

'''Overview:''' The GXP2170 is a powerful High-End IP phone that is ideal for busy users who handle high call volumes. Receptionists, administrators, sales staff and other call-intensive rolls can enjoy efficiency by utilizing the GXP2170’s 12 line keys, 4.3 inch color display LCD and 48 digital, on-screen speed dial/BLF keys.

[[Grandstream_GXP2170|See Configuration Details]]
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====Grandstream GXP2200====

[[File:Grandstream_GXP2200.png|300px|thumb|left|Grandstream GXP2200]]

'''Product:''' Grandstream GXP2200

'''Company:''' Grandstream

'''Overview:''' The GXP2200 is one of the most advanced AndroidTM desktop IP phones available on the market today. The innovative phone includes the AndroidTM version 2.3 operating system with a 4.3 inch capacitive touchscreen LCD and the ability to host 6 SIP accounts. Web applications such as news, social media sites, and games can be downloaded directly via Google Play Store, and applications can be created to fit any need and downloaded directly to the phone for customized use.

[[Grandstream_GXP2200|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

===Innomedia===

====BuddyTalk BT110====

[[File:Buddy_Talk_BT110_Innomedia.jpg|250px|thumb|left|Innomedia BuddyTalk BT110]]

'''Product:''' BuddyTalk BT110

'''Company:''' Innomedia

'''Overview:''' Powered by the Amazon Alexa Voice Service (AVS), the BuddyTalk series of products supports a broad suite of AVS enabled smart speaker features.
Equipped with advanced audio processing and VoIP technologies, BuddyTalk devices are intelligent speakerphones that deliver unprecedented flexibility in calling, superior voice quality, and high levels of security.

[[BuddyTalk_-_BT110|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

===Konftel===
==== Konftel 300Wx IP ====
[[File:Konftel-300Wx-IP.png|300px|thumb|left|Konftel 300Wx IP]]

'''Product:''' Konftel 300Wx IP

'''Company:''' Konftel

'''Overview:''' The Konftel 300Wx IP wireless conference phone allows you to hold conference calls in HD quality wherever is convenient for you – without worrying about network and power outlets. Reliable and secure DECT technology. The accompanying IP DECT base can handle up to 20 registered Konftel 300Wx devices and five ongoing calls.

The rechargeable battery ensures more than 60 hours of call time, so you can talk for a full working week without recharging! A USB port makes the Konftel 300Wx ready for all the apps and services we use to communicate and collaborate via computers. Combine meeting apps and regular phone calls. OmniSound® delivers superb sound quality

[[Konftel 300Wx|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==== Konftel 300IPx ====

[[File:Konftel300ipx-conference-phone.jpg|300px|thumb|left|Konftel 300IPx]]

'''Product:''' Konftel 300IPx

'''Company:''' Konftel

'''Overview:''' The [https://www.konftel.com/en/products/konftel-300ipx KONFTEL 300IPx] together with the Konftel Unite app brings a whole new easiness to conference calls. It is highly intuitive and based on our natural mobile behavior. The new generation of IP conference phone is – The Art of Easiness.

[[Konftel 300IPx|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

===Panasonic===

====Panasonic KX-TGP 550====

[[File:Pana550b.jpg|300px|thumb|left|Panasonic KX-TGP 550]]

'''Product:''' Panasonic KX-TGP 550

'''Company:''' Panasonic

'''Overview:''' The Panasonic KX-TGP550 responses to the needs of SIP IP-Centrix/Hosted PBX systems and Asterisk users. Conveniently, no need to set up a system telephone at every base. This system also enables you to use a range of convenient services provided by the carrier such as Call Forward, Voice Mail, etc.

*Support for 3 simultaneous network conversations
*Up to 6 DECT cordless handsets
*Support for up to 8 SIP registrations (e.g. up to 8 DID lines or extensions)
*Hands-free speakerphone on a cordless handset

[[Panasonic KX-TGP 550|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====Panasonic KX-HDV130C====

[[File:Panasonic_KXHDV130_01.jpg|300px|thumb|left|Panasonic KX-HDV130C]]

'''Product:''' Panasonic KX-HDV130C

'''Company:''' Panasonic

'''Overview:''' The Panasonic KX-HDV130 SIP desk phone delivers the ideal balance of low cost and high quality, along a range of value added features.

*Support for up to 2 SIP registrations (e.g. up to 2 DID lines or extensions)
*Support for 3 simultaneous network conversations (3-way conferencing)*
*2 Programmable keys / Line keys
*Hands-free speakerphone on a cordless handset

[[Panasonic KX-HDV130C|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====Panasonic KX-HDV230====

[[File:Panasonic_KXHDV230_01.jpg|300px|thumb|left|Panasonic KX-HDV230]]

'''Product:''' Panasonic KX-HDV230

'''Company:''' Panasonic

'''Overview:''' The Panasonic KX-HDV230 IP phone offers streamlined functions and the high definition voice quality that's essential for effective communication.

*Support for up to 6 SIP registrations (e.g. up to 6 DID lines or extensions)
*24 Flexible Function Keys
*2 ethernet ports 10/100/1000 Base -T
*Large LCD with Backlight

[[Panasonic KX-HDV130C|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====Panasonic KX-HDV330====

[[File:Panasonic_KXHDV330_01.jpg|300px|thumb|left|Panasonic KX-HDV330]]

'''Product:''' Panasonic KX-HDV330

'''Company:''' Panasonic

'''Overview:''' The Panasonic KX-HDV330 is a multi-functional business SIP phone equipped with a colour touch panel for intuitive operation.

*Large LCD with Backlight
*Built-in Bluetooth®
*Support for up to 12 SIP registrations (e.g. up to 12 DID lines or extensions)
*24 Flexible Function Keys
*Hands-free speakerphone on a cordless handset

[[Panasonic KX-HDV130C|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

===Pirelli DP-L10===

[[File:Pirelli1.jpg|300px|thumb|left|Pirelli DP-L10]]

'''Product:''' Pirelli DP-L10

'''Company:''' Pirelli

'''Overview:''' Pirelli's DualPhone DP-L10 phone combines GSM tri-band capabilities with SIP-based Wireless VoIP via an integrated WLAN 802.11b/g interface, combining cell phone technology with the added advantage of transporting phone calls over the Internet.

The Dual Phone enables mobile phone features such as Internet browsing, e-mail, SMS, and MMS, which can also be used through a fixed-broadband connection, providing the same mobile services with the added benefits of greater bandwidth and cost savings.

[[Pirelli DP-L10|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

===Polycom===

====Poly Edge B10====

[[File:PolyEdgeB10.png|300px|thumb|left|Poly Edge B10]]

'''Product:''' Poly Edge B10

'''Company:''' Poly

'''Overview:''' Sleek design meets Poly pro-grade audio at a shockingly affordable price: That’s what makes the Poly Edge B Series IP Phones the genius choice for any growing business. Easy to use with illuminated keys where you need them. Plug-and-play provisioning and hardcore reliability make it pure value in a low-cost business phone.

[[Poly_Edge_B10|See Configuration Details]]

<div style="width:100%;overflow:hidden;clear:both"></div>

====Polycom SoundStation IP 4000 Conference Phone====

[[File:Polycom-4000-0.gif|300px|thumb|left|Polycom SoundStation IP 4000 Conference Phone]]

'''Product:''' Polycom SoundStation IP 4000 Conference Phone

'''Company:''' Polycom

'''Overview:''' the SoundStation IP 4000 SIP voice conferencing unit is the answer for organizations that are ready for the
benefits and versatility of a SIP enabled business. Designed for offices or small to medium-sized conference rooms, the SoundStation IP 4000 SIP provides remarkable room coverage. You can speak naturally from up to 10-feet away from a microphone and still be heard clearly on the far end of the call. Need coverage for a larger room? The optional extension microphones offer an increased pickup for larger rooms. Plus, with gated microphone technology, echo and background noise is almost entirely eliminated.

The SoundStation IP 4000 SIP also provides familiar features that are easy to use. The menu-driven user interface, viewable on a high-resolution backlit LCD, offers convenient access to business telephony functions you use most including transfer, hold, redial, and conference.

[[Polycom_SoundStation_IP_4000_Conference_Phone|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====Polycom SoundPoint IP 501, 550, 650, etc.====

[[File:Polycom501.jpg|258px|thumb|left|Polycom SoundPoint IP 501]]

'''Product:''' Polycom SoundPoint IP 501, 550, 650, etc.

'''Company:''' Polycom

'''Overview:''' The SoundPoint IP series are multi-line (3-6 lines depending on model) Voice over IP telephones that seamlessly integrate with IP PBX and softswitch vendors’ IP solutions.

As protocols develop and standards evolve, it’s easy to update the phone in the field via a software download, thus enabling new features and functionality for the phone and protecting your investment.

Whether you deploy MGCP or SIP standards, Polycom offers a solution that fits all your business communication needs.

[[Polycom_SoundPoint_IP_501|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====Polycom SoundPoint IP 601====

[[File:Voipms-polycom601.jpg|258px|thumb|left|Polycom SoundPoint IP 601]]

'''Product:''' Polycom SoundPoint IP 601

'''Company:''' Polycom

'''Overview:''' The 6-line Polycom® SoundPoint IP™ 601 offers industry-leading functionality and call handling unmatched voice quality an intuitive user interface & expandability to 12 lines!

[[Polycom_SoundPoint_IP_601|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====Poly VVX-D230====

[[File:Poly-vvx-d230-base.jpg|250px|thumb|left|Poly VVX-D230 cordless base]]

'''Product:''' Poly VVX-D230 cordless DECT phone

'''Company:''' Poly (Obahai, Polycom)

'''Overview:''' The VVX series are SIP desk phones aimed at small to mid-sized business; the VVX-D230 is a cordless system on which one base may control up to ten handsets.

[[Poly VVX-D230|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====Polycom VVX 300, 400, etc====

[[File:Vvx300.png|250px|thumb|left|Polycom VVX 300 Series]]

'''Product:''' Polycom VVX Series

'''Company:''' Poly

'''Overview:''' The VVX series provides high-quality audio (HD Voice) and video communications from 6 lines and up.

[[Polycom VVX 300, 400, etc|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

=== Positron IP phones ===

[[File:PositronLogo.jpeg|250px|thumb|left|Positron IP phones]]

'''Product:''' Positron

'''Company:''' [http://www.positrontelecom.com/products.html Positron]

'''Overview:''' Positron IP Phones is an affordable next-generation SIP phone including wideband audio support, ethernet ports and integrated Power over Ethernet.

All the IP Phones are optimized for executives, administrative assistants and those working with bandwidth-intensive applications on collocated PCs. Line keys can also be configured as IP PBX features such as BLF, SCA, Intercom, Call Pickup, Call Park, and many others. The high-resolution screen and the HD voice features provide a high quality visual and authentic audio experience.

==== Positron IP304 ====

[[File:IP304.png |250px|thumb|left|Positron IP304]]

'''Product:''' Positron IP304

'''Company:''' [http://www.positrontelecom.com/products.html Positron]

'''Overview:''' Positron IP304 is an affordable next-generation SIP phone with wideband audio support, dual Ethernet port and integrated Power over Ethernet.

The IP304 enterprise VoIP phone is Positron’s entry-level phone with 3 VoIP accounts. Its functionalities make it easy to place calls on hold, to transfer calls and to initiate a conference call.

With HD audio for both headset and speakerphone, IP304 provides a clear authentic audio experience

[[Positron IP304 | View configuration for Positron IP304]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==== Positron IP304C ====

[[File:IP304C.png |250px|thumb|left|Positron IP304C]]

'''Product:''' Positron IP304C

'''Company:''' [http://www.positrontelecom.com/products.html Positron]

'''Overview:''' Positron IP304C is an innovative enterprise-level IP Phone that features 4 line keys, color display, 3.5” TFT-LCD with 480 x 320 pixel. It supports up to a 5-way conference.

IP304C is optimized for executives, administrative assistants and those working with bandwidth-intensive applications on collocated PCs. Line keys can also be configured as IP PBX features such as BLF, SCA, Intercom, Call Pickup, Call Park, and many others.

The high-resolution TFT-LCD screen and the HD voice features provide a high quality visual and authentic audio experience.

[[Positron IP304C | View configuration for Positron IP304C]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==== Positron IP408 ====

[[File:IP408.png |250px|thumb|left|Positron IP408]]

'''Product:''' Positron IP408

'''Company:''' [http://www.positrontelecom.com/products.html Positron]

'''Overview:''' Positron IP408 is an affordable next-generation SIP 2.0 phone including wideband audio support and WAN/LAN Ethernet ports with route and bridge mode.

The IP408 enterprise VoIP phone supports 4 SIP lines. Its functionalities make it easy to place calls on hold, to transfer calls and to initiate a conference call.

With HD audio for both headset and speakerphone, IP408 provides a clear authentic audio experience

[[Positron IP408 | View configuration for Positron IP408]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==== Positron IP410C ====

[[File:IP410C.png |250px|thumb|left|Positron IP410C]]

'''Product:''' Positron IP410C

'''Company:''' [http://www.positrontelecom.com/products.html Positron]

'''Overview:''' Positron IP410C is an affordable next-generation SIP Phone that features 4 line keys, 10 programmable extension keys, color display, wideband audio support and dual Ethernet port with integrated Power over Ethernet.

IP410C is optimized for executives, administrative assistants and those working with bandwidth-intensive applications on collocated PCs. Line keys can also be configured as IP PBX features such as BLF, SCA, Intercom, Call Pickup, Call Park, and many others.

The high-resolution TFT-LCD screen and the HD voice features provide a high quality visual and authentic audio experience.

[[Positron IP410C | View configuration for Positron IP410C]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==== Positron IP410G ====

[[File:IP410G.png |250px|thumb|left|Positron IP410G]]

'''Product:''' Positron IP410G

'''Company:''' [http://www.positrontelecom.com/products.html Positron]

'''Overview:''' Positron IP410G is an innovative enterprise-level color IP Phone that features 4 line keys, 10 programmable extension keys, color display, 3.5” TFT-LCD with 480*320 pixel, wideband audio support and dual Gigabit Ethernet port with integrated Power over Ethernet.

IP410G is optimized for executives, administrative assistants and those working with bandwidth-intensive applications on collocated PCs. Ten programmable keys can also be configured as IP PBX features such as BLF, SCA, Intercom, Call Pickup, Call Park, and many others.

The high-resolution TFT-LCD screen and the HD voice features provide a high quality visual and authentic audio experience.

[[Positron IP410G | View configuration for Positron IP410G]]
<div style="width:100%;overflow:hidden;clear:both"></div>

===Siemens Gigaset C450-Ip===

[[File:Siemens_gigaset_c450_IP.jpg|300px|thumb|left|Siemens Gigaset C450-Ip]]

'''Product:''' Siemens Gigaset C450-Ip

'''Company:''' Siemens

'''Overview:''' The Gigaset C450 is an entry phone in the Siemens Gigaset product line, the addition of IP connectivity via a standard Ethernet 10/100 BaseT interface and SIP protocol within a reasonable price bracket, opens large applications where wireless operation and SIP are required.
The Ethernet connectivity allows the unit to work without a PC.

The dual mode (SIP/PSTN) enables incoming call on a legacy interface with existing directory number and safe emergency outgoing calls.

The presentation of the unit with an independent base and a dedicated phone charger/support will simplify cabling.

[[Siemens Gigaset C450-Ip|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

===Snom===

====Snom 320 VoIP Phone====

[[File:Snom320.png|300px|frame|left|Snom 320 VoIP Phone]]

'''Product:''' Snom 320 VoIP Phone

'''Company:''' Snom

'''Overview:'''Ideal for the office and everyone who spends a lot of time on the phone, the snom 320 is an affordable, yet powerful SIP business phone with a built-in, full-duplex speakerphone and three-party conference bridging.

[[SNOM 320|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====Snom m3 VoIP Phone====

[[File:Snom_m3.png|300px|thumb|left|Snom m3 VoIP Phone]]

'''Product:''' Snom m3 VoIP Phone

'''Company:''' Snom

'''Overview:''' The snom m9 is the next generation DECT handheld (CAT-iq) that empowers users with the convenience of wireless communication along with the widely accepted benefits and feature richness of Voice-over-IP telephony.

The snom m9 promises to deliver excellent speech quality through digital and wideband audio (klarVOICE).

By combining professional functions of versatile business communication with the intuitive features of the mobile-carrier world, the snom m9 is ideally suited for professional and private use alike. With features such as hands-free mode, calling line identification (CLI) by displaying name, number, and image of the caller as well as typical mobile-phone features such as Address Book, Calendar, Calculator and Alarm function, the snom m9 provides the perfect blend of mobility and accessibility.

[[Snom_m3_VoIP_Phone|See Configuration Details]]
<br clear="all" />

====SNOM C520====

[[File:snom_c520.png|300px|thumb|left|C520]]

'''Product:''' SNOM C520 Conferencing

'''Company:''' Snom

'''Overview:''' With its modern and sleek design, the C520 fits seamlessly into your working day. Two detachable DECT microphones can be positioned freely or carried in the room as required to ensure the best sound and voice quality.

Built-in charging stations with magnetic bays directly on the base station mean both microphones are always charged and ready for use in the next meeting. The conference phone also features automatic volume control and digital noise reduction so that all call participants can be understood in best sound quality.

[[Snom C520|See Configuration Details]]
<br clear="all" />

====SNOM professional D7XX====

[[File:Snom.jpg|300px|thumb|left|Snom D7XX]]

'''Products:''' D120, D717, D735, D785

'''Company:''' Snom

'''Overview:''' The [https://www.snom.com/en/ip-phones/desk-phones/d7xx-series/ professional D7XX] Series telephones are both aesthetically appealing and highly practical, meeting business requirements when a telephone is a key tool in daily work.

These high-performance devices are future-proofed and provide the best in Wideband HD audio, ensuring crystal clear sound quality. They are Bluetooth compatible to meet the connectivity requirements of today’s offices.

[[Snom IP Phones|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====SNOM M100 KLE====

[[File:M100.jpg|300px|thumb|left|M100 KLE]]

'''Products:''' M100 KLE SIP DECT 4-Line Base Station

'''Company:''' Snom

'''Overview:''' The [https://www.snomamericas.com/en/pd/ip-phones/m-series/m-kle-series/m100-kle/ M100 KLE SIP DECT 4-Line Base Station] supports up to 10 phones in the Snom KLE DECT 4-Line Series, including the M10 and M10R SIP DECT 4-Line handsets and the M18 KLE SIP DECT 4-Line deskset. This cordless family of phones features four programmable LED backlit line keys on the handsets and desk sets.

With key system emulation, the M100 KLE Series handles shared line appearances locally without the need for SCA (shared called appearances) support from your provider. This allows an easy and intuitive method for your customers to see incoming calls, hold calls, and resume calls from any handset or deskset with a simple press of a button.

[[M100_KLE_SIP_DECT_4-Line_Base_Station|See Configuration Details]]

<div style="width:100%;overflow:hidden;clear:both"></div>

=== Vtech ===

==== Vtech Conference Station ====

[[File:VCS754-thumb.PNG|300px|thumb|left|Vtech VCS Series]]

'''Product:''' VCSV752 & CS754

'''Company:''' Vtech

'''Overview:''' The [https://businessphones.vtech.com/pd/3439/VCS754-ErisStation-SIP-Conference-Phone-with-Four-Wireless-Mics Vtech VCS754 ErisStation] conference phone features a compact, all-in-one design makes it easy to keep everything together—no clutter, no hassle. Built-in charging stations with magnetic bays ensure the microphones are charged and available for the next meeting.

[[Vtech Conference Station|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==== Vtech VSP Series ====

[[File:VSP736 ErisTerminal.jpg|300px|thumb|left|VSP Series]]

'''Products:''' VSP600 - VSP715 - VSP725 - VSP726 - VSP736

'''Company:''' Vtech

'''Overview:''' The [https://businessphones.vtech.com/products/sip-phones/vsp700 Vtech VSP700 Series] comes with all the essential features you need to keep pace with your business and your budget. Depending on the model, support two to six SIP accounts with these easy-to-use phones.

[[Vtech VSP Series|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

===Yealink===

====Yealink Voice Solutions====

[[File:Yealink_easyVoip.png|300px|thumb|left|Yealink SIP-T28P]]

'''Product:''' Yealink W60B, Yealink T21, Yealink T42S

'''Company:''' Yealink

'''Overview:''' Yealink offers solutions for each customer's needs, starting from basic to more complex ones.

[[Yealink Voice Solutions|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====Yealink SIP-T28P (VSRF)====

[[File:Yealink-sip-t28p.jpg|300px|thumb|left|Yealink SIP-T28P]]

'''Product:''' Yealink SIP-T28P

'''Company:''' Yealink

'''Overview:''' Yealink SIP-T28P represents the next generation VoIP phone which designed for the business user who needs rich telephony features, friendly UI and super voice quality. It is equipped with the TI TITAN chipset, offers high definition voice quality through TI voice engine, HD handset, HD speaker and HD codec (G.722).

By the large, high-resolution graphical display, and together with all the 48 keys, SIP-T28P offers an excellent user experience to configure, make calls, express XML browser, etc. Moreover, to avoid problems with unwanted violations of your audio data, Yealink SIP-T28P supports the security standards TLS, SRTP, HTTPS, 802.1x, Open VPN and AES encryption which are necessary to protect against electronic eavesdropping and data theft.

[[Yealink_SIP-T28P|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>
[[Category:Devices]]

===Zycoo ZP502===

[[File:ZP502.jpg|left|Zycoo ZP502]]

'''Product:''' Zycoo ZP502

'''Company:''' Zycoo ZP502

'''Overview:''' ZP502 VoIP Phone is ZYCOO business IP phone terminal which adopts SIP and IAX2 protocols.

With Broadcom solution, compatible with various Platforms such as Asterisk, FreePBX, Broadsoft, Cisco call manager, etc.

[[Zycoo ZP502|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

IP Phones

2023-09-25T06:49:39Z

Carlb: /* Polycom */

== Blog Articles ==

* '''VoIP – Bring Your Own Device (BYOD)''': To be able to place or receive calls using VoIP, you may need a hardware setup that will allow you to use your regular phone (ATA) or a special phone that connects directly to VoIP networks (IP Phone). For more information, Take a peek at our blog article [https://wiki.voip.ms/article/VoIP_%E2%80%93_Bring_Your_Own_Device_(BYOD) VoIP – Bring Your Own Device (BYOD)]

* '''IP Phone:''' An IP Phone uses voice over IP (VoIP) technologies allowing telephone calls to be made over an IP network such as the Internet instead of the ordinary PSTN system. Still not clear? Take a peek at our blog article : [https://wiki.voip.ms/article/Back_to_Basics:_What_is_an_IP_Phone%3F Back to Basics - What is an IP Phone?]

'''Looking for a Analog Telephone Adapter (ATA)? [[ATA_Devices | Click here]] to see all models.'''

__TOC__

==IP Phones ==

===3COM 3108 Wireless Phone===

[[File:3com_3108_1.jpg|300px|thumb|left|3COM 3108 Wireless Phone]]

'''Product:''' 3COM 3108 Wireless Phone

'''Company:''' 3COM

'''Overview:''' The 3Com 3108 Wireless Telephone is a Session Initiation Protocol (SIP)-based wireless Voice over Internet Protocol (VoIP) telephone. SIP is an internationally recognized standard <nowiki>(IETF RFC 3261)</nowiki> for implementing VoIP. You can make and receive VoIP calls as long as your Wireless Telephone is registered with a SIP proxy server and you are operating it within range of an IEEE 802.11b/g enabled wireless network (WLAN). The SIP proxy server can belong to a wireless Internet Telephony Service Provider (ITSP) or corporate VoIP PBX system, such as the 3Com NBX® System.

[[3COM_3108_Wireless_Phone|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

===Aastra 6730i/6731i VoIP Phone===

[[File:Aastra6730i.jpg|300px|frame|left|Aastra 6730i VoIP Phone]]

'''Product:''' Aastra 6730i/6731i VoIP Phone

'''Company:''' Aastra

'''Overview:'''The Aastra 6730i, a new member of the carrier-grade, open-standards-based 67xi SIP portfolio, offers exceptional features and flexibility in an enterprise grade IP telephone. With a sleek, elegant design and a compact footprint, this multi-line SIP telephone delivers the advanced features and performance traditionally found only in higher priced products. Featuring a 3 line LCD display, the 6730i supports up to 6 lines with call appearances, offers advanced XML capability to access custom applications and is fully interoperable with leading IP-PBX platforms.

Supported by a host of Aastra configuration options, XML development tools, and continuous product enhancements via software releases, the value offered by the 6730i makes it is ideally suited for light telephone use for the small and home-based business applications.

