VoIP.ms Wiki - User contributions [en]

Musique d'attente personnalisée

2026-07-30T20:07:51Z

Neocypher: /* Article de blog */

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

__TOC__

== Vidéo tutoriel ==

[[Image:ThumbnailCMOH.jpg|200px|link=https://www.youtube.com/watch?v=6d5b_SQT9d8]]

== Article de blog ==
[https://voip.ms/blog/how-to-improve-your-customer-experience-with-custom-music-on-hold/ How to Improve Your Customer Experience with Custom Music on Hold?]

[https://voip.ms/blog/5-examples-of-how-to-use-custom-music-on-hold-in-your-business/ 5 Examples of How to Use Custom Music on Hold in Your Business]

Cette fonctionnalité vous permet de télécharger et d'utiliser vos propres fichiers audio à reproduire lorsque votre client est mis «en attente» ou pendant la «durée de sonnerie» lorsque vous recevez un appel entrant.

Vous pouvez télécharger vos fichiers audio personnalisés à partir de la section '' 'Enregistrements' '' puis les convertir en '' 'fichiers MOH' ''. Par la suite, vous pouvez utiliser vos fichiers MOH dans certaines de vos fonctionnalités telles que:

*'''Files d'attentes'''
*'''Groupe de sonneries'''
*'''Recherche d'appels'''
*'''Audio conférence'''
*'''Compte principal ou sous-comptes'''

'''Point important à prendre en compte avant de créer vos propres enregistrements de musique d'attente'''
La Société autorise le Client à choisir et à diffuser la musique de son choix, pendant que ses clients sont dans une file d'attente d'appels, à la condition expresse que
le Client dispose des licences et droits appropriés sur une telle musique.

En diffusant de la musique, le Client garantit qu'il possède ou dispose des droits et licences appropriés pour diffuser une telle musique, et qu'il comprend que la
Société ne possède aucun droit et licence pour diffuser la musique, mais ne fournit qu'une fonctionnalité permettant au Client de faire ladite diffusion.

En cas de réclamation d'un tiers directement ou indirectement liée à la musique en file d'attente fournie par le Client, le Client défendra, indemnisera et dégagera la
Société de toute responsabilité et contre toute réclamation.

==Téléversement de vos fichiers personnalisés==

Dirigez-vous dans la section '' 'Numéros DID >> Enregistrements' '' et cliquez sur le bouton '' '' Télécharger un nouvel enregistrement '' ''. Dans la fenêtre contextuelle, vous devrez remplir les éléments suivants:

* '' 'Nom:' '' Attribuez un nom à votre enregistrement
* '' 'Fichier:' '' Cliquez sur le bouton '' '"Choisir un fichier"' '' pour rechercher le fichier audio que vous souhaitez ajouter à votre compte.
* '' 'Télécharger:' '' Cliquez sur le bouton '' '' Téléverser un fichier '' '' pour téléverser votre fichier audio.

[[File:MoHFR1.png|thumb|none|600px]]

==Conversion d'enregistrements en fichiers MOH==

Vous devrez maintenant convertir le fichier '' 'Enregistrement' '' que vous venez de télécharger en un '' 'Fichier MOH' ''. Cela peut être fait à partir de 2 sections différentes.

1. À partir de la section '' 'Numéros DID >> Enregistrements' ''

Une fois que vous téléchargez votre enregistrement, vous pouvez cliquer sur l'icône du casque pour convertir facilement cet enregistrement en un fichier MOH, il vous suffit d'attribuer un nom au fichier MOH et de cliquer sur '' '"Enregistrer"' ''

[[File:MoHFR2.png|thumb|none|600px]]

[[File:MoHFR3.png|thumb|none|600px]]

2. À partir de la section '' 'Numéros DID >> Musique d'attente personnalisée' ''

De là, vous pouvez convertir n'importe quel fichier d'enregistrement en un fichier MOH en cliquant sur le bouton '' '"Ajouter une musique en attente personnalisée"' ''.

[[File:MoHFR4.png|thumb|none|600px]]

Dans la fenêtre contextuelle, remplissez la '' 'Description' '' qui est le nom du fichier MOH et faites glisser et déposez les enregistrements de la colonne de gauche (Enregistrements disponibles) vers la colonne de droite (Enregistrements sélectionnés).

Le fichier MOH est réglé par défaut sur "Volume élevé" si vous souhaitez diminuer le volume, vous pouvez activer le bouton "Tout à fait volume". Cliquez sur le bouton '' 'Enregistrer'''.

'''Ordre''' est utilisé soit pour lire les enregistrements que vous ajoutez dans l'ordre que vous les avez défini, soit dans un ordre aléatoire selon ce que vous préférez.

[[File:CMOHnewFR2.png|thumb|none|600px]]

==Utilisation de votre fichier MOH avec une fonctionnalité==

Votre fichier MOH est maintenant prêt à être utilisé avec l'une des fonctionnalités suivantes:

* '' '"Musique en attente"' '' pour un sous-compte ou le compte principal
Pour le compte principal, dirigez-vous sous '' 'Menu Principal >> Paramètres du compte >> Généralités' '' et rechercher le paramètre '' '"Musique d'attente"' ''.

Pour le sous-compte, dirigez-vous sous '' 'Sous-comptes >> Gérer les sous-comptes' '', cliquer sur l'icône avec un stylo pour modifier votre sous-compte et rechercher le paramètre '' '"Musique d'attente"' ''.

[[File:MoHFR6.png|thumb|none|600px]]

* '' '"Musique en attente"' '' pour une file d'attente d'appels
Ceci est disponible sous '' 'Numéros DID >> Files d'attente d'appels' '', cliquez sur l'icône avec un stylo pour modifier votre recherche d'appel ou vous pouvez en créer une nouvelle. Recherchez le paramètre '' '"Musique d'attente"' ''.

[[File:MoHFR6.png|thumb|none|600px]]

* '' '"Musique que l'appelant entendra pendant que l'appel est répondu"' '', votre fichier MOH sera disponible pour un '' 'Groupe de sonnerie' '' ou '' 'Recherche d'appel' ''

Pour un groupe de sonnerie, il est disponible dans '' 'Numéros DID >> Groupes de sonnerie' '', cliquez sur l'icône avec un stylo pour modifier votre groupe de sonnerie ou vous pouvez en créer une nouvelle. Recherchez le paramètre "Musique de l'appelant entendra pendant que l’appel est répondu "

[[File:MoHFR7.png|thumb|none|600px]]

Pour une recherche d'appel, elle est disponible sous '' 'Numéros DID >> Recherche d'appel' '', cliquez sur l'icône avec un stylo pour modifier votre recherche d'appel ou vous pouvez en créer une nouvelle. Recherchez le paramètre "Musique de l'appelant entendra, tandis qu'il est répondu: "

[[File:MohFR8.png|thumb|none|600px]]

Pour l'audioconférence, dirigez-vous dans Numéros DID >> Audioconférence, cliquez sur l'icône avec un stylo pour modifier votre conférence ou vous pouvez en créer une nouvelle. Allez dans l'onglet «participants et paramètres de la conférence», puis créez ou modifiez le profil de votre choix. Dans les '' 'options avancées' '', choisissez votre fichier dans la section "MoH quand vide". Lorsque le profil est activé et qu'il n'y a aucun participant à la conférence, votre musique d'attente sera lue.

=Besoin d'un type de musique spécifique pour votre entreprise ? Ne cherchez pas plus loin=

A la recherche de musique pour votre musique d'attente ? Voici quelques suggestions de sites Web d'IA qui créent de la musique et vous permettent de télécharger pour votre propre usage.
*[https://soundraw.io/create_music Soundraw.io]
*[https://www.epidemicsound.com Epicemicsound]

= Utilisation de la fonction à l'aide de l'interface du revendeur =

La fonctionnalité est disponible pour vos clients via l'interface Revendeur. Vous devez activer cette fonctionnalité dans votre forfait afin de leur donner la possibilité d'utiliser cette fonctionnalité.

Dirigez-vous sous la barre de navigation sur '''[Revendeur]''' puis cliquez sur '''[Gestion des tarifs et des forfaits]'''

: [[File:SequenceReseller1Fr.png|thumb|none|300px]]

Cliquez sur le bouton Modifier afin de modifier votre forfait, ou cliquez sur '''[Créez un nouveau forfait]''' afin d'en créer un nouveau.

: [[File:SequenceReseller2Fr.png|thumb|none|700px]]

Dirigez-vous dans l'onglet '''[Configuration du système du revendeur]''' et dans la section "Type de configuration" selectionner: '''[Configuration du forfait]''',

: [[File:SequenceReseller3Fr.png|thumb|none|700px]]

Défiler ensuite vers le bas afin de trouver la fonction «Musique d'attente personnalisée», puis activez-la.

: [[File:MusicAttentePersonnaliséReseller4Fr.png|thumb|none|500px]]

1) Pour ajouter une musique en attente pour votre client ou pour aider votre client à en ajouter une. Allez sous les [Services] dans la barre de navigation de gauche, puis sur [Custom Music on Hold]

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

2) Si certains enregistrements audio sont disponibles dans la section [Available recordings] de la section [recording], vous pouvez sélectionner un enregistrement et le faire glisser dans la section [Selected Recordings].

Si vous ne voyez aucun enregistrement disponible ou si votre enregistrement n'est pas présent, cliquez sur le bouton [+ Add New] afin d'ajouter un nouveau fichier d'enregistrement.

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

3) Cliquez sur l'onglet [Add New recording], indiquez le nom de l'enregistrement, téléversez-le.

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

4) Une fois téléchargé, vous serez redirigé vers l'onglet avec tous vos enregistrements. Trouvez l'enregistrement que vous souhaitez convertir en MOH personnalisé. Ensuite, cliquez sur le bouton [Écouteurs].

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

5) Donnez un nom à votre nouvelle musique d'attente personnalisée.

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

Votre musique d'attente personnalisée a été créée avec succès. Il sera disponible pour être sélectionné dans une fonction qu'une musique d'attente peut être utilisé comme le groupe de sonnerie, la recherche d'appels, la file d'attente, le compte SIP, etc.

SMS-MMS-ES

2026-07-30T20:06:42Z

Neocypher: /* Artículos del Blog */

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

SMS significa '''S'''hort '''M'''essage '''S'''ervice ('''SMS'''). Con la incorporación del Servicio de Mensajes Multimedia '''M'''ultimedia '''M'''essage '''S'''ervice ('''MMS''') al servicio, los usuarios pueden ahora enviar y recibir archivos multimedia ('''fotos, videos, y audio''') así como enviar mensajes de texto con más de 160 caracteres (que es el límite de un SMS normal). El servicio MMS también permite al usuario enviar un mensaje a varios destinatarios al mismo tiempo.

= Artículos del Blog =
:* [https://voip.ms/blog/benefits-of-sms-mms-campaigns-vs-email-marketing/ Beneficios de las campañas de SMS/MMS vs. Email Marketing]
:* [https://voip.ms/blog/how-to-generate-leads-with-sms-mms/ ¿Cómo generar clientes potenciales con SMS/MMS?]
:* [https://voip.ms/blog/instantly-boost-your-customer-experience-allow-your-customers-to-text-you/ Aumente al instante su experiencia como cliente, permita que sus clientes le envíen mensajes de texto.]

= Limitaciones del Servicio =

<div style="margin-left:50px;
padding:5px;
border-left: 6px solid rgb(208, 56, 45, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(208, 56, 45, 0.08);
width:70vw;">

🏗️ ''' Por favor tenga en cuenta que el servicio de MMS está en fase Beta.'''
: Es importante para nosotros que reporte cualquier problema con este servicio enviando un correo a [mailto:support@voip.ms soporte] para que los desarrolladores puedan intervenir si es necesario.
</div>

<div style="margin-left:50px;
padding:5px;
border-left: 6px solid rgb(208, 56, 45, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(51, 51, 51, 0.08);
width:70vw;">

⚠️ '''LIMITACIONES'''
: ➜ '''El límite de la capacidad de envío de SMS/MMS a través del portal es ilimitado. '''
: ➜ El límite de envío de SMS/MMS a través de la API está limitado a '''<u>100 por dia</u>'''. Para cualquier modificación del límite, por favor póngase en contacto con soporte o abra un ticket a través del portal de clientes.
: ➜ Por el momento, la recepción y lectura de MMS sólo es posible a través del centro de mensajes o utilizando la opción de reenvío de SMS/MMS. '''Los MMS a través del protocolo SIP no están disponibles.'''

: ➜ El servicio de SMS/MMS sólo está disponible para los <u>'''números DID locales de Estados Unidos y Canadá'''</u> marcados con el '''icono de SMS/MMS'''.
: ➜ El servicio SMS/MMS sólo puede utilizarse para enviar mensajes MMS a números <u>'''canadienses y estadounidenses'''</u> de 10 dígitos en este momento..
: ➜ Los mensajes SMS SIP sólo serán entregados y aceptados desde troncales SIP utilizando la autenticación de nombre de usuario/contraseña.
: ➜ Los mensajes SIP SMS utilizan el método SIP MESSAGE. Consulte el [https://tools.ietf.org/html/rfc3428 RFC 3428] para más detalles.
: ➜ En este momento '''<u>no podemos garantizar</u> que los mensajes SMS de código corto''', que suelen tener 6 dígitos o menos ''(por ejemplo: códigos bancarios, anuncios de televisión, etc...)'' funcionen.
:➜ En este momento, '''<u>no podemos garantizar</u> que los mensajes SMS de autenticación de dos factores (2FA), que suelen utilizarse para confirmar su identidad e iniciar sesión en su portal de servicios '''(por ejemplo: Microsoft, entre otros proveedores que ofrecen esta función)''', funcionarán.
: ➜ En este momento, los mensajes SMS enviados/recibidos desde/hacia números de código corto (short codes) '''<u>no son soportados</u> con los Números de Lada sin costo (Toll-Free)'''.
: ➜ No podemos garantizar '''que los acentos o los caracteres especiales''' incluidas las letras no latinas, se entreguen correctamente.

: ➜ Public URL shorteners such as '''''bit.ly, tinyurl.com''''' etc, <font style="color:red">'''están ampliamente restringidos'''</font> por los proveedores de telefonía móvil como respuesta de la industria para combatir las campañas de SPAM y proteger al consumidor. Incluir este tipo de enlaces en tus mensajes '''<u>puede dar lugar a un mensaje bloqueado o no entregado. </u>'''.
: ➜ Para los '''números porteados''' compatibles con la función: La Funcionalidad SMS/MMS estará disponible hasta '''48 horas después de que el proceso de portabilidad se marque como Completado'''.

: ➜ VoIP.ms se reserva el derecho, a su sola discreción, de limitar o deshabilitar el servicio de SMS/MMS de cualquier cuenta que pueda presentar patrones de comercialización/de automatización/de envío o recepción masiva.

</div>

<div style="margin-left:50px;
clear:left;
padding:5px;
border-left: 6px solid rgb(66, 192, 251, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(66, 192, 251, 0.05);
width:70vw;">

'''💬 NOTA''' Si tiene más preguntas, no dude en ponerse en contacto con el personal de soporte en el chat en vivo o en el sistema de tickets.
:➜ '''Los mensajes MMS tienen un costo de $0,02 por mensaje entrante y saliente. '''

: ➜ '''Adicionalmente, tenga en cuenta las siguientes restricciones para el envío de MMS:'''
:: ➜ Los tipos de archivos adjuntos permitidos son JPG, GIF, JPEG, PNG, MP3, WAV, MIDI, MP4 y 3GP.
:: ➜ El tamaño máximo de cada archivo adjunto es de 1300 KB. ''(1.3 MB)''
:: ➜ Puede adjuntar hasta 3 archivos.
:: ➜ El máximo de caracteres en un mensaje de texto es de 2048.

</div>

=Usando el SMS/MMS=
==Identificando un número DID con capacidad de SMS/MMS==

Lo primero que debe saber es que esta función sólo está disponible para los números locales de Estados Unidos '''(incluidos Alaska, Hawái y Puerto Rico).''' y Canadá en este momento. Notará que algunos números tienen un pequeño[[File:Sms-mms-icon.png|15px]] icono, esto indica que el número soporta esta función.

Puede empezar a pedir un número DID desde su portal >> Números DID >> Pedir DID >> Números Locales >> Números (de Estados Unidos o Canadá) y podrá comprar el número que desee, sólo asegúrese de que soporte la función MMS.

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

==Configuración del servicio SMS/MMS==

Una vez adquirido, el Servicio SMS/MMS debe ser activado en la configuración del DID, desde su portal >> > DID Numbers >> Manage DID >> Edit Selection >> sección de configuración de SMS/MMS.

En esa sección, puede activar y configurar algunas opciones de reenvío para el servicio de SMS/MMS.

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

:* '''Message Service (SMS/MMS)''': Marque la casilla para activar el servicio.
:* '''VoIP.ms SMS/MMS Portal''': Está habilitado por defecto y no se puede modificar, esto significa que podrás crear y enviar mensajes desde el Centro de Mensajes SMS/MMS.
:* '''SMS/MMS Email Address''': Activando esta opción todos los Mensajes SMS serán enviados a la dirección de correo electrónico que configure en este campo. Puede recibirlos en más de una dirección de correo electrónico separando cada correo electrónico con una coma. La ventaja de esta opción es que recibirá sus Mensajes SMS directamente a su correo electrónico y podrá responder a estos mensajes desde su correo electrónico también. Sólo tiene que hacer clic en '''responder''' al correo electrónico. Tenga en cuenta que las respuestas al correo electrónico también están limitadas a 160 caracteres y, si superan este límite, se dividirán en dos (o más mensajes si procede) cuando se envíen.

<div style="margin-left:50px;
padding:5px;
border-left: 6px solid rgb(208, 56, 45, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(51, 51, 51, 0.08);
width:70vw;">

'''NOTA''' 💬: ➜ Para responder al Mensaje SMS a través de su correo electrónico es importante hacer clic en '''Responder''' y '''NO cambiar o modificar''' el correo de destino "PARA" (sms@voip.ms) y el asunto, por ejemplo: ''[#USXXXXXX] Message sent to 5555555555''.
</div>

:*'''SMS/MMS Forward''': Puede hacer que sus Mensajes SMS sean reenviados a su celular o a cualquier otro número que soporte SMS, con esta opción recibirá los Mensajes SMS de nuestro sistema al número telefónico configurado.
<div style="margin-left:50px;
padding:5px;
border-left: 6px solid rgb(208, 56, 45, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(51, 51, 51, 0.08);
width:70vw;">

'''NOTA''' 💬: ➜ El identificador de llamadas enviado al teléfono móvil desviado será el número DID que recibió el mensaje SMS. No es posible reenviar el ID de llamada original en este momento.
</div>

:*'''SMS/MMS SIP Account''': Puede hacer que sus mensajes SMS sean reenviados a la cuenta SIP que haya seleccionado. Se enviarán a través de nuestra vía SMS/SIP como un SIP MESSAGE.
<div style="margin-left:50px;
padding:5px;
border-left: 6px solid rgb(208, 56, 45, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(51, 51, 51, 0.08);
width:70vw;">

'''NOTA''' 💬: ➜ El destino campo principal "Para" ("To" header field) del SIP MESSAGE será el nombre de su cuenta, en lugar del número de destino del SMS original.
</div>

:'''SMS/MMS URL Callback''': Devolución de llamada de URL de SMS/MMS: Activando esta opción podrá enviar los mensajes a otro destino, por ejemplo a su propio servidor. Si se selecciona, los mensajes SMS recibidos por su DID enviarán una solicitud de obtención "GET" a la URL Callback (devolución de llamada) proporcionada. Variables disponibles para su URL:
<div style="margin-left:50px;
padding:5px;
font-family: Consolas, Monaco, Lucida Console, Liberation Mono, DejaVu Sans Mono, Bitstream Vera Sans Mono, Courier New, monospace;
color: silver;
border-left: 6px solid rgb(208, 56, 45, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: black;
width:58vw;">
:: '''Ejemplo''': <nowiki>https://SuNombreDeDominio.tld/sms.php?to={TO}&from={FROM}&message={MESSAGE}&id={ID}&date={TIMESTAMP}</nowiki>

: '''{ID}''' = Identificación del mensaje SMS.
: '''{TIMESTAMP}''' = La fecha y la hora en que se recibió el mensaje.
: '''{FROM}''' = l número de teléfono que le envió el mensaje.
: '''{TO}''' = El número DID que recibió el mensaje.
: '''{MESSAGE}''' = El contenido del mensaje.
: '''{MEDIA}''' = Lista de archivos multimedia separados por comas.

</div>

:*'''URL Callback Retry''': Reintento de devolución de llamada de URL: Cuando se selecciona, esperaremos un "ok" (sin comillas) de su página de devolución de llamada URL como indicador de que ha recibido el mensaje correctamente. Si no recibimos el "ok" (sin comillas) de su página de devolución de llamada, seguiremos enviándole el mismo mensaje cada 30 minutos.

Una vez configurada la opción deseada, haga clic en Apply changes.

==Enviando un SMS/MMS (Interfaz web)==

'''Paso 1'''

: Haga clic en el botón "enviar nuevo mensaje" para que se muestre una lista de contactos o escriba directamente el número en el campo de texto del contacto. Si este número está registrado en la agenda, se mostrará su nombre en su lugar.

:: [[File:MMS-Webportal1.png|border|400px]]

'''Paso 2'''

: Haga clic para mostrar la lista de números DID disponibles. Elija el número desde el que desea enviar el mensaje.

:: [[File:MMS-Webportal3.png|border|400px]]

'''Paso 3'''

: Escriba su mensaje en el campo de texto y pulse la flecha roja para enviarlo. Los mensajes de texto de hasta 160 caracteres se considerarán SMS. Los mensajes multimedia o los mensajes de texto de más de 160 caracteres se considerarán MMS y verá que el icono pasa de SMS a MMS para indicarlo.
:: [[File:MMS-Webportal4.png|border|400px]]

'''Paso 4'''

: Para enviar un archivo multimedia a través de un MMS, haga clic en el botón de clip para buscar y elegir un archivo multimedia.

:: [[File:MMS-Webportal5.2.png|border|400px]]

: Haga clic en el botón "Browse" para buscar el archivo multimedia en su ordenador o arrastre y suelte el archivo directamente en el recuadro. .

:: [[File:MMS-Webportal6.PNG|border|400px]]

: Los archivos adjuntos se mostrarán debajo del cuadro de texto.

:: [[File:MMS-Webportal5.PNG|border|400px]]

: Una vez enviado el mensaje, se abrirá una nueva conversación a la izquierda. Al seleccionar esta conversación se mostrará el historial de mensajes en el recuadro.

:: [[File:MMS_WINDOW_CONVO.png|border|400px]]

==Recibiendo un SMS/MMS (Interfaz web)==

: Mientras que los mensajes SMS pueden ser recibidos directamente en una aplicación a través del Protocolo SIP y una correcta configuración, por el momento, '''la recepción y lectura de MMS sólo es posible a través del centro de mensajes.'''

: Puede acceder al centro de comunicación desde su portal de cliente en DID numbers > SMS/MMS Message Center. Cualquier mensaje recibido aparecerá aquí en la parte izquierda del panel.

: Utilice DID Filter para elegir y mostrar los mensajes de un número DID específico o de todos los números DID.

: También puede filtrar los mensajes por fecha y zona horaria.

:: [[File:SMSMMSfilter1.png|border|300px]]

:: [[File:MMS-Inbound-1.png|border|500px]]

: Choose a conversation from the left panel. An option to delete the conversation is also available. Choose the DID number from the text box to reply.

:: [[File:MMS-Inbound_DELETE.png|border|500px]]

== Historial de SMS / MMS ==

Ahora también puede obtener acceso a un historial general de SMS / MMS que se envió y recibió para todos sus DID, o para uno específico haciendo clic en '''History'''.

:: [[File:SMSMMSHistoryLoc.png|border|500px]]

:: [[File:History.png|border|500px]]

= Correo Electrónico a SMS =

A partir del 14 de enero de 2025, ahora es posible iniciar una conversación SMS a través de correo electrónico. Para habilitar esta función, diríjase a su portal de VoIP.ms, DID Numbers, SMS/MMS Message Center, y luego haga clic en « Email to SMS ».

: [[File:EmailtoSMS1.png|border|500px]]

Luego, haga clic en « Add New Email to SMS » y complete la siguiente información:

*Activado: Sí
*Dirección de correo electrónico: Su dirección de correo electrónico
*Código de seguridad activado: Si desea activar un código de seguridad para enviar SMS/MMS.
*Código de seguridad: El código de seguridad que se usará.
*Número remitente predeterminado: El número DID que desea usar para enviar el mensaje SMS/MMS.

: [[File:EmailtoSMS2.png|border|500px]]

Una vez que haya completado la información, presione « Save » ¡y listo!

== Cómo iniciar una conversación SMS/MMS a través de correo electrónico ==
1. Utilice la cuenta de correo electrónico que proporcionó al activar el servicio de correo electrónico a SMS.

2. Enviar el correo a sms@voip.ms.

3. En el campo de asunto, ingrese el número SMS del destinatario (Ejemplo: 5552340000).

4. Adjunte un documento al mensaje de correo electrónico si desea enviar multimedia.

5. Tipos de archivos adjuntos permitidos: JPG, GIF, JPEG, PNG, MP3, WAV, MIDI, MP4 y 3GP.

6. El tamaño máximo de cada archivo adjunto es de 1300 KB.

7. Puede adjuntar hasta 3 archivos.

8. Enviar el correo electrónico.

===Número de seguridad y número remitente:===

1. Si el código de seguridad está activado, necesitará añadir un punto (.) y el código de seguridad después del número SMS de destino (Ejemplo: 5552340000.Az09).

2. Si tiene más de un número SMS, puede cambiar el número remitente añadiendo un punto (.) y el número remitente que desea utilizar después del número SMS de destino y el código de seguridad (ejemplo: 5552340000.Az09.5557890000).

3. Si no ha activado el código de seguridad y desea cambiar el número remitente, puede añadir un punto (.) y el número remitente después del número de SMS de destino (Ejemplo: 5552340000.5557890000).

= SMS utilizando un servidor SMPP. =

Un servidor SMPP, abreviatura de '''Short Message Peer to Peer''' es un método diseñado para enviar y recibir grandes volúmenes de mensajes SMS a través de un servidor dedicado.

En otras palabras, esto le permite controlar mejor la cantidad de mensajes que se enviarán a su público objetivo, evitando pasar por una aplicación de softphone o por su portal de VoIP.ms para enviar/recibir estos SMS, eliminando así múltiples tareas repetitivas y ahorrando tiempo.

Por ejemplo, si tiene una promoción para su negocio que ofrece un descuento durante un determinado lapso de tiempo y que sus clientes han dado su conformidad para recibir este tipo de promociones vía SMS, podría enviar su promoción a sus clientes de una manera más rápida que si lo hiciera desde su portal de clientes, aumentando así la eficiencia.
<div style="margin-left:50px;
padding:5px;
border-left: 6px solid rgb(208, 56, 45, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(51, 51, 51, 0.08);
width:70vw;">

'''NOTA''' 💬: ➜ Tenga en cuenta que el envío y la recepción de mensajes SMS se cobrará a nuestra tarifa habitual de 0.0075 dólares por mensaje.'''
</div>

==Requisitos==

: '''Esto requerirá tener su propio servidor SMPP.'''
: Una vez configurado correctamente, podrá enviar y recibir mensajes SMS a su propio servidor.