[[Aastra_6730i_VoIP_Phone|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

===Audiocodes===

====400HD Series====

[[File:Audiocodes 420HD.jpg|300px|thumb|left|Audiocodes 420HD IP Phone]]

'''Product:''' Audiocodes 400HD Series

'''Company:''' Audiocodes

'''Overview:''' The [https://www.audiocodes.com/solutions-products/products/ip-phones AudioCodes 400HD series] of IP phones is a range of easy-to-use, feature-rich desktop devices for the service provider hosted services, enterprise IP telephony and contact center markets. Based on the same advanced, field-proven underlying technology as our other VoIP products, AudioCodes high quality IP phones enable systems integrators and end customers to build end-to-end VoIP solutions.

[[Audiocodes 400HD|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

===Cisco===

====Cisco 88XX & 68XX series====

[[File:8800_Series.png|300px|thumb|left|Cisco 8800 Series]]

'''Product:''' 88XX & 68XX series

'''Company:''' Cisco

'''Overview:''' The ''Cisco IP Phone 6800'' Series multiplatform phones are designed for affordability. They deliver reliable, business-grade audio, with Gigabit Ethernet integration and low power usage.

Ideal for customers with moderate to active VoIP needs, the 6800 Series phones are supported on Cisco-approved third-party unified communications as a service (UCaaS) providers.

The ''Cisco IP Phone 8800'' Series is a great fit for businesses of all sizes seeking secure, high-quality, full-featured VoIP. Select models provide affordable entry to HD video and support for highly-active, in-campus mobile workers. This advanced series provides flexible deployment options: on-premises, cloud and Cisco pre-approved third-party UCaaS providers.

[[Cisco IP Phone 68XX and 88XX|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====Cisco Linksys SPA942 NA====

[[File:SPA942-0.jpg|300px|thumb|left|Cisco Linksys SPA942 NA]]

'''Product:''' Linksys SPA942 NA

'''Company:''' Cisco

'''Overview:''' Four active lines and dual switched Ethernet ports offer incredible flexibility to the home or small-business Internet telephony user. Part of Cisco Small Business IP Phone Series, the SPA942 IP Phone can be configured to carry a unique phone number or extension for each of its lines or can be configured to use a shared number over multiple phones.

Based on the SIP standard, the SPA942 is able to easily integrate new features and services offered by providers. A high-resolution display makes for an easy menu- and web-based configuration.

[[Cisco_Linksys_SPA942_NA|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====Cisco SPA525G====

[[File:525g.jpg|300px|thumb|left|Cisco SPA525g Phone Adapter with Router]]

'''Product:''' Cisco SPA525G

'''Company:''' Cisco

'''Overview:''' The Cisco SPA525G 5-line IP Phone with Color Display is a full-featured VoIP (Voice over Internet Protocol) phone that provides voice communication over an IP network. It provides traditional features, such as call forwarding, redialing, speed dialing, transferring calls, conference calling, and accessing voice mail. Calls can be made or received with a handset, headset or speaker.
Your Cisco IP Phone provides a web interface for the phone user that allows you to configure some features of your phone by using a web browser.
This article will guide you through the steps for basic configuration to make it work with VoIP.ms

[[Cisco SPA525G|See Configuration Details]]
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====Cisco IP Phone 7940/7960====

[[File:Cisco_7960g.png|300px|thumb|left|Cisco IP Phone 7940/7960]]

'''Product:''' Cisco IP Phone 7940/7960

'''Company:''' Cisco

'''Overview:''' The Cisco IP Phone 7940/7960 is a full-featured telephone that provides voice communication over an IP network. This phone functions as a traditional analog phone, allowing you to place and receive telephone calls. This phone also supports features, such as network call forwarding, redialing, speed dialing, transferring calls, placing conference calls, and accessing voice mail. Phones require Power Over Ethernet (PoE) or a 48V AC Adapter to power up.

[[Cisco_IP_Phone_7940/7960|See Configuration Details]]
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====Cisco SPA30x and SPA50x series IP phones====

[[File:Cisco_spa504g.jpg|300px|thumb|left|Cisco SPA504G Phone]]

'''Product:''' Cisco SPA504G Phone

'''Company:''' Cisco

'''Overview:'''
The Cisco SPA504G is an office-style desk telephone with built-in voice over the Internet.

It is one in a series of similar models (SPA30x and SPA50x) which vary primarily in the number of lines (extensions) on the 'phone, power source (some models use power-over-Ethernet) and the availability of a second Ethernet connector. These devices are well-suited to offices and IP PBX applications. These do not provide a virtual line for connecting analog devices such as standard telephone handsets; they are instead self-contained to connect directly to VoIP.

[[Cisco_SPA504G_Phone|See Configuration Details]]
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=== Dinstar===

==== Dinstar C60 Series ====

[[File:C60U_F.png|300px|thumb|left|Dinstar C60UP]]

C60 series are based on high innovative SIP technology, which is ideal for all kinds of business communication. It integrates with 132x64-pixel graphical LCD with back-light, elegant and intuitive user interface, which indicate you can enjoy good user experience.

[[Dinstar_C60_series|See Configuration Details]]
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===Fanvil ===

====Fanvil X4G====

[[File:FanfillX4g.jpg|300px|thumb|left|Fanvil X4G]]

'''Product:''' Fanvil X4G

'''Company:''' Fanvil

'''Overview:''' The Fanvil X4G has a 2.8" main color screen and a secondary 2.4" DSS color screen. The user interface is sleek, colorful and easy to navigate. It has a one button call function and a call log and the ability to store 500 phonebook entries. The X4G's high compatibility supports various systems including 3CX, Avaya, OpenVox, NEC, Elastix, Asterisk, Matrix, Broadsoft, Epygi and more.

[[Fanvill X4G|See Configuration Details]]
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====Fanvil V62====

[[File:Fanvil-v62-VoIPms.png|300px|thumb|left|Fanvil V62]]

'''Product:''' Fanvil V62

'''Company:''' Fanvil

'''Overview:''' V62 is more than an efficient telephone but a delicate work of art, providing a smart and smooth business communication experience for enterprises. As the essential business phone featuring a graphical Dot-matrix screen with backlight and necessary VoIP features and other extended features, V62 is a great combination of elegant outside and powerful inside.

[[Fanvil V62|See Configuration Details]]
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====Fanvil V63====

[[File:Fanvil-v63-VoIPms.png|300px|thumb|left|Fanvil V63]]

'''Product:''' Fanvil V63

'''Company:''' Fanvil

'''Overview:''' V63 is more than an efficient telephone but a delicate work of art, providing a smart and smooth business communication experience for enterprises. As the essential business phone featuring 2.8 color screen and necessary VoIP features and other extended features, V63 is a perfect combination of elegant outside and powerful inside.

[[Fanvil V63|See Configuration Details]]
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====Fanvil V65====

[[File:Fanvil-v65-VoIPms.jpg|300px|thumb|left|Fanvil V65]]

'''Product:''' Fanvil V65

'''Company:''' Fanvil

'''Overview:''' V65 is more than an efficient telephone but a delicate work of art, providing a smart and smooth business communication experience for executives and managers. As the prime business phone featuring an adjustable screen and built-in Bluetooth 4.2 and 2.4G/5G Wi-Fi, V65 is a perfect combination of elegant outside and powerful inside.

[[Fanvil V62|See Configuration Details]]
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====Fanvil V67====

[[File:Fanvil-V67-VoIPms.jpg|300px|thumb|left|Fanvil V67]]

'''Product:''' Fanvil V67

'''Company:''' Fanvil

'''Overview:''' V67 is more than an efficient telephone but also a delicate work of art, which provides a more intelligent and elegant office operation experience for executives, managers and teleworkers. With brand new design, V67 features an adjustable touch screen and a keypad with colorful light effect that improve the beauty and comfort of office desktop.

[[Fanvil V67|See Configuration Details]]
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====Fanvil X4U-V2====

[[File:Fanvil-X4U-V2-VoIPms.jpg|300px|thumb|left|Fanvil X4U-V2]]

'''Product:''' Fanvil X4U-V2

'''Company:''' Fanvil

'''Overview:''' The X4/X4G is a feature-rich sip phone for business. The 4-Line IP Phone has been designed by pursuing ease of use in even the tiniest details. Dual 10/100 Mbps(X4G: 10/100/1000 Mbps) network ports with integrated PoE are ideal for extended network use. Delivering a superb sound quality as well as rich visual experience. With second DSS color screen, the IP Phone supports up to 30 DSS keys which improve work efficiency. Using standard encryption protocols to perform highly secure remote provisioning and software upgrades.

[[Fanvil X4U-V2|See Configuration Details]]
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====Fanvil X6U-V2====

[[File:Fanvil-X6U-V2-VoIPms.jpg|300px|thumb|left|Fanvil X6U-V2]]

'''Product:''' Fanvil X6U-V2

'''Company:''' Fanvil

'''Overview:''' Featuring three color displays, newly added line keys with LED light and built-in Bluetooth, Fanvil X6U provides the direct access to instructions, aiming to offering greater flexibility, productivity, to exceed the different demands of businesses. Wideband codec of G.722 and Opus in this device delivers you an immersive HD audio experience in both high band and low band with the network.

[[Fanvil X6U-V2|See Configuration Details]]
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===Linkvil by Fanvil===

====Linkvil W611W====

[[File:Linkvil_W611W_by_Fanvil.jpg|300px|thumb|left|Linkvil W611W by Fanvil]]

'''Product:''' Linkvil W611W by Fanvil

'''Company:''' Fanvil

'''Overview:''' LINKVIL W611W is a portable, elegant Wi-Fi phone designed for mobile communication applications. Certified to IP67 standard, W611W is highly waterproof, dustproof, and drop-safe from 1.8-meter height. It has an excellent performance in different environments with humidity and dust. W611W integrates Wi-Fi 6, bringing a superb wireless communication experience. Moreover, it integrates Bluetooth 5.0 for pairing with headsets and mobile devices. Installed with a rechargeable 1900mAh battery, W611W is ready for 9 hours’ talk time or 200 hours standby time. W611W is widely used in various wireless scenarios such as enterprises, shopping malls, residential areas, hotels and warehouses, providing users with a high-quality mobile communication experience.

[[Linkvil_W611W_by_Fanvil | See Configuration Details]]
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[[Category:Devices]]

=== Flyingvoice ===

====Flyingvoice====

[[File:Flyingvoice_Phones.jpg|300px|thumb|left|Flyingvoice]]

'''Product:''' '''FIP11C / FIP11CP''', '''FIP13G''', '''FIP14G''', '''FIP15G''', '''FIP10/FIP10P''', '''FIP12WP''', '''FIP16'''

'''Company:''' Flyingvoice

'''Overview:''' Flyingvoice is a provider of communication terminal equipment and one-stop VoIP CPE solutions. They offer a full range of VoIP products, such as VoIP phones, ATAs, gateways and routers for businesses and consumers. Their WiFI IP phones offer a wireless option, so you do not need a wired internet connection to either make or receive VoIP calls.

[[Flyingvoice|See Configuration Details]]
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[[Category:Devices]]

===Fortinet===

====FortiFone FON-570====

[[File:FON-570_Phone.jpg|300px|thumb|left|Fortinet FortiFone FON-570]]

'''Product:''' FortiFone FON-570

'''Company:''' Fortinet

'''Overview:''' Featuring a large 7” color touchscreen and premium HD call quality, this IP phone is great for efficient communications. Combined dedicated feature keys and programable keys expandable to 109, you have the flexibility to control your calls within your fingertips.

*7" color screen
*7 dedicated feature keys
*109 programable phone keys
*Full duplex speakerphone
*2 x 10/100/1000 Ethernet ports
*Integrated Power over Ethernet (PoE) support

[[Fortinet_FortiFone_FON-570|See Configuration Details]]
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====FortiFone FON-375====

[[File:FON-375_Phone.jpg|300px|thumb|left|Fortinet FortiFone FON-375]]

'''Product:''' FortiFone FON-375

'''Company:''' Fortinet

'''Overview:''' A reliable IP phone delivers HD sound quality, ideal for office workers who need efficient communications. An easy-to-read color screen and a programable second screen make it easy to display which lines are in use and who is on a call.

:*Dual color screens: 2.8" + 2.4”
:*8 dedicated feature keys
:*30 programable phone keys
:*Full duplex speakerphone
:*2 x 10/100/1000 Ethernet ports
:*Integrated Power over Ethernet (PoE) support

[[Fortinet_FortiFone_FON-375|See Configuration Details]]
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====FortiFone FON-175====

[[File:FON-175_Phone.jpg|300px|thumb|left|Fortinet FortiFone FON-175]]

'''Product:''' FortiFone FON-175

'''Company:''' Fortinet

'''Overview:''' A quality, two-line IP phone delivers reliable communications with HD audio quality. This entry-level business phone is easy to use that works in any office.

*2.4" color screen
*5 dedicated feature keys
*Full duplex speakerphone
*2 x 10/100 Ethernet ports
*Integrated Power over Ethernet (PoE) support

[[Fortinet_FortiFone_FON-175|See Configuration Details]]
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===Gigaset A510 IP===

[[File:Gigaset_a510_IP.jpg#file|300px|thumb|left]]

'''Product:'''Gigaset A510 IP

'''Company:'''Gigaset

'''Overview:'''The Gigaset A510 and C610 IP phones are fitting solutions if you are looking for the flexibility of VoIP and the convenience of using a cordless handset.

[[Gigaset_A510_IP| See Configuration Details]]
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===Grandstream===

====DP Series====

=====Grandstream DP715/DP710=====

[[File:Grandstream 715-710.png|300px|thumb|left]]

'''Product:''' Grandstream DP715/DP710

'''Company:''' Grandstream

'''Overview:''' The DP715/710 is the next generation of powerful, affordable, high quality and simple to configure DECT Cordless IPPhone for small business and residential users. Their compact size, superb voice quality, rich feature set, market leading price-performance and wide range radio coverage enable consumers to maximize the power of IP voice application and mobility for a minimum investment.

[[Grandstream_DP715/DP710| See Configuration Details]]
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=====Grandstream DP750/DP720=====

[[File:DP750-720.png|300px|thumb|left]]

'''Product:''' Grandstream DP750/DP720

'''Company:''' Grandstream

'''Overview:''' The DP750/720 base station and handsets allows you to deploy an immersive DECT environment that allows users to communicate free from their desktop using Grandstream’s DP720 DECT handsets. The DP750 pairs with up to 5 DP720s to create a powerful and mobile network solution with up to 10 lines per handset, and 5 concurrent calls per DECT system.

[[Grandstream_DP750/DP720| See Configuration Details]]
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====GXP Series====

=====Grandstream GXP1630 IP Phone=====

[[File:Gxp1630.jpg|300px|thumb|left|Grandstream GXP2120 IP Phone]]

'''Product:''' Grandstream GXP1630 IP Phone

'''Company:''' Grandstream

'''Overview:''' The GXP1630 comes equipped with a suite of VoIP features that are deployed in a clear and easy-to-use fashion. Focused primarily for low to medium call volumes and efficient call handling, its 3 line/SIP account design and 8 dual-colored BLF/speed dial keys gives this versatility. The GXP1630 also supports the best possible connection speeds and call quality with its dual Gigabit ports and HD audio on both speakerphone and handset. With other features such as its integrated PoE, 3 XML programmable soft keys and 4-way conferencing support, the GXP1630 is a high-quality and versatile Basic IP phone.

[[Grandstream_GXP1630|See Configuration Details]]
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=====Grandstream GXP2120 IP Phone=====

[[File:Gxp2110.png|300px|thumb|left|Grandstream GXP2120 IP Phone]]

'''Product:''' Grandstream GXP2120 IP Phone

'''Company:''' Grandstream

'''Overview:''' The Grandstream GXP2120 is a 6 line SIP Phone which features HD Voice hardware and software support and a large 320 x 160 backlit graphical LCD. The GXP2120 can handle 6 SIP accounts represented by 6 dual-color line keys and 4 XML programmable context-sensitive soft keys. In addition, the GXP2120 has 7 dual-color BLF extension keys for the most common calls and transfers making it an ideal phone for an office user with moderate to heavy interoffice calling needs.

[[Grandstream_GXP2120_IP_Phone|See Configuration Details]]
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=====Grandstream GXP2135 IP Phone=====

[[File:GXP2135-device.jpg|300px|thumb|left|Grandstream GXP2135 IP Phone]]

'''Product:''' Grandstream GXP2135 IP Phone

'''Company:''' Grandstream

'''Overview:''' The GXP2135 is the ideal selection for busy users who value call control, productivity and usability, and manage medium to heavy call volumes. Equipped with 8 lines and 4 SIP accounts, a 2.8-inch color LCD display, and 32 digital speed dial/BLF keys, the GXP2135 enables quick and powerful usability.

As all Grandstream IP phones do, the GXP2135 features state-of-the-art security encryption technology (SRTP and TLS). The GXP2135 supports a variety of automated provisioning options, including zero-configuration with Grandstream’s UCM series IP PBXs, encrypted XML files and TR-069, to make mass deployment extremely easy.

[[Grandstream GXP2135|See Configuration Details]]
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=====Grandstream GXP2170=====

[[File:Grandstream_GXP2170.png|300px|thumb|left|Grandstream GXP2170]]

'''Product:''' Grandstream GXP2170

'''Company:''' Grandstream

'''Overview:''' The GXP2170 is a powerful High-End IP phone that is ideal for busy users who handle high call volumes. Receptionists, administrators, sales staff and other call-intensive rolls can enjoy efficiency by utilizing the GXP2170’s 12 line keys, 4.3 inch color display LCD and 48 digital, on-screen speed dial/BLF keys.

[[Grandstream_GXP2170|See Configuration Details]]
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=====Grandstream GXP2200=====

[[File:Grandstream_GXP2200.png|300px|thumb|left|Grandstream GXP2200]]

'''Product:''' Grandstream GXP2200

'''Company:''' Grandstream

'''Overview:''' The GXP2200 is one of the most advanced AndroidTM desktop IP phones available on the market today. The innovative phone includes the AndroidTM version 2.3 operating system with a 4.3 inch capacitive touchscreen LCD and the ability to host 6 SIP accounts. Web applications such as news, social media sites, and games can be downloaded directly via Google Play Store, and applications can be created to fit any need and downloaded directly to the phone for customized use.

[[Grandstream_GXP2200|See Configuration Details]]
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====GRP Series====

: '''Note that the GRP2615 configurations are the same to all GRP series.'''

=====Grandstream GRP2615=====

[[File:Grandstream_GRP2615.png|300px|thumb|left|Grandstream GRP2615]]

'''Product:''' Grandstream GRP2615

'''Company:''' Grandstream

'''Overview:''' The GRP2615 is a high-end carrier-grade IP phone featuring a sleek design and a suite of next-generation features including integrated Wi-Fi, Bluetooth support, 40 multipurpose keys (MPKs), an available extension module, dual Gigabit ports and more. This device features a large 4.3 inch color LCD with swappable face plates to allow for easy logo customization. For cloud provisioning and centralized management, the GRP2615 is supported by Grandstream’s Device Management System (GDMS), which provides a centralized interface to configure, provision, manage and monitor deployments of Grandstream endpoints.

[[Grandstream_GRP2615|See Configuration Details]]
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====WP Series====
:''There are multiple IP phones in this series, which are all very similar in configuration. All are self-contained IP handsets which connect directly to Wi-Fi and register directly with any of the VOiP.ms servers.''

=====Grandstream WP810=====

[[File:Grandstream WP810.png|150px|thumb|left|Grandstream WP810]]

'''Product:''' Grandstream WP810

'''Company:''' Grandstream

'''Overview:''' The WP810, WP820, WP822 and WP825 are cordless SIP IP phones which connect directly to the local network using dual-band Wi-Fi, eliminating the need for a cordless base station or other hardware. Effectively, the WP810 (with its charging cradle) is a self-contained two-line SIP solution.

[[Grandstream WP810, WP820, WP822 and WP825|See Configuration Details]]
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===Innomedia===

====BuddyTalk BT110====

[[File:Buddy_Talk_BT110_Innomedia.jpg|250px|thumb|left|Innomedia BuddyTalk BT110]]

'''Product:''' BuddyTalk BT110

'''Company:''' Innomedia

'''Overview:''' Powered by the Amazon Alexa Voice Service (AVS), the BuddyTalk series of products supports a broad suite of AVS enabled smart speaker features.
Equipped with advanced audio processing and VoIP technologies, BuddyTalk devices are intelligent speakerphones that deliver unprecedented flexibility in calling, superior voice quality, and high levels of security.

[[BuddyTalk_-_BT110|See Configuration Details]]
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===Konftel===

==== Konftel 300Wx IP ====
[[File:Konftel-300Wx-IP.png|300px|thumb|left|Konftel 300Wx IP]]

'''Product:''' Konftel 300Wx IP

'''Company:''' Konftel

'''Overview:''' The Konftel 300Wx IP wireless conference phone allows you to hold conference calls in HD quality wherever is convenient for you – without worrying about network and power outlets. Reliable and secure DECT technology. The accompanying IP DECT base can handle up to 20 registered Konftel 300Wx devices and five ongoing calls.

The rechargeable battery ensures more than 60 hours of call time, so you can talk for a full working week without recharging! A USB port makes the Konftel 300Wx ready for all the apps and services we use to communicate and collaborate via computers. Combine meeting apps and regular phone calls. OmniSound® delivers superb sound quality

[[Konftel 300Wx|See Configuration Details]]
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==== Konftel 300IPx ====

[[File:Konftel300ipx-conference-phone.jpg|300px|thumb|left|Konftel 300IPx]]

'''Product:''' Konftel 300IPx

'''Company:''' Konftel

'''Overview:''' The [https://www.konftel.com/en/products/konftel-300ipx KONFTEL 300IPx] together with the Konftel Unite app brings a whole new easiness to conference calls. It is highly intuitive and based on our natural mobile behavior. The new generation of IP conference phone is – The Art of Easiness.

[[Konftel 300IPx|See Configuration Details]]
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===Panasonic===

====Panasonic KX-TGP 550====

[[File:Pana550b.jpg|300px|thumb|left|Panasonic KX-TGP 550]]

'''Product:''' Panasonic KX-TGP 550

'''Company:''' Panasonic

'''Overview:''' The Panasonic KX-TGP550 responses to the needs of SIP IP-Centrix/Hosted PBX systems and Asterisk users. Conveniently, no need to set up a system telephone at every base. This system also enables you to use a range of convenient services provided by the carrier such as Call Forward, Voice Mail, etc.

*Support for 3 simultaneous network conversations
*Up to 6 DECT cordless handsets
*Support for up to 8 SIP registrations (e.g. up to 8 DID lines or extensions)
*Hands-free speakerphone on a cordless handset

[[Panasonic KX-TGP 550|See Configuration Details]]
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====Panasonic KX-HDV130C====

[[File:Panasonic_KXHDV130_01.jpg|300px|thumb|left|Panasonic KX-HDV130C]]

'''Product:''' Panasonic KX-HDV130C

'''Company:''' Panasonic

'''Overview:''' The Panasonic KX-HDV130 SIP desk phone delivers the ideal balance of low cost and high quality, along a range of value added features.

*Support for up to 2 SIP registrations (e.g. up to 2 DID lines or extensions)
*Support for 3 simultaneous network conversations (3-way conferencing)*
*2 Programmable keys / Line keys
*Hands-free speakerphone on a cordless handset

[[Panasonic KX-HDV130C|See Configuration Details]]
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====Panasonic KX-HDV230====

[[File:Panasonic_KXHDV230_01.jpg|300px|thumb|left|Panasonic KX-HDV230]]

'''Product:''' Panasonic KX-HDV230

'''Company:''' Panasonic

'''Overview:''' The Panasonic KX-HDV230 IP phone offers streamlined functions and the high definition voice quality that's essential for effective communication.

*Support for up to 6 SIP registrations (e.g. up to 6 DID lines or extensions)
*24 Flexible Function Keys
*2 ethernet ports 10/100/1000 Base -T
*Large LCD with Backlight

[[Panasonic KX-HDV130C|See Configuration Details]]
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====Panasonic KX-HDV330====

[[File:Panasonic_KXHDV330_01.jpg|300px|thumb|left|Panasonic KX-HDV330]]

'''Product:''' Panasonic KX-HDV330

'''Company:''' Panasonic

'''Overview:''' The Panasonic KX-HDV330 is a multi-functional business SIP phone equipped with a colour touch panel for intuitive operation.

*Large LCD with Backlight
*Built-in Bluetooth®
*Support for up to 12 SIP registrations (e.g. up to 12 DID lines or extensions)
*24 Flexible Function Keys
*Hands-free speakerphone on a cordless handset

[[Panasonic KX-HDV130C|See Configuration Details]]
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===Pirelli DP-L10===

[[File:Pirelli1.jpg|300px|thumb|left|Pirelli DP-L10]]

'''Product:''' Pirelli DP-L10

'''Company:''' Pirelli

'''Overview:''' Pirelli's DualPhone DP-L10 phone combines GSM tri-band capabilities with SIP-based Wireless VoIP via an integrated WLAN 802.11b/g interface, combining cell phone technology with the added advantage of transporting phone calls over the Internet.

The Dual Phone enables mobile phone features such as Internet browsing, e-mail, SMS, and MMS, which can also be used through a fixed-broadband connection, providing the same mobile services with the added benefits of greater bandwidth and cost savings.

[[Pirelli DP-L10|See Configuration Details]]
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===Poly (Polycom, Obahai, Plantronics)===
Poly.com has been acquired by Hewlett-Packard.

====Poly Edge B10====

[[File:PolyEdgeB10.png|300px|thumb|left|Poly Edge B10]]

'''Product:''' Poly Edge B10

'''Company:''' Poly

'''Overview:''' Sleek design meets Poly pro-grade audio at a shockingly affordable price: That’s what makes the Poly Edge B Series IP Phones the genius choice for any growing business. Easy to use with illuminated keys where you need them. Plug-and-play provisioning and hardcore reliability make it pure value in a low-cost business phone.

[[Poly_Edge_B10|See Configuration Details]]

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====Polycom SoundStation IP 4000 Conference Phone====

[[File:Polycom-4000-0.gif|300px|thumb|left|Polycom SoundStation IP 4000 Conference Phone]]

'''Product:''' Polycom SoundStation IP 4000 Conference Phone

'''Company:''' Poly

'''Overview:''' the SoundStation IP 4000 SIP voice conferencing unit is the answer for organizations that are ready for the
benefits and versatility of a SIP enabled business. Designed for offices or small to medium-sized conference rooms, the SoundStation IP 4000 SIP provides remarkable room coverage. You can speak naturally from up to 10-feet away from a microphone and still be heard clearly on the far end of the call. Need coverage for a larger room? The optional extension microphones offer an increased pickup for larger rooms. Plus, with gated microphone technology, echo and background noise is almost entirely eliminated.

The SoundStation IP 4000 SIP also provides familiar features that are easy to use. The menu-driven user interface, viewable on a high-resolution backlit LCD, offers convenient access to business telephony functions you use most including transfer, hold, redial, and conference.

[[Polycom_SoundStation_IP_4000_Conference_Phone|See Configuration Details]]
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====Polycom SoundPoint IP 501, 550, 650, etc.====

[[File:Polycom501.jpg|258px|thumb|left|Polycom SoundPoint IP 501]]

'''Product:''' Polycom SoundPoint IP 501, 550, 650, etc.

'''Company:''' Poly

'''Overview:''' The SoundPoint IP series are multi-line (3-6 lines depending on model) Voice over IP telephones that seamlessly integrate with IP PBX and softswitch vendors’ IP solutions.

As protocols develop and standards evolve, it’s easy to update the phone in the field via a software download, thus enabling new features and functionality for the phone and protecting your investment.

Whether you deploy MGCP or SIP standards, Polycom offers a solution that fits all your business communication needs.

[[Polycom_SoundPoint_IP_501|See Configuration Details]]
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====Polycom SoundPoint IP 331====

[[File:Polycom_Soundpoint_IP_331.png|258px|thumb|left|Polycom SoundPoint IP 331]]

'''Product:''' Polycom SoundPoint IP 331

'''Company:''' Poly

'''Overview:''' The Polycom IP 331 is engineered to make installation, configuration, and upgrades as simple and efficient as possible. The phones' standard base stand can be reversed to become a wall mount, eliminating the need for a separate accessory. Built-in IEEE 802.3af PoE circuitry and a dual-port Ethernet switch enables flexible deployment options and savings on cabling expenses.

[[Polycom_Soundpoint_IP_331|See Configuration Details]]
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====Polycom SoundPoint IP 601====

[[File:Voipms-polycom601.jpg|258px|thumb|left|Polycom SoundPoint IP 601]]

'''Product:''' Polycom SoundPoint IP 601

'''Company:''' Poly

'''Overview:''' The 6-line Polycom® SoundPoint IP™ 601 offers industry-leading functionality and call handling unmatched voice quality an intuitive user interface & expandability to 12 lines!

[[Polycom_SoundPoint_IP_601|See Configuration Details]]
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====Poly VVX-D230====

[[File:Poly-vvx-d230-base.jpg|250px|thumb|left|Poly VVX-D230 cordless base]]

'''Product:''' Poly VVX-D230 cordless DECT phone

'''Company:''' Poly (Obahai, Polycom)

'''Overview:''' The VVX series are SIP desk phones aimed at small to mid-sized business; the VVX-D230 is a cordless system on which one base may control up to ten handsets.

[[Poly VVX-D230|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====Poly VVX 300, 400, etc====

[[File:Vvx300.png|250px|thumb|left|Polycom VVX 300 Series]]

'''Product:''' Polycom VVX Series

'''Company:''' Poly

'''Overview:''' The VVX series provides high-quality audio (HD Voice) and video communications from 6 lines and up.

[[Polycom VVX 300, 400, etc|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

=== Positron IP phones ===

[[File:PositronLogo.jpeg|250px|thumb|left|Positron IP phones]]

'''Product:''' Positron

'''Company:''' [http://www.positrontelecom.com/products.html Positron]

'''Overview:''' Positron IP Phones is an affordable next-generation SIP phone including wideband audio support, ethernet ports and integrated Power over Ethernet.

All the IP Phones are optimized for executives, administrative assistants and those working with bandwidth-intensive applications on collocated PCs. Line keys can also be configured as IP PBX features such as BLF, SCA, Intercom, Call Pickup, Call Park, and many others. The high-resolution screen and the HD voice features provide a high quality visual and authentic audio experience.

==== Positron IP304 ====

[[File:IP304.png |250px|thumb|left|Positron IP304]]

'''Product:''' Positron IP304

'''Company:''' [http://www.positrontelecom.com/products.html Positron]

'''Overview:''' Positron IP304 is an affordable next-generation SIP phone with wideband audio support, dual Ethernet port and integrated Power over Ethernet.

The IP304 enterprise VoIP phone is Positron’s entry-level phone with 3 VoIP accounts. Its functionalities make it easy to place calls on hold, to transfer calls and to initiate a conference call.

With HD audio for both headset and speakerphone, IP304 provides a clear authentic audio experience

[[Positron IP304 | View configuration for Positron IP304]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==== Positron IP304C ====

[[File:IP304C.png |250px|thumb|left|Positron IP304C]]

'''Product:''' Positron IP304C

'''Company:''' [http://www.positrontelecom.com/products.html Positron]

'''Overview:''' Positron IP304C is an innovative enterprise-level IP Phone that features 4 line keys, color display, 3.5” TFT-LCD with 480 x 320 pixel. It supports up to a 5-way conference.

IP304C is optimized for executives, administrative assistants and those working with bandwidth-intensive applications on collocated PCs. Line keys can also be configured as IP PBX features such as BLF, SCA, Intercom, Call Pickup, Call Park, and many others.

The high-resolution TFT-LCD screen and the HD voice features provide a high quality visual and authentic audio experience.

[[Positron IP304C | View configuration for Positron IP304C]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==== Positron IP408 ====

[[File:IP408.png |250px|thumb|left|Positron IP408]]

'''Product:''' Positron IP408

'''Company:''' [http://www.positrontelecom.com/products.html Positron]

'''Overview:''' Positron IP408 is an affordable next-generation SIP 2.0 phone including wideband audio support and WAN/LAN Ethernet ports with route and bridge mode.

The IP408 enterprise VoIP phone supports 4 SIP lines. Its functionalities make it easy to place calls on hold, to transfer calls and to initiate a conference call.

With HD audio for both headset and speakerphone, IP408 provides a clear authentic audio experience

[[Positron IP408 | View configuration for Positron IP408]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==== Positron IP410C ====

[[File:IP410C.png |250px|thumb|left|Positron IP410C]]

'''Product:''' Positron IP410C

'''Company:''' [http://www.positrontelecom.com/products.html Positron]

'''Overview:''' Positron IP410C is an affordable next-generation SIP Phone that features 4 line keys, 10 programmable extension keys, color display, wideband audio support and dual Ethernet port with integrated Power over Ethernet.

IP410C is optimized for executives, administrative assistants and those working with bandwidth-intensive applications on collocated PCs. Line keys can also be configured as IP PBX features such as BLF, SCA, Intercom, Call Pickup, Call Park, and many others.

The high-resolution TFT-LCD screen and the HD voice features provide a high quality visual and authentic audio experience.

[[Positron IP410C | View configuration for Positron IP410C]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==== Positron IP410G ====

[[File:IP410G.png |250px|thumb|left|Positron IP410G]]

'''Product:''' Positron IP410G

'''Company:''' [http://www.positrontelecom.com/products.html Positron]

'''Overview:''' Positron IP410G is an innovative enterprise-level color IP Phone that features 4 line keys, 10 programmable extension keys, color display, 3.5” TFT-LCD with 480*320 pixel, wideband audio support and dual Gigabit Ethernet port with integrated Power over Ethernet.

IP410G is optimized for executives, administrative assistants and those working with bandwidth-intensive applications on collocated PCs. Ten programmable keys can also be configured as IP PBX features such as BLF, SCA, Intercom, Call Pickup, Call Park, and many others.

The high-resolution TFT-LCD screen and the HD voice features provide a high quality visual and authentic audio experience.

[[Positron IP410G | View configuration for Positron IP410G]]
<div style="width:100%;overflow:hidden;clear:both"></div>

===Siemens Gigaset C450-Ip===

[[File:Siemens_gigaset_c450_IP.jpg|300px|thumb|left|Siemens Gigaset C450-Ip]]

'''Product:''' Siemens Gigaset C450-Ip

'''Company:''' Siemens

'''Overview:''' The Gigaset C450 is an entry phone in the Siemens Gigaset product line, the addition of IP connectivity via a standard Ethernet 10/100 BaseT interface and SIP protocol within a reasonable price bracket, opens large applications where wireless operation and SIP are required.
The Ethernet connectivity allows the unit to work without a PC.

The dual mode (SIP/PSTN) enables incoming call on a legacy interface with existing directory number and safe emergency outgoing calls.

The presentation of the unit with an independent base and a dedicated phone charger/support will simplify cabling.

[[Siemens Gigaset C450-Ip|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

===Snom===

====Snom 320 VoIP Phone====

[[File:Snom320.png|300px|frame|left|Snom 320 VoIP Phone]]

'''Product:''' Snom 320 VoIP Phone

'''Company:''' Snom

'''Overview:'''Ideal for the office and everyone who spends a lot of time on the phone, the snom 320 is an affordable, yet powerful SIP business phone with a built-in, full-duplex speakerphone and three-party conference bridging.

[[SNOM 320|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====Snom m3 VoIP Phone====

[[File:Snom_m3.png|300px|thumb|left|Snom m3 VoIP Phone]]

'''Product:''' Snom m3 VoIP Phone

'''Company:''' Snom

'''Overview:''' The snom m9 is the next generation DECT handheld (CAT-iq) that empowers users with the convenience of wireless communication along with the widely accepted benefits and feature richness of Voice-over-IP telephony.

The snom m9 promises to deliver excellent speech quality through digital and wideband audio (klarVOICE).

By combining professional functions of versatile business communication with the intuitive features of the mobile-carrier world, the snom m9 is ideally suited for professional and private use alike. With features such as hands-free mode, calling line identification (CLI) by displaying name, number, and image of the caller as well as typical mobile-phone features such as Address Book, Calendar, Calculator and Alarm function, the snom m9 provides the perfect blend of mobility and accessibility.

[[Snom_m3_VoIP_Phone|See Configuration Details]]
<br clear="all" />

====SNOM C520====

[[File:snom_c520.png|300px|thumb|left|C520]]

'''Product:''' SNOM C520 Conferencing

'''Company:''' Snom

'''Overview:''' With its modern and sleek design, the C520 fits seamlessly into your working day. Two detachable DECT microphones can be positioned freely or carried in the room as required to ensure the best sound and voice quality.

Built-in charging stations with magnetic bays directly on the base station mean both microphones are always charged and ready for use in the next meeting. The conference phone also features automatic volume control and digital noise reduction so that all call participants can be understood in best sound quality.

[[Snom C520|See Configuration Details]]
<br clear="all" />

====SNOM professional D7XX====

[[File:Snom.jpg|300px|thumb|left|Snom D7XX]]

'''Products:''' D120, D717, D735, D785

'''Company:''' Snom

'''Overview:''' The [https://www.snom.com/en/ip-phones/desk-phones/d7xx-series/ professional D7XX] Series telephones are both aesthetically appealing and highly practical, meeting business requirements when a telephone is a key tool in daily work.

These high-performance devices are future-proofed and provide the best in Wideband HD audio, ensuring crystal clear sound quality. They are Bluetooth compatible to meet the connectivity requirements of today’s offices.

[[Snom IP Phones|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====SNOM M100 KLE====

[[File:M100.jpg|300px|thumb|left|M100 KLE]]

'''Products:''' M100 KLE SIP DECT 4-Line Base Station

'''Company:''' Snom

'''Overview:''' The [https://www.snomamericas.com/en/pd/ip-phones/m-series/m-kle-series/m100-kle/ M100 KLE SIP DECT 4-Line Base Station] supports up to 10 phones in the Snom KLE DECT 4-Line Series, including the M10 and M10R SIP DECT 4-Line handsets and the M18 KLE SIP DECT 4-Line deskset. This cordless family of phones features four programmable LED backlit line keys on the handsets and desk sets.

With key system emulation, the M100 KLE Series handles shared line appearances locally without the need for SCA (shared called appearances) support from your provider. This allows an easy and intuitive method for your customers to see incoming calls, hold calls, and resume calls from any handset or deskset with a simple press of a button.

[[M100_KLE_SIP_DECT_4-Line_Base_Station|See Configuration Details]]

<div style="width:100%;overflow:hidden;clear:both"></div>

=== Vtech ===

==== Vtech Conference Station ====

[[File:VCS754-thumb.PNG|300px|thumb|left|Vtech VCS Series]]

'''Product:''' VCSV752 & CS754

'''Company:''' Vtech

'''Overview:''' The [https://businessphones.vtech.com/pd/3439/VCS754-ErisStation-SIP-Conference-Phone-with-Four-Wireless-Mics Vtech VCS754 ErisStation] conference phone features a compact, all-in-one design makes it easy to keep everything together—no clutter, no hassle. Built-in charging stations with magnetic bays ensure the microphones are charged and available for the next meeting.

[[Vtech Conference Station|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==== Vtech VSP Series ====

[[File:VSP736 ErisTerminal.jpg|300px|thumb|left|VSP Series]]

'''Products:''' VSP600 - VSP715 - VSP725 - VSP726 - VSP736

'''Company:''' Vtech

'''Overview:''' The [https://businessphones.vtech.com/products/sip-phones/vsp700 Vtech VSP700 Series] comes with all the essential features you need to keep pace with your business and your budget. Depending on the model, support two to six SIP accounts with these easy-to-use phones.

[[Vtech VSP Series|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

===Yealink===

====Yealink Voice Solutions====

[[File:Yealink_easyVoip.png|300px|thumb|left|Yealink SIP-T28P]]

'''Product:''' Yealink W60B, Yealink T21, Yealink T42S

'''Company:''' Yealink

'''Overview:''' Yealink offers solutions for each customer's needs, starting from basic to more complex ones.

[[Yealink Voice Solutions|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====Yealink SIP-T28P (VSRF)====

[[File:Yealink-sip-t28p.jpg|300px|thumb|left|Yealink SIP-T28P]]

'''Product:''' Yealink SIP-T28P

'''Company:''' Yealink

'''Overview:''' Yealink SIP-T28P represents the next generation VoIP phone which designed for the business user who needs rich telephony features, friendly UI and super voice quality. It is equipped with the TI TITAN chipset, offers high definition voice quality through TI voice engine, HD handset, HD speaker and HD codec (G.722).

By the large, high-resolution graphical display, and together with all the 48 keys, SIP-T28P offers an excellent user experience to configure, make calls, express XML browser, etc. Moreover, to avoid problems with unwanted violations of your audio data, Yealink SIP-T28P supports the security standards TLS, SRTP, HTTPS, 802.1x, Open VPN and AES encryption which are necessary to protect against electronic eavesdropping and data theft.

[[Yealink_SIP-T28P|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====Yealink SIP-W73P (W70B) DECT====

[[File:Yealink_SIP-W73P.png|300px|thumb|left|Yealink SIP-W73P DECT IP Phone]]

'''Product:''' Yealink SIP-W73P DECT IP PHONE

'''Company:''' Yealink

'''Overview:''' The Yealink W70B is the DECT IP base station for small and medium-sized businesses. Paring with up to a total of 10 Yealink W73H/W56H/W59R/CP930W/color screen DDPhone (T54W+DD10K) DECT handsets, W70B allows you to enjoy superb mobility and efficient flexibility immediately as well as significantly eliminates additional wiring troubles and charges. A powerful chip ensuring a better and higher performance, this DECT IP base station not only supports up to 10 VoIP accounts and 20 simultaneous calls, but also speeds up its startup and signal connection, provides no perception upgrade, slashes its upgrade downtime as well.

[[Yealink_SIP-W73P|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====Yealink SIP-T54W====

[[File:Yealink_SIP-T54W.png|300px|thumb|left|Yealink SIP-T54W Prime Business Phone]]

'''Product:''' Yealink SIP-T54W

'''Company:''' Yealink

'''Overview:''' Yealink SIP-T54W is an easy-to-use Prime Business Phone with an adjustable 4.3-inch color LCD screen that you can easily and flexibly find the comfortable viewing angle according to the personal and environmental needs. With the built-in Bluetooth 4.2 and the built-in dual band 2.4G/5G Wi-Fi, the SIP-T54W IP Phone ensures you to keep up with the modern wireless technology and take the first chance in the future wireless age. Its built-in USB 2.0 port allows for USB recording or a direct wired/wireless USB headset or up to three Yealink EXP50 expansion modules connection. Benefitting from these features, the Yealink SIP-T54W is a powerful and expandable office phone that delivers optimum desktop efficient and productivity

[[Yealink_SIP-T54W|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====Yealink SIP-T46U====

[[File:Yealink_SIP-T46U.png|300px|thumb|left|Yealink SIP-T46U Ultra-elegant Gigabit IP Phone]]

'''Product:''' Yealink SIP-T46U

'''Company:''' Yealink

'''Overview:''' The SIP-T46U IP phone is an ultimate communication tool that has the better overall performance. The phone employs an appealing high-resolution TFT color display that looks brighter and more vibrant. United Yealink Optima HD Voice technology and wideband codec of Opus, the T46U awards you the superb audio quality and crystal-clear voice communications. Moreover, the T46U puts dual USB ports in a phone that makes Bluetooth, Wi-Fi, USB headset and USB recording come true, and you can use any two of them freely according to your needs.

[[Yealink_SIP-T46U|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====Yealink SIP-T33G====

[[File:Yealink_SIP-T33G.png|300px|thumb|left|Yealink SIP-T33G Classic Business IP Phone]]

'''Product:''' Yealink SIP-T33G

'''Company:''' Yealink

'''Overview:''' Yealink SIP-T33G offers support for 4 lines and includes local 5-way
conferencing. For its fashionable appearance as well as an extra-large 320x240-pixel color display with backlight, it brings comfortable operation experience and clear visual experience for users. Designed with a new powerful chip, it helps greatly improved work efficiency. Additional features include a dual-port Gigabit Ethernet with integrated PoE, EHS35 support for Yealink wireless headset, and adjustable multi-angle stand support. These features allow the SIP-T33G to be a high-quality but cost-effective classic IP phone that maximizes productivity in both small and large office environments.

[[Yealink_SIP-T33G|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

[[Category:Devices]]

===Zycoo ZP502===

[[File:ZP502.jpg|left|Zycoo ZP502]]

'''Product:''' Zycoo ZP502

'''Company:''' Zycoo ZP502

'''Overview:''' ZP502 VoIP Phone is ZYCOO business IP phone terminal which adopts SIP and IAX2 protocols.

With Broadcom solution, compatible with various Platforms such as Asterisk, FreePBX, Broadsoft, Cisco call manager, etc.

[[Zycoo ZP502|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

IP Phones

2023-09-25T06:47:36Z

Carlb: /* Polycom VVX 300, 400, etc */

== Blog Articles ==

* '''VoIP – Bring Your Own Device (BYOD)''': To be able to place or receive calls using VoIP, you may need a hardware setup that will allow you to use your regular phone (ATA) or a special phone that connects directly to VoIP networks (IP Phone). For more information, Take a peek at our blog article [https://wiki.voip.ms/article/VoIP_%E2%80%93_Bring_Your_Own_Device_(BYOD) VoIP – Bring Your Own Device (BYOD)]

* '''IP Phone:''' An IP Phone uses voice over IP (VoIP) technologies allowing telephone calls to be made over an IP network such as the Internet instead of the ordinary PSTN system. Still not clear? Take a peek at our blog article : [https://wiki.voip.ms/article/Back_to_Basics:_What_is_an_IP_Phone%3F Back to Basics - What is an IP Phone?]

'''Looking for a Analog Telephone Adapter (ATA)? [[ATA_Devices | Click here]] to see all models.'''

__TOC__

==IP Phones ==

===3COM 3108 Wireless Phone===

[[File:3com_3108_1.jpg|300px|thumb|left|3COM 3108 Wireless Phone]]

'''Product:''' 3COM 3108 Wireless Phone

'''Company:''' 3COM

'''Overview:''' The 3Com 3108 Wireless Telephone is a Session Initiation Protocol (SIP)-based wireless Voice over Internet Protocol (VoIP) telephone. SIP is an internationally recognized standard <nowiki>(IETF RFC 3261)</nowiki> for implementing VoIP. You can make and receive VoIP calls as long as your Wireless Telephone is registered with a SIP proxy server and you are operating it within range of an IEEE 802.11b/g enabled wireless network (WLAN). The SIP proxy server can belong to a wireless Internet Telephony Service Provider (ITSP) or corporate VoIP PBX system, such as the 3Com NBX® System.

[[3COM_3108_Wireless_Phone|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

===Aastra 6730i/6731i VoIP Phone===

[[File:Aastra6730i.jpg|300px|frame|left|Aastra 6730i VoIP Phone]]

'''Product:''' Aastra 6730i/6731i VoIP Phone

'''Company:''' Aastra

'''Overview:'''The Aastra 6730i, a new member of the carrier-grade, open-standards-based 67xi SIP portfolio, offers exceptional features and flexibility in an enterprise grade IP telephone. With a sleek, elegant design and a compact footprint, this multi-line SIP telephone delivers the advanced features and performance traditionally found only in higher priced products. Featuring a 3 line LCD display, the 6730i supports up to 6 lines with call appearances, offers advanced XML capability to access custom applications and is fully interoperable with leading IP-PBX platforms.

Supported by a host of Aastra configuration options, XML development tools, and continuous product enhancements via software releases, the value offered by the 6730i makes it is ideally suited for light telephone use for the small and home-based business applications.

[[Aastra_6730i_VoIP_Phone|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

===Audiocodes===

====400HD Series====

[[File:Audiocodes 420HD.jpg|300px|thumb|left|Audiocodes 420HD IP Phone]]

'''Product:''' Audiocodes 400HD Series

'''Company:''' Audiocodes

'''Overview:''' The [https://www.audiocodes.com/solutions-products/products/ip-phones AudioCodes 400HD series] of IP phones is a range of easy-to-use, feature-rich desktop devices for the service provider hosted services, enterprise IP telephony and contact center markets. Based on the same advanced, field-proven underlying technology as our other VoIP products, AudioCodes high quality IP phones enable systems integrators and end customers to build end-to-end VoIP solutions.

[[Audiocodes 400HD|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

===Cisco===

====Cisco 88XX & 68XX series====

[[File:8800_Series.png|300px|thumb|left|Cisco 8800 Series]]

'''Product:''' 88XX & 68XX series

'''Company:''' Cisco

'''Overview:''' The ''Cisco IP Phone 6800'' Series multiplatform phones are designed for affordability. They deliver reliable, business-grade audio, with Gigabit Ethernet integration and low power usage.

Ideal for customers with moderate to active VoIP needs, the 6800 Series phones are supported on Cisco-approved third-party unified communications as a service (UCaaS) providers.

The ''Cisco IP Phone 8800'' Series is a great fit for businesses of all sizes seeking secure, high-quality, full-featured VoIP. Select models provide affordable entry to HD video and support for highly-active, in-campus mobile workers. This advanced series provides flexible deployment options: on-premises, cloud and Cisco pre-approved third-party UCaaS providers.