== Configuración en su servidor SMPP ==

: Para que su servidor se comunique y se verifique adecuadamente con el servidor SMPP de VoIP.ms, deberá configurar lo siguiente: :

::* Un '''nombre de usuario''' y '''una contraseña ''' de su elección para que el servidor SMPP de VoIP.ms le verifique.
::* Haga que todas sus comunicaciones se envíen a '''smpp.voip.ms''' a través del puerto '''2775''', para el SMPP normal, o del puerto '''3550''' que es nuestro puerto SMPP cifrado.
::* Abra una conexión utilizando el comando '''bind_transceiver''' Puede mantener esta conexión abierta todo el tiempo que necesite y consultar sobre ella mediante el comando '''enquire_link'''.
::* Envíe sus mensajes con los comandos '''deliver_sm''' o '''submit_sm''' . Asegúsere de incluir el '''source_addr''' de su PDU como uno de los DIDs habilitados para SMS de su cuenta. Su número de SMS de destino debe establecerse como '''destination_addr''' mientras que su mensaje de texto se incluirá en short_message. '''short_message'''.

== Configurando SMPP en su Portal VoIP.ms ==

Una vez que haya configurado lo anterior en su servidor, tendrá que habilitar y especificar qué nombre de usuario y contraseña serán utilizados por VoIP.ms para autenticar sus solicitudes SMPP. Para ello, diríjase a su portal de VoIP.ms, DID Numbers, Manage DIDs, edite el DID que soporta mensajes SMS y bajando por la página, verá las siguientes opciones:

:* '''SMPP Enabled:''' SMPP Enabled: Si se selecciona, se permitirá el envío y la recepción de mensajes SMS mediante SMPP.
:: Puede especificar la URL que se utilizará para enviar una copia de los mensajes entrantes a su servidor SMPP. Déjela vacía si sólo desea enviar mensajes. Véase el ejemplo siguiente. La parte del '''esquema''' es obligatoria y debe establecerse como smpp para '''smpp''' no encriptado, o '''ssmpp''' para SMPP encriptado usando TLS. La parte '''user:password@''' es opcional y será por defecto el mismo nombre de usuario y contraseña definidos para la autenticación en nuestro lado como se define a continuación. La parte ''':port''' también es opcional y por defecto será 2775 para '''smpp''' y'''3550''' para ssmpp.

:* '''SMPP Username:''' El nombre de usuario que utilizará para autentificarse en nuestro servidor SMPP para el envío de mensajes.
:* '''SMPP Password:''' La contraseña que utilizará para autentificarse en nuestro servidor SMPP para el envío de mensajes.

<div style="margin-left:50px;
padding:5px;
border-left: 6px solid rgb(208, 56, 45, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(51, 51, 51, 0.08);
width:70vw;">

'''NOTAS''' 💬
: ➜ Para evitar cualquier confusión al configurar el nombre de usuario en su servidor, puede utilizar el ID de su cuenta de VoIP.ms como nombre de usuario (los 6 dígitos de su cuenta).
: ➜ Asegúrese de tener activada la opción Servicio de '''Message Service (SMS/MMS)'''.
</div>

Active la opción '''SMPP Enabled''', rellene el nombre de usuario y la contraseña según lo configurado en su servidor SMPP y una vez hecho esto, aplique los cambios pulsando '''Click here to apply changes'''.

: [[File:SMPPconfig.png|border|750px]]

Ahora puede empezar a enviar y recibir mensajes SMS desde su servidor SMPP.

= Enviar y recibir mensajes de texto (Interfaz Web) =

==Centro de mensajes SMS para móviles==

Para usar el servicio deberá utilizar el Centro de Mensajes SMS, desde su portal >> DID Numbers >> [https://www.voip.ms/m/sms.php SMS Message Center].

Para los dispositivos móviles, utilice [https://sms.voip.ms/ sms.voip.ms] como una solución completa todo en uno.

Desde el Centro de Mensajes SMS (Message Center) puede consultar su historial de SMS, enviar y recibir mensajes SMS.

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

:* '''Search Range (Rango de búsqueda)''': Seleccione un rango de fechas de su historial de SMS. Puede seleccionar un periodo de tiempo marcando el campo "Show Details" y se mostrarán los mensajes de texto.
:* '''Search Filter (Filtro de búsqueda)''': Puede filtrar su búsqueda por DID, Contacto y Tipo (Enviado, Recibido y Ambos).
:* '''Send New SMS Messages (Enviar nuevos mensajes SMS)''': Al hacer clic aquí se abrirá otra ventana para crear un nuevo mensaje SMS.
:* '''Delete Selected Messages (Eliminar los mensajes seleccionados)''': Esta acción borrará los mensajes que haya seleccionado, haciendo clic en la casilla que hay junto a ellos. '''Esta acción no se puede deshacer.'''
:* '''Delete All Messages (Borrar todos los mensajes)''': Borre TODOS sus mensajes SMS, tanto los enviados como los recibidos '''Esta acción no se puede deshacer. '''
:* '''Checking a Message (Comprobación de un mensaje)''': Para comprobar un mensaje SMS recibido basta con hacer clic directamente en él para que aparezca la información.

== Crear un nuevo mensaje SMS (Legacy) ==

: [[File:SMS41.png|thumb|none|600px]]

Puede crear un nuevo '''Mensaje SMS''' le explicaremos todos los parámetros que puede encontrar en esta sección:

* '''DID''': Si tiene más de un número DID puede seleccionar cualquiera de los números disponibles para enviar su mensaje.
* '''Contact''': El número de teléfono de destino o si tiene una entrada en el [[Directorio_(Phone_Book)|directorio telefónico]] comience a escribir el nombre y el sistema mostrará la información si está disponible.
* '''Message''': Puede crear su mensaje en ese campo, tenga en cuenta que tiene hasta 160 caracteres.

A continuación, haga clic en "'''Send Message'''" para enviar el SMS.

= Envío y recepción de mensajes (mediante el protocolo SIP MESSAGE) =
<div style="margin-left:50px;
padding:5px;
border-left: 6px solid rgb(208, 56, 45, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(51, 51, 51, 0.08);
width:70vw;">

'''NOTAS''' 💬
: ➜ El servicio de '''SMS via SIP''' sólo está disponible a través del '''Protocolo SIP y la autenticación de usuario/contraseña'''. '''IP Auth e IAX2 no están disponibles en este momento.'''
: ➜ '''Consulte nuestra guía [[SIP/SMS with FreePBX]] para obtener instrucciones sobre cómo configurar el servicio con un FreePBX'''
</div>

== Enviando ==

:* DID debe tener habilitada la opción de servicio '''"Short Message Service (SMS)"''' service option enabled. Esta opción se puede habilitar desde la página '''Manage DIDs > Edit DID'''.

:* Para enviar un SMS desde su cuenta SIP, es necesario que configure su número de identificación de llamadas (Caller ID) en uno de sus DID habilitados para SMS. Este será el número desde el que enviará el mensaje. Puede configurar el número de identificación de llamadas desde su portal de clientes para la subcuenta específica (Sub-Account) si está usando un softphone o directamente desde su extensión o troncal si está usando un servidor Asterisk o PBX.

:* Es importante tener en cuenta que si configura su nombre de identificación de llamadas (CNAM) en su cliente SIP para que sea un número de 10 dígitos, esto anulará su número de identificación de llamadas (Caller ID). Si el nombre del identificador de llamadas es diferente a un número de 10 dígitos, se descartará y se utilizará el número de identificador de llamadas.

:* Si es un usuario de asterisk o PBX, asegúrese de usar la última versión de '''Asterisk (v12 o superior)''' y use '''chan_pjsip''' para el troncal. Esta es una limitación de Asterisk.

:* La versión de escritorio de Zoiper requiere que el usuario tenga activada la '''versión PRO''' para poder enviar mensajes SMS y habilitar la '''SIP Presence'''. Comuníquese con soporte técnico para solicitar que se habilite la presencia SIP en su cuenta para el POP de VoIP.ms que está utilizando. Aunque normalmente no es necesario, puede haber otros casos no documentados de softphones o aplicaciones que requieran que se habilite SIP Presence. Póngase en contacto con soporte técnico si tiene algún problema con su software específico.

:* La '''versión de escritorio gratuita''' de Zoiper no puede enviar mensajes SMS y solo es capaz de recibir SMS.

:* La '''versión móvil''' de Zoiper está completamente libre de estas limitaciones. La versión gratuita ya funciona para enviar y recibir sin la necesidad de "SIP Presence".

:* Al enviar un mensaje SMS, el número del destinatario debe tener 10 dígitos. Evite usar el prefijo '"1"' ya que no es compatible en este momento. El sistema reconocerá los números como en 10 dígitos.

:* El límite de caracteres actual para '''un solo SMS es 160'''. Si un mensaje tiene más de 160 caracteres, se dividirá en bloques de hasta 160 caracteres cada uno y se enviará como mensajes independientes. La longitud total máxima de mensaje para SMS a través de SIP es de 2000 caracteres.

== Recepción ==
<div style="margin-left:50px;
padding:5px;
border-left: 6px solid rgb(208, 56, 45, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(51, 51, 51, 0.08);
width:70vw;">

'''NOTES''' 💬: ➜ EL ENVÍO DE MMS UTILIZANDO LA CUENTA SIP NO ES POSIBLE EN ESTE MOMENTO.'''
</div>

:* El DID debe tener habilitada la opción de servicio '''"Message Service (SMS/MMS)"''' Esta opción se puede habilitar desde la página '''Manage DIDs > Edit DID page'''.
:* Habilite la opción '''"SMS SIP Account"''' y seleccione el destino de la cuenta para sus mensajes SMS a este número.
:* La cuenta que recibe SMS debe estar registrada con éxito en cualquiera de nuestros POP.
:* Para garantizar los mejores resultados al recibir los mensajes SMS, no registre la misma cuenta SIP en más de un softphone o aplicación al mismo tiempo, esto puede llevar a resultados inesperados.
:* Para los dispositivos SIP que requieren esta información, como una troncal o PBX, codificamos el número de destino original en un campo de encabezado SIP personalizado llamado "X-SMS-To".
:* El softphone 3CX actualmente no admite ningún servicio de SMS de terceros. Solo funcionará si tiene una conexión directa al sistema PBX 3CX. '''Comuníquese con 3CX para obtener más detalles.'''.

== Enviar SIP / SMS con FreePBX ==

<div style="margin-left:50px;
padding:5px;
border-left: 6px solid rgb(208, 56, 45, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(51, 51, 51, 0.08);
width:70vw;">

'''NOTES''' 💬: ➜ AEL ENVÍO DE MMS UTILIZANDO LA CUENTA SIP NO ES POSIBLE EN ESTE MOMENTO.'''
</div>

Es posible configurar un sistema FreePBX para enviar SMS, consulte aquí para obtener instrucciones detalladas.

= SMS/MMS a través de la Interfaz para Revendedores (Reseller Interface) =

La función está disponible para su cliente a través de la interfaz para revendedores. Debe habilitar esta función en su paquete para que sus clientes puedan aprovecharla.

Vaya bajo la barra de navegación en '''[Reseller]''' y luego haga clic en '''[Manage Rates & Packages]'''
: [[File:CallForwarding_Reseller_1.png|thumb|none|300px]]

Haga clic en el botón Edit para editar su paquete, o haga clic en '''[Create a new package]''' para crear uno nuevo.

: [[File:CallForwarding_Reseller_2.png|thumb|none|700px]]

Diríjase a la pestaña '''[Reseller System Configuration]''' y en la sección "Type of configuration" seleccione: '''[Package Configuration]''',

: [[File:CallForwarding_Reseller_3.png|thumb|none|700px]]

A continuación, desplácese hacia abajo y busque la función "'''SMS/MMS'''", y actívela.

: [[File:SMSMMSreseller4.png|thumb|none|500px]]

= Activación/desactivación de SMS/MMS en un DID a través de la interfaz del cliente =
Para que su cliente pueda activar o desactivar los SMS/MMS desde su portal, su cliente deberá dirigirse a Services, My Services.
: [[File:SMSMMSreseller5.png|thumb|none|500px]]

Después, diríjase a la pestaña '''"DID Lists"''' y edit DID.

: [[File:SMSMMSreseller6.png|thumb|none|500px]]
A continuación, podrá habilitar los SMS/MMS, recibirlos por correo electrónico y mucho más.

: [[File:SMSMMSreseller7.png|thumb|none|500px]]

= Acceder a sus SMS/MMS a través de la interfaz del cliente =
1) Para acceder a sus mensajes SMS/MMS, diríjase a Services y luego a SMS/MMS.
: [[File:SMSMMSreseller8.png|thumb|none|500px]]

[[category:guides]]

Sequence

2026-07-30T20:05:10Z

Neocypher:

{| class="wikitable"
|+
|-
! Article en Français !! Artículo en Español
|-
| [https://wiki.voip.ms/article/Sequence_Fr Français] ||
[https://wiki.voip.ms/article/Secuencia Español]
|}

__TOC__

Sequences is an IVR enhancement feature that allows you to create very specific scenarios where you can make use of any combination of the following options:

:*DTMF
:*Recordings
:*SMS

A single sequence can contain any amount of the previous options in any pre-defined order to be executed one step after another.

A call sequence is often and mainly used (but not limited) to operate an automatic door’s system where the user will call into a DID number routing to an IVR, where it will dial the option that will trigger the sequence. This sequence will contain specific DTMF tones that the system will send to the door for it to be opened.<br>

'''Important''': A sequence can only be used as an IVR option, it is '''not''' possible to route a DID directly to a sequence's entry.

'''Find the Sequence feature within the customer portal at the DID Numbers menu.'''<br>

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

== Creating a Sequence entry ==

To create a Sequence entry you simply need to go from your customer portal to: DID numbers >> [https://www.voip.ms/m/sequence.php Sequences]

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

Once there, click " '''''+ Add Sequence''''' ", a popup window will appear with the following options:

*Name: Set a name for your call Sequence's entry
*"Add Step": Click to add a Step, you can add as many as you need.
*Action: You can move up, down or delete a step

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

=== Adding a DTMF step ===
This option will play the specified DTMF digits to the caller.

When you add a step and select '''DTMF''' at "Parameter(s)" you can set a specific number that will be sent as DTMF (tones), for example, if you set "1234" the system will send signals as if you were dialling 1,2,3,4.

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

=== Adding a Recording step ===

This will play any specific recording you have previously uploaded to at your [[Recordings]] section in your account

Once you select "Recording" as the step's type, the "Parameter(s)" field will display a drop-down list with the recordings you currently have in your account.

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

=== Adding a SMS step ===

This will instruct the system to send an SMS to a specific number. You will need to have a DID with SMS enabled and to set a local 10 digit number as the receiver (local Canada or USA number) All the [[SMS | SMS limitations]] also apply for this.

In order to use this option as a step, you will need to select an SMS compatible DID from the list '''from''', you need to set the receiving number in the '''TO''' and the message in the '''message''' text field

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

== Setting it in an IVR ==

Once you have saved the entry, you just need to add it as an option for your [https://www.voip.ms/m/ivr.php IVR]

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

If you need further information about how to create or edit an IVR, please check our wiki entry available [[ DigitalReceptionist IVR | Here]]

== Sequence using the Reseller Interface ==

'''Note that Sequence feature for the reseller portal does not contain the possibility to send SMS messages at this time.'''

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:cMOH_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:cMOH_Reseller_2.png|thumb|none|700px]]

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

: [[File:cMOH_Reseller_3.png|thumb|none|700px]]

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

: [[File:cMOH_Sequence_4.png|thumb|none|500px]]

1) To add a new Sequence for your client, or to help your client adding one. Go under the [Services] at the left navigation bar, then on [Sequence]. Once on the page, click on [Add sequence]

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

2) You will need to enter a description, and click on [Add Step] button. You can add many step as wished. You will have the ability to choose between "Recording" in order to play a recording to your caller, or sending a DTMF sounds (Per eg: like to open a door).

If you don't see any recording available or your recording is not present, you can upload it on the "Service > Recording" menu, and add a new recording file. Or simply use the option "No recording" or the Default by the system.

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

3) Click [Save]

Your Sequence feature has been created successfully. It will be available to be selected in the an IVR.

[[category:Guides]]

Mensajes de texto

2026-07-30T20:03:02Z

Neocypher: /* Artículos de blog */

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

Servicio de mensajes cortos (SMS). Esta función le permitirá enviar y recibir mensajes con su número DID (solo números DID de USA y Canadá).
Tenga en cuenta que esta función es para uso habitual de los clientes.

'''No se permitirá la automatización, telemarketing, envío o recepción masiva.'''

Nuestro servicio de SMS ya no está en BETA. A partir del 1 de agosto de 2020, los mensajes SMS se cobrarán a una tarifa de $ 0.0075 por mensaje.

==Artículos de blog==
[https://voip.ms/blog/benefits-of-sms-mms-campaigns-vs-email-marketing/ Benefits of SMS/MMS Campaigns vs. Email Marketing]

==Información importante a conocer sobre el servicio SMS==

*El servicio SMS solo está disponible para números DID locales de E.U.A. y Canadá marcados con el ícono SMS distintivo.
*Los mensajes SMS SIP solo se entregarán y aceptarán desde troncales SIP mediante autenticación de nombre de usuario / contraseña.
*Los mensajes SMS SIP utilizan el método MESSAGE SIP. Consulte RFC 3428 para obtener más detalles.
*El servicio SMS solo se puede utilizar para enviar mensajes SMS a números canadienses y estadounidenses de 10 dígitos en este momento. Tampoco podemos garantizar que los SMS internacionales se recibirán correctamente.
*VoIP.ms se reserva el derecho, a su sola discreción, de limitar o deshabilitar el servicio de SMS de cualquier cuenta que pueda presentar patrones de marketing / patrones de automatización / envío o recepción masiva.
*En este momento, no podemos garantizar que los mensajes SMS de código corto, que generalmente tienen 6 dígitos o menos (por ejemplo, Skype, códigos bancarios, comerciales de televisión, etc.) funcionarán.
*No podemos garantizar que los acentos o caracteres especiales, incluidas las letras no latinas, se entreguen correctamente.
*Para los números porteados o transferidos compatibles con la función SMS estará disponible hasta 48 horas después de que el proceso de portabilidad o transferencia se marque como Completado.

Si tiene más preguntas, no dude en ponerse en contacto con el personal de soporte en el chat en vivo o el sistema de tickets.

==Identificación de un número DID de SMS==

Lo primero que debe saber es que esta función solo está disponible para números locales de USA y Canadá en este momento. Notarás que algunos números tienen un pequeño ícono de un dispositivo de teléfono celular, esto indica que el número admite SMS. Tenga en cuenta que no todas las ubicaciones admiten SMS en este momento.

Puede comenzar a solicitar un número DID desde su portal >>DID Numbers >> Order DID >>Local Numbers >> Números (de E.U.A. o Canadá) y podrá comprar el número deseado, solo asegúrese de que admita la función SMS.

[[File:SMS 1.png|thumb|none|600px]]

== Números de lada sin costo de Canadá ==

Los números de lada sin costo de Canadá son un caso especial en el que, aunque los mesajes de texto están disponibles, se debe solicitar y habilitar manualmente. Identifique si el número de lada sin costo soporta mensajes por el icono gris de teléfono celular.

[[File:Tfsms.png]]

Al comprar un número de lada sin costo de Canadá, se debe enviar una solicitud directamente al operador. Comuníquese con el servicio de soporte técnico a través del chat en vivo o por correo electrónico a la direccón '' 'support@voip.ms' '', y solicite habilitar los mensajes en el número de lada sin costo. El proceso de activación puede demorar hasta '''3 días hábiles' '', el departamento de soporte técnico le confirmará una vez que al número se le hayan habilitado los mensajes.

==Configurar el servicio SMS==

Una vez comprado, el Servicio SMS debe activarse en DID settings, desde su portal >> DID Numbers >> Manage DID >> Edit Selection >> SMS.

En esa sección puede activar y configurar algunas opciones de reenvío para el servicio de SMS.

[[File:SMSSIPAccount.png]]

La primera acción a realizar es marcar el '''servicio de Mensajes Cortos (SMS)''', al habilitar este campo el Servicio de SMS estará activo y tendrá un cargo de $ 0.0075 por SMS.

*'''VoIP.ms SMS Portal:''' Esto está habilitado de forma predeterminada y no se puede modificar, lo que significa que podrá crear y enviar mensajes desde el Centro de mensajes SMS.
*'''SMS Email Address:''' Al activar esta opción todos los Mensajes SMS serán enviados a la dirección de correo electrónico que configure en este campo. La ventaja de esta opción es que recibirá sus mensajes SMS directamente en su correo electrónico y también podrá responder a estos mensajes desde su correo electrónico. Solo tiene que hacer clic en reply to the email. Tenga en cuenta que las respuestas por correo electrónico también están limitadas a 160 caracteres y si exceden este límite, se dividirán en dos (o más mensajes si corresponde) cuando se envíen.

Para responder el mensaje SMS a través de su correo electrónico, es importante hacer clic en "Reply" y NO cambiar ni modificar el correo de destino "TO" (sms@voip.ms) y el asunto E.G [#USXXX] Mensaje enviado al 5555555555

*'''SMS Forward:''' Puede hacer que sus Mensajes SMS se reenvíen a su teléfono celular o cualquier otro número que admita SMS, con esta opción recibirá los Mensajes SMS de nuestro sistema al número de teléfono configurado.

El Identificador de llamadas enviado al teléfono celular de llamada transferida será el número DID que recibió el mensaje SMS. No es posible reenviar el identificador de llamadas original en este momento.

*'''SMS SIP Account:''' Puede hacer que sus mensajes SMS se reenvíen a su cuenta SIP seleccionada. Se enviarán a través de nuestra puerta de enlace SMS / SIP como un MENSAJE SIP.

El destino (campo de encabezado "To") del MENSAJE SIP será el nombre de su cuenta, en lugar del número de destino del SMS original.

*'''SMS URL Callback:''' Al habilitar esta opción, podrá enviar los mensajes a otro destino, p. Ej. su propio servidor. Los mensajes SMS seleccionados recibidos por su DID enviarán una solicitud GET a la URL de devolución de llamada (Callback) proporcionada. Variables disponibles para su URL:

{ID} El ID del mensaje SMS.
{TIMESTAMP} La fecha y hora en que se recibió el mensaje.
{FROM} El número de teléfono que le envió el mensaje.
{TO} El número DID que recibió el mensaje
{MESSAGE} El contenido del mensaje
Example: http: //mysite.com/sms.php?to={TO}&from={FROM}&message={MESSAGE}&id={ID}&date={TIMESTAMP}

*'''URL Callback Retry:''' Cuando se selecciona, esperaremos un resultado "ok" (sin comillas) de su página de devolución de llamada URL como un indicador de que ha recibido el mensaje correctamente. Si no recibimos el "ok" (sin comillas) de su página de devolución de llamada (callback), seguiremos enviándole el mismo mensaje cada 30 minutos.

Una vez que configure la opción deseada, haga clic en "Apply Changes".

==Enviar y recibir mensajes (interfaz web)==

===Centro de mensajes SMS===

Para comenzar a utilizar el servicio, deberá utilizar el Centro de mensajes SMS, desde su portal >> DID Numbers>> SMS Message Center.

Para dispositivos móviles, utilice sms.voip.ms como una solución completa todo en uno.

Desde el Centro de mensajes SMS puede consultar su historial de SMS, enviar y recibir mensajes SMS.

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

*'''Search Range:''' Seleccione un intervalo de fechas de su "SMS History". Puede seleccionar un período de tiempo marcando el campo "Show Details" y se mostrarán los mensajes.
*'''Search Filter:''' Puede filtrar su búsqueda por DID, Contact y Type (enviado, recibido y ambos).
*'''Send New SMS Messages:''' Se abrirá otra ventana haciendo clic aquí para crear un nuevo mensaje SMS.
*'''Delete Selected Messages:''' Esta acción eliminará cualquier mensaje que haya seleccionado, haciendo clic en el cuadro junto a ellos. Esta acción no se puede deshacer.
*'''Delete All Messages:''' Elimina TODOS sus mensajes SMS, tanto enviados como recibidos. Esta acción no se puede deshacer.
*'''Checking a Message:''' Para verificar un mensaje SMS recibido, simplemente haga clic en él directamente para mostrar la información.

===Crear un nuevo mensaje SMS===

[[File:SMS41.png|thumb|none|600px]]

Puedes crear un nuevo Mensaje SMS le explicaremos todos los parámetros que puedes encontrar en este apartado:
*'''DID:''' Si tiene más de un número DID, puede seleccionar cualquiera de los números disponibles para enviar su mensaje.
*'''Contact:''' El número de teléfono de destino o si tiene una entrada en la agenda de teléfonos, comience a escribir el nombre y el sistema mostrará la información si está disponible.
*'''Message:''' Puede crear su mensaje en ese campo, tenga en cuenta que tiene hasta 160 caracteres.

Luego haga clic en 'Send Message' para enviar el SMS.

==Enviar y recibir mensajes (protocolo SIP)==

Los SMS a través del servicio SIP solo están disponibles mediante el protocolo SIP y la autenticación de usuario / contraseña. IP Auth e IAX2 no están disponibles en este momento.
'''Consulte nuestra guía SIP / SMS con FreePBX para obtener instrucciones sobre cómo configurar el servicio con FreePBX'''

===Enviando===

*DID debe tener habilitada la opción de servicio "Servicio de mensajes cortos (SMS)". Esta opción se puede habilitar desde la página Manage DID's > Edit DID.
*Para enviar un SMS desde su cuenta SIP, es necesario que configure su número de identificación de llamadas en uno de sus DID habilitados para SMS. Este será el número desde el que enviará el mensaje. Puede configurar el número de identificación de llamadas desde su portal de clientes para la subcuenta específica si está usando un softphone o directamente desde su extensión o troncal si está usando un servidor Asterisk o PBX.
*Es importante tener en cuenta que si configura su nombre de identificación de llamadas en su cliente SIP para que sea un número de 10 dígitos,esto anulará su número de identificación de llamadas. Si el nombre del identificador de llamadas es diferente a un número de 10 dígitos, se descartará y se utilizará el número de identificador de llamadas.
*Si es un usuario de asterisk o PBX, asegúrese de usar la última versión de Asterisk (v12 o superior) y use chan_pjsip para el tronco. Esta es una limitación de asterisco.
*La versión de escritorio de Zoiper requiere que el usuario tenga activada la versión PRO para poder enviar mensajes SMS y habilitar la presencia SIP. Comuníquese con el Soporte técnico para solicitar que se habilite la presencia SIP en su cuenta para el POP de VoIP.ms que está utilizando. Aunque normalmente no es necesario, puede haber otros casos no documentados de softphones o aplicaciones que requieran que se habilite SIP Presence. Póngase en contacto con el soporte técnico si tiene algún problema con su software específico.
*La versión de escritorio gratuita de Zoiper no puede enviar mensajes SMS y solo es capaz de recibir SMS.
*La versión móvil de Zoiper está completamente libre de estas limitaciones. La versión gratuita ya funciona para enviar y recibir sin la necesidad de presencia SIP.
*Al enviar un mensaje SMS, el número del destinatario debe tener 10 dígitos. Evite usar el prefijo '"1"' ya que no es compatible en este momento.El sistema reconocerá los números como en 10 dígitos.
*El límite de caracteres actual para un solo SMS es 160. Si un mensaje tiene más de 160 caracteres, se dividirá en bloques de hasta 160 caracteres cada uno y se enviará como mensajes independientes. La longitud total máxima de mensaje para SMS a través de SIP es de 2000 caracteres.