[[Cisco IP Phone 68XX and 88XX|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====Cisco Linksys SPA942 NA====

[[File:SPA942-0.jpg|300px|thumb|left|Cisco Linksys SPA942 NA]]

'''Product:''' Linksys SPA942 NA

'''Company:''' Cisco

'''Overview:''' Four active lines and dual switched Ethernet ports offer incredible flexibility to the home or small-business Internet telephony user. Part of Cisco Small Business IP Phone Series, the SPA942 IP Phone can be configured to carry a unique phone number or extension for each of its lines or can be configured to use a shared number over multiple phones.

Based on the SIP standard, the SPA942 is able to easily integrate new features and services offered by providers. A high-resolution display makes for an easy menu- and web-based configuration.

[[Cisco_Linksys_SPA942_NA|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====Cisco SPA525G====

[[File:525g.jpg|300px|thumb|left|Cisco SPA525g Phone Adapter with Router]]

'''Product:''' Cisco SPA525G

'''Company:''' Cisco

'''Overview:''' The Cisco SPA525G 5-line IP Phone with Color Display is a full-featured VoIP (Voice over Internet Protocol) phone that provides voice communication over an IP network. It provides traditional features, such as call forwarding, redialing, speed dialing, transferring calls, conference calling, and accessing voice mail. Calls can be made or received with a handset, headset or speaker.
Your Cisco IP Phone provides a web interface for the phone user that allows you to configure some features of your phone by using a web browser.
This article will guide you through the steps for basic configuration to make it work with VoIP.ms

[[Cisco SPA525G|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====Cisco IP Phone 7940/7960====

[[File:Cisco_7960g.png|300px|thumb|left|Cisco IP Phone 7940/7960]]

'''Product:''' Cisco IP Phone 7940/7960

'''Company:''' Cisco

'''Overview:''' The Cisco IP Phone 7940/7960 is a full-featured telephone that provides voice communication over an IP network. This phone functions as a traditional analog phone, allowing you to place and receive telephone calls. This phone also supports features, such as network call forwarding, redialing, speed dialing, transferring calls, placing conference calls, and accessing voice mail. Phones require Power Over Ethernet (PoE) or a 48V AC Adapter to power up.

[[Cisco_IP_Phone_7940/7960|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====Cisco SPA30x and SPA50x series IP phones====

[[File:Cisco_spa504g.jpg|300px|thumb|left|Cisco SPA504G Phone]]

'''Product:''' Cisco SPA504G Phone

'''Company:''' Cisco

'''Overview:'''
The Cisco SPA504G is an office-style desk telephone with built-in voice over the Internet.

It is one in a series of similar models (SPA30x and SPA50x) which vary primarily in the number of lines (extensions) on the 'phone, power source (some models use power-over-Ethernet) and the availability of a second Ethernet connector. These devices are well-suited to offices and IP PBX applications. These do not provide a virtual line for connecting analog devices such as standard telephone handsets; they are instead self-contained to connect directly to VoIP.

[[Cisco_SPA504G_Phone|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

=== Dinstar===

==== Dinstar C60 Series ====

[[File:C60U_F.png|300px|thumb|left|Dinstar C60UP]]

C60 series are based on high innovative SIP technology, which is ideal for all kinds of business communication. It integrates with 132x64-pixel graphical LCD with back-light, elegant and intuitive user interface, which indicate you can enjoy good user experience.

[[Dinstar_C60_series|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

===Fanvil ===

====Fanvil X4G====

[[File:FanfillX4g.jpg|300px|thumb|left|Fanvil X4G]]

'''Product:''' Fanvil X4G

'''Company:''' Fanvil

'''Overview:''' The Fanvil X4G has a 2.8" main color screen and a secondary 2.4" DSS color screen. The user interface is sleek, colorful and easy to navigate. It has a one button call function and a call log and the ability to store 500 phonebook entries. The X4G's high compatibility supports various systems including 3CX, Avaya, OpenVox, NEC, Elastix, Asterisk, Matrix, Broadsoft, Epygi and more.

[[Fanvill X4G|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====Fanvil V62====

[[File:Fanvil-v62-VoIPms.png|300px|thumb|left|Fanvil V62]]

'''Product:''' Fanvil V62

'''Company:''' Fanvil

'''Overview:''' V62 is more than an efficient telephone but a delicate work of art, providing a smart and smooth business communication experience for enterprises. As the essential business phone featuring a graphical Dot-matrix screen with backlight and necessary VoIP features and other extended features, V62 is a great combination of elegant outside and powerful inside.

[[Fanvil V62|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====Fanvil V63====

[[File:Fanvil-v63-VoIPms.png|300px|thumb|left|Fanvil V63]]

'''Product:''' Fanvil V63

'''Company:''' Fanvil

'''Overview:''' V63 is more than an efficient telephone but a delicate work of art, providing a smart and smooth business communication experience for enterprises. As the essential business phone featuring 2.8 color screen and necessary VoIP features and other extended features, V63 is a perfect combination of elegant outside and powerful inside.

[[Fanvil V63|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====Fanvil V65====

[[File:Fanvil-v65-VoIPms.jpg|300px|thumb|left|Fanvil V65]]

'''Product:''' Fanvil V65

'''Company:''' Fanvil

'''Overview:''' V65 is more than an efficient telephone but a delicate work of art, providing a smart and smooth business communication experience for executives and managers. As the prime business phone featuring an adjustable screen and built-in Bluetooth 4.2 and 2.4G/5G Wi-Fi, V65 is a perfect combination of elegant outside and powerful inside.

[[Fanvil V62|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====Fanvil V67====

[[File:Fanvil-V67-VoIPms.jpg|300px|thumb|left|Fanvil V67]]

'''Product:''' Fanvil V67

'''Company:''' Fanvil

'''Overview:''' V67 is more than an efficient telephone but also a delicate work of art, which provides a more intelligent and elegant office operation experience for executives, managers and teleworkers. With brand new design, V67 features an adjustable touch screen and a keypad with colorful light effect that improve the beauty and comfort of office desktop.

[[Fanvil V67|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====Fanvil X4U-V2====

[[File:Fanvil-X4U-V2-VoIPms.jpg|300px|thumb|left|Fanvil X4U-V2]]

'''Product:''' Fanvil X4U-V2

'''Company:''' Fanvil

'''Overview:''' The X4/X4G is a feature-rich sip phone for business. The 4-Line IP Phone has been designed by pursuing ease of use in even the tiniest details. Dual 10/100 Mbps(X4G: 10/100/1000 Mbps) network ports with integrated PoE are ideal for extended network use. Delivering a superb sound quality as well as rich visual experience. With second DSS color screen, the IP Phone supports up to 30 DSS keys which improve work efficiency. Using standard encryption protocols to perform highly secure remote provisioning and software upgrades.

[[Fanvil X4U-V2|See Configuration Details]]
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====Fanvil X6U-V2====

[[File:Fanvil-X6U-V2-VoIPms.jpg|300px|thumb|left|Fanvil X6U-V2]]

'''Product:''' Fanvil X6U-V2

'''Company:''' Fanvil

'''Overview:''' Featuring three color displays, newly added line keys with LED light and built-in Bluetooth, Fanvil X6U provides the direct access to instructions, aiming to offering greater flexibility, productivity, to exceed the different demands of businesses. Wideband codec of G.722 and Opus in this device delivers you an immersive HD audio experience in both high band and low band with the network.

[[Fanvil X6U-V2|See Configuration Details]]
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===Linkvil by Fanvil===

====Linkvil W611W====

[[File:Linkvil_W611W_by_Fanvil.jpg|300px|thumb|left|Linkvil W611W by Fanvil]]

'''Product:''' Linkvil W611W by Fanvil

'''Company:''' Fanvil

'''Overview:''' LINKVIL W611W is a portable, elegant Wi-Fi phone designed for mobile communication applications. Certified to IP67 standard, W611W is highly waterproof, dustproof, and drop-safe from 1.8-meter height. It has an excellent performance in different environments with humidity and dust. W611W integrates Wi-Fi 6, bringing a superb wireless communication experience. Moreover, it integrates Bluetooth 5.0 for pairing with headsets and mobile devices. Installed with a rechargeable 1900mAh battery, W611W is ready for 9 hours’ talk time or 200 hours standby time. W611W is widely used in various wireless scenarios such as enterprises, shopping malls, residential areas, hotels and warehouses, providing users with a high-quality mobile communication experience.

[[Linkvil_W611W_by_Fanvil | See Configuration Details]]
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[[Category:Devices]]

=== Flyingvoice ===

====Flyingvoice====

[[File:Flyingvoice_Phones.jpg|300px|thumb|left|Flyingvoice]]

'''Product:''' '''FIP11C / FIP11CP''', '''FIP13G''', '''FIP14G''', '''FIP15G''', '''FIP10/FIP10P''', '''FIP12WP''', '''FIP16'''

'''Company:''' Flyingvoice

'''Overview:''' Flyingvoice is a provider of communication terminal equipment and one-stop VoIP CPE solutions. They offer a full range of VoIP products, such as VoIP phones, ATAs, gateways and routers for businesses and consumers. Their WiFI IP phones offer a wireless option, so you do not need a wired internet connection to either make or receive VoIP calls.

[[Flyingvoice|See Configuration Details]]
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[[Category:Devices]]

===Fortinet===

====FortiFone FON-570====

[[File:FON-570_Phone.jpg|300px|thumb|left|Fortinet FortiFone FON-570]]

'''Product:''' FortiFone FON-570

'''Company:''' Fortinet

'''Overview:''' Featuring a large 7” color touchscreen and premium HD call quality, this IP phone is great for efficient communications. Combined dedicated feature keys and programable keys expandable to 109, you have the flexibility to control your calls within your fingertips.

*7" color screen
*7 dedicated feature keys
*109 programable phone keys
*Full duplex speakerphone
*2 x 10/100/1000 Ethernet ports
*Integrated Power over Ethernet (PoE) support

[[Fortinet_FortiFone_FON-570|See Configuration Details]]
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====FortiFone FON-375====

[[File:FON-375_Phone.jpg|300px|thumb|left|Fortinet FortiFone FON-375]]

'''Product:''' FortiFone FON-375

'''Company:''' Fortinet

'''Overview:''' A reliable IP phone delivers HD sound quality, ideal for office workers who need efficient communications. An easy-to-read color screen and a programable second screen make it easy to display which lines are in use and who is on a call.

:*Dual color screens: 2.8" + 2.4”
:*8 dedicated feature keys
:*30 programable phone keys
:*Full duplex speakerphone
:*2 x 10/100/1000 Ethernet ports
:*Integrated Power over Ethernet (PoE) support

[[Fortinet_FortiFone_FON-375|See Configuration Details]]
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====FortiFone FON-175====

[[File:FON-175_Phone.jpg|300px|thumb|left|Fortinet FortiFone FON-175]]

'''Product:''' FortiFone FON-175

'''Company:''' Fortinet

'''Overview:''' A quality, two-line IP phone delivers reliable communications with HD audio quality. This entry-level business phone is easy to use that works in any office.

*2.4" color screen
*5 dedicated feature keys
*Full duplex speakerphone
*2 x 10/100 Ethernet ports
*Integrated Power over Ethernet (PoE) support

[[Fortinet_FortiFone_FON-175|See Configuration Details]]
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===Gigaset A510 IP===

[[File:Gigaset_a510_IP.jpg#file|300px|thumb|left]]

'''Product:'''Gigaset A510 IP

'''Company:'''Gigaset

'''Overview:'''The Gigaset A510 and C610 IP phones are fitting solutions if you are looking for the flexibility of VoIP and the convenience of using a cordless handset.

[[Gigaset_A510_IP| See Configuration Details]]
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===Grandstream===

====DP Series====

=====Grandstream DP715/DP710=====

[[File:Grandstream 715-710.png|300px|thumb|left]]

'''Product:''' Grandstream DP715/DP710

'''Company:''' Grandstream

'''Overview:''' The DP715/710 is the next generation of powerful, affordable, high quality and simple to configure DECT Cordless IPPhone for small business and residential users. Their compact size, superb voice quality, rich feature set, market leading price-performance and wide range radio coverage enable consumers to maximize the power of IP voice application and mobility for a minimum investment.

[[Grandstream_DP715/DP710| See Configuration Details]]
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=====Grandstream DP750/DP720=====

[[File:DP750-720.png|300px|thumb|left]]

'''Product:''' Grandstream DP750/DP720

'''Company:''' Grandstream

'''Overview:''' The DP750/720 base station and handsets allows you to deploy an immersive DECT environment that allows users to communicate free from their desktop using Grandstream’s DP720 DECT handsets. The DP750 pairs with up to 5 DP720s to create a powerful and mobile network solution with up to 10 lines per handset, and 5 concurrent calls per DECT system.

[[Grandstream_DP750/DP720| See Configuration Details]]
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====GXP Series====

=====Grandstream GXP1630 IP Phone=====

[[File:Gxp1630.jpg|300px|thumb|left|Grandstream GXP2120 IP Phone]]

'''Product:''' Grandstream GXP1630 IP Phone

'''Company:''' Grandstream

'''Overview:''' The GXP1630 comes equipped with a suite of VoIP features that are deployed in a clear and easy-to-use fashion. Focused primarily for low to medium call volumes and efficient call handling, its 3 line/SIP account design and 8 dual-colored BLF/speed dial keys gives this versatility. The GXP1630 also supports the best possible connection speeds and call quality with its dual Gigabit ports and HD audio on both speakerphone and handset. With other features such as its integrated PoE, 3 XML programmable soft keys and 4-way conferencing support, the GXP1630 is a high-quality and versatile Basic IP phone.

[[Grandstream_GXP1630|See Configuration Details]]
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=====Grandstream GXP2120 IP Phone=====

[[File:Gxp2110.png|300px|thumb|left|Grandstream GXP2120 IP Phone]]

'''Product:''' Grandstream GXP2120 IP Phone

'''Company:''' Grandstream

'''Overview:''' The Grandstream GXP2120 is a 6 line SIP Phone which features HD Voice hardware and software support and a large 320 x 160 backlit graphical LCD. The GXP2120 can handle 6 SIP accounts represented by 6 dual-color line keys and 4 XML programmable context-sensitive soft keys. In addition, the GXP2120 has 7 dual-color BLF extension keys for the most common calls and transfers making it an ideal phone for an office user with moderate to heavy interoffice calling needs.

[[Grandstream_GXP2120_IP_Phone|See Configuration Details]]
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=====Grandstream GXP2135 IP Phone=====

[[File:GXP2135-device.jpg|300px|thumb|left|Grandstream GXP2135 IP Phone]]

'''Product:''' Grandstream GXP2135 IP Phone

'''Company:''' Grandstream

'''Overview:''' The GXP2135 is the ideal selection for busy users who value call control, productivity and usability, and manage medium to heavy call volumes. Equipped with 8 lines and 4 SIP accounts, a 2.8-inch color LCD display, and 32 digital speed dial/BLF keys, the GXP2135 enables quick and powerful usability.

As all Grandstream IP phones do, the GXP2135 features state-of-the-art security encryption technology (SRTP and TLS). The GXP2135 supports a variety of automated provisioning options, including zero-configuration with Grandstream’s UCM series IP PBXs, encrypted XML files and TR-069, to make mass deployment extremely easy.

[[Grandstream GXP2135|See Configuration Details]]
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=====Grandstream GXP2170=====

[[File:Grandstream_GXP2170.png|300px|thumb|left|Grandstream GXP2170]]

'''Product:''' Grandstream GXP2170

'''Company:''' Grandstream

'''Overview:''' The GXP2170 is a powerful High-End IP phone that is ideal for busy users who handle high call volumes. Receptionists, administrators, sales staff and other call-intensive rolls can enjoy efficiency by utilizing the GXP2170’s 12 line keys, 4.3 inch color display LCD and 48 digital, on-screen speed dial/BLF keys.

[[Grandstream_GXP2170|See Configuration Details]]
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=====Grandstream GXP2200=====

[[File:Grandstream_GXP2200.png|300px|thumb|left|Grandstream GXP2200]]

'''Product:''' Grandstream GXP2200

'''Company:''' Grandstream

'''Overview:''' The GXP2200 is one of the most advanced AndroidTM desktop IP phones available on the market today. The innovative phone includes the AndroidTM version 2.3 operating system with a 4.3 inch capacitive touchscreen LCD and the ability to host 6 SIP accounts. Web applications such as news, social media sites, and games can be downloaded directly via Google Play Store, and applications can be created to fit any need and downloaded directly to the phone for customized use.

[[Grandstream_GXP2200|See Configuration Details]]
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====GRP Series====

: '''Note that the GRP2615 configurations are the same to all GRP series.'''

=====Grandstream GRP2615=====

[[File:Grandstream_GRP2615.png|300px|thumb|left|Grandstream GRP2615]]

'''Product:''' Grandstream GRP2615

'''Company:''' Grandstream

'''Overview:''' The GRP2615 is a high-end carrier-grade IP phone featuring a sleek design and a suite of next-generation features including integrated Wi-Fi, Bluetooth support, 40 multipurpose keys (MPKs), an available extension module, dual Gigabit ports and more. This device features a large 4.3 inch color LCD with swappable face plates to allow for easy logo customization. For cloud provisioning and centralized management, the GRP2615 is supported by Grandstream’s Device Management System (GDMS), which provides a centralized interface to configure, provision, manage and monitor deployments of Grandstream endpoints.

[[Grandstream_GRP2615|See Configuration Details]]
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====WP Series====
:''There are multiple IP phones in this series, which are all very similar in configuration. All are self-contained IP handsets which connect directly to Wi-Fi and register directly with any of the VOiP.ms servers.''

=====Grandstream WP810=====

[[File:Grandstream WP810.png|150px|thumb|left|Grandstream WP810]]

'''Product:''' Grandstream WP810

'''Company:''' Grandstream

'''Overview:''' The WP810, WP820, WP822 and WP825 are cordless SIP IP phones which connect directly to the local network using dual-band Wi-Fi, eliminating the need for a cordless base station or other hardware. Effectively, the WP810 (with its charging cradle) is a self-contained two-line SIP solution.

[[Grandstream WP810, WP820, WP822 and WP825|See Configuration Details]]
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===Innomedia===

====BuddyTalk BT110====

[[File:Buddy_Talk_BT110_Innomedia.jpg|250px|thumb|left|Innomedia BuddyTalk BT110]]

'''Product:''' BuddyTalk BT110

'''Company:''' Innomedia

'''Overview:''' Powered by the Amazon Alexa Voice Service (AVS), the BuddyTalk series of products supports a broad suite of AVS enabled smart speaker features.
Equipped with advanced audio processing and VoIP technologies, BuddyTalk devices are intelligent speakerphones that deliver unprecedented flexibility in calling, superior voice quality, and high levels of security.

[[BuddyTalk_-_BT110|See Configuration Details]]
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===Konftel===

==== Konftel 300Wx IP ====
[[File:Konftel-300Wx-IP.png|300px|thumb|left|Konftel 300Wx IP]]

'''Product:''' Konftel 300Wx IP

'''Company:''' Konftel

'''Overview:''' The Konftel 300Wx IP wireless conference phone allows you to hold conference calls in HD quality wherever is convenient for you – without worrying about network and power outlets. Reliable and secure DECT technology. The accompanying IP DECT base can handle up to 20 registered Konftel 300Wx devices and five ongoing calls.

The rechargeable battery ensures more than 60 hours of call time, so you can talk for a full working week without recharging! A USB port makes the Konftel 300Wx ready for all the apps and services we use to communicate and collaborate via computers. Combine meeting apps and regular phone calls. OmniSound® delivers superb sound quality

[[Konftel 300Wx|See Configuration Details]]
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==== Konftel 300IPx ====

[[File:Konftel300ipx-conference-phone.jpg|300px|thumb|left|Konftel 300IPx]]

'''Product:''' Konftel 300IPx

'''Company:''' Konftel

'''Overview:''' The [https://www.konftel.com/en/products/konftel-300ipx KONFTEL 300IPx] together with the Konftel Unite app brings a whole new easiness to conference calls. It is highly intuitive and based on our natural mobile behavior. The new generation of IP conference phone is – The Art of Easiness.

[[Konftel 300IPx|See Configuration Details]]
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===Panasonic===

====Panasonic KX-TGP 550====

[[File:Pana550b.jpg|300px|thumb|left|Panasonic KX-TGP 550]]

'''Product:''' Panasonic KX-TGP 550

'''Company:''' Panasonic

'''Overview:''' The Panasonic KX-TGP550 responses to the needs of SIP IP-Centrix/Hosted PBX systems and Asterisk users. Conveniently, no need to set up a system telephone at every base. This system also enables you to use a range of convenient services provided by the carrier such as Call Forward, Voice Mail, etc.

*Support for 3 simultaneous network conversations
*Up to 6 DECT cordless handsets
*Support for up to 8 SIP registrations (e.g. up to 8 DID lines or extensions)
*Hands-free speakerphone on a cordless handset

[[Panasonic KX-TGP 550|See Configuration Details]]
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====Panasonic KX-HDV130C====

[[File:Panasonic_KXHDV130_01.jpg|300px|thumb|left|Panasonic KX-HDV130C]]

'''Product:''' Panasonic KX-HDV130C

'''Company:''' Panasonic

'''Overview:''' The Panasonic KX-HDV130 SIP desk phone delivers the ideal balance of low cost and high quality, along a range of value added features.

*Support for up to 2 SIP registrations (e.g. up to 2 DID lines or extensions)
*Support for 3 simultaneous network conversations (3-way conferencing)*
*2 Programmable keys / Line keys
*Hands-free speakerphone on a cordless handset

[[Panasonic KX-HDV130C|See Configuration Details]]
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====Panasonic KX-HDV230====

[[File:Panasonic_KXHDV230_01.jpg|300px|thumb|left|Panasonic KX-HDV230]]

'''Product:''' Panasonic KX-HDV230

'''Company:''' Panasonic

'''Overview:''' The Panasonic KX-HDV230 IP phone offers streamlined functions and the high definition voice quality that's essential for effective communication.

*Support for up to 6 SIP registrations (e.g. up to 6 DID lines or extensions)
*24 Flexible Function Keys
*2 ethernet ports 10/100/1000 Base -T
*Large LCD with Backlight

[[Panasonic KX-HDV130C|See Configuration Details]]
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====Panasonic KX-HDV330====

[[File:Panasonic_KXHDV330_01.jpg|300px|thumb|left|Panasonic KX-HDV330]]

'''Product:''' Panasonic KX-HDV330

'''Company:''' Panasonic

'''Overview:''' The Panasonic KX-HDV330 is a multi-functional business SIP phone equipped with a colour touch panel for intuitive operation.

*Large LCD with Backlight
*Built-in Bluetooth®
*Support for up to 12 SIP registrations (e.g. up to 12 DID lines or extensions)
*24 Flexible Function Keys
*Hands-free speakerphone on a cordless handset

[[Panasonic KX-HDV130C|See Configuration Details]]
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===Pirelli DP-L10===

[[File:Pirelli1.jpg|300px|thumb|left|Pirelli DP-L10]]

'''Product:''' Pirelli DP-L10

'''Company:''' Pirelli

'''Overview:''' Pirelli's DualPhone DP-L10 phone combines GSM tri-band capabilities with SIP-based Wireless VoIP via an integrated WLAN 802.11b/g interface, combining cell phone technology with the added advantage of transporting phone calls over the Internet.

The Dual Phone enables mobile phone features such as Internet browsing, e-mail, SMS, and MMS, which can also be used through a fixed-broadband connection, providing the same mobile services with the added benefits of greater bandwidth and cost savings.

[[Pirelli DP-L10|See Configuration Details]]
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===Polycom===

====Poly Edge B10====

[[File:PolyEdgeB10.png|300px|thumb|left|Poly Edge B10]]

'''Product:''' Poly Edge B10

'''Company:''' Poly

'''Overview:''' Sleek design meets Poly pro-grade audio at a shockingly affordable price: That’s what makes the Poly Edge B Series IP Phones the genius choice for any growing business. Easy to use with illuminated keys where you need them. Plug-and-play provisioning and hardcore reliability make it pure value in a low-cost business phone.

[[Poly_Edge_B10|See Configuration Details]]

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====Polycom SoundStation IP 4000 Conference Phone====

[[File:Polycom-4000-0.gif|300px|thumb|left|Polycom SoundStation IP 4000 Conference Phone]]

'''Product:''' Polycom SoundStation IP 4000 Conference Phone

'''Company:''' Polycom

'''Overview:''' the SoundStation IP 4000 SIP voice conferencing unit is the answer for organizations that are ready for the
benefits and versatility of a SIP enabled business. Designed for offices or small to medium-sized conference rooms, the SoundStation IP 4000 SIP provides remarkable room coverage. You can speak naturally from up to 10-feet away from a microphone and still be heard clearly on the far end of the call. Need coverage for a larger room? The optional extension microphones offer an increased pickup for larger rooms. Plus, with gated microphone technology, echo and background noise is almost entirely eliminated.