===Recepción===
*El DID debe tener habilitada la opción de servicio "Servicio de mensajes cortos (SMS)". Esta opción se puede habilitar desde la página Manage DID's> Edit DID.
*Habilite la opción "SMS SIP Account" y seleccione el destino de la cuenta para sus mensajes SMS a este número.
*La cuenta que recibe SMS debe estar registrada con éxito en cualquiera de nuestros POP.
*Para garantizar los mejores resultados al recibir los mensajes SMS, no registre la misma cuenta SIP en más de un softphone o aplicación al mismo tiempo, esto puede llevar a resultados inesperados.
*Para los dispositivos SIP que requieren esta información, como una troncal o PBX, codificamos el número de destino original en un campo de encabezado SIP personalizado llamado "X-SMS-To".
*El softphone 3CX actualmente no admite ningún servicio de SMS de terceros. Solo funcionará si tiene una conexión directa al sistema PBX 3CX.Comuníquese con 3CX para obtener más detalles.

===Enviar SIP / SMS con FreePBX===

Es posible configurar un sistema FreePBX para enviar SMS, consulte [[SIP/SMS with FreePBX | aquí]] para obtener instrucciones detalladas.

= SMS/MMS usando la interfaz de revendedor =

La función está disponible para su cliente a través de la interfaz de revendedor. Debe habilitar esta función en su paquete para que sus clientes puedan aprovecharla.
Vaya debajo de la barra de navegación en '''[Reseller]''' luego haga clic en '''[Manage Rates & Packages]'''
: [[File:CallForwarding_Reseller_1.png|thumb|none|300px]]

Haga clic en el botón Editar para editar su paquete, o haga clic en '' '[Crear un nuevo paquete]' '' para crear uno nuevo.

: [[File:CallForwarding_Reseller_2.png|thumb|none|700px]]

Pasa por debajo del '''[Reseller System Configuration]''' y en la sección "Type of configuration" Seleccione: '''[Package Configuration]''',

: [[File:CallForwarding_Reseller_3.png|thumb|none|700px]]

Luego, desplácese hacia abajo y busque la función "SMS / MMS" "", y habilítela.

: [[File:SMSMMSreseller4.png|thumb|none|500px]]

=== Habilitar / deshabilitar SMS / MMS en su DID ===

Para que su cliente habilite o deshabilite SMS / MMS desde su portal, su cliente deberá dirigirse a Servicios, Mis servicios.
: [[File:SMSMMSreseller5.png|thumb|none|500px]]

Luego proceda dirigiéndose a ‘’’DID Lists’’’ y edite el DID.

: [[File:SMSMMSreseller6.png|thumb|none|500px]]

Luego podrá habilitar SMS / MMS, recibir por correo electrónico y másé

: [[File:SMSMMSreseller7.png|thumb|none|500px]]

=== Sus clientes acceden a sus SMS / MMS a través de su portal ===
1) Para obtener acceso a sus mensajes SMS / MMS, diríjase a Servicios y luego a SMS / MMS. : [[File:SMSMMSreseller8.png|thumb|none|500px]]

[[category:guides]]

TOTP Authentication

2026-07-30T20:02:09Z

Neocypher:

{| class="wikitable"
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! Article en Français !! Artículo en Español
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| [https://wiki.voip.ms/article/Authentification_TOTP Français] ||
[https://wiki.voip.ms/article/Autentificaci%C3%B3n_TOTP Español]
|}

Two-factor authentication is a method for securing the account, there are several methods available however in VoIP.ms we offer 2 options.

:* 2 step authentication using a one time code sent via email (OTP) You can find the guide for this feature in our wiki page [[Two-step Verification | Two-Step Verification]]
:* Authentication based on a time-based one-time password (TOTP) using an authenticator app like FreeOTP, Google Authenticator, Microsoft Authenticator, Authy, etc.

Using TOTP as the authentication method allows an extra layer of security on your VoIP.ms account which requires a device with access to such authenticator app as those mentioned above.

Enabling TOTP will disable 2 step Authentication (OTP) via email as both are mutually exclusive.

__TOC__

== Activate This Option for your Account ==

1) To active this feature go to your Customer portal home and click on “'''Main Menu'''” > “'''<font color="#FF7C21">Account Settings</font>'''”

:[[File:TLS-SRTP-Steps0001.png|border|thumb|none|600px]]

2) Once you are in the "Account settings" section, navigate to the Security tab and scroll down to find the Account security options "2FA/TOTP Authentication" and click on the "Enable" button.

:[[File:TOTP_enable.PNG|border|thumb|none|600px]]

3) A confirmation window will pop up, click on "Yes, get my QR code" button.

:[[File:TOTP_OK.PNG|border|thumb|none|600px]]

4) A QR code will show up in the new window, you can download this image, scan it with your authenticator app (Google Authenticator in the image below) or input the text string below the image. Either will work.

:[[File:TOTP_code.PNG|border|thumb|none|600px]]

== Pairing your authenticator app ==

1) Open your preferred authenticator app, for the purpose of this wiki, will be using Google authenticator. Click on the begin button if you haven't used the app before or click on add an account button otherwise.

:[[File:TOTP_GAUTH.png|border|thumb|none|400px]]

2) Use the scanner or input manually the code using your keyboard. After added you'll see a timed six-digit code on the screen

:[[File:TOTP_GAUTH2.png|border|thumb|none|400px]]

3) You can now input in the VoIP.ms website the code in the TOTP window below were the QR code is displayed then click on the "Validate" button to complete the validation.

The code in the screen is timed and will only work while the timer is shown after it's done the code will change and you'll need to input the new code instead.

:[[File:TOTP_code.PNG|border|thumb|none|600px]]

4) After the last step is done and if everything is correct a warning will show up letting you know the code will appear only once more in your account which you can download again. Click either download or accept to finish.

:[[File:TOTP_warning.PNG|border|thumb|none|600px]]

Numéros sans frais

2026-07-30T20:01:24Z

Neocypher:

{| class="wikitable"
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|-
! Article in English !! Artículo en Español
|-
| [https://wiki.voip.ms/article/Toll-free_number English] ||
[https://wiki.voip.ms/article/Num%C3%A9ros_sans_frais Español]
|}

Un numéro sans frais ou un numéro de téléphone gratuit permet à d'autres de vous joindre à partir d'un pays (ou de l'Amérique du Nord) sans encourir de péages interurbains. Comme les frais d'interurbain sont payés par le destinataire, ces DIDs sont par minute et entraînent un coût légèrement plus élevé que les numéros locaux entrants. Voir [[Commander un numéro DID#Commander un numéro sans frais | Commande d'un numéro sans frais]] et [[Transfert d'un numéro sans frais]] pour plus de détails sur l'obtention d'un numéro sans frais.

== Amérique du nord ==

Depuis 2017, les numéros sans frais nord-américains commencent par + 1-800, 1-833, 1-844, 1-855, 1-866, 1-877 et 1-888.Dans l'indicatif régional sans frais d'origine nord-américain, 1-800, les nouveaux numéros sans frais qui épellent des mots de téléphone significatifs ont longtemps été difficiles à trouver en raison de l'utilisation abusive du système. Un rapport de 2011 de l'Associated Press indique qu'une entreprise à elle seule accumule [http://www.smh.com.au/technology/ap-impact-porn-company-is-amassing-1800-numbers-20110419-1dnlb.html un quart de l'indicatif régional 1-800], mais rien n'a été fait, en grande partie en raison du refus de la FCC d'appliquer les réglementations existantes contre la thésaurisation, l'entreposage et le courtage. Dans la plupart des cas, la base de données nord-américaine «Service Management System (SMS / 800)» derrière https://www.voip.ms/m/customtollfree.php n'aura pas de numéros 1-800 ou 1-888 disponibles (bien que quelques 1-855 et d'autres codes plus récents restent disponibles).

Cela laisse quelques options:
*Un 1-800 existant peut être transféré depuis un autre opérateur.

*Un numéro (y compris un numéro personnalisé) peut être obtenu plus facilement dans un nouvel indicatif régional, tel que 1-833 ou 1-844.

*Si 1-855- (votre nom) est parti, essayez quelques autres alternatives - 1-855- (nom de votre produit), 1-855- (les sept derniers chiffres de votre numéro local), tout ce qui semble vaguement mémorable. Essayez les 1-844 et 1-833. En cas d'échec, achetez un numéro DID sans frais ordinaire qui n'épelle rien car ils sont disponibles pour une activation immédiate à partir de l'inventaire.

*Comme il n'y a plus de "nouveaux" vrais numéros 1-800, quelques entreprises agissent en tant que RespOrgs (soi-disant, "organisation responsable" - dans les numéros sans frais, un rôle similaire à un registraire pour les noms Internet) spécifiquement pour "drop-catch" déconnecté + 1-800 numéros dès qu'ils sont disponibles. Il est possible d'obtenir un numéro d'un RespOrg indépendant (comme tollfreenumbers.com) et de le porter dans voip.ms; ces services ne sont pas des transporteurs et ne fournissent pas le service continu des transporteurs intercirconscriptions. Cela peut être une option coûteuse et longue, car le RespOrg drop-catch facture des centaines de dollars (s'ils obtiennent le numéro), le numéro doit devenir disponible et le numéro doit être porté une fois qu'il fonctionne.

*Les nouveaux indicatifs régionaux sans frais ont tendance à apparaître tous les quelques années; le plus récent était le + 1-833 qui a ouvert le 3 juin 2017. Il est conseillé de demander des numéros dès qu'un nouvel indicatif régional est disponible, car le processus se transforme rapidement en une ruée vers les meilleurs nombres. Les termes génériques les plus précieux tels que FLEURS ou VACANCES ont tendance à disparaître dans les premières secondes.

*Si vous pensez que vous devez absolument engager un avocat américain pour contacter la Commission fédérale des communications (Federal Communications Commission) et exiger que certains spammeurs sexuels par téléphone cessent de stocker "votre" nom de société 1-800 (avec un million d'autres),cela pourrait être une bonne idée pour saisir et publier un simple + 1-844 ou 1-855 entre-temps. Le processus est lent et le résultat n'est pas garanti. Dans certains cas (comme + 1-800-HOLIDAY vs + 1-800-H0LIDAY, avec un zéro à la place du «O»), un combat impliquant un numéro distinctif a été traîné en vain jusqu'aux cours d'appel.En général, une fois que vous avez votre DID sans frais, il fonctionne à peu près de la même manière que tout autre numéro entrant.
Quelques considérations pratiques:

*Si vous vendez principalement dans une grande ville (et sa banlieue), l'obtention d'un numéro local dans cette ville peut être une alternative viable à un numéro sans frais. Il n'a pas de coût supplémentaire, il semble local et tout appel entrant avec un numéro erroné proviendra généralement d'une seule zone locale. À l'inverse, un numéro national sans frais peut être un moyen d'éviter d'avoir besoin de plusieurs numéros locaux de plusieurs villes et est facilement personnalisable.

*Il peut être préférable de laisser un numéro local sur votre identification de l'appelant sortant. Il n'y a aucune restriction technique sur l'affichage du + 1-855-555-5555 ou similaire, mais trop de ces appels sont actuellement des télévendeurs, des collecteurs de factures à la volée ou d'autres demandes de renseignements indésirables. De nombreux appelés hésiteront donc à répondre à un appel provenant d'un numéro non géographique inconnu.

*Les appels provenant de téléphones publics américains seraient notoirement coûteux, lorsqu'ils sont activés, car le propriétaire du téléphone public est payé généreusement par appel (généralement soixante cents en plus du coût par minute de l'appel), mais achemine les appels des téléphones publics vers la VoIP. Le fonctionnement des numéros sans frais ms n'est pas garanti.

*Les appels en provenance d'Alaska, d'Hawaï, des possessions américaines et des trois territoires canadiens (+ 1-867) peuvent également être exclus en raison de leur coût premium; https://www.voip.ms/m/ordertollfree.php répertorie l'Alaska comme "déverrouillé sur demande"à un coût par minute plus élevé.

*Les numéros sans frais des Caraïbes couvrent normalement un seul pays, malgré l'utilisation du format de numéro de code régional + 1 apparemment familier en Amérique du Nord.

*Les appels vers des numéros sans frais d'outre-mer, à de rares exceptions près, échoueront. "Le numéro que vous avez composé n'est pas disponible dans votre zone d'appel." Dans les rares cas où ils terminent, ils ne sont pas gratuits. La plupart des annonceurs indiqueront un numéro local et un numéro sans frais pour éviter de devenir inaccessible de cette manière.

*Les faux numéros entrants sont un problème potentiel, surtout si votre numéro est à un chiffre d'une énorme franchise nationale. Il peut être préférable d'attendre quelques jours après l'activation et le fonctionnement de votre numéro avant de l'imprimer sur des publicités et des cartes de visite; si le téléphone sonne décroché avec des appels erronés, il n'est pas trop tard pour demander un autre numéro. En règle générale, ce problème est le pire en vrai + 1-800 et moins un problème pour les abonnés avec des indicatifs régionaux plus récents et moins utilisés tels que 1-833, 1-844 ou 1-855.

*Sachez quelle zone d'appel vous souhaitez avant de commander; un numéro sans frais pour couvrir tous les États-Unis et le Canada est possible, mais peut ajouter des coûts car il augmente le nombre de faux numéros entrants. Bien que https://www.voip.ms/m/callerid_filtering.php soit une option pour limiter les indicatifs régionaux pouvant appeler,il peut être judicieux de décider si les appels d'un autre pays sont souhaités lors du choix initial d'un opérateur.

*Si vous vous trompez de numéros et que vous ne souhaitez pas modifier votre numéro sans frais ou votre zone de couverture, l'envoi d'appels à la réponse vocale interactive donnera aux appelants qui entendent l'annonce du nom de votre entreprise la possibilité de raccrocher en silence avant que les appels ne soient envoyés à une personne vivante.

== International ==
Un code de pays +800 existe nominalement pour les appels internationaux mais est rarement utilisé. En règle générale, les numéros sans frais sont obtenus séparément pour chaque pays dans lequel ils doivent être utilisés - de sorte qu'un appel gratuit australien ne couvre que l'Australie.
Voir [[Commander un numéro DID#Commander un numéro DID international | Commande d'un numéro DID international]]'''

[[Category:Configuration]]

Toll-free number

2026-07-30T20:00:41Z

Neocypher:

{| class="wikitable"
|+
|-
! Article en Français !! Artículo en Español
|-
| [https://wiki.voip.ms/article/Num%C3%A9ros_sans_frais Français] ||
[https://wiki.voip.ms/article/N%C3%BAmeros_de_Lada_sin_costo Español]
|}

A '''toll-free number''' or '''freephone number''' allows others to reach you from within one country (or within North America) without incurring long-distance tolls. As the long-distance charges are paid by the recipient, these DIDs are per-minute and carry a slightly higher cost than local inbound numbers. See '''[[Order a DID Number#Ordering a Toll Free Number|Ordering a Toll Free Number]]''' and '''[[Porting a Toll Free Number]]''' for details on obtaining a toll-free number.

__TOC__

== North America ==
As of 2017, North American toll-free numbers begin with +1-800, 1-833, 1-844, 1-855, 1-866, 1-877 and 1-888. In the original North American toll-free area code, 1-800, new toll-free numbers which spell meaningful phonewords have long been hard to come by due to widespread abuse of the system. A 2011 Associated Press report indicates one company alone is hoarding [http://www.smh.com.au/technology/ap-impact-porn-company-is-amassing-1800-numbers-20110419-1dnlb.html a quarter of the entire 1-800 area code] but nothing has been done, largely due to FCC unwillingness to enforce existing regulations against hoarding, warehousing and brokering. At most times, the North American "Service Management System (SMS/800)" database behind https://www.voip.ms/m/customtollfree.php will have no available 1-800 or 1-888 numbers (although quite a few 1-855's and other newer codes remain available).

This leaves a few options:
* An existing 1-800 can be ported in from another carrier.
* A number (including a vanity number) may be obtained more easily in a newer area code, such as 1-833 or 1-844.
* If 1-855-(yourname) is gone, try a few other alternatives - 1-855-(your product name), 1-855-(the last seven digits of your local number), anything which looks vaguely memorable. Try 1-844 and 1-833's. If these fail, buy an ordinary toll-free DID number which spells nothing as these are available for immediate activation from inventory.
* As there are no "new" true 1-800 numbers left, a few companies act as RespOrgs (supposedly, "responsible organisation" - in tollfree numbers, a role similar to a registrar for Internet names) specifically to "drop-catch" disconnected +1-800 numbers as they become available. It is possible to obtain a number from an independent RespOrg (like tollfreenumbers.com) and port it into voip.ms; these services are not carriers and don't provide the ongoing interexchange carrier service. This can be an expensive, time-consuming option as the drop-catch RespOrg charges hundreds of dollars (if they get the number), the number has to become available and the number must be ported once it's working.
* New toll-free area codes tend to appear every few years; the most recent was +1-833 which opened on June 3, 2017. Requesting numbers as soon as any new area code becomes available is advisable, as the process rapidly devolves into a landrush for the best numbers. The most valuable generic vanity terms like FLOWERS or HOLIDAY tend to be gone in the first few seconds.
* If you feel you absolutely must engage a US lawyer to contact the Federal Communications Commission and demand that some phone-sex spammer stop hoarding "your" 1-800-company name (along with a million others), it might be a good idea to grab and publish a plain +1-844 or 1-855 in the meantime. The process is slow and the outcome not guaranteed. In some cases (like +1-800-HOLIDAY vs. +1-800-H0LIDAY, with a zero in place of the 'O') a fight involving a distinctive number has been dragged in vain all the way to the appeal courts.

In general, once you have your toll-free DID, it works in much the same way as any other inbound number.

A few practical considerations:
* If you market primarily to one large city (and its suburbs), obtaining a local number in that city may be a viable alternative to a toll-free number. It has no premium cost, it looks local and any inbound wrong number calls will usually be from just the one local area. Conversely, a national toll-free number can be a means to avoid needing multiple local numbers from multiple cities and is easily brandable.
* It may be best to leave a local number on your outbound [[caller ID]]. There's no technical restriction on displaying +1-855-555-5555 or similar, but too many of these calls currently are telemarketers, fly-by-night bill collectors or other undesired enquiries. Many called parties will therefore hesitate to answer a call from any unfamiliar non-geographic number.
* Calls from US pay phones would be notoriously expensive, where enabled, as the owner of the pay phone is paid generously per call (typically sixty cents on top of the per-minute cost of the call), however originating calls from payphones to VoIP.ms toll-free numbers is not guaranteed to work.
* Calls from Alaska, Hawaii, US possessions and the three Canadian territories (+1-867) may also be excluded due to their premium cost; https://www.voip.ms/m/ordertollfree.php lists Alaska as "unlocked on request" at a higher per-minute cost.
* Caribbean toll-free numbers normally cover one country only, despite their use of the seemingly-familiar North American +1-areacode-number format.
* Calls to toll-free numbers from overseas, with rare exception, will fail. "The number you have dialled is not available from your calling area." In the rare instance where they do complete, they're not free. Most advertisers will list a local number and a toll-free number to avoid becoming unreachable in this manner.
* Inbound wrong numbers are a potential problem, especially if your number is one digit away from some huge national franchise. It may be best to wait a few days after your number is activated and working before printing it on advertisements and business cards; if the 'phone rings off the hook with misdial calls, it's not too late to request some other number. Typically, this issue is worst in true +1-800 and less of a problem for subscribers with newer, less-used area codes like 1-833, 1-844 or 1-855.
* Know what calling area you want before you order; one toll-free number to cover all of the US and Canada is possible, but may add cost as it increases the number of inbound wrong numbers. While https://www.voip.ms/m/callerid_filtering.php is one option to limit what area codes can call, it may make sense to decide whether calls from another country are desired when initially choosing a carrier.
* If you are getting wrong numbers and don't want to change your toll-free number or coverage area, sending calls to [[Digital Receptionist (IVR)|interactive voice response]] will give misdial callers who hear your company name announcement an opportunity to silently hang up before the calls are sent to a live person.

==International==
A country code +800 nominally exists for international calls but is rarely used. Typically, toll-free numbers are obtained separately for each country in which they are to be used - so an Australian toll-free covers only Australia.

See '''[[Order a DID Number#Ordering an International DID number | Ordering an International DID number]]'''
[[Category:Configuration]]

Comprar un número DID

2026-07-30T19:56:54Z

Neocypher: /* Artículo de blog */

{| class="wikitable"
|+
|-
! Article in English !! Article en Français
|-
| [https://wiki.voip.ms/article/Order_a_DID_Number English] ||
[https://wiki.voip.ms/article/Commander_un_num%C3%A9ro_DID Français]
|}

Los números DID (Direct Inward Dial) se utilizan para recibir llamadas entrantes mediante un servicio de voz sobre IP (VoIP).Estos números se pueden llamar desde cualquier teléfono regular, incluidos teléfonos fijos y celulares. Ya sea que sea una empresa local, nacional o global, puede usar sus números DID.
Los números DID solo son necesarios para recibir llamadas. Para realizar llamadas salientes, solo necesita configurar sus credenciales VoIP.ms en un dispositivo o softphone y mantener un saldo positivo en su cuenta.

== Artículo de blog ==
[https://voip.ms/blog/benefits-of-having-a-toll-free-number-for-your-business/ Beneficios de tener un número gratuito para su empresa (Ingles)]

== Comprar un DID ==

Puede comprar un número DID en su portal de clientes, DID Numbers >> Order DID(s).

[[File:Order.JPG|thumb|none|750px|Order DID]]
<br clear="all" />

Ahí puede encontrar las siguientes opciones:

* '''Local Numbers:''' Puede elegir números de USA y Canadá.
* '''International Numbers:''' Estos números pertenecen a un país y una ciudad en particular
* '''Toll Free Numbers:''' Puede elegir números de lada sin costo de E.U.A, Canadá, E.U.A + Canadá o números internacionales. A diferencia de un número local (nacional o internacional), un número de lada sin costo generalmente funciona dentro de un país o un código de país, y es lada sin costo para llamar desde un teléfono fijo.
* '''National Numbers (Non Geographical):''' Estos números pertenecen a un país en particular, pero no a una ciudad en particular, y solo se puede acceder a ellos con un teléfono fijo desde ese país.
* '''Virtual SIP Numbers:''' Solo se puede acceder a estos números a través de "SIP URI" a través de Internet.
* '''DIDs with Virtual PRI (Per Channel):''' Disponible a solicitud: envíe un correo electrónico a sales@voip.ms para solicitar una cotización.

[[File:Order2.JPG|thumb|none|750px]]

==Elija un número DID==

Una vez que haya seleccionado el tipo de número deseado, puede seleccionar el país / estado para comprar su DID (cuando corresponda) y luego seleccionar un número de la lista si están disponibles, y podrá ver el plan entrante y la tarifa de instalación antes de comprarlo.

También puede refinar una búsqueda de número DID utilizando las opciones "Contains"(Continene), "Starts with" (Empieza con) o "Ends with" (Termina con) en el campo de búsqueda de esa misma pantalla. Los números disponibles que cumplan con estos criterios deben mostrarse si los hay.

[[File:Order3.JPG|thumb|none|750px]]

==Elija un plan entrante para su DID==

Puede elegir entre 2 planes diferentes para llamadas entrantes:
*El plan al por minuto, que cobra una tarifa mensual y una tarifa fija por cada minuto que dura la llamada.
*Y el plan de tarifa fija, que está diseñado para uso residencial únicamente e incluye 3500 minutos entrantes por mes.

[[File:Order4.JPG|thumb|none|750px]]

==Finalize su compra DID==

Después de ésto, solo tiene que seleccionar el Enrutamiento principal, el Punto de presencia, las medidas de Conmutación por error y otras características opcionales para su nuevo número y hacer clic en el botón "Order DID" para finalizar este proceso.
Una vez que haya comprado con éxito un número DID, se enviará un correo electrónico al correo electrónico asociado con su cuenta VoIP.ms confirmando esta acción.

[[File:Order5.JPG|thumb|none|750px]]

==Iniciar una orden pendiente de DID==

En caso de que no haya un número disponible en stock, puede iniciar un proceso de pedido pendiente y se puede enviar un número aleatorio a su cuenta.

[[File:backorder.JPG|thumb|none|750px]]

Después de esto, solo tiene que elegir entre los diferentes planes de llamadas entrantes y seleccionar la configuración predeterminada para su nuevo número, tal como se explica en la sección anterior.

[[File:quantity.JPG|thumb|none|750px]]

==Comprar un número DID internacional==

Al solicitar un número DID internacional, podrá seleccionar entre diferentes países.

[[File:OrderINT-1.png|thumb|none|750px]]

Una vez que seleccione un país, aparecerá una nueva página que mostrará la tarifa de configuración, la tarifa mensual, la cantidad de llamadas entrantes y los canales (entrantes) para ese país específico.

En la siguiente sección se mostrarán los centros de tarifas disponibles, el código de área y la cantidad de números existentes para ese país.

Hay un campo en blanco donde puede ingresar la cantidad de números DID de ese centro de tarifas que desea comprar.

'''La entrega de números en stock tarda en promedio 1 (un) día hábil pero puede variar según la ubicación.'''

Una vez que haga clic en el botón "Order", aparecerá la siguiente pantalla.

[[File:OrderINT-3.png|thumb|none|750px]]

Esta es la página de confirmación de la compra DID. Aquí podrá preestablecer su número DID antes de que lo agreguemos a su cuenta. Podrá cambiar las siguientes opciones:
*Búsqueda de nombre de identificador de llamadas (Caller ID Name lookup).
*Punto de presencia DID (DID Point of Presence).
*Enrutamiento (Routing).
*Opciones de conmutación por error adicionales (Additional Failover Options).
Una vez que termine con el ajuste preestablecido, haga clic en el botón "Click here to confirm your order".

[[File:orderINT-2.png|thumb|none|750px]]

'''La entrega de números en stock tarda en promedio 1 (un) día hábil pero puede variar según la ubicación.'''