The SoundStation IP 4000 SIP also provides familiar features that are easy to use. The menu-driven user interface, viewable on a high-resolution backlit LCD, offers convenient access to business telephony functions you use most including transfer, hold, redial, and conference.

[[Polycom_SoundStation_IP_4000_Conference_Phone|See Configuration Details]]
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====Polycom SoundPoint IP 501, 550, 650, etc.====

[[File:Polycom501.jpg|258px|thumb|left|Polycom SoundPoint IP 501]]

'''Product:''' Polycom SoundPoint IP 501, 550, 650, etc.

'''Company:''' Polycom

'''Overview:''' The SoundPoint IP series are multi-line (3-6 lines depending on model) Voice over IP telephones that seamlessly integrate with IP PBX and softswitch vendors’ IP solutions.

As protocols develop and standards evolve, it’s easy to update the phone in the field via a software download, thus enabling new features and functionality for the phone and protecting your investment.

Whether you deploy MGCP or SIP standards, Polycom offers a solution that fits all your business communication needs.

[[Polycom_SoundPoint_IP_501|See Configuration Details]]
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====Polycom SoundPoint IP 331====

[[File:Polycom_Soundpoint_IP_331.png|258px|thumb|left|Polycom SoundPoint IP 331]]

'''Product:''' Polycom SoundPoint IP 331

'''Company:''' Polycom

'''Overview:''' The Polycom IP 331 is engineered to make installation, configuration, and upgrades as simple and efficient as possible. The phones' standard base stand can be reversed to become a wall mount, eliminating the need for a separate accessory. Built-in IEEE 802.3af PoE circuitry and a dual-port Ethernet switch enables flexible deployment options and savings on cabling expenses.

[[Polycom_Soundpoint_IP_331|See Configuration Details]]
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====Polycom SoundPoint IP 601====

[[File:Voipms-polycom601.jpg|258px|thumb|left|Polycom SoundPoint IP 601]]

'''Product:''' Polycom SoundPoint IP 601

'''Company:''' Polycom

'''Overview:''' The 6-line Polycom® SoundPoint IP™ 601 offers industry-leading functionality and call handling unmatched voice quality an intuitive user interface & expandability to 12 lines!

[[Polycom_SoundPoint_IP_601|See Configuration Details]]
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====Poly VVX-D230====

[[File:Poly-vvx-d230-base.jpg|250px|thumb|left|Poly VVX-D230 cordless base]]

'''Product:''' Poly VVX-D230 cordless DECT phone

'''Company:''' Poly (Obahai, Polycom)

'''Overview:''' The VVX series are SIP desk phones aimed at small to mid-sized business; the VVX-D230 is a cordless system on which one base may control up to ten handsets.

[[Poly VVX-D230|See Configuration Details]]
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====Poly VVX 300, 400, etc====

[[File:Vvx300.png|250px|thumb|left|Polycom VVX 300 Series]]

'''Product:''' Polycom VVX Series

'''Company:''' Polycom

'''Overview:''' The VVX series provides high-quality audio (HD Voice) and video communications from 6 lines and up.

[[Polycom VVX 300, 400, etc|See Configuration Details]]
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=== Positron IP phones ===

[[File:PositronLogo.jpeg|250px|thumb|left|Positron IP phones]]

'''Product:''' Positron

'''Company:''' [http://www.positrontelecom.com/products.html Positron]

'''Overview:''' Positron IP Phones is an affordable next-generation SIP phone including wideband audio support, ethernet ports and integrated Power over Ethernet.

All the IP Phones are optimized for executives, administrative assistants and those working with bandwidth-intensive applications on collocated PCs. Line keys can also be configured as IP PBX features such as BLF, SCA, Intercom, Call Pickup, Call Park, and many others. The high-resolution screen and the HD voice features provide a high quality visual and authentic audio experience.

==== Positron IP304 ====

[[File:IP304.png |250px|thumb|left|Positron IP304]]

'''Product:''' Positron IP304

'''Company:''' [http://www.positrontelecom.com/products.html Positron]

'''Overview:''' Positron IP304 is an affordable next-generation SIP phone with wideband audio support, dual Ethernet port and integrated Power over Ethernet.

The IP304 enterprise VoIP phone is Positron’s entry-level phone with 3 VoIP accounts. Its functionalities make it easy to place calls on hold, to transfer calls and to initiate a conference call.

With HD audio for both headset and speakerphone, IP304 provides a clear authentic audio experience

[[Positron IP304 | View configuration for Positron IP304]]
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==== Positron IP304C ====

[[File:IP304C.png |250px|thumb|left|Positron IP304C]]

'''Product:''' Positron IP304C

'''Company:''' [http://www.positrontelecom.com/products.html Positron]

'''Overview:''' Positron IP304C is an innovative enterprise-level IP Phone that features 4 line keys, color display, 3.5” TFT-LCD with 480 x 320 pixel. It supports up to a 5-way conference.

IP304C is optimized for executives, administrative assistants and those working with bandwidth-intensive applications on collocated PCs. Line keys can also be configured as IP PBX features such as BLF, SCA, Intercom, Call Pickup, Call Park, and many others.

The high-resolution TFT-LCD screen and the HD voice features provide a high quality visual and authentic audio experience.

[[Positron IP304C | View configuration for Positron IP304C]]
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==== Positron IP408 ====

[[File:IP408.png |250px|thumb|left|Positron IP408]]

'''Product:''' Positron IP408

'''Company:''' [http://www.positrontelecom.com/products.html Positron]

'''Overview:''' Positron IP408 is an affordable next-generation SIP 2.0 phone including wideband audio support and WAN/LAN Ethernet ports with route and bridge mode.

The IP408 enterprise VoIP phone supports 4 SIP lines. Its functionalities make it easy to place calls on hold, to transfer calls and to initiate a conference call.

With HD audio for both headset and speakerphone, IP408 provides a clear authentic audio experience

[[Positron IP408 | View configuration for Positron IP408]]
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==== Positron IP410C ====

[[File:IP410C.png |250px|thumb|left|Positron IP410C]]

'''Product:''' Positron IP410C

'''Company:''' [http://www.positrontelecom.com/products.html Positron]

'''Overview:''' Positron IP410C is an affordable next-generation SIP Phone that features 4 line keys, 10 programmable extension keys, color display, wideband audio support and dual Ethernet port with integrated Power over Ethernet.

IP410C is optimized for executives, administrative assistants and those working with bandwidth-intensive applications on collocated PCs. Line keys can also be configured as IP PBX features such as BLF, SCA, Intercom, Call Pickup, Call Park, and many others.

The high-resolution TFT-LCD screen and the HD voice features provide a high quality visual and authentic audio experience.

[[Positron IP410C | View configuration for Positron IP410C]]
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==== Positron IP410G ====

[[File:IP410G.png |250px|thumb|left|Positron IP410G]]

'''Product:''' Positron IP410G

'''Company:''' [http://www.positrontelecom.com/products.html Positron]

'''Overview:''' Positron IP410G is an innovative enterprise-level color IP Phone that features 4 line keys, 10 programmable extension keys, color display, 3.5” TFT-LCD with 480*320 pixel, wideband audio support and dual Gigabit Ethernet port with integrated Power over Ethernet.

IP410G is optimized for executives, administrative assistants and those working with bandwidth-intensive applications on collocated PCs. Ten programmable keys can also be configured as IP PBX features such as BLF, SCA, Intercom, Call Pickup, Call Park, and many others.

The high-resolution TFT-LCD screen and the HD voice features provide a high quality visual and authentic audio experience.

[[Positron IP410G | View configuration for Positron IP410G]]
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===Siemens Gigaset C450-Ip===

[[File:Siemens_gigaset_c450_IP.jpg|300px|thumb|left|Siemens Gigaset C450-Ip]]

'''Product:''' Siemens Gigaset C450-Ip

'''Company:''' Siemens

'''Overview:''' The Gigaset C450 is an entry phone in the Siemens Gigaset product line, the addition of IP connectivity via a standard Ethernet 10/100 BaseT interface and SIP protocol within a reasonable price bracket, opens large applications where wireless operation and SIP are required.
The Ethernet connectivity allows the unit to work without a PC.

The dual mode (SIP/PSTN) enables incoming call on a legacy interface with existing directory number and safe emergency outgoing calls.

The presentation of the unit with an independent base and a dedicated phone charger/support will simplify cabling.

[[Siemens Gigaset C450-Ip|See Configuration Details]]
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===Snom===

====Snom 320 VoIP Phone====

[[File:Snom320.png|300px|frame|left|Snom 320 VoIP Phone]]

'''Product:''' Snom 320 VoIP Phone

'''Company:''' Snom

'''Overview:'''Ideal for the office and everyone who spends a lot of time on the phone, the snom 320 is an affordable, yet powerful SIP business phone with a built-in, full-duplex speakerphone and three-party conference bridging.

[[SNOM 320|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====Snom m3 VoIP Phone====

[[File:Snom_m3.png|300px|thumb|left|Snom m3 VoIP Phone]]

'''Product:''' Snom m3 VoIP Phone

'''Company:''' Snom

'''Overview:''' The snom m9 is the next generation DECT handheld (CAT-iq) that empowers users with the convenience of wireless communication along with the widely accepted benefits and feature richness of Voice-over-IP telephony.

The snom m9 promises to deliver excellent speech quality through digital and wideband audio (klarVOICE).

By combining professional functions of versatile business communication with the intuitive features of the mobile-carrier world, the snom m9 is ideally suited for professional and private use alike. With features such as hands-free mode, calling line identification (CLI) by displaying name, number, and image of the caller as well as typical mobile-phone features such as Address Book, Calendar, Calculator and Alarm function, the snom m9 provides the perfect blend of mobility and accessibility.

[[Snom_m3_VoIP_Phone|See Configuration Details]]
<br clear="all" />

====SNOM C520====

[[File:snom_c520.png|300px|thumb|left|C520]]

'''Product:''' SNOM C520 Conferencing

'''Company:''' Snom

'''Overview:''' With its modern and sleek design, the C520 fits seamlessly into your working day. Two detachable DECT microphones can be positioned freely or carried in the room as required to ensure the best sound and voice quality.

Built-in charging stations with magnetic bays directly on the base station mean both microphones are always charged and ready for use in the next meeting. The conference phone also features automatic volume control and digital noise reduction so that all call participants can be understood in best sound quality.

[[Snom C520|See Configuration Details]]
<br clear="all" />

====SNOM professional D7XX====

[[File:Snom.jpg|300px|thumb|left|Snom D7XX]]

'''Products:''' D120, D717, D735, D785

'''Company:''' Snom

'''Overview:''' The [https://www.snom.com/en/ip-phones/desk-phones/d7xx-series/ professional D7XX] Series telephones are both aesthetically appealing and highly practical, meeting business requirements when a telephone is a key tool in daily work.

These high-performance devices are future-proofed and provide the best in Wideband HD audio, ensuring crystal clear sound quality. They are Bluetooth compatible to meet the connectivity requirements of today’s offices.

[[Snom IP Phones|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====SNOM M100 KLE====

[[File:M100.jpg|300px|thumb|left|M100 KLE]]

'''Products:''' M100 KLE SIP DECT 4-Line Base Station

'''Company:''' Snom

'''Overview:''' The [https://www.snomamericas.com/en/pd/ip-phones/m-series/m-kle-series/m100-kle/ M100 KLE SIP DECT 4-Line Base Station] supports up to 10 phones in the Snom KLE DECT 4-Line Series, including the M10 and M10R SIP DECT 4-Line handsets and the M18 KLE SIP DECT 4-Line deskset. This cordless family of phones features four programmable LED backlit line keys on the handsets and desk sets.

With key system emulation, the M100 KLE Series handles shared line appearances locally without the need for SCA (shared called appearances) support from your provider. This allows an easy and intuitive method for your customers to see incoming calls, hold calls, and resume calls from any handset or deskset with a simple press of a button.

[[M100_KLE_SIP_DECT_4-Line_Base_Station|See Configuration Details]]

<div style="width:100%;overflow:hidden;clear:both"></div>

=== Vtech ===

==== Vtech Conference Station ====

[[File:VCS754-thumb.PNG|300px|thumb|left|Vtech VCS Series]]

'''Product:''' VCSV752 & CS754

'''Company:''' Vtech

'''Overview:''' The [https://businessphones.vtech.com/pd/3439/VCS754-ErisStation-SIP-Conference-Phone-with-Four-Wireless-Mics Vtech VCS754 ErisStation] conference phone features a compact, all-in-one design makes it easy to keep everything together—no clutter, no hassle. Built-in charging stations with magnetic bays ensure the microphones are charged and available for the next meeting.

[[Vtech Conference Station|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==== Vtech VSP Series ====

[[File:VSP736 ErisTerminal.jpg|300px|thumb|left|VSP Series]]

'''Products:''' VSP600 - VSP715 - VSP725 - VSP726 - VSP736

'''Company:''' Vtech

'''Overview:''' The [https://businessphones.vtech.com/products/sip-phones/vsp700 Vtech VSP700 Series] comes with all the essential features you need to keep pace with your business and your budget. Depending on the model, support two to six SIP accounts with these easy-to-use phones.

[[Vtech VSP Series|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

===Yealink===

====Yealink Voice Solutions====

[[File:Yealink_easyVoip.png|300px|thumb|left|Yealink SIP-T28P]]

'''Product:''' Yealink W60B, Yealink T21, Yealink T42S

'''Company:''' Yealink

'''Overview:''' Yealink offers solutions for each customer's needs, starting from basic to more complex ones.

[[Yealink Voice Solutions|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====Yealink SIP-T28P (VSRF)====

[[File:Yealink-sip-t28p.jpg|300px|thumb|left|Yealink SIP-T28P]]

'''Product:''' Yealink SIP-T28P

'''Company:''' Yealink

'''Overview:''' Yealink SIP-T28P represents the next generation VoIP phone which designed for the business user who needs rich telephony features, friendly UI and super voice quality. It is equipped with the TI TITAN chipset, offers high definition voice quality through TI voice engine, HD handset, HD speaker and HD codec (G.722).

By the large, high-resolution graphical display, and together with all the 48 keys, SIP-T28P offers an excellent user experience to configure, make calls, express XML browser, etc. Moreover, to avoid problems with unwanted violations of your audio data, Yealink SIP-T28P supports the security standards TLS, SRTP, HTTPS, 802.1x, Open VPN and AES encryption which are necessary to protect against electronic eavesdropping and data theft.

[[Yealink_SIP-T28P|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====Yealink SIP-W73P (W70B) DECT====

[[File:Yealink_SIP-W73P.png|300px|thumb|left|Yealink SIP-W73P DECT IP Phone]]

'''Product:''' Yealink SIP-W73P DECT IP PHONE

'''Company:''' Yealink

'''Overview:''' The Yealink W70B is the DECT IP base station for small and medium-sized businesses. Paring with up to a total of 10 Yealink W73H/W56H/W59R/CP930W/color screen DDPhone (T54W+DD10K) DECT handsets, W70B allows you to enjoy superb mobility and efficient flexibility immediately as well as significantly eliminates additional wiring troubles and charges. A powerful chip ensuring a better and higher performance, this DECT IP base station not only supports up to 10 VoIP accounts and 20 simultaneous calls, but also speeds up its startup and signal connection, provides no perception upgrade, slashes its upgrade downtime as well.

[[Yealink_SIP-W73P|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====Yealink SIP-T54W====

[[File:Yealink_SIP-T54W.png|300px|thumb|left|Yealink SIP-T54W Prime Business Phone]]

'''Product:''' Yealink SIP-T54W

'''Company:''' Yealink

'''Overview:''' Yealink SIP-T54W is an easy-to-use Prime Business Phone with an adjustable 4.3-inch color LCD screen that you can easily and flexibly find the comfortable viewing angle according to the personal and environmental needs. With the built-in Bluetooth 4.2 and the built-in dual band 2.4G/5G Wi-Fi, the SIP-T54W IP Phone ensures you to keep up with the modern wireless technology and take the first chance in the future wireless age. Its built-in USB 2.0 port allows for USB recording or a direct wired/wireless USB headset or up to three Yealink EXP50 expansion modules connection. Benefitting from these features, the Yealink SIP-T54W is a powerful and expandable office phone that delivers optimum desktop efficient and productivity

[[Yealink_SIP-T54W|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====Yealink SIP-T46U====

[[File:Yealink_SIP-T46U.png|300px|thumb|left|Yealink SIP-T46U Ultra-elegant Gigabit IP Phone]]

'''Product:''' Yealink SIP-T46U

'''Company:''' Yealink

'''Overview:''' The SIP-T46U IP phone is an ultimate communication tool that has the better overall performance. The phone employs an appealing high-resolution TFT color display that looks brighter and more vibrant. United Yealink Optima HD Voice technology and wideband codec of Opus, the T46U awards you the superb audio quality and crystal-clear voice communications. Moreover, the T46U puts dual USB ports in a phone that makes Bluetooth, Wi-Fi, USB headset and USB recording come true, and you can use any two of them freely according to your needs.

[[Yealink_SIP-T46U|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====Yealink SIP-T33G====

[[File:Yealink_SIP-T33G.png|300px|thumb|left|Yealink SIP-T33G Classic Business IP Phone]]

'''Product:''' Yealink SIP-T33G

'''Company:''' Yealink

'''Overview:''' Yealink SIP-T33G offers support for 4 lines and includes local 5-way
conferencing. For its fashionable appearance as well as an extra-large 320x240-pixel color display with backlight, it brings comfortable operation experience and clear visual experience for users. Designed with a new powerful chip, it helps greatly improved work efficiency. Additional features include a dual-port Gigabit Ethernet with integrated PoE, EHS35 support for Yealink wireless headset, and adjustable multi-angle stand support. These features allow the SIP-T33G to be a high-quality but cost-effective classic IP phone that maximizes productivity in both small and large office environments.

[[Yealink_SIP-T33G|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

[[Category:Devices]]

===Zycoo ZP502===

[[File:ZP502.jpg|left|Zycoo ZP502]]

'''Product:''' Zycoo ZP502

'''Company:''' Zycoo ZP502

'''Overview:''' ZP502 VoIP Phone is ZYCOO business IP phone terminal which adopts SIP and IAX2 protocols.

With Broadcom solution, compatible with various Platforms such as Asterisk, FreePBX, Broadsoft, Cisco call manager, etc.

[[Zycoo ZP502|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

Poly VVX-D230

2023-09-25T06:21:47Z

Carlb: Created page with "The VVX-D230's base can control ten DECT handsets Aimed at small business users, the Poly (Polycom, Obahai) VVX-D230 is a DECT [[IP Phones|..."

[[Image:Poly-vvx-d230-base.jpg|thumb|The VVX-D230's base can control ten DECT handsets]]
Aimed at small business users, the Poly (Polycom, Obahai) VVX-D230 is a DECT [[IP Phones|SIP phone]] with support for up to ten cordless handsets — including the one handset which is packaged with the base.

This system (OBAHAI/VVXD230) is intended to complement the [[Polycom VVX 300, 400, etc|VVX-series]] desk phones and the [[Poly Trio 8800]] conference room speakerphone.

==Configuration==
# If the cordless handsets are not already paired to the base, press and hold the 'find' button on the base for at least five seconds. Then, from the handset, go to Menu→Settings→Registration→Register.
# Configuration of the system is done via a web interface. From any paired handset, go to Menu→Settings→Basestation Info to obtain the IPv4 address of the base.
# Open a web browser to this address and log in with username 'admin' and default password 'admin'.

From the "Setup Wizard" page, change the following parameters (changes will not be applied until you save them and reboot the device). Be sure to uncheck the "default" check box for each item you intend to modify:
*LocalTimeZone - set to your local time zone, for instance GMT-5:00 for (Eastern Time). Factory default is GMT-8:00 (Pacific Time, California).
*ITSP A SIPProxyServer - atlanta.voip.ms (or one of the other multiple [[servers]] - this must match the same server as your [[DID Troubleshooting|DID]] configuration).
*ITSP A DigitMap - (optional) (1xxxxxxxxxx|<1555>[2-9]xxxxxxS3|<1>[2-9]xxxxxxxxx|011xx.|xx.|(Mipd)|[^*#]@@.) (as an example, replacing 1-555 with your local area code) will enable both 7 and 10-digit dial for North American numbers. See the manufacturer's technical reference manual for details of the syntax.
*Phone1 PrimaryLine - "SP1 service"
*SP1 ITSP Profile - A
*SP1 AuthUserName - 123456_1 (your voip.ms user number and a [[Sub Accounts|subaccount]] number, in this example user #123456 subaccount #1
*SP1 AuthPassword - the password associated with your voip.ms subsccount
If you have additional virtual lines, repeat the same configuration steps for each subaccount using SP2, SP3... up to a maximum of eight lines and ten handsets.

From the "SP1 Service" page, change the following settings for the first virtual "line" (the others will be similar):
*X_DisplayNumber - A short text string identifying this line on the local handset screen. Typically the seven-digit NXX-XXXX local number, although any value will do.
*X_AcceptSipFromRegistrarOnly - YES (to keep [[Sip Scanner Ghost Calls|spurious]] calls out)
*X_KeepAliveEnable - YES (if you are behind NAT)
*X_KeepAliveExpires - 180 (three minutes) is reasonable
*DirectoryNumber - NPA-NXX-XXXX, the directory number associated with this line
*AuthUserName, AuthPassword - should already be set from the "Setup Wizard" above; if not, set them here.
*CallerIDName - your name (15 alphanumeric chars max, no punctuation but spaces are allowed) as you want it to appear on [[Caller ID]] for outbound calls.
*X_CheckVoiceMailNumber - *97
*MessageWaiting - YES

Save everything before leaving each page of the setup. When you are finished making changes, reboot the device.

Try a test call, such as the [[Dialing Codes|echo test]] (press [4] [4] [4] [3] and the [green] button). You should hear your own voice played back just as it was received.

==See alos==
* [[Polycom VVX 300, 400, etc|VVX-series]]
* [[Poly Trio 8800]]

==Documentation==
This page is merely a brief overview with just enough information to connect the handsets to VoIP.ms; see the manufacturer's website (poly.com) for more extensive documentation:

* Quick start: [https://www.poly.com/content/dam/www/products/support/voice/vvx-dect-ip-phones/setup/vvx-d230-hs-bs-class-a-qsg-access.pdf handset + base], [https://www.poly.com/content/dam/www/products/support/voice/vvx-dect-ip-phones/setup/vvx-d230-hs-chg-class-a-qsg-access.pdf expansion handset + charger]
* [https://www.poly.com/content/dam/www/products/support/voice/vvx-dect-ip-phones/user/vvxd230-adminguide-710.pdf Admin guide] and [https://www.poly.com/content/dam/www/products/support/voice/vvx-dect-ip-phones/user/vvxd230-tech-ref-710.pdf technical reference] manual from Poly.com

[[Category:SIP phones]]

File:Poly-vvx-d230-base.jpg

2023-09-25T05:15:15Z

Carlb: Polycom (Obahai) VVX-D230 cordless DECT handset with base. As one base can control ten DECT cordless handsets, the other handsets in the system come with just a charging cradle, not the full base. Poly is a division of Hewlett-Packard. image is copyright

Polycom (Obahai) VVX-D230 cordless DECT handset with base. As one base can control ten DECT cordless handsets, the other handsets in the system come with just a charging cradle, not the full base. Poly is a division of Hewlett-Packard. image is copyright to the manufacturer.

Grandstream GXP2135

2023-09-16T10:02:21Z

Carlb:

[[File:GXP2135-device.jpg|300px|thumb|Grandstream GXP2135]]
High-profile desktop phones built for the busy user. The GXP21xx series are sleek high-end [[IP Phones|IP phones]] that will bring higher levels of communication productivity to a network. Their modern design, high call capacity, and rich functionality make them the ideal choice for the call-intensive worker.

IP phones in Grandstream's [https://documentation.grandstream.com/article-categories/gxp21xx/ GXP21xx series]. all of which are similar in configuration, include:

{|
||
||GXP2130
||GXP2140
||GXP2160
||GXP2170
||GXP2135
|-
||LCD Display
||320 × 240
||480 x 272
||480 x 272
||480 x 272
||320 × 240
|-
||LCD Backlight
||Yes
||Yes
||Yes
||Yes
||Yes
|-
||Number of Lines
||3
||4
||6
||12
||8
|-
||Programmable Hard Keys
||8
||N/A
||24
||48
||32
|-
||Programmable Softkeys
||4
||5
||5
||5
||4
|-
||Extension Module
||No
||Up to 4 EXT Boards
||No
||Up to 4 EXT Boards
||No
|}

== Configuration ==

'''SIP ACCOUNT'''

Once you have connected your device to your local connection, it will be necessary to obtain the local IP address that was designated for it.

-From your IP phone navigate through the menus to obtain its IP address:

'''Menu >> Status >> Network Status >> IPv4 Address'''

-Now that you have the Local IP address, open it in your web browser of preference.

[[File:Pasteipchromegxp2135.PNG|thumb|none|600px]]

-Once you have opened the login page, use the following credentials to access to the configuration:

'''Username: admin'''

'''Password: admin'''

[[File:Loginadmingxp2135.PNG|thumb|none|600px]]

=== General Settings ===

-Once you see the settings page, go to the following:

'''Account >> Account # >> General Settings'''

<div><ul>
<li style="display: inline-block;">[[File:Accountsgxp2135.PNG|thumb|none|600px]]</li>
<li style="display: inline-block;">[[File:Account1gxp2135.PNG|thumb|none|600px]]</li>
<li style="display: inline-block;">[[File:Generalgxp2135.PNG|thumb|none|600px]]</li>
</ul></div>

-Change the following settings according to the information of your account:

'''A. Account Name''': The name for this account that will be displayed on the LCD screen of your phone.

'''B. SIP Server''': The server that you will use for the registration of this device. (one of VoIP.ms multiple [[Choosing_Server#Choosing_a_Server | servers]], you can choose the one closest to your location.)

'''Notice that it is necessary to use the same server for both the device and the DID number in order to get incoming calls correctly'''

'''C. SIP User ID''': The account or sub-account number that you will be using for this line.

'''D. Authenticate Password''': The password for the account or sub-account you will be using for this line.

'''E. Name''': The name that will be used as Caller-ID Name. ('''See the requirements below.''')

'''F. Save and Apply'''

'''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.

[[File:Accountgeneralset.PNG|thumb|none|600px]]

=== Basic SIP settings ===

Once this is done, we will need to set the time that the device will send the information for its registration status.

-Please go to the following:

'''Account # >> SIP Settings >> Basic Settings'''

<div><ul>
<li style="display: inline-block;">[[File:Account1gxp2135.PNG|thumb|none|600px]] </li>
<li style="display: inline-block;">[[File:Sipsettingsgxp1.PNG|thumb|none|600px]] </li>
<li style="display: inline-block;">[[File:Basicsettingsgxp01.PNG|thumb|none|600px]] </li>
</ul></div>

- Change the following:

'''SIP Registration:''' Yes

'''Register Expiration:''' 5 (this is in minutes)

'''Enable OPTIONS Keep Alive:''' Yes

Finally, click '''Save and Apply'''

===Common errors===

====Outgoing calls issue====
If you are able to receive incoming calls but outgoing calls are failing with error "No response", make the following changes to fix it:

1. Login into your device's settings and head to '''Accounts > Account X > SIP > Custom SIP Header''' and disable:

*'''Use X-Grandstream-PBX Header'''
*'''Use P-Access-Network-Info Header'''
*'''Use P-Emergency-Info Header'''

[[File:SIP Headers GXP.png|thumb|none|600px]]

2. Go to '''Accounts > Account X > SIP > Audio Settings''', and choose codec '''G.722''' as preferred Vocoder and the rest with '''PCMU'''.

[[File:Audio codecs GXP.png|thumb|none|600px]]

==See also==
* [[Busy Lamp Field]], is available on this model if a request is made to enable the feature on the server.

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

User Manual : [http://www.grandstream.com/sites/default/files/Resources/gxp2130_gxp2140_gxp2160_gxp2135_gxp2170_quick_user_guide_english..pdf Download PDF]

Admin Manual : [http://www.grandstream.com/sites/default/files/Resources/gxp21xx_administration_guide.pdf Download PDF]

[[Category:SIP phones]]

Grandstream GXP2135

2023-09-16T10:00:25Z

Carlb: /* Guide Links */

[[File:GXP2135-device.jpg|300px|thumb|Grandstream GXP2135]]
High-profile desktop phones built for the busy user. The GXP21xx series are sleek high-end [[IP Phones|IP phones]] that will bring higher levels of communication productivity to a network. Their modern design, high call capacity, and rich functionality make them the ideal choice for the call-intensive worker.

IP phones in Grandstream's [https://documentation.grandstream.com/article-categories/gxp21xx/ GXP21xx series]. all of which are similar in configuration, include:

{|
||Features
||GXP2130
||GXP2140
||GXP2160
||GXP2170
||GXP2135
|-
||LCD Display
||320 × 240
||480 x 272
||480 x 272
||480 x 272
||320 × 240
|-
||LCD Backlight
||Yes
||Yes
||Yes
||Yes
||Yes
|-
||Number of Lines
||3
||4
||6
||12
||8
|-
||Programmable Hard Keys
||8
||N/A
||24
||48
||32
|-
||Programmable Softkeys
||4
||5
||5
||5
||4
|-
||Extension Module
||No
||Up to 4 EXT Boards
||No
||Up to 4 EXT Boards
||No
|}

== Configuration ==

'''SIP ACCOUNT'''

Once you have connected your device to your local connection, it will be necessary to obtain the local IP address that was designated for it.

-From your IP phone navigate through the menus to obtain its IP address:

'''Menu >> Status >> Network Status >> IPv4 Address'''

-Now that you have the Local IP address, open it in your web browser of preference.

[[File:Pasteipchromegxp2135.PNG|thumb|none|600px]]

-Once you have opened the login page, use the following credentials to access to the configuration:

'''Username: admin'''

'''Password: admin'''

[[File:Loginadmingxp2135.PNG|thumb|none|600px]]

=== General Settings ===

-Once you see the settings page, go to the following:

'''Account >> Account # >> General Settings'''

<div><ul>
<li style="display: inline-block;">[[File:Accountsgxp2135.PNG|thumb|none|600px]]</li>
<li style="display: inline-block;">[[File:Account1gxp2135.PNG|thumb|none|600px]]</li>
<li style="display: inline-block;">[[File:Generalgxp2135.PNG|thumb|none|600px]]</li>
</ul></div>

-Change the following settings according to the information of your account:

'''A. Account Name''': The name for this account that will be displayed on the LCD screen of your phone.

'''B. SIP Server''': The server that you will use for the registration of this device. (one of VoIP.ms multiple [[Choosing_Server#Choosing_a_Server | servers]], you can choose the one closest to your location.)

'''Notice that it is necessary to use the same server for both the device and the DID number in order to get incoming calls correctly'''

'''C. SIP User ID''': The account or sub-account number that you will be using for this line.

'''D. Authenticate Password''': The password for the account or sub-account you will be using for this line.

'''E. Name''': The name that will be used as Caller-ID Name. ('''See the requirements below.''')

'''F. Save and Apply'''

'''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.

[[File:Accountgeneralset.PNG|thumb|none|600px]]

=== Basic SIP settings ===

Once this is done, we will need to set the time that the device will send the information for its registration status.

-Please go to the following:

'''Account # >> SIP Settings >> Basic Settings'''

<div><ul>
<li style="display: inline-block;">[[File:Account1gxp2135.PNG|thumb|none|600px]] </li>
<li style="display: inline-block;">[[File:Sipsettingsgxp1.PNG|thumb|none|600px]] </li>
<li style="display: inline-block;">[[File:Basicsettingsgxp01.PNG|thumb|none|600px]] </li>
</ul></div>

- Change the following:

'''SIP Registration:''' Yes

'''Register Expiration:''' 5 (this is in minutes)

'''Enable OPTIONS Keep Alive:''' Yes

Finally, click '''Save and Apply'''

===Common errors===

====Outgoing calls issue====
If you are able to receive incoming calls but outgoing calls are failing with error "No response", make the following changes to fix it:

1. Login into your device's settings and head to '''Accounts > Account X > SIP > Custom SIP Header''' and disable:

*'''Use X-Grandstream-PBX Header'''
*'''Use P-Access-Network-Info Header'''
*'''Use P-Emergency-Info Header'''

[[File:SIP Headers GXP.png|thumb|none|600px]]

2. Go to '''Accounts > Account X > SIP > Audio Settings''', and choose codec '''G.722''' as preferred Vocoder and the rest with '''PCMU'''.

[[File:Audio codecs GXP.png|thumb|none|600px]]

==See also==
* [[Busy Lamp Field]], is available on this model if a request is made to enable the feature on the server.

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

User Manual : [http://www.grandstream.com/sites/default/files/Resources/gxp2130_gxp2140_gxp2160_gxp2135_gxp2170_quick_user_guide_english..pdf Download PDF]

Admin Manual : [http://www.grandstream.com/sites/default/files/Resources/gxp21xx_administration_guide.pdf Download PDF]

[[Category:SIP phones]]

Grandstream GXP2135

2023-09-16T09:55:07Z

Carlb: GXP2130, |GXP2140. GXP2160, GXP2170 and GXP2135 are same series and similar in configuration

[[File:GXP2135-device.jpg|300px|thumb|Grandstream GXP2135]]
High-profile desktop phones built for the busy user. The GXP21xx series are sleek high-end [[IP Phones|IP phones]] that will bring higher levels of communication productivity to a network. Their modern design, high call capacity, and rich functionality make them the ideal choice for the call-intensive worker.

IP phones in Grandstream's [https://documentation.grandstream.com/article-categories/gxp21xx/ GXP21xx series]. all of which are similar in configuration, include:

{|
||Features
||GXP2130
||GXP2140
||GXP2160
||GXP2170
||GXP2135
|-
||LCD Display
||320 × 240
||480 x 272
||480 x 272
||480 x 272
||320 × 240
|-
||LCD Backlight
||Yes
||Yes
||Yes
||Yes
||Yes
|-
||Number of Lines
||3
||4
||6
||12
||8
|-
||Programmable Hard Keys
||8
||N/A
||24
||48
||32
|-
||Programmable Softkeys
||4
||5
||5
||5
||4
|-
||Extension Module
||No
||Up to 4 EXT Boards
||No
||Up to 4 EXT Boards
||No
|}

== Configuration ==

'''SIP ACCOUNT'''

Once you have connected your device to your local connection, it will be necessary to obtain the local IP address that was designated for it.

-From your IP phone navigate through the menus to obtain its IP address:

'''Menu >> Status >> Network Status >> IPv4 Address'''

-Now that you have the Local IP address, open it in your web browser of preference.

[[File:Pasteipchromegxp2135.PNG|thumb|none|600px]]

-Once you have opened the login page, use the following credentials to access to the configuration:

'''Username: admin'''

'''Password: admin'''

[[File:Loginadmingxp2135.PNG|thumb|none|600px]]

=== General Settings ===

-Once you see the settings page, go to the following:

'''Account >> Account # >> General Settings'''

<div><ul>
<li style="display: inline-block;">[[File:Accountsgxp2135.PNG|thumb|none|600px]]</li>
<li style="display: inline-block;">[[File:Account1gxp2135.PNG|thumb|none|600px]]</li>
<li style="display: inline-block;">[[File:Generalgxp2135.PNG|thumb|none|600px]]</li>
</ul></div>

-Change the following settings according to the information of your account:

'''A. Account Name''': The name for this account that will be displayed on the LCD screen of your phone.

'''B. SIP Server''': The server that you will use for the registration of this device. (one of VoIP.ms multiple [[Choosing_Server#Choosing_a_Server | servers]], you can choose the one closest to your location.)

'''Notice that it is necessary to use the same server for both the device and the DID number in order to get incoming calls correctly'''

'''C. SIP User ID''': The account or sub-account number that you will be using for this line.

'''D. Authenticate Password''': The password for the account or sub-account you will be using for this line.

'''E. Name''': The name that will be used as Caller-ID Name. ('''See the requirements below.''')

'''F. Save and Apply'''

'''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.

[[File:Accountgeneralset.PNG|thumb|none|600px]]

=== Basic SIP settings ===

Once this is done, we will need to set the time that the device will send the information for its registration status.

-Please go to the following:

'''Account # >> SIP Settings >> Basic Settings'''

<div><ul>
<li style="display: inline-block;">[[File:Account1gxp2135.PNG|thumb|none|600px]] </li>
<li style="display: inline-block;">[[File:Sipsettingsgxp1.PNG|thumb|none|600px]] </li>
<li style="display: inline-block;">[[File:Basicsettingsgxp01.PNG|thumb|none|600px]] </li>
</ul></div>

- Change the following:

'''SIP Registration:''' Yes

'''Register Expiration:''' 5 (this is in minutes)

'''Enable OPTIONS Keep Alive:''' Yes

Finally, click '''Save and Apply'''

===Common errors===

====Outgoing calls issue====
If you are able to receive incoming calls but outgoing calls are failing with error "No response", make the following changes to fix it:

1. Login into your device's settings and head to '''Accounts > Account X > SIP > Custom SIP Header''' and disable:

*'''Use X-Grandstream-PBX Header'''
*'''Use P-Access-Network-Info Header'''
*'''Use P-Emergency-Info Header'''

[[File:SIP Headers GXP.png|thumb|none|600px]]

2. Go to '''Accounts > Account X > SIP > Audio Settings''', and choose codec '''G.722''' as preferred Vocoder and the rest with '''PCMU'''.

[[File:Audio codecs GXP.png|thumb|none|600px]]

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

User Manual : [http://www.grandstream.com/sites/default/files/Resources/gxp2130_gxp2140_gxp2160_gxp2135_gxp2170_quick_user_guide_english..pdf Download PDF]

Admin Manual : [http://www.grandstream.com/sites/default/files/Resources/gxp21xx_administration_guide.pdf Download PDF]

[[Category:SIP phones]]

Grandstream WP810, WP820, WP822 and WP825

2023-09-16T07:57:05Z

Carlb: /* Notes */

The '''Grandstream WP810, WP820, WP822 and WP825''' are two-line cordless [[IP Phone|IP phones]] 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 ==
[[Image:Grandstream WP810.png|thumb|right|Grandstream WP810]]
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.

1. Insert the supplied lithium-ion battery and place the handset in the charger.

2. 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".

3. The handset will show a list of available wi-fi networks. Select one, then select "Password" and enter your local wi-fi password.

4. Press [Connect]; the display should show "Wi-Fi Networks" with the list of network names and "Connected" for the selected network.

5. Return to the main menu. Select [status] (the ''i'' icon) then "Network Status". The handset will display IPv4 info, which includes the IPv4 address.

6. 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.

7. 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.

8. 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.

[[Image:Grandstream WP810 Account 1 General Settings.png|Acccount 1 - General Settings]]

9. 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

10. 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.

11. 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.

12. 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.

13. 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.

14. 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.

15. 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.

{||
|| +,1,2,3,4,5,6,7,8,9,0, *, #, A,a,B,b,C,c,D,d
|| Individual digit, to be passed through verbatim.
|-
|| x
|| matches any digit from 0-9;
|-
|| xx+
|| at least 2-digit number;
|-
|| xx
|| exactly 2-digit number;
|-
|| xx?
|| 1 or 2-digit numbers from 0-9
|-
|| ^
|| exclude;
|-
|| .
|| wildcard, matches one or more characters
|-
|| [3-5]
|| any digit of 3, 4, or 5.
|-
|| [147]
|| any digit 1, 4, or 7.
|-
|| <2=011>
||replace digit 2 with 011 when dialling.
|-
|| <=1>
|| add a 1 to dialled number
|-
|| <1=>
|| remove 1 from the number dialled
|-
|| |
|| 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 }
|-
|| \+
|| + sign
|-
|| 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 '' { x+ | \+x+ | *x+ | *xx*x+ | x+*x+*x+*x+ | x+*x+*x+*x+#x+ } ''

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.

== Guide 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: SIP phones]]

Sip Scanner Ghost Calls

2023-09-14T00:48:08Z

Carlb: /* Grandstream WP810/WP820 */

{| class="wikitable"
|+
|-
! Article en Français !! Artículo en Español
|-
| [https://wiki.voip.ms/article/Appels_fant%C3%B4mes_en_provenance_d%27un_scanner_SIP Français] || [https://wiki.voip.ms/article/Llamadas_fantasmas_de_Scanner_SIP Español]
|}

__TOC__

Some people from time to time experience calls on their IP phones from unknown numbers or extensions and when they pick up they just hear silence. This is unfortunately a well known problem in regular Telephone & VoIP and has nothing to do with the service provider. We call these types of calls SIP Scanner Ghost Calls and besides being extremely annoying they don’t pose any significant risk to your phones or network. Providing you make sure the firmware on your phone is up to date.

These calls are not coming from our service but they are generated by “port scans” performed by hackers trying to find a vulnerable phone network to gain access to. They do large series of automated tests against IP addresses on the internet, to find systems that respond. The good news is that there are several ways you can prevent these types of calls.

==Change Local SIP Port==

Changing the local SIP port on your phone will make it harder for the scanners to guess the way into your device. You can try and set this to something like 5080 or 42872. The place to do this is usually in the Line/EXT config page for that device and it will say either Sip Port or Local Sip Port in most cases. By Default the SIP Port is usually set to 5060.

==Use a Firewall==
Some firewalls are able to filter these port scans from legit traffic. Look in the manual for your router/firewall to see how to do this, or contact your internet provider and ask for their assistance.

==Change Your IP==
If you don’t have a specific reason to have a static IP address, you can ask your internet provider to assign you a new IP address. This may not be a permanent solution to the problem, but it can definitely stop the calls for some time.

==Only Allow Calls from VoIP.ms Servers==
Some IP phones can disable direct calls from other devices than a specific server. This means that the phone will reject all calls that are not coming from the VoIP.ms server. The setting’s location and name varies from phone to phone, so check your manual to see if your phone supports it.

== Here are a few models that have a resolution for this issue: ==

===Cisco/Linksys SPAxxx===
Please look under the Voice>> Line/EXT # page in your SPA device for the following setting: Restrict Source IP and make sure it's enabled.

This way the ATA device will block any traffic not coming from our servers.

[[File:VL_1_restrictSourceIP.png|800px|thumb|left|Restrict IP - click to enlarge]]
<br clear="all" />

===Cisco/Linksys Pap2t===
Please look under the Voice>> Line/EXT # page in your Linksys device for the following setting: Restrict Source IP and make sure it's enabled.

This way the ATA device will block any traffic not coming from our servers.

[[File:RestrictSourceIP.png|800px]]

===Grandstream GXP2200===
Advanced Settings -> Call Features -> Disable Direct IP Calls

===Grandstream GXP2130/40/60===
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"

===Grandstream HT50X/HT70X===
To prevent direct IP calls to your device and only allow calls from our service please enable the following 2 options in your FXS Port Configuration Page.
* '''Check SIP User ID for incoming INVITE''' - Default is No. Check the incoming SIP User ID in Request URI. If they don’t match, the call will be rejected. If this option is enabled, the device will not be able to make direct IP calls.
* '''Allow Incoming SIP Messages from SIP Proxy Only''' - Default is No. Check the incoming SIP messages. If they don’t come from the SIP proxy, they will be rejected. If this option is enabled, the device will not be able to make direct IP calls.

===Grandstream WP810/WP820===
To prevent direct IP calls to your device and only allow calls from our service please go to the "Account" section of the configuration. For each account (1 and 2, using the [[Grandstream WP810, WP820, WP822 and WP825|WP820]]'s settings as an example):
* Change "Account #n" → "SIP settings" → "SIP basic settings" → "'''Local SIP Port'''" to be "0" (use random port) instead of "5060" or any other well-known value. [Save] this and [Apply].
* Go to "Account #n" → "Advanced settings" → "Security" and check the checkbox for "'''Only Accept SIP Requests from Known Servers'''". Again, [Save] and [Apply].

===Obi 1xx/2xx===
*You can just disable (by unchecking Enable) for SP2 and OBiTALK under your Voice Tab (If you are using our service as SP1).

*You can restrict which IP addresses that can connect to your OBi. Going to "Service Providers -> ITSP Profile A -> SIP -> X_AccessList" : voip.ms_ip_address. You can see the IP address of the server you are currently using from this link: [http://wiki.voip.ms/article/Choosing_Server#IPs Server's IPs]

*You can also change your Obi Firewall Setting X_InboundCallRoute to : {(?|x|xx|xxx|xxxx|xxxxx|xxxxxx|un@@.|anon@@.):}, ph
This will only allow 7 digit or greater numbers through.

*Another alternative: OBi Interface>> Voice Services>> SP1 Service>> X_InboundCallRoute: {>('Insert your AuthUserName here'):ph}, example:

{>('100000'):ph} where 100000 is replaced with your own six digit SIP account UserID or the sub-account registered with your device.

By default, OBi devices accept calls destined for any username. The above syntax rejects calls that are not intended for whatever you have configured as AuthUserName.

===Panasonic KX-TGP 500/550===
To turn off IP Dialing function on a TGP 500/550 you go to Line 1 > Enable SSAF (SIP Source Address Filter). That should stop the random dialing.

===Polycom Phones===
Try to utilize the Incoming Signaling Validation where you would be able to add security to the phone to validating incoming network signaling in the GUI.
All of this is described in the <requestValidation/> section of the Admin Guide matching your software version.

===Yealink===

Look for these settings on the Line config page.

Allow Direct IP Call - this means the phone will respond to calls coming in to it from any IP address, to any number. Sometimes used for internal intercom systems or basic phone testing without using a PBX. Set it to disabled. This setting is found in the "Features" setting tab, "General Information" page.

Accept SIP Trust Server Only - this is whether the phone accepts calls to the correct phone number but from a different place than it is Registered to. Sometimes needed for certain SIP providers but you want to set this to enabled wherever possible so the phone only accepts calls from your service provider. This setting is found either in the "Features" tab, "General Information" page or the "Account" tab depending on the phone model or firmware version.

You can also try to add below syntaxes to your cfg template(M7 template) and auto-provision it.

1. You can try this syntax in CFG template.
---------------------------------------------------------------------------
#!version:1.0.0.1

#The x of the parameter "account.x.sip_trust_ctrl " ranges from 1 to max accounts. For example, x ranges from 1 to 6 of T28.

account.x.sip_trust_ctrl=1
------------------------------------------------------------------------------------------

When you want to enable this sip trust control for account 1, fill 1 to “account.1.sip_trust_ctrl”.
Then SIP messages from other servers will refuse by the phone.

2. If not, you can disable the “Allow IP Call” in webpage or auto-provisioning and try again.

-------------------------------------------------------------------------------------------------
#!version:1.0.0.1

#Enable or disable the phone to dial the IP address directly; 0-Disabled, 1-Enabled (default);
features.direct_ip_call_enable = 0

-------------------------------------------------------------------------------------------------

Grandstream WP810, WP820, WP822 and WP825

2023-09-14T00:36:01Z

Carlb: /* Notes */

The '''Grandstream WP810, WP820, WP822 and WP825''' are two-line cordless [[IP Phone|IP phones]] 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 ==
[[Image:Grandstream WP810.png|thumb|right|Grandstream WP810]]
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.

1. Insert the supplied lithium-ion battery and place the handset in the charger.

2. 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".

3. The handset will show a list of available wi-fi networks. Select one, then select "Password" and enter your local wi-fi password.

4. Press [Connect]; the display should show "Wi-Fi Networks" with the list of network names and "Connected" for the selected network.

5. Return to the main menu. Select [status] (the ''i'' icon) then "Network Status". The handset will display IPv4 info, which includes the IPv4 address.

6. 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.

7. 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.

8. 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.

[[Image:Grandstream WP810 Account 1 General Settings.png|Acccount 1 - General Settings]]

9. 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

10. 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.

11. 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.

12. 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.

13. 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.

14. 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.

15. 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]]. 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.

{||
|| +,1,2,3,4,5,6,7,8,9,0, *, #, A,a,B,b,C,c,D,d
|| Individual digit, to be passed through verbatim.
|-
|| x
|| matches any digit from 0-9;
|-
|| xx+
|| at least 2-digit number;
|-
|| xx
|| exactly 2-digit number;
|-
|| xx?
|| 1 or 2-digit numbers from 0-9
|-
|| ^
|| exclude;
|-
|| .
|| wildcard, matches one or more characters
|-
|| [3-5]
|| any digit of 3, 4, or 5.
|-
|| [147]
|| any digit 1, 4, or 7.
|-
|| <2=011>
||replace digit 2 with 011 when dialling.
|-
|| <=1>
|| add a 1 to dialled number
|-
|| <1=>
|| remove 1 from the number dialled
|-
|| |
|| 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 }
|-
|| \+
|| + sign
|-
|| 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 '' { x+ | \+x+ | *x+ | *xx*x+ | x+*x+*x+*x+ | x+*x+*x+*x+#x+ } ''

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.