==Comprar un número de lada sin costo==

Ofrecemos 4 opciones al solicitar un número de lada sin costo:
*'''Estados Unidos (United States):''' Verifique la disponibilidad y compre un DID sin cargo para llamadas estadounidenses, Canadá es desbloqueable.
*'''Canadá + Estados Unidos (Canada+United States):''' Verifique la disponibilidad y compre un DID sin cargo para las personas que llaman en Canadá y Estados Unidos.
*'''Números Personalizados (Vanity Numbers):''' Verifique la disponibilidad de números de lada sin costo personalizados en la base de datos SMS / 800.
*'''Internacional (International):''' Verifique la disponibilidad y solicite números de lada sin costo internacionales.

[[File:orderTFN-1.png|thumb|none|750px]]

En la sección de Estados Unidos, lo primero que verá es el filtro para buscar números de lada sin costo. Puede filtrarlos por '''800 | 833 | 844 | 855 | 866 | 877 | 888 | Todo''', o incluso puede ingresar en '''el campo''' en blanco un número específico que desee buscar.

La sección Canadá + Estados Unidos '''no incluye este filtro'''. Todo lo demás en esta sección es exactamente igual que en la sección 'Estados Unidos'.

[[File:orderTFN-2.png|thumb|none|750px]]

Una vez que haya seleccionado el número que desee comprar, desplácese hacia abajo y podrá ver la tarifa mensual,el costo al por minuto, la tarifa de instalación y el incremento de facturación para ese número en específico.

En la misma sección, puede preestablecer su número de lada sin costo asignando una ruta, un punto de presencia y opciones adicionales de conmutación por error.

Desde esta pantalla, debe "hacer clic para confirmar su pedido" y el número se agregará a su cuenta.

[[File:orderTFN-3.png|thumb|none|750px]]

==Comprar un número personalizado==

Los números personalizados son números de lada sin costo personalizables, por ejemplo: 888-421-HELLO. Esto ayudará a las personas que llaman o al cliente a recordar fácilmente su número. Un número de lada sin costo "personalizado" normalmente tardará más en configurarse inicialmente que un número aleatorio que ya está en el inventario de voip.ms, pero puede servir como una valiosa herramienta de marketing debido a su facilidad de memoria.

[[File:orderTFN-4.png|thumb|none|750px]]

Una vez que ingresó un número válido y disponible, deberá seleccionar un operador.
*'''Proveedor Americano:''' Su número de lada sin costo funcionará para las personas que llaman de los E.U.A. por solo a una tarifa de $1.9 centavos por minuto. Esto se puede cambiar más tarde a E.U.A. /Canadá (ver más abajo). La tarifa mensual es de $0.99 a partir de la fecha de entrega. Hay una tarifa de instalación única de $15 para solicitar cada DID personalizado.Plazo de entrega aproximado: 1-5 días laborales.

*'''Proveedor Americano, llamadas desde E.U.A. y Canadá permitidas:''' Su número de lada sin costo funcionará para las personas que llaman de E.U.A. y Canadá a una tarifa de $1.9 centavos por minuto y $8 centavos por minuto para las personas que llaman de Canadá. La tarifa mensual es de $0.99 por mes a partir de la fecha de entrega. Hay una tarifa de instalación única de $15 para solicitar cada DID personalizado. Plazo de entrega aproximado: 1-5 días laborales.

*'''Proveedor Canadiense:''' Sus números de lada sin costo funcionarán para las personas que llaman de de E.U.A. y Canadá a una tarifa de $2.7 centavos por minuto para ambos países. La tarifa mensual es de $1.25 por mes a partir de la fecha de entrega. Hay una tarifa de instalación única de $30 para solicitar cada DID personalizado. Plazo de entrega aproximado: 1-3 días laborales.

Ahora solo necesita hacer clic en el botón '''"Order Now"''' para continuar.

'''Los pedidos de números personalizados pueden tardar hasta 2 horas en activarse completamente.'''

[[File:SPVanityTFNorder.png|thumb|none|750px]]

==Comprar un número de lada sin costo internacional==

Este proceso es básicamente el mismo que para solicitar un número DID internacional. '''Consulte esa [[Order_a_DID_Number#Ordering_an_International_DID_number | sección]]'''

==Comprar números nacionales (no geográficos)==

Estos números pertenecen a un país en particular pero no a una ciudad en particular; a menudo, solo se puede acceder a ellos desde teléfonos fijos en ese país.

Este proceso es básicamente el mismo que para solicitar un número DID internacional. '''Consulte esa [[Order_a_DID_Number#Ordering_an_International_DID_number | sección ]]'''

==Compra de números virtuales==

Los números SIP virtuales son similares a los números DID estándar. La principal diferencia es que los números SIP virtuales no son accesibles a través de '''"PSTN"'''. Solo se puede acceder a ellos a través de '''"SIP URI"''' a través de Internet. Por ejemplo, si tiene un número DID con otro proveedor y este admite el reenvío de '''SIP URI''', puede reenviar su número a un número virtual en voip.ms como si fuera uno de nuestros números.

Todos los números virtuales constan de los siguientes dígitos: '''11 + Código de cuenta + 3 dígitos de su elección para un total de 11 dígitos'''. El URI final será ese número seguido del signo @ en uno de nuestros servidores. Si tiene la intención de enviar las llamadas a un teléfono o adaptador, deberá apuntar al servidor adecuado. URI de SIP de ejemplo: 11139808123@houston.voip.ms

[[File:orderVN-1.png|thumb|none|750px]]

== Comprar un números DID en la interfaz de revendedor ==

La función está disponible para su cliente a través de la interfaz de revendedor. Debe habilitar esta función en su paquete para que puedan aprovecharla.

Vaya debajo de la barra de navegación en '''[Reseller]''' y luego haga clic en '''[Manage Rates and Packages]'''
: [[File:OrderDID_Reseller_1.png|thumb|none|300px]]

Haga clic en el botón Editar para editar su paquete, o haga clic en '''[Create a new package]''' para crear uno nuevo.

: [[File:OrderDID_Reseller_2.png|thumb|none|700px]]

Vaya a la pestaña [Reseller System Configuration] y, en la sección '''"Type of configuration'''", seleccione: '''[Package Configuration]''',

: [[File:OrderDID_Reseller_3.png|thumb|none|700px]]

Luego, desplácese hacia abajo y busque la función "Order DIDS" y habilítela.

: [[File:OrderDID_Reseller_4.png|thumb|none|500px]]

[[File:OrderDID_Add.png|thumb|none|300px]]

[[File:OrderDID_Add_2.png|thumb|none|500px]]

[[File:OrderDID_Add_3.png|thumb|none|500px]]

[[File:OrderDID_Add_4.png|thumb|none|500px]]

El nuevo DID solicitado estará disponible en la sección '''My Services''' en '''Services'''.

DISA

2026-07-30T19:55:54Z

Neocypher: /* Blog Article */

{| class="wikitable"
|+
|-
! Article en Français !! Artículo en Español
|-
| [https://wiki.voip.ms/article/Acc%C3%A8s_direct_en_entr%C3%A9e_au_syst%C3%A8me_-_DISA Français] ||
[https://wiki.voip.ms/article/Acceso_directo_al_sistema_interno_(DISA) Español]
|}

'''What is DISA ?'''
----
Direct Inward System Access (DISA) allows you to use our system for placing outgoing calls, event if you are far away from any device where you are registering your account or sub account. In this case you just would need to dial to your DID number and to provide a 4-digit PIN number, then you can dial out to any number in the world under our termination rates.

__TOC__

== Blog Article ==

[https://voip.ms/blog/direct-inward-system-access-disa/ Direct Inward System Access (DISA)]

== Setup DISA ==

=== How you can create a DISA entry ===
----

1) Please login on the customer portal, then go to DID numbers -> [https://www.voip.ms/m/disa.php DISA]. There you will see this:

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

2) Once you click on the button "Add DISA" you will see this:

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

- '''DISA Name''': Here you can enter the name which best describes this entry.

- '''DISA PIN''': It is a 4-digit number that you will need to provide in order to get dial tone.

- '''Digit Timeout''': The maximum amount of time( in seconds ) our system will wait between digits.

- '''CallerID Number Override''': Here you can set the number you want people to receive when you dial out from DISA.

3) Once you enter all this information, you would need to click on the button "Save" to see DISA entry in this way:

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

=== Where you can use a DISA entry ===
----
You can use a DISA entry for:

1) '''Routing your DID number.'''

2) '''Pointing one extension of a Digital Receptionist ( [[Digital Receptionist (IVR)|IVR]] ).'''

3) '''Setting as destination into a [[Time Conditions|Time Condition]], or'''

4) '''Routing a call according to what you set on the [[CallerID Filtering|CallerID Filtering]].'''

IMPORTANT NOTE : From DISA you can dial out to any number, including a Speed Dial Entry you have created in your [[Phone book|Phone
book]]. For example: *75XX and then press the pound key (#) or simply wait for the duration of the "Digit Time-out", and our system will
place the call.
After you enter your PIN Number and get the Dial Tone on a phone please just dial your number and wait for the call to initiate.
Do not hit your call button on the phone this may result in the call going over your other Provider's Network.

=== How calls will be billed using DISA? ===
----
The cost of these calls will depend on the billing plan of your DID number (per minute or flat rate), the destination of the outgoing call and the time of the call. For example:

1) If you have a DID number under per minute plan with a rate for the incoming of $0.009 (per minute) and this number is routed to DISA, when someone calls and places an outgoing call for 1 minute to a Canadian number the cost will be:

A -> $0.009 + $0.0052 = $0.0142 [value routing], or

B -> $0.009 + $0.009 = $0.018 [premium routing].

2) If you have a DID number under flat rate plan with a rate for incoming of $0.00 (per minute) and this number is routed to DISA, when someone calls and places an outgoing call for 1 minute to a Canadian number the cost will be:

C -> $0.00 + $0.0052 = $0.0052 [value routing], or

D -> $0.00 + $0.009 = $0.009 [premium routing].

= Reseller Configuration =

Also if you're using the [[Reseller Basic Guide|Reseller Interface]], you can associate each DISA feature with one of your reseller client.

'''Reseller Client''': Here you can select your reseller client that you want to associate this feature. You need first to create the account of your customer using the [[Reseller Basic Guide|Reseller section]] in your Customer Portal.

: [[File:ResellerClient_SelectClient_Only.png|500px]]

=DISA 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:DISA_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:DISA_Reseller_2.png|thumb|none|700px]]

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

: [[File:DISA_Reseller_3.png|thumb|none|700px]]

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

: [[File:DISA_Reseller_4.png|thumb|none|500px]]

1) To add a DISA entry for your client, or to help your client adding one. Go under the '''[Services]''' at the left navigation bar, then on '''[DISA]'''
: [[File:DISA_Add.png|thumb|none|400px]]

2) Click on the tab Add new DISA and enter a '''Name''' and a 4-digits '''PIN'''. You can also set the other settings if necessairy.
: [[File:DISA_Add2.png|thumb|none|600px]]

3) Complete the form and click the '''[Save DISA]''' button.

When your DISA is created, you will be able to select it as a routing option, such as DID Routing, Failover, IVR, Time Condition etc.

[[category:guides]]

Autentificación TOTP

2026-07-30T19:55:07Z

Neocypher: /* Artículos del Blog */

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

La autenticación de dos factores es un método para proteger la cuenta, hay varios métodos disponibles, sin embargo, en VoIP.ms ofrecemos 2 opciones.

:* Autenticación en 2 pasos mediante un código de un solo uso enviado por correo electrónico (OTP). Puede encontrar la guía para esta función en nuestra página wiki. [[Two-step Verification | Two-Step Verification]]
:* Autenticación basada en una contraseña de un solo uso basada en el tiempo (TOTP) mediante una aplicación de autenticación como FreeOTP, Google Authenticator, Microsoft Authenticator, Authy, etc.

El uso de TOTP como método de autenticación permite una nivel adicional de seguridad en su cuenta VoIP.ms que requiere un dispositivo con acceso a una aplicación de autenticación como las mencionadas anteriormente.

Habilitar TOTP deshabilitará la Autenticación de 2 pasos (OTP) por correo electrónico ya que ambos son mutuamente excluyentes.

__TOC__

== Artículos del Blog ==
[https://voip.ms/blog/back-to-basics-what-is-totp/ Back-to-Basics: What is a TOTP]

== Activar esta opción para su cuenta ==

1) Para activar esta función, vaya a la página de inicio de su portal de clientes y haga clic en “'''Main Menu'''” > “'''<font color="#FF7C21">Account Settings</font>'''”

:[[File:TLS-SRTP-Steps0001.png|border|thumb|none|600px]]

2) Una vez que ya este en la sección para editar la cuenta "Account settings", navegue hasta la pestaña seguridad "Security" y desplácese hacia abajo para encontrar las opciones de seguridad de la cuenta "2FA/TOTP Authentication" y haga clic en el botón activar "Enable".

:[[File:TOTP_enable.PNG|border|thumb|none|600px]]

3) Aparecerá una ventana de confirmación, haga clic en el botón "Sí, obtener mi código QR" ("Yes, get my QR code").

:[[File:TOTP_OK.PNG|border|thumb|none|600px]]

4) Aparecerá un código QR en la nueva ventana, puede descargar esta imagen, escanearla con su aplicación de autenticación (Google Authenticator en la imagen a continuación) o ingresar la cadena de texto debajo de la imagen. Cualquiera funcionará.

:[[File:TOTP_code.PNG|border|thumb|none|600px]]

== Emparejamiento de su aplicación de autenticación ==

1) Abra su aplicación de autenticación preferida, para el propósito de esta wiki, utilizará el autenticador de Google. Haga clic en el botón comenzar si no ha usado la aplicación antes o haga clic en el botón Agregar una cuenta de lo contrario.

:[[File:TOTP_GAUTH.png|border|thumb|none|400px]]

2) Use el escáner o ingrese manualmente el código usando su teclado. Después de agregar, verá un código cronometrado de seis dígitos en la pantalla.

:[[File:TOTP_GAUTH2.png|border|thumb|none|400px]]

3) Ahora puede ingresar en el sitio web VoIP.ms el código en la ventana TOTP a continuación donde se muestra el código QR y luego haga clic en el botón validar "Validate" para completar la validación.

El código en la pantalla está cronometrado y solo funcionará mientras se muestre el temporizador después de que esté hecho, el código cambiará y deberá ingresar el nuevo código en su lugar.

:[[File:TOTP_code.PNG|border|thumb|none|600px]]

4) Una vez realizado el último paso y si todo está correcto, aparecerá una advertencia que le informará que el código aparecerá solo una vez más en su cuenta, que puede descargar nuevamente. Haga clic en descargar o aceptar para finalizar.

:[[File:TOTP_warning.PNG|border|thumb|none|600px]]

Audio Conferencing

2026-07-30T19:54:01Z

Neocypher: /* Blog Articles */

__TOC__

{| class="wikitable"
|+
|-
! Article en Français !! Artículo en Español
|-
| [https://wiki.voip.ms/article/Audioconf%C3%A9rence Français] || [https://wiki.voip.ms/article/Audioconferencia Español]
|}

== Tutorial Video ==

[[Image:CallConfThumbnail.png|200px|link=https://www.youtube.com/watch?v=DcpmW0vIcng]]

== Blog Articles ==

[https://voip.ms/blog/ways-to-hold-effective-conference-call-meetings/ Ways to Hold Effective Conference Call Meetings]

[https://voip.ms/blog/benefits-of-voip-conference-call/ 3 Benefits of VoIP Conference Call]

== What is the Audio Conferencing ==

The Audio Conferencing feature allows several people to dial into your DID number to be bridged together in the same conversation.<br />
Audio Conferencing provides the host with a dashboard monitor that allows organizers to monitor and control different aspects of the conference call in real time.

Press the '''“Start Recording”''' button to get a recording of your conference call that you can listen and download after for any purposes.<br />
Start using the Audio Conferencing feature by clicking the '''“Add Conference”''' button to create an entry.

Once your conference entry is ready, you can route your DID number to it from the navigation menu '''DID Numbers > Manage DIDs > Edit''' the number you want to route, to start receiving calls into your conference.

'''⚠ Note: Conference feature is only available to DID Numbers in Per Minute plan.'''

'''Calls entering the Audio Conferencing are billed each one independently according to the regular per minute rate of the DID number
'''for the whole duration that participant remained in the conference.
'''The Call recording service has a rate of $0.0025 per minute, for the final duration of the recording.
'''The per minute charges can be consulted at the Call Detail Records.

== Add a New Conference ==

[[File:AudioConference1.png|border]]

Click on the '''"Add Conference"''' button to start creating your Audio Conferencing entry.

=== Information ===

[[File:AudioConference2.png|border]]

'''Name:''' This is the unique name of your conference. (e.g. Sales Meeting) <br />
'''Description:''' This can be used as a note or description to easily identify your conference. <br />
'''Maximum Participants:''' Specifies the maximum numberof callers that can participate in this conference. Leave empty for no limit. Once the limit is reached, the conference will be locked until a person leaves. Admin-level users are exempt from this limit and will still be able to access to join a locked conference. <br />
'''Language:''' Language for system messages, such as “Invalid Option”.

For your new conference, you need to create a member profile, click on the other TAB "Participants and conference settings"

=== Participants and conference settings ===
This is the list of all member profiles for all conferences. To add a member profile to this conference, simply check the box next to his/her name.

[[File:AudioConference3.png|border]]

==== Add Member Profile ====
For your conference, you need to link a profile to your new conferences. In this way, you will apply some settings.

What you need to know before creating a profile.

''' IMPORTANT: We highly suggest you to use a PIN # to connect the user into your conferences in order to prevent undesirable caller

''' NOTE: You can have more than one profile per conference, but ONLY 1, can have a unlocked profile ''(Profile without PIN #)''. a PIN number can only be used by ONE profile.

''' When the caller gets into your conference, if a PIN number is asked, the PIN entered will be reflected by the profile created.

''' If you have created a pinless profile and profiles with a PIN are existent, at the prompt, if the user does not enter his PIN, he will automatically join the pinless profile.

===== Basic Options =====

[[File:AudioConference4.png|border]]

'''Name:''' This is the unique name of your Member Profile.<br />
'''Description:''' This can be used as a note or description to easily identify your Member Profile.<br />
'''PIN:''' Sets if the user must enter a PIN before joining the conference. The user will be prompted for the PIN. This also serves for selecting the member profile when the user joins the conference.<br />

'''Note: The usage of the pound key after the PIN number to join the conference can be included
or excluded depending on your configuration, for instance, if all the members to the conference
are granted with a PIN code with the same amount of digits, they won't be requested to dial pound
key, however, if one of the members does have a larger PIN code, other members will require to dial
the pound key to access the conference.'''

'''Admin:''' Sets if the user is an Admin or not.<br />
'''Start Muted:''' Sets if the user should start out muted after entering the conference.<br />
'''Announce Join Leave:''' When enabled, this option prompts the user for his name while entering the conference. After the name is recorded, it will be played as the user enters and exits the conference.<br />
'''Announce User Count:''' Sets if the number of users in the conference should be announced to the caller as he joins.<br />
'''Announce Only User:''' Sets if the '''“only user”''' announcement should be played when a caller enters an empty conference.<br />

===== Advanced Options =====

[[File:AudioConference5.png|border]]

'''MoH When Empty:''' Sets whether music on hold (MoH) should be played when only one person is in the conference. <br />
'''Quiet:''' When set to '''"YES"''', enter/leave prompts and user introductions are not played.<br />
'''Announcement:''' If set, this recording will be heard only by the user as he joins the conference.<br />
'''Drop Silence:''' The system will drop what is detected as silence from entering into the conference. Enabling this option will drastically improve performance and help remove the buildup of background noise from the call. This option is highly recommended for large conferences, due to its performance improvements.<br />
'''Talking Threshold:''' The time, in milliseconds, that a users needs to be sending sound or voice before the system can consider them to be talking. '''Recommended value is 160 MS.'''
** This setting affects different options:
*** ''1) Audio is only mixed out of a user's incoming audio stream if talking is detected. If this value is set too loose, the users will hear themselves briefly each time they begin talking until enough time has passed for the system to establish that they are in fact talking.''<br />
*** ''2) This value determines when talking has begun, which causes events to trigger in the conference dashboard. If this value is set too tight, events may be falsely triggered by the background noise of the caller.''<br />
*** ''3) The '''"Drop Silence"''' option depends on this value to determine when the user's audio should be mixed into the conference after periods of silence. If this value is too loose, the beginning of a user's speech will get cut off as they transition from silence to talking.''

'''Silence Threshold:''' The time, in milliseconds, that silence needs to be present in the user’s sound stream before the system can consider it to be in fact silent and close the audio. The best way to use this option is to set it slightly above the maximum amount of milliseconds of silence a user may generate during natural speech, i.e. the regular pauses while speaking. '''Recommended value is 2500 MS'''.
** This setting affects different options:<br />
*** ''1) This value determines when the user has stopped talking after a period of talking. If this value is set too low, events in the conference dashboard indicating that the user has stopped talking may get falsely sent out when the user briefly pauses during mid sentence.''<br />
*** ''2) The '''"Drop Silence"''' option depends on this value to determine when the user's audio should begin to be dropped from the conference, after the user stops talking. If this value is set too low, the user's audio stream may sound choppy to other participants.''

'''Talk Detection:''' If set to '''YES''', the conference dashboard will display a notification when a participant starts and stops talking.
'''Jitter Buffer:''' When set to '''YES''', the system will place a jitter buffer on the caller's audio stream before any audio mixing is performed. This option is highly recommended for maximum voice quality, but will add a slight delay to the audio and slightly impact system’s performance.

==== Select your profiles ====

After creating your profile.
'''🛈 Note:''' Depending on your needs, you can create more than one profile, with different settings on each one of them.

[[File:AudioConference6.png|border]]

=== Customize recordings prompt ===

In this section, you can customize each recording prompt used in the conference. Beforehand, you must have uploaded a recording file. <br /> For further information on proceeding, please visit [[Recordings]].

[[File:AudioConference7.png|border]]

{| class="wikitable"
|-
| '''Join:''' || The recording played when a user joins, typically some kind of beep sound
|-
| '''Leave:''' || The recording played when a user leaves, typically some kind of beep sound
|-
| '''Has joined:''' || The recording played as a user intro, e.g. "xxxx has joined the conference."
|-
| '''Has left:''' || The recording played as a user parts the conference, e.g. "xxxx has left the conference."
|-
| '''Kicked:''' || The recording played to a user who has been kicked from the conference.
|-
| '''Muted:''' || The recording played to a user when the mute option is toggled on.
|-
| '''Unmuted:''' || The recording played to a user when the mute option is toggled off.
|-
| '''Only person:''' || The recording played when a user is the only person in the conference.
|-
| '''Only one:''' || The recording played to a user when there is only one other person in the conference.
|-
| '''There are:''' || The recording played when announcing how many users there are in a conference.
|-
| '''Participants muted:''' || The recording played when all non-admin participants are muted.
|-
| '''Other in party:''' || The recording used in conjunction with the "There are" option, used like "There are" (number of participants) "Other in party”
|-
| '''Get PIN:''' || The recording played when prompting for a conference PIN.
|-
| '''Invalid PIN:''' || The recording played when an invalid PIN is entered too many (3) times.
|-
| '''Locked:''' || The recording played to a user trying to join a locked conference.
|-
| '''Locked now:''' || The recording played to an Admin-level user after toggling the conference to locked mode
|-
| '''Unlocked now:''' || The recording played to an Admin-level user after toggling the conference to unlocked mode.
|-
| '''Error menu:''' || The recording played when an invalid menu option is entered.
|-
| '''Participants unmuted:''' || The recording played when all non-admin participants are unmuted.
|}

== Link a DID number to your conference ==

You need to link a DID to your new conference to use this feature. You need to navigate through the navigation bar and select '''"DID Numbers"''' > '''"Manage DID(s)"'''.

Then, in the column "Actions" click on the button edit [[File:Edit_icon.png]], for the row of the DID you need to use with this conference.

[[File:AudioConference8.png|border|767px]]

After you chose the DID you want to link to your conference. You need to route it to a conference. Then, save your choice at the bottom of the page, by clicking on the button '''"Click here to apply changes"'''.

[[File:AudioConference9.png|border]]

== Conference Dashboard ==

The dashboard will allow the user to have a graphical interface of the participants in the conference call, providing additional options for the admins.

[[File:AudioConference10.png|border]]

The Dashboard will also offer the '''"Start Recording"''' option, making it possible to record the call by the whole duration of the call recording service was active. <br />

'''Call recording service has a rate of $0.0025 per minute''', for the duration of the recording. The per minute charges will appear in the CDR.

You also have the '''"Lock conference"''' option, which allows you to block access to this conference. Otherwise, no new members can join the conference. <br />

''' View statistics''' will allow you to see the conference statistics such as numbers or parties if the conference is locked and the maximum of members allowed.

On each participant, you will be able to see his caller ID name and number if it is available. Also, you have the possibility to '''mute''' the participant or kick-out the participant of the conference.

[[File:AudioConference11.png|border|767px]]
<br />

== Record your conference ==

During a conference, if you have activated the '''Start recording''' option, you will be able to listen to it in this section by clicking on the option '''"Downloads"'''

[[File:AudioConference12.png|border]]

The '''"Downloads"''' button opens a screen that allows the user to download, listen and send the recordings directly from the user interface.

[[File:AudioConference13.png|border]]

The option [[File:AudioConference-play.png|19px]] allows you to listen to the recording directly from your navigator. <br />

The option [[File:AudioConference-cloud.png|19px]] allows you to download the recording directly to your device. <br />

The option [[File:AudioConference-email.png|19px]] if you want to send the recording over email. Simply input the target email address and click '''"Send Email"''' button.

[[File:AudioConference14.png|border]]

[[File:AudioConference15.png|border]]
<br /><br />
[[File:AudioConference16.png|border]]

You also have the possibility to select multiple entries and delete them in one click by selecting them on the left row and then pressing *Delete Selected Records.*
[[File:AudioConfDel.png|border]]

== Direct Phone Key options In The Conference ==

The following option can be done directly in the conference, by the phone directly. This will depend if setting of the user profile, has been set as admin or not.

=== For a Regular user ===
'''* to hear options'''
[1] Mute or Unmute yourself
[4] Decrease the conference volume
[6] Increase the conference volume
[7] Decrease your volume
[9] Increase your volume
[8] Exit

=== For an Admin user ===
'''* to hear options'''
[1] Mute or Unmute yourself
[2] Lock or Unlock the conference
[3] Eject the last user
[4] Decrease the conference volume
[6] Increase the conference volume
[7] Decrease your volume
[9] Increase your volume
[8] Exit

SMS-MMS-FR

2026-07-30T19:52:15Z

Neocypher: /* Articles de blog */

{| class="wikitable"
|+
|-
! Article in English !! Artículo en Español
|-
| [https://wiki.voip.ms/article/SMS-MMS English] ||
[https://wiki.voip.ms/article/SMS-MMS-ES Español]
|}

SMS signifie '''S'''hort '''M'''essage '''S'''ervice ('''SMS'''). Avec l'ajout du service '''M'''ultimedia '''M'''essage '''S'''ervice ('''MMS'''), les utilisateurs sont désormais capables d'envoyer et de recevoir des fichiers multimédias ('''photos, vidéos, et audio''') ainsi que d'envoyer des messages texte de plus de 160 caractères (ce qui est la limite pour un SMS standard). Le service MMS permet également à l'utilisateur d'envoyer un message à plusieurs destinataires en même temps.