== Guide 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: SIP phones]]

Grandstream GXP1630

2023-09-12T16:04:47Z

Carlb:

[[File:Gxp1630.jpg|300px|thumb|Grandstream GXP1630]]
The GXP1630 is one of a series of Basic IP phones which provide SIP telephony for small and medium-size businesses. According to the manufacturer, the [https://www.grandstream.com/products/ip-voice-telephony/gxp-series-ip-phones/gxp-series-basic-ip-phones GXP1600 series] deliver interactive communications to a desktop for the user who needs access to VoIP but does not require advanced features.

The GXP1600 series delivers an effective communications platform 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.

Most of the phones in this series (except for the GXP1610/GXP1610P/GXP1615) support at least two lines with individual SIP registrations for each. All provide wired-Ethernet connections, HD audio, speakerphone and three-way conferencing.
* The GXP1630 (3 SIP accounts, 3 line keys, 4-way conferencing) and GXP1628 (2 SIP accounts, 2 line keys) are higher-end models with features including dual-switched Gigabit ports, integrated [[Power over Ethernet]], a backlit LCD and eight [[Busy Lamp Field]] indicators.
* The GXP1620/GXP1625 are two-line office phones with dual switched 10/100 mbps ports (integrated PoE on GXP1625 only)
* The GXP1610/GXP1610P/GXP1615 are single-line base model phones with no backlight (integrated PoE on GXP1610P/PGXP1615 only)

== Configuration ==

'''SIP ACCOUNT'''

Once you have connected your device to your local connection, it will be necessary to obtain the local IP address that was designated for it.

-From your IP phone navigate through the menus to obtain its IP address:

'''Menu >> Status >> Network Status >> IPv4 Address'''

-Now that you have the Local IP address, open it in your web browser of preference.

-Once you have opened the login page, use the following credentials to access to the configuration:

'''Username: admin'''

'''Password: admin'''

Note: Grandstream is gradually phasing out "admin"/"admin" as default webpage credential in new designs. While these appear to still be working for this model, should this become deprecated, 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|WP810 series]], in the battery compartment under the battery). The username will remain "admin" and the password will be noted as P/W followed by s0m3g1bb3r!$h as a random password.

=== General Settings ===

-Once you see the settings page, go to the following:

'''Account >> Account # >> General Settings'''

-Change the following settings according to the information of your account:

'''A. Account Name''': The name for this account that will be displayed on the LCD screen of your phone.

'''B. SIP Server''': The server that you will use for the registration of this device. (one of VoIP.ms multiple [[Choosing_Server#Choosing_a_Server | servers]], you can choose the one closest to your location.)

'''Notice that it is necessary to use the same server for both the device and the DID number in order to get incoming calls correctly'''

'''C. SIP User ID''': The account or sub-account number that you will be using for this line.

'''D. Authenticate Password''': The password for the account or sub-account you will be using for this line.

'''E. Name''': The name that will be used as Caller-ID 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.

'''F. Save and Apply'''

[[File:General_GXP1630.png|thumb|none|600px]]

=== Basic SIP settings ===

Once this is done, we will need to set the time that the device will send the information for its registration status.

-Please go to the following:

'''Account # >> SIP Settings >> Basic Settings'''

- Change the following:

'''SIP Registration:''' Yes

'''Register Expiration:''' 5 (this is in minutes)

'''Enable OPTIONS Keep Alive:''' Yes

'''SIP Transport:''' UDP or TCP

Finally, click '''Save and Apply'''

<div><ul>
<li style="display: inline-block;">[[File:SIP Basic Grandstream GXP1630.png|thumb|none|600px]] </li>
</ul></div>

===Common errors===

====Outgoing calls issue====
If you are able to receive incoming calls but outgoing calls are failing with error "No response", make the following changes to fix it:

1. Login into your device's settings and head to '''Accounts > Account X > SIP > Custom SIP Header''' and disable:

*'''Use X-Grandstream-PBX Header'''
*'''Use P-Access-Network-Info Header'''
*'''Use P-Emergency-Info Header'''

[[File:SIP_Custom_SIP_Headers_Grandstream_GXP1630.png|thumb|none|600px]]

2. Go to '''Accounts > Account X > SIP > Audio Settings''', and choose codec '''G729A/B''' as preferred Vocoder and the rest with '''PCMU'''.

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

User Manual : [http://www.grandstream.com/sites/default/files/Resources/gxp16xx_user_guide.pdf]

Admin Manual : [http://www.grandstream.com/sites/default/files/Resources/gxp16xx_administration_guide.pdf]

[[Category:SIP phones]]

Power over Ethernet

2023-09-12T16:02:32Z

Carlb:

'''Power over Ethernet''' (PoE) is a standard which allows both DC power and wired network data to be carried to a device over the same wiring. It is most commonly used to power desktop [[IP Phones]] in an office environment, although some other hardware (such as wireless access points and security cameras) also makes use of the standard.

The power is normally inserted using either a PoE-enabled local area network hub or by a small device packaged specifically as a PoE power inserter.

See [[How to Troubleshoot PoE Issues In Your VoIP Phone System]].

[[Category:Networking Devices]]

Grandstream GXP1630

2023-09-10T23:22:21Z

Carlb: /* Configuration */

[[File:Gxp1630.jpg|300px|thumb|Grandstream GXP1630]]
The GXP1630 is one of a series of Basic IP phones which provide SIP telephony for small and medium-size businesses. According to the manufacturer, the [https://www.grandstream.com/products/ip-voice-telephony/gxp-series-ip-phones/gxp-series-basic-ip-phones GXP1600 series] deliver interactive communications to a desktop for the user who needs access to VoIP but does not require advanced features.

The GXP1630 delivers an effective communications platform 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.

Most of the phones in this series (except for the GXP1610/GXP1610P/GXP1615) support at least two lines with individual SIP registrations for each. All provide wired-Ethernet connections, HD audio, speakerphone and three-way conferencing.
* The GXP1630 (3 SIP accounts, 3 line keys, 4-way conferencing) and GXP1628 (2 SIP accounts, 2 line keys) are higher-end models with features including dual-switched Gigabit ports, integrated [[Power over Ethernet]], a backlit LCD and eight [[Busy Lamp Field]] indicators.
* The GXP1620/GXP1625 are two-line office phones with dual switched 10/100 mbps ports (integrated PoE on GXP1625 only)
* The GXP1610/GXP1610P/GXP1615 are single-line base model phones with no backlight (integrated PoE on GXP1610P/PGXP1615 only)

== Configuration ==

'''SIP ACCOUNT'''

Once you have connected your device to your local connection, it will be necessary to obtain the local IP address that was designated for it.

-From your IP phone navigate through the menus to obtain its IP address:

'''Menu >> Status >> Network Status >> IPv4 Address'''

-Now that you have the Local IP address, open it in your web browser of preference.

-Once you have opened the login page, use the following credentials to access to the configuration:

'''Username: admin'''

'''Password: admin'''

Note: Grandstream is gradually phasing out "admin"/"admin" as default webpage credential in new designs. While these appear to still be working for this model, should this become deprecated, 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|WP810 series]], in the battery compartment under the battery). The username will remain "admin" and the password will be noted as P/W followed by s0m3g1bb3r!$h as a random password.

=== General Settings ===

-Once you see the settings page, go to the following:

'''Account >> Account # >> General Settings'''

-Change the following settings according to the information of your account:

'''A. Account Name''': The name for this account that will be displayed on the LCD screen of your phone.

'''B. SIP Server''': The server that you will use for the registration of this device. (one of VoIP.ms multiple [[Choosing_Server#Choosing_a_Server | servers]], you can choose the one closest to your location.)

'''Notice that it is necessary to use the same server for both the device and the DID number in order to get incoming calls correctly'''

'''C. SIP User ID''': The account or sub-account number that you will be using for this line.

'''D. Authenticate Password''': The password for the account or sub-account you will be using for this line.

'''E. Name''': The name that will be used as Caller-ID 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.

'''F. Save and Apply'''

[[File:General_GXP1630.png|thumb|none|600px]]

=== Basic SIP settings ===

Once this is done, we will need to set the time that the device will send the information for its registration status.

-Please go to the following:

'''Account # >> SIP Settings >> Basic Settings'''

- Change the following:

'''SIP Registration:''' Yes

'''Register Expiration:''' 5 (this is in minutes)

'''Enable OPTIONS Keep Alive:''' Yes

'''SIP Transport:''' UDP or TCP

Finally, click '''Save and Apply'''

<div><ul>
<li style="display: inline-block;">[[File:SIP Basic Grandstream GXP1630.png|thumb|none|600px]] </li>
</ul></div>

===Common errors===

====Outgoing calls issue====
If you are able to receive incoming calls but outgoing calls are failing with error "No response", make the following changes to fix it:

1. Login into your device's settings and head to '''Accounts > Account X > SIP > Custom SIP Header''' and disable:

*'''Use X-Grandstream-PBX Header'''
*'''Use P-Access-Network-Info Header'''
*'''Use P-Emergency-Info Header'''

[[File:SIP_Custom_SIP_Headers_Grandstream_GXP1630.png|thumb|none|600px]]

2. Go to '''Accounts > Account X > SIP > Audio Settings''', and choose codec '''G729A/B''' as preferred Vocoder and the rest with '''PCMU'''.

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

User Manual : [http://www.grandstream.com/sites/default/files/Resources/gxp16xx_user_guide.pdf]

Admin Manual : [http://www.grandstream.com/sites/default/files/Resources/gxp16xx_administration_guide.pdf]

[[Category:SIP phones]]

Grandstream GXP1630

2023-09-10T23:12:31Z

Carlb:

[[File:Gxp1630.jpg|300px|thumb|Grandstream GXP1630]]
The GXP1630 is one of a series of Basic IP phones which provide SIP telephony for small and medium-size businesses. According to the manufacturer, the [https://www.grandstream.com/products/ip-voice-telephony/gxp-series-ip-phones/gxp-series-basic-ip-phones GXP1600 series] deliver interactive communications to a desktop for the user who needs access to VoIP but does not require advanced features.

The GXP1630 delivers an effective communications platform 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.

Most of the phones in this series (except for the GXP1610/GXP1610P/GXP1615) support at least two lines with individual SIP registrations for each. All provide wired-Ethernet connections, HD audio, speakerphone and three-way conferencing.
* The GXP1630 (3 SIP accounts, 3 line keys, 4-way conferencing) and GXP1628 (2 SIP accounts, 2 line keys) are higher-end models with features including dual-switched Gigabit ports, integrated [[Power over Ethernet]], a backlit LCD and eight [[Busy Lamp Field]] indicators.
* The GXP1620/GXP1625 are two-line office phones with dual switched 10/100 mbps ports (integrated PoE on GXP1625 only)
* The GXP1610/GXP1610P/GXP1615 are single-line base model phones with no backlight (integrated PoE on GXP1610P/PGXP1615 only)

== Configuration ==

'''SIP ACCOUNT'''

Once you have connected your device to your local connection, it will be necessary to obtain the local IP address that was designated for it.

-From your IP phone navigate through the menus to obtain its IP address:

'''Menu >> Status >> Network Status >> IPv4 Address'''

-Now that you have the Local IP address, open it in your web browser of preference.

-Once you have opened the login page, use the following credentials to access to the configuration:

'''Username: admin'''

'''Password: admin'''

Note: Grandstream is gradually phasing out "admin"/"admin" as default webpage credential in new designs. While these appear to still be working for this model, should this become deprecated, check for a model/serial number sticker on the underside or back (or, for self-contained handsets like the [[Grandstream WP810, WP820, WP822 and WP825|WP810 series]], in the battery compartment under the battery). The username will remain "admin" and the password will be noted as P/W followed by s0m3g1bb3r!$h as a random password.

=== General Settings ===

-Once you see the settings page, go to the following:

'''Account >> Account # >> General Settings'''

-Change the following settings according to the information of your account:

'''A. Account Name''': The name for this account that will be displayed on the LCD screen of your phone.

'''B. SIP Server''': The server that you will use for the registration of this device. (one of VoIP.ms multiple [[Choosing_Server#Choosing_a_Server | servers]], you can choose the one closest to your location.)

'''Notice that it is necessary to use the same server for both the device and the DID number in order to get incoming calls correctly'''

'''C. SIP User ID''': The account or sub-account number that you will be using for this line.

'''D. Authenticate Password''': The password for the account or sub-account you will be using for this line.

'''E. Name''': The name that will be used as Caller-ID 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.

'''F. Save and Apply'''

[[File:General_GXP1630.png|thumb|none|600px]]

=== Basic SIP settings ===

Once this is done, we will need to set the time that the device will send the information for its registration status.

-Please go to the following:

'''Account # >> SIP Settings >> Basic Settings'''

- Change the following:

'''SIP Registration:''' Yes

'''Register Expiration:''' 5 (this is in minutes)

'''Enable OPTIONS Keep Alive:''' Yes

'''SIP Transport:''' UDP or TCP

Finally, click '''Save and Apply'''

<div><ul>
<li style="display: inline-block;">[[File:SIP Basic Grandstream GXP1630.png|thumb|none|600px]] </li>
</ul></div>

===Common errors===

====Outgoing calls issue====
If you are able to receive incoming calls but outgoing calls are failing with error "No response", make the following changes to fix it:

1. Login into your device's settings and head to '''Accounts > Account X > SIP > Custom SIP Header''' and disable:

*'''Use X-Grandstream-PBX Header'''
*'''Use P-Access-Network-Info Header'''
*'''Use P-Emergency-Info Header'''

[[File:SIP_Custom_SIP_Headers_Grandstream_GXP1630.png|thumb|none|600px]]

2. Go to '''Accounts > Account X > SIP > Audio Settings''', and choose codec '''G729A/B''' as preferred Vocoder and the rest with '''PCMU'''.

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

User Manual : [http://www.grandstream.com/sites/default/files/Resources/gxp16xx_user_guide.pdf]

Admin Manual : [http://www.grandstream.com/sites/default/files/Resources/gxp16xx_administration_guide.pdf]

[[Category:SIP phones]]

Power over Ethernet

2023-09-10T23:10:14Z

Carlb: soft-redirect

'''Power over Ethernet''' (PoE) is a standard which allows both DC power and wired network data to be carried to a device over the same wiring. It is most commonly used to power desktop [[IP Phones]] in an office environment, although some other hardware (such as wireless access points) also makes use of the standard.

The power is normally inserted using either a PoE-enabled local area network hub or by a small device packaged specifically as a PoE power inserter.

See [[How to Troubleshoot PoE Issues In Your VoIP Phone System]].

[[Category:Networking Devices]]

Grandstream GXP1630

2023-09-10T23:02:27Z

Carlb: this is a series with multiple GXP16xx models

[[File:Gxp1630.jpg|300px|thumb|Grandstream GXP1630]]
The GXP1630 is one of a series of Basic IP phones which provide SIP telephony for small and medium-size businesses. According to the manufacturer, the [https://www.grandstream.com/products/ip-voice-telephony/gxp-series-ip-phones/gxp-series-basic-ip-phones GXP1600 series] deliver interactive communications to a desktop for the user who needs access to VoIP but does not require advanced features.

The GXP1630 delivers an effective communications platform 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.

Most of the phones in this series (except for the GXP1610/GXP1610P/GXP1615) support at least two lines with individual SIP registrations for each. All provide wired-Ethernet connections, HD audio, speakerphone and three-way conferencing.
* The GXP1630 (3 SIP accounts, 3 line keys, 4-way conferencing) and GXP1628 (2 SIP accounts, 2 line keys) are higher-end models with features invlufinh dual-switched Gigabit ports, integrated [[Power over Ethernet]], a backlit LCD and eight [[Busy Lamp Field]] indicators.
* The GXP1620/GXP1625 are two-line office phones with dual switched 10/100 mbps ports (integrated PoE on GXP1625 only)
* The GXP1610/GXP1610P/GXP1615 are single-line base model phones with no backlight (integrated PoE on GXP1610P/PGXP1615 only)

== Configuration ==

'''SIP ACCOUNT'''

Once you have connected your device to your local connection, it will be necessary to obtain the local IP address that was designated for it.

-From your IP phone navigate through the menus to obtain its IP address:

'''Menu >> Status >> Network Status >> IPv4 Address'''

-Now that you have the Local IP address, open it in your web browser of preference.

-Once you have opened the login page, use the following credentials to access to the configuration:

'''Username: admin'''

'''Password: admin'''

Note: Grandstream is gradually phasing out "admin"/"admin" as default webpage credential in new designs. While these appear to still be working for this model, should this become deprecated, check for a model/serial number sticker on the underside or back (or, for self-contained handsets like the [[Grandstream WP810, WP820, WP822 and WP825|WP810 series]], in the battery compartment under the battery). The username will remain "admin" and the password will be noted as P/W followed by s0m3g1bb3r!$h as a random password.

=== General Settings ===

-Once you see the settings page, go to the following:

'''Account >> Account # >> General Settings'''

-Change the following settings according to the information of your account:

'''A. Account Name''': The name for this account that will be displayed on the LCD screen of your phone.

'''B. SIP Server''': The server that you will use for the registration of this device. (one of VoIP.ms multiple [[Choosing_Server#Choosing_a_Server | servers]], you can choose the one closest to your location.)

'''Notice that it is necessary to use the same server for both the device and the DID number in order to get incoming calls correctly'''

'''C. SIP User ID''': The account or sub-account number that you will be using for this line.

'''D. Authenticate Password''': The password for the account or sub-account you will be using for this line.

'''E. Name''': The name that will be used as Caller-ID 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.

'''F. Save and Apply'''

[[File:General_GXP1630.png|thumb|none|600px]]

=== Basic SIP settings ===

Once this is done, we will need to set the time that the device will send the information for its registration status.

-Please go to the following:

'''Account # >> SIP Settings >> Basic Settings'''

- Change the following:

'''SIP Registration:''' Yes

'''Register Expiration:''' 5 (this is in minutes)

'''Enable OPTIONS Keep Alive:''' Yes

'''SIP Transport:''' UDP or TCP

Finally, click '''Save and Apply'''

<div><ul>
<li style="display: inline-block;">[[File:SIP Basic Grandstream GXP1630.png|thumb|none|600px]] </li>
</ul></div>

===Common errors===

====Outgoing calls issue====
If you are able to receive incoming calls but outgoing calls are failing with error "No response", make the following changes to fix it:

1. Login into your device's settings and head to '''Accounts > Account X > SIP > Custom SIP Header''' and disable:

*'''Use X-Grandstream-PBX Header'''
*'''Use P-Access-Network-Info Header'''
*'''Use P-Emergency-Info Header'''

[[File:SIP_Custom_SIP_Headers_Grandstream_GXP1630.png|thumb|none|600px]]

2. Go to '''Accounts > Account X > SIP > Audio Settings''', and choose codec '''G729A/B''' as preferred Vocoder and the rest with '''PCMU'''.

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

User Manual : [http://www.grandstream.com/sites/default/files/Resources/gxp16xx_user_guide.pdf]

Admin Manual : [http://www.grandstream.com/sites/default/files/Resources/gxp16xx_administration_guide.pdf]

[[Category:SIP phones]]

Busy Lamp Field

2023-09-10T21:45:43Z

Carlb: Redirected page to BLF Example Scenario

#REDIRECT [[BLF Example Scenario]]

FortiVoice FVE-20E

2023-09-08T01:34:50Z

Carlb: /* Add an Outbound Route */

The FortiVoice<sup>TM</sup> solutions accommodate efficient employee collaboration within a centralized, safe, and secured environment so your organization can provide the best customer service through a variety of our unique and advanced communication features.

First, you will have to log into the Admin of the PBX.
[[File:FortiVoice_20E4_Admin_Login.png|thumb|none|600px]]

= Add a VoIP.ms trunk =

Once you are in your admin console, Go the left navigation bar and click on '''[Trunk]''', then '''[VoIP]'''. At the top, click on '''[+ New]'''
[[File:FortiVoice_20E4_Trunk.png|thumb|none|600px]]

In the first section, you will have to give a Name to your Trunk.
:* '''Display Name''' will be the name you would like to display when you do an outbound call.
:: - Capital letters,
:: - Not longer than 15 Characters
:: - No Special Characters
:: - Spaces are allowed.
:*'''Main Number''' is the DID you would like to display with the outbound call.

'''SIP Setting'''
:* '''SIP server:''' YourPreferredPoP.voip.ms
:* '''SIP port:''' 5060 or 5080 or 42872
:* '''SIP server:''' Your Sub Account username ######_SubAccountUsername
:* '''Password:''' Your sub-account password.
:* '''Auth.user name:''' Your Sub Account username ######_SubAccountUsername
:* '''Realm/Domain:''' YourPreferredPoP.voip.ms
:* '''SIP server:''' YourPreferredPoP.voip.ms
:* Max channel: ''(You can define a Max of channel for this trunk)''
:* Max outgoing channel: ''(You can define a Max of outgoing channel for this trunk)''

'''Caller ID Option'''
:* '''From header:''' Select '''[SIP user name]'''
:* '''P-Asserted-Identity header:''' Select '''[Caller ID Priority rule]'''

'''Phone Number''': You can leave it empty. You can use the '''Inbound Route''' section to add your DIDs. ''This option is especially useful if you have a global inbound route with no DID assigned to it.''

[[File:FortiVoice_20E4_Add_Trunk.png|thumb|none|600px]]

= Add an Inbound Route =
To get incoming calls and manage the incoming call from one of your DID you will need to create an Inbound Route.

[[File:FortiVoice_20E4_Inbound_Route.png|thumb|none|600px]]

Give a name to your inbound route, and enable it. In the Trunk section, you will have to '''select the VoIPms trunk''' you have created, once is selected from the section '''"Available"''' you need to add it to the selected section by clicking the '''[➡]''' button.

:- '''Dialed Number Match''', should be your DID, without Dot/Special characters.<br/>
:- '''Call Handling''', should where your incoming call will be routed in your PBX.
[[File:FortiVoice_20E4_Inbound_Route_Add.png|thumb|none|600px]]

= Add an Outbound Route =
In order to be able to do outgoing call with VoIP.ms you will need to associate the VoIP.ms Trunk to an existing outbound route.

Go the left navigation bar and click on '''[Call Routing]''', then '''[Outbound]'''. Choose an existing Route that you would like to edit, then press '''[Edit]''', Or create a '''[+New]''' one.

[[File:FortiVoice_20E4_Outbound_Route.png|thumb|none|600px]]

Give the name of the outgoing route and activate it. You can customize the '''Dialed Number Match''' to this route and then, in the '''Call Handling''' section, you will have to '''select VoIPms Trunk''' according to your programming rules.
[[File:FortiVoice_20E4_Outbound_Route_Add_Dialplan.png|thumb|none|600px]]

[[Category:PBXes]]

Softphone.Pro

2023-09-08T01:28:29Z

Carlb: /* Guide Links */

[[File:SoftphoneProLogoNW.png|450px|thumb|right|Softphone.Pro]]



Softphone.Prop is a great solution for call center agents, sales pros and support teams.

Offers instant click-to-call, screen pop-up integration with any 3rd party CRM and Helpdesk software, online reporting, call and screen recording, as well as agent's personal stats in a special window.

You can use it either [https://softphone.pro/en/help/download alone application] or together with a dashboard [https://softphone.pro/en/select-product-trial/ addon].

==Basic setup==

===Setting up Softphone.Pro with your SIP credentials===

''' Step 1 -''' Download and install '''Softphone.Pro''' from this [https://softphone.pro/en/select-product-trial/ link]. If you don't have a license yet you can try it for 14 days.

'''Step 2 -''' Start Softphone.Pro, click on '''Outbound Account''', then click on the '''+ Add Account''' option.

[[File:SoftphonePro_Add_Account.png|thumb|none|600px|Click to enlarge]]

'''Step 3 -''' Make sure you are on the '''SIP Account''' section and set up the new account using the following configuration:

* '''Account name:''' Any name of your choice. For example, it can be just "VoIP.ms", or "My VoIP.ms Account".
* '''SIP server:''' Any of the multiple servers available with VoIP.ms (E.g. houston1.voip.ms). You can check the list [https://wiki.voip.ms/article/Choosing_Server here]. We recommend using one close to your location.
* '''Login:''' Your Main account or sub account username (six digit number) E.g. 123456 / 123456_XXXX (the underscore has to be used for sub-accounts)
* '''Password:''' The password you set for the account / sub account.
* '''Display name:''' You can set here your name or your company name to be passed along with the Caller ID number (Please check below important information about the Display Name).
* '''Authorization name:''' This can be the same set as your Login or you can leave it blank.
* '''Domain:''' The same server set as SIP Server.
* '''SIP Proxy:''' The same server set as SIP Server.