= Articles de blog =
:* [https://voip.ms/blog/benefits-of-sms-mms-campaigns-vs-email-marketing/ Avantages des campagnes SMS/MMS par rapport au marketing par courriel]
:* [https://voip.ms/blog/how-to-generate-leads-with-sms-mms/ Comment engendrer des prospects avec les SMS/MMS ?]
:* [https://voip.ms/blog/instantly-boost-your-customer-experience-allow-your-customers-to-text-you/ Améliorez instantanément votre expérience client, permettez à vos clients de vous envoyer des SMS!]

= Limites du service =

<div style="margin-left:50px;
padding:5px;
border-left: 6px solid rgb(208, 56, 45, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(208, 56, 45, 0.08);
width:70vw;">

🏗️ ''' Veuillez noter que le service MMS est en phase bêta. '''
: Il est important pour nous que vous signaliez tout problème avec ce service en envoyant un courriel à [mailto:support@voip.ms support] afin que les développeurs puissent intervenir si nécessaire.
</div>

<div style="margin-left:50px;
padding:5px;
border-left: 6px solid rgb(208, 56, 45, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(51, 51, 51, 0.08);
width:70vw;">

⚠️ '''LIMITATIONS'''
: ➜ '''La capacité d'envoi de SMS/MMS via le portail est illimitée.'''
: ➜ La limite de l'envoi de SMS/MMS via l'API est limitée à '''<u>100 par jour</u>'''. Pour toute modification de la limite, veuillez contacter le support ou ouvrir un ticket via le portail client.
: ➜ Pour l'instant, la réception et la lecture de MMS ne sont possibles que par le centre de messagerie ou en utilisant l'option de renvoi des SMS/MMS. '''MMS via le protocole SIP n'est pas disponible.'''

: ➜ Le Service SMS/MMS est uniquement disponible pour les <u>''''numéros DID locaux américains et canadiens'''</u> marqués avec l'icône '''SMS/MMS'''.
: ➜ Pour l'instant, le service SMS/MMS ne peut être utilisé que pour envoyer des messages MMS à des numéros à 10 chiffres <u>'''canadiens et américains'''</u>.
: ➜ Les messages SMS SIP ne seront délivrés et acceptés qu'à partir de troncs SIP utilisant une authentification par nom d'utilisateur/mot de passe.
: ➜ Les messages SMS SIP utilisent la méthode SIP MESSAGE. Voir [https://tools.ietf.org/html/rfc3428 RFC 3428] pour plus de détails.
: ➜ Pour l'instant, '''<u>nous ne pouvons pas garantir</u> que les messages SMS à numéro court''', qui comportent généralement 6 chiffres ou moins ''(ex: Skype, codes bancaires, publicités télévisées, etc.)'' fonctionneront.
: ➜ En ce moment, '''<u>nous ne pouvons pas garantir</u> que les messages SMS pour la double authentification (2FA)''', habituellement utilisés pour confirmer votre identité et vous connecter à votre portail de service '''(par exemple : Microsoft, parmi d'autres fournisseurs proposant cette fonctionnalité)''', fonctionneront."
: ➜ Pour le moment, les messages SMS envoyés/reçu à partir de numéros à code court (short codes) '''<u>ne sont pas pris en charge</u> pour les Numéros sans frais (Toll-Free)'''.
: ➜ Nous ne pouvons pas garantir que les '''accents ou caractères spéciaux''' y compris les lettres non latines, seront correctement transmis.

: ➜ Les raccourcisseurs d'URL publics tels que '''''bit.ly, tinyurl.com''''' etc, <font style="color:red">'''sont fortement restreint'''</font> par les fournisseurs de téléphonie mobile en lien à une réponse de l'industrie pour combattre les campagnes de SPAM et protéger les consommateurs. L'utilisation de ce type de liens dans vos messages '''<u>peut entraîner le blocage ou la non-délivrance du message</u>'''.
: ➜ Pour les '''Numéros transférés''' compatibles avec la fonctionnalité : La fonctionnalité SMS/MMS sera disponible jusqu'à '''48 heures après que le processus de transfert soit marqué comme terminé'''.

: ➜ VoIP.ms se réserve le droit, à sa seule discrétion, de limiter ou de désactiver le service SMS/MMS de tout compte pouvant présenter des tendances de marketing/automatisation/envoi ou réception en masse.

</div>

<div style="margin-left:50px;
clear:left;
padding:5px;
border-left: 6px solid rgb(66, 192, 251, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(66, 192, 251, 0.05);
width:70vw;">

'''💬 NOTE ''' Si vous avez d'autres questions, n'hésitez pas à contacter le personnel d'assistance sur le Live Chat ou le système de tickets.
:➜ '''Les messages MMS ont un coût de 0,02 $ par message entrant et sortant..'''

: ➜ '''De plus, veuillez noter les restrictions suivantes pour l'envoi de MMS:'''
:: ➜ Les types de pièces jointes autorisés sont les suivants : JPG, GIF, JPEG, PNG, MP3, WAV, MIDI, MP4 et 3GP.
:: ➜ La taille maximale de chaque pièce jointe est de 1300 KB. ''(1.3 MB)''
:: ➜ Vous pouvez joindre jusqu'à 3 fichiers.
:: ➜ Le nombre maximum de caractères dans un message texte est de 2048.

</div>

=Utilisation de la fonction SMS/MMS=
==Identifier un numéro DID compatible avec les SMS/MMS==

La première chose que vous devez savoir est que cette fonction n'est disponible que pour les numéros locaux américains '''(incluant Alaska, Hawaii et Puerto Rico)''' et canadiens pour le moment. Vous remarquerez que certains numéros ont une petite icône [[File:Sms-mms-icon.png|15px]] ce qui indique que le numéro supporte cette fonction.

Vous pouvez commencer à commander un numéro DID à partir de votre portail >> Numéros DID >> Commander DID >> Numéros locaux >> Numéros (US ou Canada) et vous pourrez acheter le numéro souhaité, assurez-vous simplement qu'il supporte la fonction SMS/MMS.

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

==Configuration du service SMS/MMS==

Une fois acheté, le service SMS/MMS doit être activé dans les paramètres DID, à partir de votre portail >> Numéros DID >> Gestion des DIDs >> Editer la sélection >> section configuration SMS/MMS.

Dans cette section, vous pouvez activer et configurer certaines options de transfert pour le service SMS/MMS.

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

:* '''Message Service (SMS/MMS)''': Cochez la case pour activer le service.
:* '''VoIP.ms SMS/MMS Portal''': Cette option est activée par défaut et ne peut être modifiée, ce qui signifie que vous pourrez créer et envoyer des messages à partir du centre de messages SMS.
:* '''SMS/MMS Email Address''': En activant cette option, tous les messages SMS seront envoyés à l'adresse email que vous configurez dans ce champ. Vous pouvez les recevoir sur plus d'une adresse e-mail en séparant chaque e-mail par une virgule. L'avantage de cette option est que vous recevrez vos messages SMS directement sur votre email et que vous pourrez également répondre à ces messages depuis votre email. Il vous suffit de cliquer sur '''répondre''' au courriel. Veuillez noter que les réponses aux e-mails sont également limitées à 160 caractères et que si elles dépassent cette limite, elles seront divisées en deux (ou plusieurs messages le cas échéant) lors de leur envoi.

<div style="margin-left:50px;
padding:5px;
border-left: 6px solid rgb(208, 56, 45, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(51, 51, 51, 0.08);
width:70vw;">

'''NOTE''' 💬: ➜ Pour répondre au message SMS par l'intermédiaire de votre messagerie électronique, il est important de cliquer sur '''Répondre''' et '''NE PAS changer ou modifier''' le courrier de destination ''À'' (sms@voip.ms) et le sujet, par exemple ''[#USXXXXXX] Message sent to 5555555555''.
</div>

:*'''SMS/MMS Forward''': Vous pouvez faire transférer vos messages SMS vers votre téléphone portable ou tout autre numéro qui supporte les SMS. Avec cette option, vous recevrez les messages SMS de notre système vers le numéro de téléphone configuré.
<div style="margin-left:50px;
padding:5px;
border-left: 6px solid rgb(208, 56, 45, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(51, 51, 51, 0.08);
width:70vw;">

'''NOTE''' 💬: ➜ Le numéro d'identification de l'appelant envoyée à un numéro mobile renvoyé sera le numéro DID qui a reçu le message SMS. Il n'est pas possible de renvoyer l'ID de l'appelant original pour le moment.
</div>

:*'''SMS/MMS SIP Account''': Vous pouvez faire transférer vos messages SMS vers le compte SIP de votre choix. Ils seront envoyés par notre passerelle SMS/SIP en tant que SIP MESSAGE.
<div style="margin-left:50px;
padding:5px;
border-left: 6px solid rgb(208, 56, 45, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(51, 51, 51, 0.08);
width:70vw;">

'''NOTE''' 💬: ➜ La destination (champ d'en-tête "To") du SIP MESSAGE sera votre nom de compte, au lieu du numéro de destination du SMS original.
</div>

:'''SMS/MMS URL Callback''': En activant cette option, vous serez en mesure d'envoyer les messages vers une autre destination, par exemple votre propre serveur. Si cette option est sélectionnée, les messages SMS reçus par votre DID enverront une requête GET à l'URL Callback fournie. Variables disponibles pour votre URL :
<div style="margin-left:50px;
padding:5px;
font-family: Consolas, Monaco, Lucida Console, Liberation Mono, DejaVu Sans Mono, Bitstream Vera Sans Mono, Courier New, monospace;
color: silver;
border-left: 6px solid rgb(208, 56, 45, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: black;
width:58vw;">
:: '''Example''': <nowiki>https://VotreNomDeDomaine.tld/sms.php?to={TO}&from={FROM}&message={MESSAGE}&id={ID}&date={TIMESTAMP}</nowiki>

: '''{ID}''' = Identification du message SMS.
: '''{TIMESTAMP}''' = La date et l'heure de réception du message.
: '''{FROM}''' = Le numéro de téléphone qui vous a envoyé le message.
: '''{TO}''' = Votre numéro DID qui a reçu le message.
: '''{MESSAGE}''' = Le contenu du message texte.
: '''{MEDIA}''' = Liste de fichiers médias séparés par des virgules.

</div>

:*'''URL de rappel (Callback Retry)''': Lorsque cette option est sélectionnée, nous attendons un "ok" (sans guillemets) de votre page web (URL de rappel) comme indicateur que vous avez reçu le message correctement. Si nous ne recevons pas le "ok" (sans guillemets) de votre page de rappel, nous continuerons à vous envoyer le même message toutes les 30 minutes.

Une fois que vous avez configuré l'option souhaitée, cliquez sur Appliquer les modifications.

==Envoi d'un SMS/MMS (Interface Web)==

'''Step 1'''

: Cliquez sur le bouton "Envoyer un nouveau message" pour faire afficher une liste de contacts ou tapez directement le numéro dans le champ de texte du contact. Si ce numéro est enregistré dans le répertoire, son nom s'affichera à la place.

:: [[File:MMS-Webportal1_FR.png|border|400px]]

'''Step 2'''

: Cliquez pour afficher la liste des numéros DID disponibles. Choisissez le numéro à partir duquel envoyer le message.

:: [[File:MMS-Webportal2_FR.png|border|400px]]

'''Step 3'''

: Ecrivez votre message dans le champ de texte et cliquez sur la flèche rouge pour l'envoyer. Les messages texte de 160 caractères maximum seront considérés comme des SMS. Les messages multimédia ou les messages texte de plus de 160 caractères seront considérés comme des MMS et vous verrez l'icône passer de SMS à MMS, ce qui indique ceci.
:: [[File:MMS-Webportal3_FR.png|border|400px]]

'''Step 4'''

: Afin d'envoyer un fichier multimédia par MMS, cliquez sur le bouton clip pour parcourir et choisir un fichier multimédia.

:: [[File:MMS-Webportal4_FR.png|border|400px]]

: Cliquez sur le bouton Parcourir pour rechercher le fichier multimédia sur votre ordinateur ou faites glisser le fichier directement dans la boîte.

:: [[File:MMS-Webportal5_FR.png|border|400px]]

: Les fichiers joints s'afficheront sous la zone de texte.

:: [[File:MMS-Webportal6_FR.png|border|400px]]

: Une fois le message envoyé, une nouvelle conversation s'ouvre sur la gauche. En sélectionnant cette conversation, l'historique des messages s'affiche dans la boîte.

:: [[File:MMS-Webportal7_FR.png|border|400px]]

==Réception d'un SMS/MMS (Interface Web)==

: Alors que les SMS peuvent être reçus directement dans une application grâce au protocole SIP et à une configuration correcte, pour l'instant, '''la réception et la lecture de MMS ne sont possibles que par le centre de messages.'''

: Vous pouvez accéder au centre de communication depuis votre portail client sous Numéros DID > Centre de messagerie SMS/MMS. Tout message reçu apparaîtra ici sur le côté gauche du panneau.

: Utilisez le filtre DID pour choisir et afficher les messages d'un numéro DID spécifique ou pour tous les numéros DID.

: Vous pouvez également filtrer les messages par date et fuseau horaire.

:: [[File:SMSmmsfilter1FR.png|border|300px]]

:: [[File:MMS-Inbound-1.png|border|500px]]

: Choisissez une conversation dans le panneau de gauche. Une option permettant de supprimer la conversation est également disponible. Choisissez le numéro DID dans la zone de texte pour répondre.

:: [[File:MMS-Inbound_DELETE.png|border|500px]]

== Historique SMS/MMS ==

Vous pouvez désormais accéder à un historique général des SMS/MMS envoyés et reçus pour tous vos DID, ou pour un spécifique en cliquant sur '''Historique'''.

:: [[File:SMSMMSHistoryLoc.png|border|500px]]

:: [[File:History.png|border|500px]]

= Email à SMS =

À partir du 14 janvier 2025, il est désormais possible d’initier une conversation SMS via courriel. Pour activer cette fonction, rendez-vous dans votre portail VoIP.ms, sous DID Numbers, SMS/MMS Message Center, puis cliquez sur « Email to SMS ».

: [[File:EmailtoSMS1.png|border|500px]]

Ensuite, cliquez sur « Add New Email to SMS » et remplissez les informations suivantes :

*Activé: Oui
*Adresse courriel: Votre adresse courriel
*Code de sécurité activé: Si vous souhaitez activer un code de sécurité pour envoyer des SMS/MMS.
*Code de sécurité: Le code de sécurité à utiliser.
* Numéro de l'expéditeur par défaut: Le numéro DID que vous souhaitez utiliser pour envoyer le message SMS/MMS.

: [[File:EmailtoSMS2.png|border|500px]]

Une fois les informations remplies, cliquez sur « Save », et tout est prêt !

==Comment initier une conversation SMS/MMS via courriel==
1. Utilisez le compte de messagerie courriel que vous avez fourni lors de l'activation du service courriel à SMS.

2. Envoyez le courriel à sms@voip.ms.

3. Dans le champ sujet, saisissez le numéro de téléphone du destinataire (ex: 5552340000).

4. Joignez le document que vous souhaitez envoyer au courriel.

5. Types de pièces jointes autorisées : JPG, GIF, JPEG, PNG, MP3, WAV, MIDI, MP4 et 3GP.

6. La taille maximale pour chaque pièce jointe est de 1300 Ko.

7. Vous pouvez joindre jusqu'à 3 fichiers.

8. Envoyez le courriel.

===Code de sécurité et numéro de l'expéditeur===

1. Si le code de sécurité est activé, vous devez ajouter un point (.) et le code de sécurité après le numéro du destinataire. (ex: 5552340000.Az09).

2. Si vous disposez plus d'un numéro DID pour l'envoi de SMS et que vous souhaitez utiliser un autre de vos DID comme "numéro d'expéditeur", vous devez ajouter un point (.) après le numéro de téléphone du destinataire de votre SMS, suivi du code de sécurité avec un autre point (.) et ensuite le numéro DID que vous souhaitez utiliser. (ex: 5552340000.Az09.5557890000).

3. Si vous n'avez pas activé le code de sécurité et que vous souhaitez utiliser un autre de vos DID comme "numéro d'expéditeur", vous devez ajouter un point (.) après le numéro de téléphone du destinataire de votre SMS suivi du numéro DID que vous souhaitez utiliser (ex: 5552340000.5557890000).

= SMS à l'aide d'un serveur SMPP =

Un serveur SMPP, abréviation de '''Short Message Peer to Peer''' est une méthode conçue pour envoyer et recevoir de gros volumes de SMS via un serveur dédié.

En d'autres termes, cela vous permet de mieux contrôler la quantité de messages qui seront envoyés à votre public cible tout en évitant de passer par une application softphone ou votre portail VoIP.ms pour envoyer/recevoir ces SMS, éliminant ainsi de multiples tâches répétitives et vous faisant gagner du temps.

Par exemple, si vous avez une promotion pour votre entreprise qui offre un rabais pendant un certain laps de temps et que vos clients ont donné leur accord pour recevoir ces types de promotions par SMS, vous pourriez envoyer votre promotion à vos clients de manière plus rapide que si vous le faisiez depuis votre portail client, augmentant ainsi votre efficacité.

<div style="margin-left:50px;
padding:5px;
border-left: 6px solid rgb(208, 56, 45, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(51, 51, 51, 0.08);
width:70vw;">

'''NOTES''' 💬: ➜ Notez que l'envoi et la réception de messages SMS seront facturés à notre tarif habituel de 0,0075 $ par message'''
</div>

==Exigences==

: '''Pour cela, vous devez disposer de votre propre serveur SMPP.'''
: Une fois configuré correctement, vous serez alors en mesure d'envoyer et de recevoir des SMS sur votre propre serveur.

== Configuration on your SMPP Server ==

: Pour que votre serveur puisse communiquer et s'authentifier en conséquence avec le serveur SMPP de VoIP.ms, vous devrez configurer ce qui suit:

::* Un '''nom d'utilisateur''' et un '''mot de passe''' de votre choix afin que le serveur SMPP de VoIP.ms puisse vous authentifier.
::* Faites en sorte que toutes vos communications soient envoyées à '''smpp.voip.ms''' via le port '''2775''', pour le SMPP normal, ou le port '''3550''' qui est notre port SMPP crypté.
::* Ouvrez une connexion en utilisant la commande '''bind_transceiver'''. Vous pouvez garder cette connexion ouverte aussi longtemps que vous le souhaitez et l'interroger en utilisant la commande '''enquire_link'''.
::* Envoyez vos messages avec les commandes '''deliver_sm''' ou '''submit_sm'''. Assurez-vous d'inclure l'adresse '''source_addr''' de votre PDU comme l'un des DID SMS activés de votre compte. Votre numéro de SMS en destination doit être défini comme '''destination_addr''' tandis que votre message texte sera inclus dans '''short_message'''.

== Configuration SMPP sur votre portail VoIP.ms ==

Une fois que vous avez configuré les éléments ci-dessus sur votre serveur, vous devrez activer et spécifier le nom d'utilisateur et le mot de passe qui seront utilisés par VoIP.ms pour authentifier vos requêtes SMPP. Pour ce faire, allez dans votre portail VoIP.ms, Numéros DID, Gérer les DID, éditez votre DID qui supporte les messages SMS et en descendant la page, vous verrez les options suivantes:

:* '''SMPP Enabled:''' Si cette option est sélectionnée, les messages SMS pourront être envoyés et reçus en utilisant SMPP.
:: Vous pouvez spécifier l'URL à utiliser pour envoyer une copie des messages entrants à votre serveur SMPP. Laissez vide si vous souhaitez uniquement envoyer des messages. Voir l'exemple ci-dessous.

:: La partie '''scheme''' est obligatoire et doit être définie comme '''smpp''' pour un SMPP non crypté, ou '''ssmpp''' pour un SMPP crypté utilisant TLS.
:: La partie '''user:password@''' est facultative et prendra par défaut le même nom d'utilisateur et le même mot de passe que ceux définis pour l'authentification de notre côté, comme défini ci-dessous.
:: La partie ''':port''' est également facultative et prendra par défaut la valeur 2775 pour '''smpp''' et '''3550''' pour ssmpp.

:* '''SMPP Username:''' Le nom d'utilisateur que vous utiliserez pour vous authentifier auprès de notre serveur SMPP pour l'envoi de messages.
:* '''SMPP Password:''' Le mot de passe que vous utiliserez pour vous authentifier auprès de notre serveur SMPP pour pour l'envoi de messages.

<div style="margin-left:50px;
padding:5px;
border-left: 6px solid rgb(208, 56, 45, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(51, 51, 51, 0.08);
width:70vw;">

'''NOTES''' 💬
: ➜ Afin d'éviter toute confusion lors de la configuration du nom d'utilisateur sur votre serveur, vous pouvez utiliser votre numéro de compte VoIP.ms comme nom d'utilisateur (les 6 chiffres de votre compte).
: ➜ Assurez-vous que l'option '''Service de messages (SMS/MMS)''' est activée.
</div>

Activer l'option '''SMPP Enabled''', remplissez le nom d'utilisateur et le mot de passe conformément à ce qui a été configuré sur votre serveur SMPP et une fois que c'est fait, appliquez les changements en appuyant sur '''Cliquez ici pour appliquer les changements'''.

: [[File:SMPPconfig.png|border|750px]]

Vous pouvez maintenant commencer à envoyer et recevoir des messages SMS à partir de votre serveur SMPP.

= Envoyer et recevoir des messages texte (interface Web) =

==Centre de messages SMS pour mobile==

Pour commencer à utiliser le service, vous devrez utiliser le Centre de messagerie SMS/MMS, à partir de votre portail >> Numéros DID >>

Pour les appareils mobiles, veuillez utiliser [https://sms.voip.ms/ sms.voip.ms] comme une solution SMS complète tout en un.

Depuis le Centre de messagerie SMS, vous pouvez consulter votre historique de SMS, envoyer et recevoir des messages SMS.

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

:* '''Plage de recherche''': Sélectionnez une plage de dates de votre historique de SMS. Vous pouvez sélectionner une période de temps en cochant le champ "Show Details" et les messages texte seront affichés.
:* '''Filtre de recherche''': Vous pouvez filtrer votre recherche par DID, contact et type (envoyé, reçu et les deux).
:* '''Envoyer de nouveaux messages SMS''': Une autre fenêtre s'ouvrira en cliquant ici afin de créer un nouveau message SMS.
:* '''Supprimer les messages sélectionnés''': Cette action supprimera tous les messages que vous avez sélectionnés, en cliquant sur la case à côté d'eux. '''Cette action ne peut pas être annulée.'''
:* '''Supprimer tous les messages''': Supprime tous vos messages SMS, qu'ils soient envoyés ou reçus. '''Cette action ne peut pas être annulée'''.
:* '''Vérifier un message''': Pour vérifier un SMS est reçu, cliquez directement dessus afin d'afficher les informations.

==Créer un nouveau message SMS (ancienne version)==

: [[File:SMS41.png|thumb|none|600px]]

Vous pouvez créer un nouveau '''Message SMS''' nous vous expliquerons tous les paramètres que vous trouverez dans cette section :

* '''DID''': Si vous avez plus d'un numéro DID, vous pouvez sélectionner n'importe lequel des numéros disponibles pour envoyer votre message.
* '''Contact''': Le numéro de téléphone de destination ou si vous avez un [[Répertoire téléphonique|Entrée dans le répertoire téléphonique]] commencez à taper le nom et le système affichera l'information si elle est disponible.
* '''Message''': Vous pouvez créer votre message dans ce champ, veuillez noter que vous avez jusqu'à 160 caractères.

Cliquez ensuite sur "Envoyer le message" pour envoyer le SMS.

= Envoyer et recevoir des messages texte (via le protocole SIP MESSAGE) =
<div style="margin-left:50px;
padding:5px;
border-left: 6px solid rgb(208, 56, 45, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(51, 51, 51, 0.08);
width:70vw;">

'''NOTES''' 💬
: ➜ '''SMS via SIP''' est uniquement disponible par le biais de '''par le protocole SIP et l'authentification utilisateur/mot de passe'''. '''L'auth. par IP et IAX2 ne sont pas disponibles pour le moment.'''
: ➜ '''Veuillez vous référer à notre guide [[SIP/SMS with FreePBX]] pour des instructions sur la façon de configurer le service avec un FreePBX'''
</div>

== Envoi ==

:* Le DID doit avoir l'option de service '''"Short Message Service (SMS)"'' activée. Cette option peut être activée à partir de la page Gestion des DIDs >> Modification d'un DID.

:* Pour envoyer un SMS à partir de votre compte SIP, vous devez définir votre numéro d'identification de l'appelant sur l'un de vos DID SMS activés. Ce sera le numéro à partir duquel vous enverrez le message. Vous pouvez configurer le numéro d'appelant à partir de votre portail client pour le sous-compte spécifique si vous utilisez un softphone ou directement à partir de votre extension ou trunk si vous utilisez un serveur Asterisk ou PBX.

:* Il est important de noter que si vous configurez le nom de l'identifiant de l'appelant dans votre client SIP pour qu'il soit un numéro à 10 chiffres, celui-ci remplacera votre numéro d'identifiant de l'appelant. Si le nom de l'identifiant de l'appelant est différent d'un numéro à 10 chiffres, il sera ignoré et le numéro de l'identifiant de l'appelant sera utilisé.

:* Si vous êtes un utilisateur d'Asterisk ou d'un PBX, assurez-vous d'utiliser la dernière version d''''Asterisk (v12 ou supérieure)''' et d'utiliser '''chan_pjsip''' pour le trunk. Il s'agit d'une limitation d'Asterisk.

:* La version Desktop de Zoiper nécessite que l'utilisateur ait la '''version PRO''' activée pour pouvoir envoyer des messages SMS ainsi que pour activer la '''Présence SIP'''. Veuillez contacter le support technique pour demander que la présence SIP soit activée pour votre compte pour le POP VoIP.ms que vous utilisez. Bien que normalement non requis, il peut y avoir d'autres cas non documentés de softphones ou d'applications nécessitant l'activation de la Présence SIP. Veuillez contacter le support technique si vous rencontrez un problème avec votre logiciel spécifique.

:* La version '''Desktop''' gratuite de Zoiper ne peut pas envoyer de SMS et est seulement capable de recevoir des SMS.

:* La '''version mobile''' de Zoiper est complètement libre de ces limitations. La version gratuite fonctionne déjà pour l'envoi et la réception sans la nécessité d'une présence SIP.

:* Lors de l'envoi d'un message SMS, le numéro du destinataire doit être composé de 10 chiffres. Evitez d'utiliser le préfixe '''1''' car il n'est pas supporté pour le moment. Le système reconnaîtra les numéros comme étant à 10 chiffres.

:* La limite actuelle de caractères pour un '''simple SMS''' est de 160 caractères. Si un message a plus de 160 caractères, il sera divisé en blocs de 160 caractères maximum chacun et sera envoyé en tant que messages indépendants. La longueur totale maximale du message pour les SMS via SIP est de 2000 caractères.

== Réception ==
<div style="margin-left:50px;
padding:5px;
border-left: 6px solid rgb(208, 56, 45, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(51, 51, 51, 0.08);
width:70vw;">

'''NOTES''' 💬: ➜ POUR LE MOMENT, L'ENVOI DE MMS À L'AIDE D'UN COMPTE SIP N'EST PAS POSSIBLE.'''
</div>

:* Le DID doit avoir l'option de service ''"Service Message (SMS/MMS)"'' activée. Cette option peut être activée à partir de la page Gestions des DIDs > Editer les DIDs.
:* Activez l'option '''"Compte SIP SMS"''' et sélectionnez le compte de destination de vos SMS vers ce numéro.
:* Le compte recevant les SMS doit être enregistré correctement dans l'un de nos POPs.
:* Pour garantir les meilleurs résultats lors de la réception des messages SMS, n'enregistrez pas le même compte SIP sur plus d'un softphone ou d'une application en même temps, cela peut entraîner des problèmes de réception.
:* Pour les appareils SIP qui requièrent cette information, tels qu'un trunk ou un PBX, nous encodons le numéro de destination original dans un champ d'en-tête SIP personnalisé nommé "X-SMS-To".

L'application 3CX ne prend actuellement pas en charge le service SMS de VoIPms. Il ne fonctionnera qu'en ayant une connexion directe avec le système PBX 3CX. '''Veuillez contacter 3CX pour plus de détails'''.

== Envoyer des SIP/SMS avec FreePBX ==

<div style="margin-left:50px;
padding:5px;
border-left: 6px solid rgb(208, 56, 45, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(51, 51, 51, 0.08);
width:70vw;">

'''NOTES''' 💬: ➜ POUR LE MOMENT, L'ENVOI DE MMS À L'AIDE D'UN COMPTE SIP N'EST PAS POSSIBLE.'''
</div>

Oui, il est possible de configurer un système FreePBX pour envoyer des SMS, veuillez vous référer [[SIP/SMS with FreePBX | ici]] pour des instructions détaillées.

=SMS/MMS à l'aide de l'interface revendeur=

Cette fonction est disponible pour votre client via l'interface revendeur. Vous devez activer cette fonction dans votre forfait afin de leur donner la possibilité d'en tirer parti.
Allez sous la barre de navigation sur ‘’’[Revendeur] ‘’’ puis cliquez sur ‘’’ [Gestion des tarifs et des forfaits] ‘’’

Cliquez sur le bouton Modifier pour modifier votre forfait, ou cliquez sur ‘’’[Créer un nouveau forfait] ‘’’ pour en créer un nouveau.

[[File:CallForwarding_Reseller_1_FR.png|thumb|none|700px]]

Allez sous l'onglet [Configuration du système du revendeur], et dans la section "Type de configuration" sélectionnez: [Configuration du forfait],

[[File:CallForwarding_Reseller_2_FR.png|thumb|none|700px]]

Ensuite, faites défiler vers le bas et recherchez la fonction "SMS/MMS", puis activez-la.
: [[File:SMSresellerFR4.png|thumb|none|500px]]

= Activer/désactiver les SMS/MMS sur un DID via l'interface du client =

Pour que votre client active ou désactive les SMS / MMS depuis son portail, votre client devra se diriger sous Services, Mes services.

: [[File:SMSMMSreseller5.png|thumb|none|500px]]

Puis continuez en vous rendant dans l’onglet ‘’ ’DID Lists’ ’’ et modifiez DID.

: [[File:SMSMMSreseller6.png|thumb|none|500px]]
Vous pourrez alors activer les SMS / MMS, recevoir par e-mail et plus encore.
: [[File:SMSMMSreseller7.png|thumb|none|500px]]

= Consultation des SMS/MMS via l'interface client =
1) Pour accéder à leurs messages SMS / MMS, rendez-vous sous Services puis SMS / MMS.
: [[File:SMSMMSreseller8.png|thumb|none|500px]]

[[category:guides]]

Numero Identificador (Caller ID)

2026-07-30T19:48:02Z

Neocypher: /* Artículo de Blog */

{| class="wikitable"
|+
|-
! Article in English !! Article en Français
|-
| [[https://wiki.voip.ms/article/Caller_ID English]] ||
[[https://wiki.voip.ms/article/ID_de_l%27appelant Français]]
|}

== Artículo de Blog ==
[https://voip.ms/blog/what-is-cnam-and-how-to-leverage-it-for-your-business/ What Is CNAM and How to Leverage It for Your Business?]

== Número de Identificación de Llamadas ==

El número identificador de llamadas (Caller ID), es un servicio que permite transmitir el número y/o nombre de la persona que llama, al teléfono de la persona que recibe la llamada.

Cuando usted hace llamadas salientes, lo más probable es que usted quiera pasar su propio número identificador, para mandar un número de contacto y asegurar que la llamada conecte correctamente. Pasar un número identificador es igualmente muy importante al momento de llamar a una linea 800 gratuita.

Hay 2 diferentes tipos de Identificador, uno es el número (Caller ID number) y otro es el nombre (Caller ID name), y es importante notar la diferencia entre ellos.

Se debe tener en cuenta en todo momento, que el identificador (para sus llamadas salientes) sólo esta garantizado usando la ruta [[Value_vs._Premium|Premium]], y solo para llamadas en US48 y Canadá, incluso si se encuentra el identificador funcionando correctamente en la ruta [[Value_vs._Premium|value]] en algún momento.

Posteriormente en este artículo se revisará el caso del Identificador de llamadas, cuando se reciben llamadas.

__TOC__

== Número de Identificación de llamadas salientes ==

Este es el tipo de identificador que usara mas comúnmente, ya que es el que envía en cada llamada saliente. Si usted usa un sistema más complejo y avanzado como un PBX o Asterisk, éstos con capaces de pasar su propio Identificador desde el trunk o la extensión.
Si en lugar utiliza un ATA - Adaptador de Teléfono ([[Welcome#Devices|Analogue Telephone Adapters]]), un Teléfono IP ([[Welcome#Devices|IP phones]]), o un [[Welcome#Softphones|softphones]] necesita configurar el identificador desde el portal de usuario de VoIP.ms.

=== Reglas del número de identificación de llamadas ===

De acuerdo con todos los fallos a favor de proteger y empoderar a los consumidores, VoIP.ms aplica identificaciones de llamadas válidas según los estándares NANPA y ITU-T E.164.

Por lo tanto, las llamadas que muestren la siguiente información de identificación de llamadas serán rechazadas:

*Llamadas internacionales con un número de identificación de llamadas que contengan menos de 7 dígitos y más de 15 dígitos.
*Llamadas de Norteamérica con un número de identificación de llamadas que no contengan exactamente 10 dígitos (se hará una excepción para los 7 dígitos para los números 310-xxx).
*Llamadas de Norteamérica con un número de identificación de llamadas que contengan 10 dígitos, pero con un NPA no asignado (los primeros 3 dígitos del número).
*Llamadas de Norteámerica con un número de identificación de llamadas que contenga 10 dígitos, pero donde se usa un NXX no asignable (es decir, el segundo bloque de 3 dígitos donde el primer dígito es cero (0) o uno (1)).

=== Menú principal ===

Para establecer el número de identificación de llamadas para su cuenta principal, haga lo siguiente:
*Ingrese al menú principal '''"Main menu"''' y seleccione la opción '''"Account_Settings"'''.
*Haga click en la pestaña de '''"General"''' y configure su número de identificación de llamadas.

<div><ul>
<li style="display: inline-block;"> [[File:AccountSettingsMenu.png|thumb|none|800px|Account settings]] </li>
<li style="display: inline-block;"> [[File:Generalnew.png|thumb|none|600px|General]]
</ul></div>

=== Subcuentas ===

Si necesita configurar el número de identificación de llamadas para una [[Sub_Cuentas_(Sub_Accounts)|subcuenta]], lo puede hacer al momento de crearla, o después, haciendo clic en Editar en la [[Sub Accounts|subaccount]].

[[File:CallerIDSubAcc.png|thumb|none|700px|CallerID field in a sub account]]

* '''Para llamadas a EE. UU. Y Canadá, se requiere un número de identificación de llamadas de 10 dígitos para garantizar la terminación adecuada de la llamada.
'''El campo de identificación de llamadas del portal solo admite dígitos numéricos.'''
* '''No se reocomienda números de identificación de llamadas anónimos o de números 800 ya que pueden causar problemas para conectar las llamadas.'''

Si está utilizando un adaptador de teléfono analógico (ATA), teléfono IP o softphone, aquí es donde debe ajustar la configuración. Es importante asegurarse de transmitir un número de identificación de llamadas válido para garantizar una terminación adecuada. Aquí tendrá la opción de que uno de sus números DID entrantes se muestre como un número de identificación de llamada saliente. También puede utilizar un número personalizado que no aparece en sus listas de números DID. No obstante, debe respetar las normas vigentes por la normativa a tal efecto. Utilice siempre un número que le pertenezca o que tenga permiso para utilizar como número de visualización de llamadas salientes. Si tiene un PBX que administra el número de identificación de la llamada saliente, tendrá una opción para esto.

'' Nota: No se recomienda usar un número de identificación de llamadas salientes gratuito, especialmente cuando se llama a números sin cargo ''.

==Diferentes métodos para pasar su número de identificación de llamadas==

'''Número de identificador de llamadas'': Puede configurar el [[Caller ID|Número de identificador de llamadas]] o su Número DID o continuar con un proceso de verificación para pasar el identificador de llamadas de un número que no está dentro de la red de VoIP.ms. Si tiene un dispositivo capaz de pasar su propio [[Caller ID|Número de CallerID]] como un interruptor suave o [[PBXs|PBX]], asegúrese de que el CallerID coincida con un DID o un número verificado dentro de su cuenta para hacer llamadas salientes con éxito.

*‘’’Usar uno de mis DID’’’: si desea pasar uno de sus números DID como el número de identificación de la persona que llama, seleccione esta opción y elija el DID, si hay más de uno, que se mostrará en la parte llamada.

*‘’’Use a Verified CallerID’’’: este es un proceso de verificación rápido para garantizar que el identificador de llamadas que pasará esté validado y autorizado para su uso. Para iniciar el proceso:

1. Simplemente haga clic en ""Haga clic aquí para comenzar"" arriba de la configuración del número de identificación de llamadas

2. Seleccione el país de su DID

3. Ingrese el número que desea verificar

4. Deje una referencia/nota para su propio uso.

5. Elija la verificación a realizar recibiendo una llamada o recibiendo un SMS (solo disponible para Canadá/Estados Unidos).

6. Una vez que presione "Enviar código de verificación", recibirá una llamada telefónica / mensaje SMS con un código de validación. Ingrese ese código en el campo "Código de validación" y presione "Verificar". Si por alguna razón no ha recibido el código, puede pedirle al sistema que lo envíe nuevamente o presione cancelar para cambiar el método en el que desea recibir el código.

*’’’I use a system capable of passing its own CallerID’’’: If you are using a PBX, you will need to pass the callerID directly from your system. Our system will verify if the callerID you are passing matches one of your DID or one of your verified callerID. If you are passing an invalid callerID and attempt to make an outgoing call, our system will send you an email advising that a call was attempted with an invalid callerID.

=== Escuche el número de identificación de llamadas de la cuenta actual ===

Para escuchar el número de identificación de llamadas configurado en la cuenta desde la que está marcando, marque '''822''' y se leerá el número de identificación de llamadas. Tenga en cuenta que dependerá del plan o patrón de marcación de su dispositivo para que esto funcione.

Estas llamadas de prueba no tendrán ningún costo para usted y aparecerán en sus reportes de llamadas grabadas [[Call_Detail_Records | CDR]] como "CallerID Testing".

===Número de identificación de llamadas salientes usando el desvío de llamadas===

Cuando elige reenviar todas sus llamadas entrantes desde su número VoIP.ms a un número de teléfono externo, el número de identificación de llamadas original se transmitirá a este número externo. Sin embargo, si la persona que llama oculta su número de teléfono, esta llamada fallará debido a las nuevas regulaciones para CID, en las que es obligatorio enviar un número de identificación de llamada válido.

Una solución alternativa es crear un filtro para llamadas "Anónimas" junto con otra entrada de desvío de llamadas, pero en esta usando la "Anulación del número de identificación de llamadas" (''"Caller ID override"'''). De esta manera, las llamadas entrantes que usen un CID válido se transmitirán a su número externo mostrando el CID original y las llamadas entrantes que oculten su número usarán el filtro y el reenvío con anulación de identificación de llamadas, mostrando el número de anulación de CID y completando la llamada.

'''Para recibir llamadas "Anónimas" al usar el desvío de llamadas, siga estos pasos:'''

1. Vaya a la sección '''DID Numbers>> Call forwarding''' y cree una nueva entrada de desvío de llamadas al mismo número al que está desviando, pero esta nueva entrada contendrá la anulación del número de identificación '''"Caller ID override"'''. Para anular el identificador de llamadas, puede escribir su número de VoIP.ms o cualquier número válido que desee ver cuando reciba una llamada anónima. Es posible que desee completar el campo de descripción '''"Description field"''' para que pueda identificar esta entrada en el siguiente paso.

2. A partir de entonces, vaya a la sección '''DID Numbers>> CallerID filtering''' y cree un nuevo filtro.

* Para el paso 1, marque la casilla de verificación número de identificación de llamadas anónimas '''"Anonymous CallerID number,"'''
* Para el paso 2, esto es opcional, puede optar por aplicar este filtro a todos sus números o solo a uno específico.
* Para el Paso 3, marque la casilla de verificación de desvio de llamadas'''"Call forwarding"''' y asegúrese de utilizar la entrada de desvío de llamadas con el número de anulación del identificador de llamadas.
* Guarde su filtro.

== Nombre de identificación de llamadas salientes ==

El identificador por nombre (Caller ID name), es información adicional que se puede enviar junto al número, esto se recibe de igual forma en el identificador de la persona a la que llama, y puede ser su nombre personal o el nombre de su negocio.

Por ejemplo: ''"John Smith"<9145551234>''

El ejemplo anterior muestra un identificador que incluye el número y el nombre, comunmente abreviado como CID o CNAM.

No es posible ingresar el nombre desde el portal de usuario de VoIP.ms

Si usted realiza llamadas a Canadá, puede pasar el nombre (Caller ID name) con sólo configurarlo desde el dispositivo o sistema que esé usando para sus llamadas (si éste lo soporta). Generalmente este campo se encuentra desde la interfaz de configuración del dispositivo o sistema que use.

El identificador por nombre (en llamadas salientes) hacia US48 funciona de forma diferente, este esta controlado por la base de datos nacional CNAM, que contiene registros de los números telefónicos, y los nombres asociados a ellos. Cuando usted realiza una llamada hacia un número de US, usted pasa un número identificador, el sistema entonces hará una búsqueda en la base de datos, por un nombre que esté asociado a este número especifico, y lo enviará junto con el número al recipiente de la llamada.

El servicio de CNAM, sólo esta disponible para números de USA y no para todas las áreas. Para ingresar su nombre y número en la base de datos de CNAM, se realiza un proceso que tiene un costo de $10 USD, esto es un único pago. Para averiguar si su número es elegible para este proceso y que información necesita proveer, contacte al soporte de VoIP.ms.

'''El CNAM sólo está disponible para algunos números locales de USA, los números 800 no pueden ser dados de alta.'''

== Número y nombre de identificación de llamadas entrantes ==

Cuando se trata de llamadas entrantes, usted recibirá la misma información, número y nombre, exactamente como VoIP.ms la reciba del usuario que llama. Usted puede verificar esta información checando las llamadas entrantes y el identificador desde su [[Registro_de_Llamadas_(CDR)|CDR]].

El nombre de identificación de llamadas entrantes funciona casi de la misma manera, excepto que esta es una configuración opcional que debe habilitar por número DID en la página de configuración DID [[Manage DID|DID settings]] por cada DID que se tenga. Esta opción se llama "CallerID Name Lookup". Cuando está habilitada esta opción, el sistema realizará una consulta en la base de datos LIBD / CNAM, para las personas que llaman con un número CID canadiense o estadounidense, con el fin de encontrar un nombre que coincida con ese número CID. Luego, el sistema mostrará el resultado de esta consulta y se desplegara en tu teléfono junto al número.

Si el número que llama ya está en su directorio telefónico, el nombre se tomará de allí en lugar de realizar una búsqueda CNAM en una base de datos externa.

'''NOTA IMPORTANTE PARA DID's CANADIENSES'''

La mayoría de los números DID canadienses admiten el paso a través de CNAM. Esto significa que para sus llamadas entrantes, el sistema no hará una consulta CNAM (y tampoco le cobrará) si la llamada entrante ya tiene un nombre de identificación de llamadas, incluso si el DID que recibe la llamada tiene habilitadas las consultas CNAM en su VoIP. portal de clientes ms.
----

==Configuración de su CallerID con el Portal de revendedores==

Una vez dentro del portal de revendedores, diríjase a Servicios, Mis servicios para ver las opciones de identificador de llamadas.
Las opciones disponibles son las mismas que las de su cuenta principal o subcuentas.

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

== Notas ==

Desde el 1/01/2020, usted puede realizar llamadas salientes con VoIP.ms de forma anónima. Para hacer esto, deberá configurar un identificador de llamadas y luego usar el código de función con estrella de su teléfono (* 67 por ejemplo) para ocultarlo.

:<span style="font-size:130%"> '''NOTA: El número identificador o nombre en llamadas salientes, no está garantizado hacia llamadas a celulares de Canadá, incluso usando la ruta [[Value_vs._Premium|premium]]. La razón de esto es básicamente la forma en la que los carriers de Canadá funcionan, de manera que muchas veces pasan un numero al azar desde su base de datos en lugar del original. Esto esta fuera de control, al ser un comportamiento común de los carriers canadienses. También hay problemas con el envío de un identificador de llamadas incorrecto en las llamadas salientes a números gratuitos de Bell Mobility (y sus revendedores); la pantalla muestra un número de timbre desconectado en un código de área correspondiente a la ubicación de la persona que llama en lugar del número de teléfono móvil de la persona que llama.'''

[[category:Guías]]

Custom Music on Hold

2026-07-30T19:46:45Z

Neocypher: /* Blogs */

{| class="wikitable"
|+
|-
! Article en Français !! Artículo en Español
|-
| [https://wiki.voip.ms/article/Musique_d%27attente_personnalis%C3%A9e Français] || [https://wiki.voip.ms/article/M%C3%BAsica_personalizada_en_espera Español]
|}

__TOC__

This feature allows you to upload and use your own music audio files to play while the customer is put "on hold" or during the "ring time" when you receive an incoming call.

You can upload your custom audio files from the '''Recordings''' section and then convert them into '''MOH-files'''. Thereafter you can use your MOH-files in some of your features like:

*'''Calling queues'''
*'''Ring groups'''
*'''Call hunting'''
*'''Audioconferencing'''
*'''Main account or subaccount settings'''

'''Important point to take into consideration before creating your own Music on Hold recordings'''
The Company allows the Customer to choose and broadcast the music of its choice while its clients are in a calling queue, to the express condition that Customer disposes
of the proper licenses and rights to such music.

By broadcasting music, the Customer warrants that it owns or disposes of the proper rights and licences to broadcast such music, and that it understands that the Company
does not own any rights and licences to broadcast the music, but only provides a feature for the Customer to do so.

In the event of a third-party claim directly or indirectly related to queue music provided by the Customer, Customer will defend, indemnify and hold harmless the Company
from any and against any and all claims.

== Tutorial video ==

[[Image:ThumbnailCMOH.jpg|200px|link=https://www.youtube.com/watch?v=6d5b_SQT9d8]]

== Blogs ==
[https://voip.ms/blog/how-to-improve-your-customer-experience-with-custom-music-on-hold/ How to Improve Your Customer Experience with Custom Music on Hold?]

[https://voip.ms/blog/5-examples-of-how-to-use-custom-music-on-hold-in-your-business/ 5 Examples of How to Use Custom Music on Hold in Your Business]

==Uploading Your Custom Music Files==

Go to '''DID Numbers>> Recordings''' section and click the button '''"Upload new recording"'''. From the pop-up window, you will need to fill the following:
*'''Name:''' Assign a name for your recording
*'''File:''' Click on '''"Choose file"''' button to look for the audio file you wish to upload.
*'''Upload:''' Click on '''"Upload file"''' button to upload your audio file.

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

==Converting Recordings into MOH-Files==
You will now need to convert the '''Recording''' file that you have just uploaded into a '''MOH-File'''. This can be donefrom 2 different sections.

1. From '''DID Numbers>> Recordings''' section

Once you upload your recording you can click on the headphones icon to easily convert this recording into a MOH-File you only need to assign a name for the MOH-File and click '''"Save"'''

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

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

2. From '''DID Numbers>> Custom Music on Hold''' section

From there, you can convert any recording file into a MOH-file by clicking on '''"Add a custom Music on Hold"''' button.

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

From the pop-up window, fill the '''Description''' which is the name of the MOH-file and drag and drop the recordings from the left column (Available Recordings) to the right column (Selected Recordings).

The MOH-file is by default set to "High volume" if you want to decrease the volume you can activate the switch button "Quite volume". Click at '''Save''' button.

'''Sort''' is used to either play the recordings you are adding in the order you set them, or in a random order depending on what you prefer.

[[File:CMOHnew2.png|thumb|none|600px]]

==Using Your MOH-file With a Feature==
Now your MOH-file is ready to be used along with one of the following features:

*'''"Music on hold"''' for a subaccount or the main account
Available under '''Subaccounts>> Manage Sub accounts''', clicking on the icon with a pen to modify your sub account and look for '''"Music on Hold"''' setting.

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

*'''"Music on hold"''' for a calling queue
This is available under '''DID Numbers>> Calling queues''', click on the icon with a pen to modify your current calling queue or while creating a new one and look for '''"Music on Hold"''' setting.

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

*'''"Music the caller will hear while the call is answered"''', your MOH-file will be available for a '''Ring group''' or '''Call hunting'''

For a Ring group, it is available at '''DID Numbers>> Ring groups''', click on the icon with a pen to modify your current ring group or while creating a new one and look for the setting "Music the caller will hear while the call is answered"

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

For a Call hunting, it is available at '''DID Numbers>> Call hunting''', click on the icon with a pen to modify your current call hunting or while creating a new one and look for the setting "Music the caller will hear, while is answered:"

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

For audio conferencing, Go to DID Numbers >> Audio conferencing, click on the icon with a pen to modify your current conference or create a new one. Go to the "participants and Conference settings" tab, then create or edit the profile of your choice. In the '''advanced options''', choose your file in the "MoH when empty" section. When the profile is activated, and there are no participants in the conference, your music on hold will play.

=Need a specific kind of music for your business? Look no further=

In search of music for your music on hold? Here is some website suggestions from AI's that creates music and allows you to download for your own use.
*[https://soundraw.io/create_music Soundraw.io]
*[https://www.epidemicsound.com Epicemicsound]

= Reseller Configuration =

Also if you're using the [[Reseller Basic Guide|Reseller Interface]], you can associate a MusicOnHold record with one of your reseller client.

'''Reseller Client''': Here you can select your reseller client that you want to associate this feature. You need first to create the account of your customer using the [[Reseller Basic Guide|Reseller section]] in your Customer Portal.

: [[File:ResellerClient_SelectClient_Only.png|500px]]

=Music on Hold 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:cMOH_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:cMOH_Reseller_2.png|thumb|none|700px]]

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

: [[File:cMOH_Reseller_3.png|thumb|none|700px]]

Then scroll down and find the feature "Custom Music On Hold", and enable it.

: [[File:cMOH_Reseller_4.png|thumb|none|500px]]

1) To add a music on hold for your client, or to help your client add one. Go under the [Services] at the left navigation bar, then on [Custom music on hold]

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

2) If some audio recordings are available on the section [Available recordings] from the section [recording], you can select one and drag it to the section [Selected Recordings].

If you don't see any recording available or your recording is not present, click on the [+ Add New] button to add a new recording file.

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

3) Click on the tab [Add New recording} tab, indicate the name of the recording, upload it
: [[File:cMOH_Add_CMOH_Recording.png|thumb|none|600px]]

4)Once it's uploaded, you will be redirected to the tab with all your recordings. Find the recording you wished to convert as a custom MOH. Then, click on the [Headphone] button.

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

5) Give a Name to your new custom Music on Hold.

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

Your custom music on hold has been created successfully. It will be available to be selected in the feature which a MoH can be used such as Ring Group, Call Hunting, Calling queue, SIP Account etc.

IP Phones

2026-07-30T19:45:13Z

Neocypher: /* Blog Articles */

== 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://voip.ms/blog/bring-your-own-device/ 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://voip.ms/blog/what-is-an-ip-phone/ Back to Basics - What is an IP Phone?]

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

=Most Popular IP Phones=
Not sure what to use? Here's our top used device across our network:
* [[IP_Phones#Grandstream_GXP2170 | Grandstream GXP2170]]
* [[IP_Phones#Grandstream_GXP2135_IP_Phone | Grandstream GXP2135]]
* [[IP_Phones#Cisco_SPA525G | Cisco SPA525G]]
* [[IP_Phones#Cisco_88XX_.26_68XX_series |Cisco 88XX & 68XX series]]

__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 H303-2 Wire====

[[File:X303-2_Wire_IP_Phone-VoIPms.png|300px|thumb|left|Fanvil X303-2 Wire IP Phones]]

'''Product:''' Fanvil X303-2 Wire IP Phone

'''Company:''' Fanvil

'''Overview:''' The Fanvil X303-2 Wired IP Phone is an enhanced version of the original, featuring a new 2-wire connection for both power and network supply. This innovative setup enables the reuse of existing wiring while delivering high-definition voice quality, a 2.4-inch color display, and support for up to 4 SIP accounts. The phone also supports multi-party conferencing, external EHS wireless headsets, and additional expansion options. With these advanced features, it is ideal for diverse enterprise environments, offering a seamless and superior user experience.

[[Fanvil_X303-2_Wire|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====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 V61G====

[[File:Fanvil_V61G_Prime_Business_Phone-VoIPms.png|300px|thumb|left|V61G Prime Business Phone]]

'''Product:''' Fanvil V61G

'''Company:''' Fanvil

'''Overview:''' The Fanvil V61G is a modern and intuitive IP phone designed for business professionals. It comes with a 2.4-inch (320x240) HD color screen, supports 4 SIP lines, and enables 6-party audio conferencing. With high-definition sound quality and dual Gigabit ports, the V61G delivers outstanding performance across various business settings, ensuring clear communication and an enhanced user experience.

[[Fanvil V61G|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 V62 Pro====

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

'''Product:''' Fanvil V62 Pro

'''Company:''' Fanvil

'''Overview:''' The Fanvil V62 Pro is a versatile business IP phone designed for efficiency and productivity. It features an HD color display and a Bluetooth handset with programmable keys to optimize daily workflows. The Bluetooth handset is interchangeable across all V Series Pro models and easy to locate if misplaced, providing exceptional convenience in office environments. Supporting 12 SIP lines, 6-way conferencing, and dual Gigabit ports, the V62 Pro delivers a high-performance communication experience tailored for modern business professionals.