[[File:SofpthonePro_SIP_Account_Settings.png|thumb|none|600px|Click to enlarge]]

'''*Note 1:''' New accounts have received these '''main SIP/IAX credentials''' by e-mail when the account was created. By default, your
'''main SIP/IAX password''' is the same you use to log into your customer portal, however, for old accounts they could be
different if the portal password has been changed in the past. If you're not sure about your '''Main SIP/IAX password''',
update it from your customer portal at '''Main Menu>> Account Settings>> Security tab>> Main SIP/IAX password'''.

'''Note 2:''' Bear in mind to use the same VoIP server your VoIP number is using. You can check what VoIP server
your VoIP number is using, from your VoIP.ms customer portal at '''DID Numbers>> Manage DIDs''', under '''POP''' column.
You can choose any server you want, as long as the one in your portal and the one in this field matches, otherwise,
incoming calls won't ring.

'''Step 4 -''' At this point your VoIP.ms account should be marked with a <font color="green">Green</font> dot next to its name. This means your Softphone.Pro is ready to start placing calls. '''In case of failure''', your account will be marked on <font color="red">Red</font> and will display an error.

[[File:SoftphonePro_ready.png|thumb|none|300px|Click to enlarge]]

'''Finally,''' you can check if your account is registered from your customer portal, at the main page. If your account is now registered and you have set your caller ID number for outgoing calls, you will be able to start placing outgoing calls.

[[File:Acc registered portal TOR1.png|600px|Click to enlarge]]

'''Note 3:''' Bear in mind to set the '''Caller ID Number''' for your outgoing calls from your customer portal. If you're using the
main account at '''Main Menu>> Account Settings>> General tab>> Caller ID Number'''. If you're using a sub account at
'''Sub accounts>> Manage Sub Accounts>> Edit Sub-account''' (orange icon with a pen)'''>> Caller ID Number'''.

'''<font style="color:#FF0000">IMPORTANT for Name/Display Name field (Outbound callerID Name)</font>''':
- We suggest entering your outbound Caller ID Name in '''capital letters'''. This will appear 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.

==Advanced setup and features==

=== Audio codecs ===

In order to use only supported codecs by VoIP.ms please go to '''Settings''' (headset icon on the top left corner) >> '''SIP Settings''' and scroll down to the section '''Audio Codecs'''. You can hover over any codec and move it to the other column or change the order by using the gray arrow buttons. We suggest keeping audio codecs '''G722''', '''G711 u-law''', and '''G729''' in '''Selected Codecs''' column and in this order.

'''Note 4:''' VoIP.ms uses G711 by default so, to use G722 or G729 as the primary codec, please disable G711u from the customer portal at
Main Menu >> Account Settings >> Advanced tab.
If you are using a Sub-Account, you have to do it from Sub Accounts >> Manage Sub Accounts >> Edit (the pencil icon) and then click on
"Advanced options - Click here to display".

[[File:SoftphonePro Codecs.png|thumb|none|600px|Click to enlarge]]

===Call Encryption TLS/SRTP===
If you decide to use '''SIP TLS - call encryption''' along with '''Softphone.Pro''' please follow these steps:

'''1.''' Make sure your Main account or sub-account has '''"Encrypted SIP Traffic"''' enabled.

Bear in mind, if this setting is enabled and your device sends '''UDP/TCP''' or '''RTP''' you will be rejected
with error code 488.

Enable this setting for the Main Account at '''Main Menu>> Account settings>> Advanced tab'''.

[[File:Mainacc encryp.png|thumb|none|600px|Click to enlarge]]

For a sub-account enable it at '''Sub accounts>> Manage sub-accounts''' by clicking on the orange icon with a pen and finally click at '''"Advanced Options (Click here to display)"'''.

[[File:Subacc encryp.png|thumb|none|600px|Click to enlarge]]

'''2.''' Now that your account/sub-account has this setting enabled, your Softphone.Pro only needs to send '''TLS''' and '''SRTP'''.

To enable '''TLS''', go to '''Settings>> SIP Accounts>> Your Account''' and change the following options there:

*'''Media encryption''': Mandatory.
*'''Transport''': TLS.

'''Note 5:''' When using TLS is very important to specify the number of the server, in case the server's name doesn't have the number "1" included, you need to add it.
Adding any of the SIP ports '''5061/5081/42873''' at the end of the '''SIP Server''' might be required too (E.g. houston1.voip.ms:5061).

[[File:SoftphonePro_Encryption_configuration.png|thumb|none|600px|Click to enlarge]]

You can check if '''Softphone.Pro''' is fully registered and using '''SIP-TLS''' protocol from your customer portal at Home page, '''Main menu>> Portal Home''' for each account/sub account registered or at '''Sub Accounts>> Manage Sub accounts''' tab to see all of your Sub Accounts registration status.

A <font color="green">green padlock</font> [[File:green_padlock.png]] will appears on the right of <font color="green">"Registered"</font> in green. If you don’t see the padlock, you need to revalidate some configuration.

[[File:TLS-SRTP-Steps0006.png|thumb|none|600px|Click to enlarge]]

[[File:TLS-SRTP-Steps0007.png|thumb|none|600px|Click to enlarge]]

==SMS Configuration==

=== Softphone.pro SMS settings ===

Softphone.pro introduced SMS messaging in their 5.3 version* and this can be configured with your VoIP.ms DIDs for outbound texting.

In Softphone.pro go to "Settings" and click on the "Messaging" section.

There, check the option '''Enable messaging and texting (SMS)'''.

Also, set '''Messaging communication protocol''' to '''SIP MESSAGE'''.

[[File:Softphonepro_nable_messaging.png|thumb|none|600px|Click to enlarge]]

*'''IMPORTANT:''' Make sure you have installed the 5.3 version or newer to be able to use SMS with Softphone.pro.

*'''IMPORTANT:''' Currently is only possible outbound messaging with Softphone.pro.
Inbound is yet to be released but you can use the alternatives offered by VoIP.ms to get your inbound messages.
Check the VoIP.ms [https://wiki.voip.ms/article/SMS-MMS SMS wiki article] for more information.

*'''IMPORTANT:''' Please note MMS is not supported via SIP.

=== VoIP.ms SMS configuration ===

To make use of the SMS service of VoIP.ms in Softphone.pro you need to make sure the following options are properly set up:

*'''SMS is enabled:''' From the settings of your DID make sure the SMS service is active. This can be done from DID numbers >> Manage DIDs >> Edit (the pencil icon). You will find the SMS settings at the end of this page.
*'''SMS SIP Account is enabled:''' Still from the DID settings you need to enable the option "Link the SMS received to this DID to a SIP Account" and select the SIP Account you used to register Softphone.pro.
*'''The Caller ID number is a DID with SMS enabled:''' In the settings of the SIP account (Main or Sub), make sure the Caller ID number set is the DID you enabled SMS for. The CallerID Number option is available for the Main Account at Main Menu >> Account Settings >> General. For a Sub-Account go to Sub Accounts >> Manage Sub Accounts >> Edit (the pencil icon).

'''IMPORTANT:''' To see more details about the SMS configuration for SIP Messaging please refer to this [https://wiki.voip.ms/article/SMS-MMS#Send_and_Receive_Messages_.28via_SIP_MESSAGE_Protocol.29 Wiki article].

==Guide Links==

Softphone.Pro Help: [https://softphone.pro/en/help/ Knowledge Base]

Softphone.Pro Contact: [https://softphone.pro/en/contact-us Contact Us]

Softphone.Pro SMS guide: [https://softphone.pro/en/help/softphone-messaging Messaging in Softphone.Pro]

[[Category:Softphones]]

Cisco SPA525G

2023-09-08T01:20:09Z

Carlb: /* Guide Links */

[[File:525.jpg]]

The Cisco SPA525G 5-line IP Phone with Color Display is a full-featured VoIP (Voice
over Internet Protocol) phone that provide voice communication over an IP
network. It provides traditional features, such as call forwarding, redialing, speed
dialing, transferring calls, conference calling, and accessing voice mail. Calls can
be made or received with a handset, headset or speaker.

Your Cisco IP Phone provides a web interface for the phone user that allows you to
configure some features of your phone by using a web browser.

This article will guide you through the steps for basic configuration to make it work with VoIP.ms

----
<br>
<font size="4">'''Step 1'''</font>
<br>
'''''Get the IP address of your phone'''''

a. Press Setup.<br>
b. Select to Status > Network Status.<br>
c. Scroll to view IP Address. This is the IP address of your phone.<br>

You should now have a number which is similar to 192.168.xxx.xxx

<font size="4">'''Step 2'''</font><br>
'''''Logging in to the Phone Web User Interface'''''

*On your PC, open a Web browser window. Your PC must be on the same subnetwork as the phone.
*Enter the IP address in the browser address bar.

You will now see this screen:<br>

[[File:525 1.gif|center]]
<br>
*Click on the '''"Admin Login"''' button near the top right side of the screen, then click on the '''"Ext 1"''' tab.

[[File:525 2.gif|center]]

<font size="4">'''Step 3'''</font>
<br>
'''''Configure with your VoIP.ms account'''''

Find the following fields on the '''"Ext"''' tab and configure accordingly.

'''Nat Keep Alive:''' Yes<br>
'''Nat Mapping/Traversal:''' Yes<br>
'''Proxy:''' atlanta.voip.ms (one of the multiple VoIP.ms servers, you can choose the one closer to your location.)<br>
'''Register Expires:''' 300<br>
'''Display Name:''' Your outbound callerID Name '''(See the requirements below)''' (Replace with your name or company name)<br>
'''User ID:''' 100000 (Replace with your 6 digits Main SIP account UserID or Sub Account user name, e.g. 123456 or 123456_sub)<br>
'''Password:''' ******** (SIP Account Password)<br>
'''Use DNS SRV:''' NO<br>
'''DNS SRV Auto Prefix:''' NO<br>

'''IMPORTANT for Display 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.

[[File:cisco-spa525g.gif|center]]

*If a second extension is needed, click on '''"Ext 2"''' and repeat Step 3. Please make sure to increment the SIP port by one. For example, Ext 1 SIP port: 5060;<br> Ext 2 SIP port: 5061. Make sure you also click on Phone Tab to route the Extensions to the proper lines.

<font size="4">'''Step 4'''</font>
<br>

'''''Configure your dial plan'''''

This step can be considered optional however this is a dial plan that is optimized to work with VoIP.ms service.

Find the dial plan section of your line and enter the following string:

'''(911S0|310xxxx|<:1555>[2-9]xxxxxx|1[2-9]xx[2-9]xxxxxxS0|[2-9]xx[2-9]xxxxxxS0|*xx|***xxx|*xx.|[3468]11|822|0|00|4xxx|**275*x.|xxxxxxxxxxxx.)'''

<font size="2" color="blue">'''If done properly, after completing all these steps your phone will now be ready to place and receive calls!'''</font>

== '''Documentation:''' ==

* You can check the most commonly used Star Codes from [[Cisco/Linksys Star Codes]]

[http://www.cisco.com/en/US/docs/voice_ip_comm/csbpipp/ip_phones/user/guide/525g_sip_user_guide_source/spa525_sip_user.pdf User´s Manual]<br>

keywords: SPA525

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

Admin Manual : [https://www.cisco.com/c/dam/en/us/td/docs/voice_ip_comm/csbpipp/ip_phones/administration/guide/spa_wip_admin.pdf Download PDF]

[[Category:SIP phones]]

Grandstream HandyTone 802 - HT802

2023-09-08T01:17:10Z

Carlb: /* Guide Links */

[[File:HT802 Device.jpg|300px|thumb|left]]

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

'''Websites:''' [https://www.grandstream.com/products/gateways-and-atas/analog-telephone-adaptors/product/ht802 Grandstream HT802]

'''Help / Support:''' [http://www.grandstream.com/support Grandstream Support]

<br clear="all" />

__TOC__

----

===Configuring the HandyTone 802===

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

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

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

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

====Plugging the HT802====

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

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

Finally plug the supplied power cable into the HandyTone.

====Getting IP address for the GUI====

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

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

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

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

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

====Loging into the device====

You should now see a page that looks like this:

[[File:HT802 Login.jpg|thumb|none|600px]]

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

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

[[File:HT802_FirstPage.PNG|thumb|none|600px]]

====Configuring device's port FXS====

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

Please use the same server in the Failover SIP Server as your Primary SIP Server or leave the Failover SIP Server field Blank.

Use the following settings to configure your VoIP.ms account: Configuration Page Settings:

* '''''Primary SIP Server''''': servername.voip.ms (one of VoIP.ms multiple [[Choosing_Server#Choosing_a_Server | '''''servers''''']], you can choose the one closest to your location.)

'''Notice that it is necessary to use the same server for both the device and the DID number in order to get incoming calls correctly'''

You can also find this information by logging into your Customer portal.

* '''''Failover SIP Server''''': (Please leave this Blank)

* '''''Outbound Proxy''''': servername.voip.ms (Use the same server you used as '''''Primary SIP Server'''''.
''For firmware 1.0.15.4 and higher we recommend leaving blank the outbound proxy field''

* '''''NAT Traversal''''': Keep-Alive

* '''''SIP User ID''''': (Replace with your Main SIP account or Subaccount UserID, e.g. 100000 or 100000_sub)

* '''''Authenticate ID''''': (Replace with your Main SIP account or Subaccount UserID, e.g. 100000 or 100000_sub)

* '''''Authenticate Password''''': ****** (Use the SIP account password - By default this is the same as the Customer Portal)

* '''''Name''''': Outbound callerID Name* '''See the requirements below.'''

* '''''DNS Mode''''': A Record

* '''''SIP Registration''''': Yes

* '''''Unregister On Reboot''''': No

* '''''Outgoing Call Without Registration''''': Yes

* '''''Register Expiration''''': 5

* '''''Allow Incoming SIP Messages from SIP Proxy Only''''': Yes

* '''''Preferred DTMF method''''': In-audio, RFC2833

* '''''Use P-Access-Network-Info Header''''': No

* '''''Use P-Emergency-info Header''''': No

* '''''Enable Call Features''''': No

* '''''Dial Plan''''': {[x*]+}

* '''''Preferred Vocoder''''': PCMU, PCMA, G729

'''IMPORTANT''': Outbound CallerID Name
- We suggest entering your outbound Caller ID Name must be in '''capital letters'''. This will appears more clearly/visible on some devices.
- You must '''NOT''' use any special characters, they will not be displayed.
- Some of regular Canadian providers will not show more than '''15 characters max'''. We suggest shrinking or adapt your caller ID.
- Spaces are allowed in a caller id name.

[[File:HT802_FXS_Port.jpg|thumb|none|600px]]

====Saving the changes====

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

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

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

Or

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

== Preventing Direct IP calls like 100 & 1000 ==

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

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

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

==Auto Provisioning==

Some newer models of the HT802 now have Auto Provisioning and will delete the changes you make in setting up the device to use our service. Please go to your Graphical user interface and go to 'Advanced Settings' tab and look for "Firmware Upgrade and Provisioning" and disable it.

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

User Manual : [http://www.grandstream.com/sites/default/files/Resources/ht80x_user_guide.pdf Download PDF]

Admin Manual : [http://www.grandstream.com/sites/default/files/Resources/ht80x_administration_guide.pdf Download PDF]

[[Category:Analog Telephone Adapters]]

Grandstream WP810, WP820, WP822 and WP825

2023-09-07T22:02:15Z

Carlb: /* Configuration */

The '''Grandstream WP810, WP820, WP822 and WP825''' are two-line cordless [[IP Phone|IP phones]] 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 ==
[[Image:Grandstream WP810.png|thumb|right|Grandstream WP810]]
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.

1. Insert the supplied lithium-ion battery and place the handset in the charger.

2. 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".

3. The handset will show a list of available wi-fi networks. Select one, then select "Password" and enter your local wi-fi password.

4. Press [Connect]; the display should show "Wi-Fi Networks" with the list of network names and "Connected" for the selected network.

5. Return to the main menu. Select [status] (the ''i'' icon) then "Network Status". The handset will display IPv4 info, which includes the IPv4 address.

6. 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.

7. 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.

8. 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.

[[Image:Grandstream WP810 Account 1 General Settings.png|Acccount 1 - General Settings]]

9. 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

10. 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.

11. 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.

12. 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.

13. 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.

14. 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.

15. 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.

== 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.

{||
|| +,1,2,3,4,5,6,7,8,9,0, *, #, A,a,B,b,C,c,D,d
|| Individual digit, to be passed through verbatim.
|-
|| x
|| matches any digit from 0-9;
|-
|| xx+
|| at least 2-digit number;
|-
|| xx
|| exactly 2-digit number;
|-
|| xx?
|| 1 or 2-digit numbers from 0-9
|-
|| ^
|| exclude;
|-
|| .
|| wildcard, matches one or more characters
|-
|| [3-5]
|| any digit of 3, 4, or 5.
|-
|| [147]
|| any digit 1, 4, or 7.
|-
|| <2=011>
||replace digit 2 with 011 when dialling.
|-
|| <=1>
|| add a 1 to dialled number
|-
|| <1=>
|| remove 1 from the number dialled
|-
|| |
|| 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 }
|-
|| \+
|| + sign
|-
|| 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 '' { x+ | \+x+ | *x+ | *xx*x+ | x+*x+*x+*x+ | x+*x+*x+*x+#x+ } ''

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.

== Guide 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: SIP phones]]

Grandstream HandyTone 502 - HT502

2023-09-07T01:47:56Z

Carlb: /* Building your own dial plan */ - looks to be less restrictive on later models like Grandstream HandyTone 802 - HT802

[[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.

'''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:''' 100000 (Replace with your Main SIP account or Subaccount UserID, e.g. 198765 or 198765_sub)
*'''Authenticate ID:''' 100000 (Replace with your Main SIP account or Subaccount UserID, e.g. 198765 or 198765_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]]

Grandstream WP810, WP820, WP822 and WP825

2023-09-06T23:46:35Z

Carlb: /* Dial plan */

The '''Grandstream WP810, WP820, WP822 and WP825''' are two-line cordless [[IP Phone|IP phones]] 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 ==
[[Image:Grandstream WP810.png|thumb|right|Grandstream WP810]]
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.

1. Insert the supplied lithium-ion battery and place the handset in the charger.

2. 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".

3. The handset will show a list of available wi-fi networks. Select one, then select "Password" and enter your local wi-fi password.

4. Press [Connect]; the display should show "Wi-Fi Networks" with the list of network names and "Connected" for the selected network.

5. Return to the main menu. Select [status] (the ''i'' icon) then "Network Status". The handset will display IPv4 info, which includes the IPv4 address.

6. Open a web browser to the displayed address, which will look like https://192.168.1.123 (or whatever address your network dynamically assigns). The browser will display a warning that the device is using an "unsigned certificate"; select "proceed anyway" to go to the login screen.

7. 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.

8. 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.

[[Image:Grandstream WP810 Account 1 General Settings.png|Acccount 1 - General Settings]]

9. 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

10. 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.

11. 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.

12. 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.

13. 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.

14. 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.

15. 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.

== 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.

{||
|| +,1,2,3,4,5,6,7,8,9,0, *, #, A,a,B,b,C,c,D,d
|| Individual digit, to be passed through verbatim.
|-
|| x
|| matches any digit from 0-9;
|-
|| xx+
|| at least 2-digit number;
|-
|| xx
|| exactly 2-digit number;
|-
|| xx?
|| 1 or 2-digit numbers from 0-9
|-
|| ^
|| exclude;
|-
|| .
|| wildcard, matches one or more characters
|-
|| [3-5]
|| any digit of 3, 4, or 5.
|-
|| [147]
|| any digit 1, 4, or 7.
|-
|| <2=011>
||replace digit 2 with 011 when dialling.
|-
|| <=1>
|| add a 1 to dialled number
|-
|| <1=>
|| remove 1 from the number dialled
|-
|| |
|| 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 }
|-
|| \+
|| + sign
|-
|| 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 '' { x+ | \+x+ | *x+ | *xx*x+ | x+*x+*x+*x+ | x+*x+*x+*x+#x+ } ''

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.

== Guide 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: SIP phones]]

Grandstream WP810, WP820, WP822 and WP825

2023-09-06T23:05:07Z

Carlb: /* Dial plan */

The '''Grandstream WP810, WP820, WP822 and WP825''' are two-line cordless [[IP Phone|IP phones]] 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 ==
[[Image:Grandstream WP810.png|thumb|right|Grandstream WP810]]
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.

1. Insert the supplied lithium-ion battery and place the handset in the charger.

2. 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".

3. The handset will show a list of available wi-fi networks. Select one, then select "Password" and enter your local wi-fi password.

4. Press [Connect]; the display should show "Wi-Fi Networks" with the list of network names and "Connected" for the selected network.

5. Return to the main menu. Select [status] (the ''i'' icon) then "Network Status". The handset will display IPv4 info, which includes the IPv4 address.

6. Open a web browser to the displayed address, which will look like https://192.168.1.123 (or whatever address your network dynamically assigns). The browser will display a warning that the device is using an "unsigned certificate"; select "proceed anyway" to go to the login screen.

7. 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.

8. 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.

[[Image:Grandstream WP810 Account 1 General Settings.png|Acccount 1 - General Settings]]

9. 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

10. 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.

11. 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.

12. 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.

13. 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.

14. 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.

15. 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.

== 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 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.

{||
|| +,1,2,3,4,5,6,7,8,9,0, *, #, A,a,B,b,C,c,D,d
|| Individual digit, to be passed through verbatim.
|-
|| x
|| matches any digit from 0-9;
|-
|| xx+
|| at least 2-digit number;
|-
|| xx
|| exactly 2-digit number;
|-
|| xx?
|| 1 or 2-digit numbers from 0-9
|-
|| ^
|| exclude;
|-
|| .
|| wildcard, matches one or more characters
|-
|| [3-5]
|| any digit of 3, 4, or 5.
|-
|| [147]
|| any digit 1, 4, or 7.
|-
|| <2=011>
||replace digit 2 with 011 when dialling.
|-
|| <=1>
|| add a 1 to dialled number
|-
|| <1=>
|| remove 1 from the number dialled
|-
|| |
|| 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 }
|-
|| \+
|| + sign
|-
|| 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 '' { x+ | \+x+ | *x+ | *xx*x+ | x+*x+*x+*x+ | x+*x+*x+*x+#x+ } ''

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.

== Guide 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: SIP phones]]

Grandstream WP810, WP820, WP822 and WP825

2023-09-06T22:48:56Z

Carlb: /* Notes */

The '''Grandstream WP810, WP820, WP822 and WP825''' are two-line cordless [[IP Phone|IP phones]] 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 ==
[[Image:Grandstream WP810.png|thumb|right|Grandstream WP810]]
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.

1. Insert the supplied lithium-ion battery and place the handset in the charger.

2. 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".

3. The handset will show a list of available wi-fi networks. Select one, then select "Password" and enter your local wi-fi password.

4. Press [Connect]; the display should show "Wi-Fi Networks" with the list of network names and "Connected" for the selected network.

5. Return to the main menu. Select [status] (the ''i'' icon) then "Network Status". The handset will display IPv4 info, which includes the IPv4 address.

6. Open a web browser to the displayed address, which will look like https://192.168.1.123 (or whatever address your network dynamically assigns). The browser will display a warning that the device is using an "unsigned certificate"; select "proceed anyway" to go to the login screen.

7. 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.

8. 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.

[[Image:Grandstream WP810 Account 1 General Settings.png|Acccount 1 - General Settings]]

9. 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

10. 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.

11. 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.

12. 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.

13. 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.

14. 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.

15. 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.

== 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. 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.

{||
|| +,1,2,3,4,5,6,7,8,9,0, *, #, A,a,B,b,C,c,D,d
|| Individual digit, to be passed through verbatim.
|-
|| x
|| matches any digit from 0-9;
|-
|| xx+
|| at least 2-digit number;
|-
|| xx
|| exactly 2-digit number;
|-
|| xx?
|| 1 or 2-digit numbers from 0-9
|-
|| ^
|| exclude;
|-
|| .
|| wildcard, matches one or more characters
|-
|| [3-5]
|| any digit of 3, 4, or 5.
|-
|| [147]
|| any digit 1, 4, or 7.
|-
|| <2=011>
||replace digit 2 with 011 when dialling.
|-
|| <=1>
|| add a 1 to dialled number
|-
|| <1=>
|| remove 1 from the number dialled
|-
|| |
|| 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 }
|-
|| \+
|| + sign
|-
|| 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 '' { x+ | \+x+ | *x+ | *xx*x+ | x+*x+*x+*x+ | x+*x+*x+*x+#x+ } ''

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.

== Guide 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: SIP phones]]