[[Fanvil_V62_Pro|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 V64====

[[File:Fanvil-v64-VoIPms.png|300px|thumb|left|Fanvil V64 Prime Business Phone]]

'''Product:''' Fanvil V64

'''Company:''' Fanvil

'''Overview:''' The V64 is not just a highly efficient business phone—it’s a refined piece of craftsmanship designed to deliver an intelligent and seamless communication experience for enterprises. Featuring a vibrant color LCD display, built-in Bluetooth 4.2, and dual-band 2.4G/5G Wi-Fi, the V64 perfectly blends elegant design with powerful performance.

[[Fanvil V64|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 V66 Pro====

[[File:Fanvil_V66_Pro_Prime_Business_Phone-VoIPms.png|300px|thumb|left|Fanvil V66 Pro Prime Business Phone]]

'''Product:''' Fanvil V66 Pro

'''Company:''' Fanvil

'''Overview:''' The Fanvil V66 Pro Prime Business Phone combines advanced audio and video communication in a single device. It comes with a programmable Bluetooth wireless handset that supports automatic pairing and location tracking. Built with antimicrobial materials, the V66 Pro features a 7-inch rotatable IPS touch screen, integrated dual-band Wi-Fi (2.4G/5G, Wi-Fi 6), and Bluetooth 5.4 connectivity—providing exceptional performance and seamless communication for premium business users.

[[Fanvil V66 Pro|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 W610W====

[[File:Linkvil_W610W_by_Fanvil.jpg|300px|thumb|left|Linkvil W611W by Fanvil]]

'''Product:''' Linkvil W6110W by Fanvil

'''Company:''' Fanvil

'''Overview:''' The Fanvil W610W is a durable Wi-Fi business handset featuring a 2.0″ color display, dual-band Wi-Fi (2.4 GHz/5 GHz) with seamless roaming, and Bluetooth 5.0 for wireless headsets. With 16 DSS keys, a PTT button, and drop resistance up to 1.8 m, it delivers secure, reliable, and flexible communication for modern mobile workplaces.

[[Linkvil_W610W_by_Fanvil | See Configuration Details]]
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====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]]

<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:''' 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]]
<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>

==== Vogtec IP Phone T370, T12, T2 ====

[[File:VogtecIPphone.png|1000px|thumb|left|Vogtec T12]]

'''Product:''' T370, T12 & T2

'''Company:''' Vogtec

'''Overview:''' Our VoIP solutions are available in two lines: Linux-based multi-feature models, which include an e-ink BLF panel, WiFi, and a digital answering machine; and Android-based multi-medium models with LTE & WiFi wireless for the mobility industries.

[[Vogtec_IP_Phone_T370,_T12,_T2|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>

DigitalReceptionist IVR

2026-07-30T19:43:29Z

Neocypher: /* Blog Articles */

{| class="wikitable"
|+
|-
! Article en Français !! Artículo en Español
|-
| [https://wiki.voip.ms/article/R%C3%A9ceptionniste_virtuelle_IVR Français] ||
[https://wiki.voip.ms/article/Recepcionista_Digital Español]
|}

Interactive Voice Response (IVR), this option can be used to present a recording to the people calling to your DID number and also give them the chance to enter the extension of the person or department they want to reach. For example, you could create an IVR and point one of your DID numbers to it, and when the IVR answers, caller will hear "Thank you for calling XYZ Inc, for Sales press 1, for Service press 2", etc.

__TOC__

== '''Blog Articles''' ==

[https://voip.ms/blog/how-to-stop-spam-calls-voip/ Harnessing VoIP.ms to Eliminate Spam Calls]

[https://voip.ms/blog/benefits-of-ivr-auto-attendants/ Benefits of Auto Attendants (IVR)]

[https://voip.ms/blog/the-dos-and-donts-of-ivr-scripts/ The Do's and Don'ts of IVR Scripts]

[https://voip.ms/blog/tips-for-creating-perfect-welcome-greeting/ Tips for Creating Perfect Welcome Greeting]

[https://voip.ms/blog/modernize-your-digital-receptionist/ Modernize Your Digital Receptionist]

== '''Tutorial Video''' ==

[[Image:IVRThumbnail.png|200px|link=https://youtu.be/KHT9VkWv-1Q]]

== '''Important Points to consider before creating Digital Receptionist (IVR)''' ==

There are some points you may want to consider before you start creating your IVR menus, like:

# Record your audio prompts and upload them using the '''[[Recordings]]''' portion of the customer portal.
# Create any destinations that don't currently exist and that you plan on using with your IVR destinations. '''([[Calling Queues]], [[Ring Groups]], etc)'''.
# Plan your call flow in advance. You should create all the destinations (child) of your IVR menus before creating the parent IVR's that lead to them.

== '''How to Create a Digital Receptionist (IVR)''' ==

To do this, you first need to enter to the customer portal, then select DID numbers menu, option "Digital Receptionist (IVR)", there you should see something like this:

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

Being in that page, you would need to click on the "Create a new IVR" button, then you should be able to see this window:

[[File:IVR2.JPG|thumb|none|600px]]

In that section you need to enter the following information:

- '''Name''': you need to set a name that describes this IVR entry

- '''Recording''': here you need to select the recording that people will hear when IVR answers the call.

- '''Timeout''': Maximum time(in seconds) allowed to enter a digit( extension, option ) once the recording is finished.
If the IVR includes routing options with more than 1 digit, the timeout will apply when the user press a 1 digit option, to allow them to dial an option with more digits.
You can use the pound key '#' to route the call instantly and bypass the timeout.

- '''Language''': here you can select among English, French or Spanish for system messages.

- '''Voicemail''': here you can select between DID voicemail(Mailbox assigned to the DID number routed to IVR) or Account Voicemail(Mailbox of the account or sub account to where points the extension)

*'''NOTE.''' If you have '''failovers''' on your DID, for '''"NO ANSWER"''' status, this will override the subaccount mailboxes, and the call will go to the Failover routing, instead of the Mailbox on the Ext. Subaccount.

Once you have set that information, now you can create the extensions of your IVR, basically you will see this

[[File:IVR3.JPG|thumb|none|600px]]

In that section you have to enter the following information:

- '''Digit(s)''': here you need to enter the "number" that people will need to press in order to reach this extension. You can enter up to 12 digits in this field.

- '''Destination''' : this is where the extension will point, as you can see in the image above, options will be shown as available if you have created at least one "entry" of each of them.

IMPORTANT NOTE: When creating an IVR menu option, apart from the standard options of 0-9 and *, you can also use 'i' and 't' destinations.
'i' is used when the caller pushes an invalid button, and 't' is used when there is no response. If those options aren't supplied, the
default 't' is to replay the menu three times and then hang up, and the default 'i' is to say 'Invalid option, please try again' and replay
the menu. After three invalid attempts, the line is hung up. Please note that the # Digit is used to confirm user's choice by bypassing the
Digit Time out option and can not be used as a choice.

As you can see, in this section you have the following options:

- '''Remove Last Choice''': use this button in case you want to delete the last extension you have created on your IVR.

- '''Add Another Choice''': By default you will see only 3 spaces to create extensions, however if you need more you can use this button.

- '''Save IVR''': use this button to save your IVR.

Once you save the IVR you have created you can see this on customer portal -> DID numbers -> Digital Receptionist(IVR):

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

As you can see now you have these options:

- '''Duplicate''': in case you want to create another IVR entry with the same options that other you have previously created.

- '''Edit''': use this option in case you want to modify an existing IVR you have created.

- '''Delete''': use this command to delete an IVR entry from your account.

== Reseller Configuration ==

Also if you're using the [[Reseller Basic Guide|Reseller Interface]], you can associate each IVR feature with one of your reseller client.

'''Reseller Client''': Here you can select your reseller client that you want to associate this feature. You need first to create the account of your customer using the [[Reseller Basic Guide|Reseller section]] in your Customer Portal.

: [[File:ResellerClient_SelectClient_Only.png|500px]]

== IVR 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:IVR_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:IVR_Reseller_2.png|thumb|none|700px]]

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

: [[File:IVR_Reseller_3.png|thumb|none|700px]]

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

: [[File:IVR_Reseller_4.png|thumb|none|500px]]

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

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

2) Once on the page, click on '''[Add new IVR]''' tab.
You will need to enter some basic information, such as the '''IVR name''', the audio '''Recording''' to be played, the voicemail destination to Account Voicemail etc.

To add DIGITS (choices) to the IVR, click on the '''[Add another routing option]''' blue button. A new tab '''[Routing]''' will appear. Click on it, then enter the DIGITS and choose the routing.

[[File:IVR_Add_2.png|thumb|none|600px]]

3) Click '''[Save IVR]'''

Your IVR has been created successfully... You can select this IVR in your DID Routing, Failover options, IVR, Time Conditions, etc.

[[category:guides]]

Ring Groups

2026-07-30T19:39:28Z

Neocypher: /* Blog Article */

{| class="wikitable"
|+
|-
! Article en Français !! Artículo en Español
|-
| [https://wiki.voip.ms/article/Groupe_de_Sonnerie Français] ||
[https://wiki.voip.ms/article/Grupo_de_timbre_/_Ring_Groups Español]
|}

__TOC__

The Ring Group feature allows you to have incoming calls (to a single DID phone number) be redirected to different destinations (i.e. multiple devices) that are included in your Ring Group, where a member of the group is able to answer. When you receive a call to a DID routed to a Ring Group, all members of that group will ring at the same time until one of them answers the call. You can add various types of members to a ring group:
Main Account,
[[Sub Accounts]],
[[SIP URI]]'s,
[[Call Forwarding]].

You can also select which voicemail should be used by the system in case none of the members answer the call. The limit of members in a Ring Group is 12: Up to 8 (SIP, IAX2, or SIP URI) members and up to 4 call forward entries per each Ring Group.

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

== Blog Article ==

[https://voip.ms/blog/the-benefits-of-ring-groups/ The Benefits of Ring Groups]

== Tutorial Video ==

[[Image:RingGroupThumbnail.png|200px|link=https://youtu.be/I__zQCqpG20]]

== Creating a Ring Group Entry ==
From your main portal please refer to DID Numbers -> Ring Group. You will have to click on the link that reads ''Click here to create a new ring group''. Then another screen will prompt and you will have to enter the following information:

'''Description:''' This can be used as a note or description to easily identify your ring groups.

'''Caller Announcement:''' This setting is optional. You can select the default announcement or one of your own recordings to be played to the caller before sending the call to the Ring Group. Please note if a Recording is selected, the call will start being charged immediately even before a member answers.

'''Music the caller will hear while the call is answered::''' This setting is also optional. The caller will hear the selected music while the members of the Ring Group are dialed. If set to 'No music', the caller will hear the normal ringing. Please note if the Music on Hold is set to "Yes", the call will start being charged immediately even before a member answers.<br/>

The following options for music are available:<br/>

{| class="wikitable" border: none;"
! colspan="3" | Music on Hold
|-
! scope="col" | Test Code
! scope="col" | Name
! scope="col" | Description
|-
| '''***89''' || Away in the Tropics || From Hawaii to the Caribbean, these tracks deliver sounds of ukulele, steel drums, and steel guitars.
|-
| '''***90''' || Coffee and Sunrise || Uplifting without being perky, and positive without being too smiley.
|-
| '''***91''' || Coffee Shop Acoustic || Soothing, acoustic guitar tracks makes for a relaxed feel and sets a great atmosphere.
|-
| '''***92''' || Easy Listening || Smooth, casual tunes.
|-
| '''***93''' || Guitar Alchemy || Clever harmonics and progressive chord sequences to create a joyful and warming musical experience.
|-
| '''***94''' || Happy Endings || Uplifting, commercial style. Guitar, drums, ukulele, harmonica and bells.
|-
| '''***95''' || Light and Casual || Soothing and peaceful songs with a light, positive feeling.
|-
| '''***96''' || Orchestral Moods || Emotional and dramatic tales spun by violins, pianos and full orchestras.
|-
| '''***97''' || Piano Mix || Smooth Piano
|-
| '''***98''' || Rock me Easy || Feel-good music to create a relaxing atmosphere.
|-
| '''***99''' || Spa Sounds || Soft, slow and serene instrumentals.
|-
| '''***100''' || No music, Intermittent bleep || Silent, but with a subtle intermittent bleep sound to let your callers know that they are still on the line while they hold.
|}

'''Language:''' Here you can select the language used for the "Default Announcement" recording selected in the Caller Announcement section presented to the Caller and the "Press One to take the Call" recording presented to the Callee. You have the choice between English, French, and Spanish.

'''Members:''' Here you can select the members of the ring group. Your account currently has a Ring Group restriction limit of up to 8 SIP, IAX2 or SIP URI members and/or up to 4 Call Forwarding entries, for a total of 12 possible members per Ring Group entry. You can set individual Ring Times for every member of your Ring Group as well as configure every member for Answer Confirmation by making them press the key "1" to take the call.

'''Voicemail:''' Optionally, you can select a Voicemail for this ring group that will override the default DID Voicemail.

Finally, just hit the '''Create''' button and you will be done creating your entry. Please keep your Ring Groups up to date by removing inactive members.

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

== Routing your DID to your Ring Group ==

After you have created your Ring Group entry, you will need to route any of your DIDs to your Ring Group entry from your main portal.

Please refer to DID Numbers -> [[Manage DID]] -> [https://www.voip.ms/m/managedid.php | Edit DID] -> Routing -> Ring Group. Also under the same menu screen, you can select your Ring Group entry for the Additional Failover Options.

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

Remember that you need to scroll down and apply the settings to have them changed.

Make sure the Point of Presence (POP) of the DID is the same server where the SIP/IAX members of the ring group are registered.

== Reseller Configuration ==

Also if you're using the [[Reseller Basic Guide|Reseller Interface]], you can associate each Ring Group feature with one of your reseller client.

'''Reseller Client''': Here you can select your reseller client that you want to associate this feature. You need first to create the account of your customer using the [[Reseller Basic Guide|Reseller section]] in your Customer Portal.

: [[File:ResellerClient_SelectClient_Only.png|500px]]

== Ring Group 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:RingGroup_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:RingGroup_Reseller_2.png|thumb|none|700px]]

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

: [[File:RingGroup_Reseller_3.png|thumb|none|700px]]

Then scroll down and find the feature '''"Ring Group"''', and enable it.

: [[File:RingGroup_Reseller_4.png|thumb|none|500px]]

1) To add a new Ring Group for your client, or to help your client adding one. Go under the '''[Services]''' at the left navigation bar, then on '''[Ring Group]'''.

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

2) Once on the page, click on '''[Add new Ring Group]''' tab.
You will need to enter some basic information, such as the '''Group Name''', Music on Hold, Voicemail when no answers, etc.

Select all the location you want to make ring at the same time.

[[File:RingGroup_Add_2.png|thumb|none|600px]]

3) Click '''[Save Ring Group]'''

Your Ring Group has been created successfully... You can select this Ring Group in your DID Routing, Failover options, IVR, Time Conditions, etc.

[[category:guides]]

Order a DID Number

2026-07-30T19:38:22Z

Neocypher: /* Blog article */

{| class="wikitable"
|+
|-
! Article en Français !! Artículo en Español
|-
| [https://wiki.voip.ms/article/Commander_un_num%C3%A9ro_DID Français] ||
[https://wiki.voip.ms/article/Comprar_un_n%C3%BAmero_DID Español]
|}

DID (Direct Inward Dial) numbers are used to receive incoming calls using a Voice over IP (VoIP) service. These numbers may be called from any regular phone, including landlines and cell phones. Whether you are a local, national, or global business, you can use your DID numbers.

DID numbers are only required for the purpose of receiving calls. To make outgoing calls, you only need to configure your VoIP.ms Credentials on a [[devices|device]] or [[softphones|softphone]] and maintain a positive balance on your account.

== Blog article ==

[https://voip.ms/blog/benefits-of-having-a-toll-free-number-for-your-business/ Benefits of Having a Toll-free Number for Your Business]

=Order a Number=

You can order a DID Number at your Customer Portal, in DID Numbers >> Order DID(s).<br>

[[File:Order.JPG|thumb|none|750px|Order DID]]
<br clear="all" />

There you can find the following options:

'''Local Numbers:''' You can choose numbers from USA and Canada.

'''International Numbers.''' These numbers belongs to a particular country and particular city.

'''Toll Free Numbers:''' You can choose [[toll-free number]]s from US, Canada, US+Canada or international numbers. Unlike a local (domestic or international) number, a toll-free or freephone number usually only works within one country or one country code but is free to call from a land line.

'''National Numbers (Non Geographical):''' These numbers belongs to a particular country but not to a particular city and they can only be reached with a land line from that country.

'''Virtual SIP Numbers:''' These numbers can only be reached via "SIP URI" over internet.

'''DIDs with Virtual PRI (Per Channel):''' Available on request - Please send an e-mail to sales@voip.ms for a quote.

[[File:Order2.JPG|thumb|none|750px]]

==Choose a DID number==
Once you have selected the desired type of number, you may select the country/state for your DID order (when it applies) and then select a number from the list if they are available, and you will be able to see the incoming plan and the setup fee before buying it.

You can also refine a DID Number search by using the "Contains", "Starts with" or "Ends with" options on the Search Field of that same screen. The available numbers fulfilling this criteria should be displayed if there is any.

[[File:Order3.JPG|thumb|none|750px]]

== Choose an incoming plan for your DID ==
You could choose between 2 different plans for incoming calls:

* The Per Minute Plan, which charges a monthly fee and a fixed fee for every minute the call lasts.
* And the Flat Rate Plan, which is intended for Residential Usage only and includes 3500 incoming Minutes per month.

[[File:Order4.JPG|thumb|none|750px]]

== Finish your DID order ==

After this you just have to select the Main Routing, Point of Presence, Failover measures and other optional features for your new number and click on the button "Order DID" to finish this process.

Once you successfully purchased a DID Number an e-mail will be sent to the e-mail associated with your VoIP.ms Account confirming this action.

[[File:Order5.JPG|thumb|none|750px]]

==Start a DID Back order==
In case there is no number available in stock, you can start a backorder process and a random number can be delivered to your account.

[[File:backorder.JPG|thumb|none|750px]]

After this you just have to choose between the different plans for incoming calls and select the default settings for your new number, just like it is explained on the section above.

[[File:quantity.JPG|thumb|none|750px]]

== Ordering an International DID number ==

When ordering an International DID Number, you will be able to select amongst different Countries.

[[File:OrderINT-1.png|thumb|none|750px]]

Once you select a Country, a new page will prompt, showing the Setup Fee, Monthly Fee, Inbound Calls amount and Channels (incoming) for that specific country.

On the section below will be shown the available rate centers, the Area code and how many number exists for that country.

There is a case in blank where you can enter the amount of DID numbers from that rate center that you wish to purchase.

'''The delivery of numbers in stock takes in average 1 (one) business day but may vary according to the location.'''

Once you click on the 'Order' button the following screen will follow.

[[File:OrderINT-3.png|thumb|none|750px]]

This is confirmation Page of the DID Purchase. Here you will be able to Preset your DID number before we add it to your account.
You will be able to change the following options:<br>
*'''Caller ID Name lookup.'''
*'''DID Point of Presence.'''
*'''Routing.'''
*'''Additional Failover Options.'''

Once you finish with the preset, click on the ''' 'Click here to confirm your order' ''' button.

[[File:orderINT-2.png|thumb|none|750px]]

'''The delivery of numbers in stock takes in average 1 (one) business day but may vary according to the location.'''

==Ordering a Toll Free Number==

We offer 4 options when ordering a Toll Free number:

*'''United States.''' (Check availability and order toll-free DIDs for American callers, Canada unlockable)
*'''Canada+United States.''' (Check availability and order toll-free DIDs for Canadian and American callers)
*'''Vanity Numbers.''' (Check availability for custom toll-free numbers in the SMS/800 database)
*'''International.''' (Check availability and order international toll-free numbers)

[[File:orderTFN-1.png|thumb|none|750px]]

On the United States section, the first thing you will see the filter to search Toll Free numbers.
You can filter them by '''800 | 833 | 844 | 855 | 866 | 877 | 888 | All''', or you can even enter in the '''blank field''' an specific number you wish to search.
*The Canada + United States section '''does not include''' this filter. Everything else in this section is exactly the '''same''' as in the 'United States' section.

[[File:orderTFN-2.png|thumb|none|750px]]

Once you have selected the number you wish to purchase, scroll down and you will be able to see the '''Monthly Fee, the Per minute Cost, Setup fee and Billing increment''' for that specific number.

On the same section you can preset your Toll Free number by assigning a '''Route, Point of Presence and Additional Failover Options.'''

From this screen, you should ''''Click to confirm your order'''' button to confirm your order and the number will be added to your account.<br><br>

[[File:orderTFN-3.png|thumb|none|750px]]

== Ordering a Vanity Number ==

Vanity numbers are '''customizable''' Toll Free Numbers, for example: 888-421-HELLO. This will help your Callers or Customer to '''easily remember''' your number. A "custom" toll-free number will normally take longer to configure initially than a random number already in the voip.ms inventory but can serve as a valuable marketing tool due to its memorability.

[[File:orderTFN-4.png|thumb|none|750px]]

Once you entered a valid and available number, you will need to select a carrier.

*'''American Carrier:''' Your toll-free number will work for US callers only at a rate of 1.9 cents per minute. This can be changed later to USA/Canada (See below). Monthly fee is $0.99 starting at delivery date. There's a one-time setup fee of $15 to order each custom DID. Approximate delivery time: 1-5 business days.<br>

*'''American Carrier, American and Canadian Callers allowed:''' Your toll-free number will work for both US and Canadian callers at a rate of 1.9 cents per minute for US callers and 8 cents per minute for Canadian Callers. Monthly fee is $0.99 per month starting at delivery date. There's a one-time setup fee of $15 to order each custom DID. Approximate delivery time: 1-5 business days.<br>

*'''Canadian Carrier:''' Your toll-free numbers will work for both American and Canadian Callers at a rate of 2.7 cents per minute for both countries. Monthly fee is $1.25 per month starting at delivery date. There's a one-time setup fee of $30 to order each custom DID. Approximate delivery time: 1-3 business days.

Now you only need to click the ''' 'Order Now' ''' button to proceed.<br><br>

'''Please note that vanity orders may take up to 2 hours to be completely activated.'''

[[File:VanityTFNorder.png|thumb|none|750px]]

==Ordering an International Toll Free Number==

This process is basically the same as Ordering an International DID Number process. '''Please refer to that [[Order_a_DID_Number#Ordering_an_International_DID_number | section ]]'''

==Ordering Virtual Numbers==

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

All virtual numbers consist of the following digits: '''11 + Accountcode + 3 digits of your choice for a total of 11 digits'''. The final URI will be that number followed by the @ sign at one of our server. If you intend to send the calls to a phone or adapter, you'll need to point it to the proper server. '''Example SIP URI: 11139808123@houston.voip.ms'''
<br><br>

[[File:orderVN-1.png|thumb|none|750px]]

== Order DIDS 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:OrderDID_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:OrderDID_Reseller_2.png|thumb|none|700px]]

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

: [[File:OrderDID_Reseller_3.png|thumb|none|700px]]

Then scroll down and find the feature '''"Order DIDS"''', and enable it.

: [[File:OrderDID_Reseller_4.png|thumb|none|500px]]

[[File:OrderDID_Add.png|thumb|none|300px]]

[[File:OrderDID_Add_2.png|thumb|none|500px]]

[[File:OrderDID_Add_3.png|thumb|none|500px]]

[[File:OrderDID_Add_4.png|thumb|none|500px]]

The new DID ordered will be available in the '''My service''' section under '''Service'''.

[[category:guides]]

Call Encryption - TLS/SRTP

2026-07-30T19:17:16Z

Neocypher: /* Blog Articles */

{| class="wikitable"
|+
|-
! Article en Français !! Artículo en Español
|-
| [https://wiki.voip.ms/article/Cryptage_des_appels_-_TLS/SRTP Français] ||
[https://wiki.voip.ms/article/Encriptado_de_llamadas-_TLS/SRTP Español]
|}

This feature allows you to encrypt the communication between your device and our server, by using the SIP-TLS ''(Transport Layer Security)'' and SRTP ''(Secure Real-Time Transport Protocol)'' protocol.

This adds a security layer when the packets are being transmitted between you and our server, it encapsulates and encrypt the transmission. In other words, when your device is configured with this encryption method, your device asks to our server a dedicated certificate to establish a trust and fully secure communication from each part.

This is ideal if you are using a softphone on a public network. ''(We strongly recommend you to use this function in this case.)''

Once encrypted calls are enabled for your account or sub account, the SIP-TLS and SRTP must be used.
Your account or sub account will no longer be able to use regular SIP communication method.

Call Encryption - TLS/SRTP is currently only available for SIP protocol. IAX is not supported.

__TOC__

== Blog Articles ==

[https://wiki.voip.ms/article/Understanding_VoIP_Encryption_%E2%80%93_SIP-TLS_%26_SRTP Understanding VoIP Encryption – SIP-TLS & SRTP]

[https://voip.ms/blog/ensuring-the-security-of-your-calls/ Ensuring the Security of Your Calls]

== Activate This Option on Your Main Account ==

1) To active this feature go to your Customer portal home and click on “'''Main Menu'''” > “'''<font color="#FF7C21">Account Settings</font>'''”

:[[File:TLS-SRTP-Steps0001.png|border]]

2) Once you are in the “'''Account Settings'''” section, navigate through the submenu and go to “'''<font color="#FF7C21">Advanced</font>'''” and find the field “'''<font color="red">Encrypted SIP Traffic</font>'''”, set to '''Yes''' and press '''Apply'''

:[[File:TLS-SRTP-Steps0002.png|border]]

'''Note:''' If enabled, all the SIP traffic will be encrypted for the main account.
''' ''' Please note that if encrypted calls are enabled then you need to configure your device to make and receive encrypted calls.

== Activate This Option on Your Sub Account ==

1) You may also activate this feature on a sub account. You need to navigate through the navigation bar and select “'''Sub Accounts'''” > “'''<font color="#FF7C21">Manage Sub accounts</font>'''”.

:''If you don't have a sub account yet, you can create one by clicking on the tab “'''Create Sub Account'''”''

:[[File:TLS-SRTP-Steps0003.png|border]]

2) In the column “'''<font color="#FF7C21">Actions</font>'''” click on the edit button [[File:edit_icon.png]], for the row of the sub account you need to activate this feature.

:[[File:TLS-SRTP-Steps0004.png|border]]

3) Once you are in the Edit screen, find the “'''<font color="#FF7C21">Advanced Options</font>'''” and “'''Click here to display'''”. Then set “'''<font color="red">Encrypted SIP Traffic</font>'''” to '''Yes''', and press '''Update Account'''

:[[File:TLS-SRTP-Steps0005.png|border]]

'''Note:''' If enabled, all SIP traffic calls will be encrypted for this sub account.
''' ''' Please note that if encrypted calls are enabled then you need to configure your device to make and receive encrypted calls.

== Configuration on SIP Client ==

Once you have activated the feature on your main account/sub account, you need to configure your SIP client.

On some devices, you will have to configure some settings to '''enable''' the SIP-TLS communication method.
In your settings you must select '''TLS''' as your transport protocol and activate ''media encryption or SRTP<sup>*</sup>'' as '''Mandatory<sup>*</sup>'''. Without mandatory ''media encryption<sup>*</sup>'', this would result in a call rejection with the SIP error 488.

'''''Some technical considerations that you need to know for using this feature.'''''
Please take note, when using encrypted calls with a server, you must always use the server name with a '''number''' at the end.
For example, you must use chicago'''1'''.voip.ms instead of chicago.voip.ms.

This also applies to cities with only one server.
For example, you select the POP server london.voip.ms in the portal, but write london'''1'''.voip.ms when you configure your [[Devices]], [[Softphones]] or [[PBXs]].

When you are using the TLS protocol, it is implied to be using TCP as packet transport. The reason is using TLS over UDP is not supported by the TLS specification.
The TLS by TCP will use the port '''5061''' instead of 5060. We also have an alternative port such as '''5081''' and '''42873'''

''*The configuration and the terminology may vary from each device/PBX.''

''*Take note; some SIP clients do not support the call encryption, in some cases is a paid feature, or is available only in the paid version.''

Some old systems/devices require to configure the certificate manually. In case is needed, please contact technical support via live chat or email to support@voip.ms (from the email address associated to your account), requesting the client certificate and advising the SIP server.

'''NOTE''': The certificate expires every 90 days, so requesting a new one every period will be necessary to keep call encryption working under these circumstances

To know more about that, please refer to the device configuration page or your device manual.

== TLS/SRTP Registration Status Validation ==

When your device is fully registered by using SIP-TLS protocol, you will be able to see the registration status in the portal Home page “'''Main menu'''” > "'''Portal Home'''" for each account/sub account registered or in “'''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|border]]

:[[File:TLS-SRTP-Steps0007.png|border]]

== Notes ==

*It has been confirmed that ATA Obi100 does not support encrypted calls.

IP Phones

2026-07-30T19:11:58Z

Neocypher: /* Blog Articles */

== 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://voip.ms/blog/bring-your-own-device/ 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.'''

=Most Popular IP Phones=
Not sure what to use? Here's our top used device across our network:
* [[IP_Phones#Grandstream_GXP2170 | Grandstream GXP2170]]
* [[IP_Phones#Grandstream_GXP2135_IP_Phone | Grandstream GXP2135]]
* [[IP_Phones#Cisco_SPA525G | Cisco SPA525G]]
* [[IP_Phones#Cisco_88XX_.26_68XX_series |Cisco 88XX & 68XX series]]

__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 H303-2 Wire====

[[File:X303-2_Wire_IP_Phone-VoIPms.png|300px|thumb|left|Fanvil X303-2 Wire IP Phones]]

'''Product:''' Fanvil X303-2 Wire IP Phone

'''Company:''' Fanvil

'''Overview:''' The Fanvil X303-2 Wired IP Phone is an enhanced version of the original, featuring a new 2-wire connection for both power and network supply. This innovative setup enables the reuse of existing wiring while delivering high-definition voice quality, a 2.4-inch color display, and support for up to 4 SIP accounts. The phone also supports multi-party conferencing, external EHS wireless headsets, and additional expansion options. With these advanced features, it is ideal for diverse enterprise environments, offering a seamless and superior user experience.

[[Fanvil_X303-2_Wire|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====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 V61G====

[[File:Fanvil_V61G_Prime_Business_Phone-VoIPms.png|300px|thumb|left|V61G Prime Business Phone]]

'''Product:''' Fanvil V61G

'''Company:''' Fanvil

'''Overview:''' The Fanvil V61G is a modern and intuitive IP phone designed for business professionals. It comes with a 2.4-inch (320x240) HD color screen, supports 4 SIP lines, and enables 6-party audio conferencing. With high-definition sound quality and dual Gigabit ports, the V61G delivers outstanding performance across various business settings, ensuring clear communication and an enhanced user experience.

[[Fanvil V61G|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]]
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====Fanvil V62 Pro====

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

'''Product:''' Fanvil V62 Pro

'''Company:''' Fanvil

'''Overview:''' The Fanvil V62 Pro is a versatile business IP phone designed for efficiency and productivity. It features an HD color display and a Bluetooth handset with programmable keys to optimize daily workflows. The Bluetooth handset is interchangeable across all V Series Pro models and easy to locate if misplaced, providing exceptional convenience in office environments. Supporting 12 SIP lines, 6-way conferencing, and dual Gigabit ports, the V62 Pro delivers a high-performance communication experience tailored for modern business professionals.

[[Fanvil_V62_Pro|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 V64====

[[File:Fanvil-v64-VoIPms.png|300px|thumb|left|Fanvil V64 Prime Business Phone]]

'''Product:''' Fanvil V64

'''Company:''' Fanvil

'''Overview:''' The V64 is not just a highly efficient business phone—it’s a refined piece of craftsmanship designed to deliver an intelligent and seamless communication experience for enterprises. Featuring a vibrant color LCD display, built-in Bluetooth 4.2, and dual-band 2.4G/5G Wi-Fi, the V64 perfectly blends elegant design with powerful performance.

[[Fanvil V64|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 V66 Pro====

[[File:Fanvil_V66_Pro_Prime_Business_Phone-VoIPms.png|300px|thumb|left|Fanvil V66 Pro Prime Business Phone]]

'''Product:''' Fanvil V66 Pro

'''Company:''' Fanvil

'''Overview:''' The Fanvil V66 Pro Prime Business Phone combines advanced audio and video communication in a single device. It comes with a programmable Bluetooth wireless handset that supports automatic pairing and location tracking. Built with antimicrobial materials, the V66 Pro features a 7-inch rotatable IPS touch screen, integrated dual-band Wi-Fi (2.4G/5G, Wi-Fi 6), and Bluetooth 5.4 connectivity—providing exceptional performance and seamless communication for premium business users.

[[Fanvil V66 Pro|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 W610W====

[[File:Linkvil_W610W_by_Fanvil.jpg|300px|thumb|left|Linkvil W611W by Fanvil]]

'''Product:''' Linkvil W6110W by Fanvil

'''Company:''' Fanvil

'''Overview:''' The Fanvil W610W is a durable Wi-Fi business handset featuring a 2.0″ color display, dual-band Wi-Fi (2.4 GHz/5 GHz) with seamless roaming, and Bluetooth 5.0 for wireless headsets. With 16 DSS keys, a PTT button, and drop resistance up to 1.8 m, it delivers secure, reliable, and flexible communication for modern mobile workplaces.

[[Linkvil_W610W_by_Fanvil | See Configuration Details]]
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====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.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]]
<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>

==== Vogtec IP Phone T370, T12, T2 ====

[[File:VogtecIPphone.png|1000px|thumb|left|Vogtec T12]]

'''Product:''' T370, T12 & T2

'''Company:''' Vogtec

'''Overview:''' Our VoIP solutions are available in two lines: Linux-based multi-feature models, which include an e-ink BLF panel, WiFi, and a digital answering machine; and Android-based multi-medium models with LTE & WiFi wireless for the mobility industries.

[[Vogtec_IP_Phone_T370,_T12,_T2|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>

Virtual Fax

2026-07-30T19:10:29Z

Neocypher: /* Blog Article */

[[File:Faxhomelogo.png|center]]

{| class="wikitable"
|+
|-
! Article en Français !! Artículo en Español
|-
| [https://wiki.voip.ms/article/T%C3%A9l%C3%A9copieur_virtuel Français] ||
[https://wiki.voip.ms/article/Fax_Virtual Español]
|}

The Virtual Fax feature is used to send and receive Faxes (facsimile) with the VoIP.ms service using a DID number specifically dedicated to Fax. You may obtain such a number from your Customer Portal in the Fax Numbers section under the ''Order DID(s)'' of the ''DID Numbers'' menu.
Regular voice DID numbers are not compatible with the Virtual Fax feature.

__TOC__

== Blog Article ==
[https://voip.ms/blog/fax-over-ip/ Fax over IP (FoIP) using T.38 Protocol]

== Tutorial Video ==

[[Image:VfaxThumbnail.png|200px|link=https://youtu.be/k0WDphvlxbk]]

== Important information about the Virtual Fax Service ==

* '''The Virtual Fax Service is only available for U.S. and Canadian DID Numbers specifically acquired from the Fax Numbers ''Order DID'' section'''
* '''It is also possible to port your VoIP.ms Voice DID Numbers and Numbers from other providers into our Virtual Fax service. For numbers from other providers, you can find this option under the ''DID Portability'' section of the Customer Portal. For VoIP.ms numbers, you can request an internal port by sending an email to our LNP department at ports@voip.ms. The porting fee is $15 per number for both options.'''
* '''The Service can currently only be used to send Faxes to Canadian and U.S. numbers. We also cannot guarantee that international will be properly received.'''
* '''Virtual Fax DID Numbers cannot receive regular voice calls nor SMS, but only faxes.'''

== Costs and Rates ==

Setup Fee: $0.00 (Currently Free)

Monthly Fee: $1.99

Per Minute Fee: $0.0290 (2.9 Cents)

[[Calls Cost|Billing Increment]] : 6 seconds

If you want to calculate how much this service could cost per page faxed, converted into per minute charges, you'll realize that this depends a lot on the destination fax speed and the content faxed (how much content is on the page compared to the blank space). Unfortunately, there are no definite ways to calculate precisely these costs . If you are sending just text, a page can take from 30s to 1 min. If you are faxing multiple pages documents, this will go faster because you don't have to handshake or negotiate with the far side's fax machine for each page. This can result in 1.5 cents to 3 cents on a 1-2 page fax. Since we also increase the charge to every 6 seconds, you are saving even more money since we do not charge you a full min for partial minute usages. You can usually get around 1-2 pages faxed per minute at 2.9 cents and $2 a month which is much less than most other Electronic Fax services currently available from other providers.

== Current Limitations ==

*Each DID Number can only send 100 messages per day. This limit can be raised upon request and verification.

*The files sent per message cannot exceed 25 MB

*The file cannot contain more than 200 pages

*Only one document per fax message can be sent

== Virtual Fax DID Number ==

Virtual Fax works specifically with Fax Numbers only acquired from the VoIP.ms Customer Portal or numbers ported in as Fax enabled. There are '''local US and Canadian numbers''' available for order. You can order a Fax DID Number from your portal >> DID Numbers >> Order DID >> Fax Numbers. You can select the desired region and a random number from the chosen area code will be assigned to you.

[[File:FaxorderDID2.jpg|700px|]]

You can also port in a number you currently own with another provider. This process can be started from the Customer Portal at DID Numbers >> DID Portability >> Porting Fax Numbers

[[File:Faxport2.png|700px]]

== Send a Fax ==

To start using the service you will need to head to your Customer Portal >> DID Numbers >> Virtual Fax. From the Home Page you can select ‘Send Fax’. There you will see:
*Fax Number or Contact Name: This is where you will put the destination number. You can start typing a name or a number from your Phone book and it will become available.
*From Name: Here you will put the name to send in the Fax header.
*From Number: Select the Fax DID number from which you will send your Fax.
*File: Choose a file to send as a Fax. The file must be in pdf, txt, jpg, gif, png or tiff
-'''IMPORTANT: The maximum file size is 25 MB'''
*Station ID: This will be the station ID you set for the header of the Fax message. It could be a specified post if your location has several stations, such as Reception, Main Office, Accounting PC, etc.
*Send Email: If selected, an email will be sent to the specified address to confirm the Fax has been sent successfully or to advise of a failed attempt.

[[File:SendFax.png|900px]]

== My Faxes ==
In this section of the Virtual Fax menu you will be able to view your Inbound and Outbound Faxes. You may select a date range and choose the folder you would like to view. Click 'Get My Faxes' to view your selection. You can view the Status of each Fax and select from several actions. You can select to View the Fax directly, download the Fax, email the Fax to an address of your choice or alter the location of the Fax by moving it to another folder.

Only the Status gets updated automatically. You will have to refresh the page to get the costs after a fax has been completed.

[[File:MyFaxes.png|900px]]

You also have the possibility to select multiple entries and delete them in one click by selecting them on the left row and then pressing *Delete Selected Records*

[[File:VfaxDel.png|900px]]

== My Folders ==

In the 'My Folders' section, you can create folders by typing in the folder name of your choice under 'New Folder' and clicking 'Create'.
You will have an overview of your folders, see the dates they were created, the number of Faxes in each folder and be able to edit the folder or delete it.
Any Faxes contained in a created folder will revert back to either the INBOX or SENT folder if the created folder is deleted.

[[File:MyFolders.png|900px]]

== My Fax Numbers ==
In the 'My Fax Numbers' section, you will see your Fax DID numbers and description, the options that have been enabled for each number, the email address if one has been configured along with the URL if configured in the URL Callback section. You can edit the number from the 'Actions' section or choose to delete it. When editing you will have the option to set an email address to receive a notification when a new Fax is received (you can also select to have the PDF file attached in the email) and set a URL Callback (you can also enable URL Callback Retry).

[[File:MyFaxNumbers.png|900px]]

'''Email Address''': If selected, a notification will be sent to the inputted email address that a new fax has been received. You can enter several email address to receive a notification by separating them with a comma ( , ).

'''Attach PDF File''': If selected, a PDF file with the contents of the fax message will be attached to the email.

'''URL Callback''': If selected, fax messages received by your number will sent a POST request to the URL callback provided.

'''Callback Retry''': When selected, we will be expecting an ''OK'' output from your URL callback page as an indicator that you have received the fax message correctly. If we do not receive the ''OK'' letters from your callback page, we will keep sending you the same fax message every 30 minutes for up to 7 days.

'''Fax to SIP''': If selected, fax messages received by your number will be sent to your device (fax machine or PBX) registered on one of our fax servers. For more information, please verify this part of the article: [[Virtual_Fax#T.38_Faxing_.28BETA.29 | T38 faxing]]

[[File:FaxInboundSetup.png|900px]]

== Email to Fax==

This feature allows you to send a Fax message using your email account.

How to send a Fax message using your email account:
*Use the email account you provided when enabling the email to Fax service.
*Send the email to fax@voip.ms
*In the subject field type the destination Fax number (example: 5148000000).
*Attach the document you wish to send to the email message. VoIP.ms supports the following formats: pdf, txt, jpg, gif, png, tif.
*Send the email.

'''Important:'''

*Only the file attached will be faxed, the body of the email won't be transmitted'''
*If you have a signature at your email, avoid using pictures (logos, photos or images) on it, as it will be set as another attachment at your email and that could cause issues at the outbound fax your email will generate.

'''Security Code and From Number:'''

If Security Code is enabled, you need to add a dot (.) and the security code after the destination Fax number (example: 5148000000.Az09).

If you have more than one Fax number, you could change the "from number" by adding a dot (.) and the "from number" you'd like to use after the destination Fax number and the security code (example: 5148000000.Az09.2268280000).

If you have not enabled the security code and want to change the from number, you can add a dot (.) and the From Number after the destination Fax number (example: 5148000000.2268280000).

[[File:EmailToFax.png|900px|]]

== Forward Inbound faxes to mail ==

To have your inbound faxes forwarded to a given email address, on your customer portal, refer to:
DID Numbers --> Manage DID's. When here, select and edit the given Virtual Fax number and on the first screen you'll be presented with, enable the tick boxes: "Email address" and "Attach PDF file", as well as filling in the email address field with the desired address for the faxes to be attached to.

== T.38 Faxing (BETA) ==

Due to the nature of the technology, '''the number of pages, as well as a copy of the fax document''',
will '''not''' be available in the customer portal when '''sending''' a fax over T.38.
'''Inbound faxing''' over T.38 will still have this information available.

'''For a list of ATA devices that support T.38 Faxing, please consult [[ATA_Devices | here]].'''

The following are the requirements in order to be able to send and receive faxes via T.38 directly from your own fax phone setup.

:* A VirtualFAX DID number
:* An analog telephone adapter (ATA) with T.38 support.
:* A Fax machine connected to an ATA
:* The ATA must be configured with a SIP account and registered to one of the fax dedicated servers: '''fax1.voip.ms, fax2.voip.ms'''
:* The caller ID number of the SIP account must be configured to match any of the Fax DIDs available in your account. This will be the number used as the sender.
:* A T.38 enabled fax machine or a T.38 enabled SIP trunk are also supported. These will not require an ATA.

To '''receive''' fax messages, edit your Fax DID number, and choose the SIP account where all the incoming faxes will be forwarded.

[[File:faxdiden.png]]

'''Additional notes:'''

:* T.38 is only supported over SIP. IAX2 is not supported.
:* TCP, UDP, TLS/SRTP are fully supported.
:* T.38 Faxing is not limited to 100 outbound faxes per day nor 25mb file size limit.
:* Because the system merely does a pass-through when sending a fax via T.38, is not possible to know if the fax was successful or not.
:* The Fax message will be charged by the duration of the entire call, from the moment it was answered until it was hung up.

== Fax 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:FaxO_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:FaxO_Reseller_2.png|thumb|none|700px]]

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

: [[File:FaxO_Reseller_3.png|thumb|none|700px]]

Then scroll down and find the feature '''"Manage Faxes"''', and enable it. (If you would like to allow
If you wish to give the possibility to your client to buy new Fax DID via the Reseller interface, enable '''"Order Faxes"'''.

: [[File:FaxO_Reseller_4.png|thumb|none|500px]]

=== Order Faxes ===
----
1) To Order a new Fax Number for your client, or to help your client adding one. Go under the '''[Virtual Faxes]''' menu, then '''[Order Faxes]'''.

[[File:FaxO_Add.png|thumb|none|300px]]

[[File:FaxO_Add_2.png|thumb|none|500px]]

[[File:FaxO_Add_3.png|thumb|none|500px]]

The new Fax DID ordered, will be available in the "Manage Fax Numbers" section.

=== Manage Fax Numbers ===
----
To manage your inbound Fax DID, and send the FAX to Email (PDF). Notifications when recieved. Go under '''[Virtual Faxes]''' then '''[Manage Fax Numbers]'''.
[[File:FaxM_Add.png|thumb|none|300px]]

Where your DID, click on the '''[Edit DID]''' button.

[[File:FaxM_Add_2.png|thumb|none|500px]]

=== Send Fax ===
----
To send a Fax using the reseller interface. Go under the '''[Virtual Faxes]''' at the left navigation bar, then '''[Send Fax]'''.

[[File:FaxS_Add.png|thumb|none|300px]]

Enter the destination number select a Fax number

[[File:FaxS_Add_2.png|thumb|none|500px]]

Click '''[Send Fax]'''

=== My Faxes ===
----
To manage the Incoming/Outgoing Faxes in order to View a fax, delete a fax, send a fax to an email, transfer a fax to a folder... Go under the '''[Virtual Faxes]''' at the left navigation bar, then '''[My Faxes]'''.

[[File:FaxF_Add.png|thumb|none|300px]]

[[File:FaxF_Add_2.png|thumb|none|500px]]

=== Email to Fax ===
----
To use the feature Email To Fax. Go under the '''[Services]''' at the left navigation bar, then on '''[Virtual Faxes]''', then '''[Email to Fax]'''.

[[File:FaxE_Add.png|thumb|none|300px]]

Click on the "'''Add Email to Fax'''" tab. Check the '''Enable''' box. Enter the email address. Select the fax number that should be associated with this email.

[[File:FaxE_Add_2.png|thumb|none|500px]]

Click '''[Save Email to Fax]'''

To send a fax via email through the reseller's "email to fax" option, the instructions and parameters are basically the same than regular email to fax with the exception of the destination email address, which for this case is fax@voipinterface.net .

Devices

2026-07-30T19:07:06Z

Neocypher: /* IP Phones */

{| cellspacing="0" style="width: 100%; background: linear-gradient(135deg, #f8fafc 0%, #f1f5f9 100%); border: 1px solid #e2e8f0; border-radius: 8px; padding: 35px; margin-bottom: 25px; box-shadow: 0 1px 3px rgba(0,0,0,0.05);"
|-
| style="text-align: center;" |

<div style="font-size: 2.8em; font-weight: 200; color: #1e293b; margin-bottom: 15px; letter-spacing: -2px;">Compatible Devices</div>

<div style="font-size: 1rem; color: #64748b; margin-bottom: 10px;">
Connect your existing equipment or choose something new to make calls with VoIP.ms.
</div>

|}

__TOC__

= Softphones =

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

A '''softphone''' is a software program for making telephone calls over the Internet using a general purpose computer, tablet or smartphone, rather than using dedicated hardware. Often designed to behave like a traditional telephone, with a display panel and buttons for user interaction.

'''Learn More:''' [https://voip.ms/blog/softphone-benefits-for-business/ Back to Basics, What is a Softphone and How to Leverage It?]
|}

{| cellspacing="0" style="width: 100%; background: white; border: 1px solid #e2e8f0; border-radius: 12px; padding: 25px; margin: 20px 0; box-shadow: 0 2px 10px rgba(0,0,0,0.06);"
|-
| style="vertical-align: top;" |
<div style="font-size: 1.1em; font-weight: 600; color: #1e293b; margin-bottom: 15px;">Popular Softphones</div>

* [[VoIP.ms_Softphone|'''VoIP.ms Softphone''']] — Official mobile app with push notifications and SMS support (Android & iOS)
* [[Zoiper_5|'''Zoiper''']] — Multi-platform SIP and IAX softphone (Windows, macOS, Android, iOS, Linux)
* [[Bria Solo|'''Bria Solo''']] / [[Bria Teams|'''Bria Teams''']] — Desktop and mobile softphones with call control and team messaging (CounterPath)
* [[Linphone|'''Linphone''']] — Open-source SIP softphone supporting voice, video, and messaging (all platforms)
* [[Microsip |'''MicroSIP''']] — Lightweight Windows/macOS/Linux SIP client
* [[Acrobits|'''Acrobits''']] — Certified mobile SIP client for iOS and Android

<div style="margin-top: 20px;">
[[Softphones|<span style="background: linear-gradient(135deg, #d0382d, #b8261a); color: white; padding: 12px 28px; border-radius: 20px; text-decoration: none; font-weight: 600; display: inline-block; box-shadow: 0 2px 10px rgba(208, 56, 45, 0.3);">Browse All Softphones →</span>]]
</div>
|}

= ATA Devices =

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

An '''Analog Telephone Adapter (ATA)''' is a device used to connect one or more standard analog telephones to a digital telephone system (such as Voice over IP) or a non-standard telephone system. ATAs allow you to keep using your existing analog phones on a VoIP network.

'''Learn More:''' [https://voip.ms/blog/what-is-an-analog-telephone-adapter/ Back to Basics - What is an ATA?]
|}

{| cellspacing="0" style="width: 100%; background: white; border: 1px solid #e2e8f0; border-radius: 12px; padding: 25px; margin: 20px 0; box-shadow: 0 2px 10px rgba(0,0,0,0.06);"
|-
| style="vertical-align: top;" |
<div style="font-size: 1.1em; font-weight: 600; color: #1e293b; margin-bottom: 15px;">Popular ATA Devices</div>

* [[Grandstream HandyTone 802 - HT802|'''Grandstream HT802''']] — 2-port ATA with T.38 faxing, SIP TLS, and SRTP encryption
* [[Grandstream_HT802v2|'''Grandstream HT802V2''']] — Updated HT802 with enhanced performance and security
* [[OBi300|'''Polycom OBi300''']] — Single-port ATA with T.38 fax support, ideal for home office use
* [[Media5 Mediatrix C7 and 4100|'''Mediatrix C7 / 4100 Series''']] — Enterprise-grade VoIP adapters, up to 24 analog ports, TLS/SRTP
* [[ReadyNet AC1000MS and AC1300MS|'''ReadyNet AC1000MS / AC1300MS''']] — Dual-band Wi-Fi router with integrated 2-line or 1-line SIP ATA

<div style="margin-top: 20px;">
[[ATA_Devices|<span style="background: linear-gradient(135deg, #d0382d, #b8261a); color: white; padding: 12px 28px; border-radius: 20px; text-decoration: none; font-weight: 600; display: inline-block; box-shadow: 0 2px 10px rgba(208, 56, 45, 0.3);">Browse All ATA Devices →</span>]]
</div>
|}

= IP Phones =

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

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. IP phones connect directly to your network and require no adapter.

'''Learn More:''' [https://voip.ms/blog/what-is-an-ip-phone/back-to-basics-what-is-an-ip-phone/ Back to Basics - What is an IP Phone?]
|}

{| cellspacing="0" style="width: 100%; background: white; border: 1px solid #e2e8f0; border-radius: 12px; padding: 25px; margin: 20px 0; box-shadow: 0 2px 10px rgba(0,0,0,0.06);"
|-
| style="vertical-align: top;" |
<div style="font-size: 1.1em; font-weight: 600; color: #1e293b; margin-bottom: 15px;">Popular IP Phones</div>

* [[Grandstream_GXP2170|'''Grandstream GXP2170''']] — High-end IP phone with color display and advanced telephony features
* [[Yealink_SIP-T46U|'''Yealink SIP-T46U''']] — Professional business phone with color display and advanced call management
* [[Fanvil_V66_Pro|'''Fanvil V66 Pro''']] — Premium phone with 7-inch rotatable touch screen and dual-band Wi-Fi 6
* [[Cisco IP Phone 68XX and 88XX|'''Cisco 68XX & 88XX Series''']] — Business phones ranging from affordable to HD video-capable models
* [[Fanvil_V67|'''Fanvil V67''']] — Executive phone with adjustable touch screen and colorful keypad lighting
* [[Grandstream_GRP2615|'''Grandstream GRP2615''']] — Carrier-grade enterprise phone with advanced security features

<div style="margin-top: 20px;">
[[IP_Phones|<span style="background: linear-gradient(135deg, #d0382d, #b8261a); color: white; padding: 12px 28px; border-radius: 20px; text-decoration: none; font-weight: 600; display: inline-block; box-shadow: 0 2px 10px rgba(208, 56, 45, 0.3);">Browse All IP Phones →</span>]]
</div>
|}

= PBXs =

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

An acronym for '''Private Branch eXchange'''. PBX telephone systems support incoming calls from the outside PSTN, placing calls between extensions, conferencing, voicemail, and a variety of other advanced telecommunication functions. PBX systems range from open-source software solutions to dedicated hardware appliances.

'''Learn More:''' [https://voip.ms/blog/back-to-basics-what-is-a-pbx/ Back to Basics - What is a PBX?]
|}

{| cellspacing="0" style="width: 100%; background: white; border: 1px solid #e2e8f0; border-radius: 12px; padding: 25px; margin: 20px 0; box-shadow: 0 2px 10px rgba(0,0,0,0.06);"
|-
| style="vertical-align: top;" |
<div style="font-size: 1.1em; font-weight: 600; color: #1e293b; margin-bottom: 15px;">Popular PBX Systems</div>

* [[VoIP.ms_PBX|'''VoIP.ms PBX''']] — VoIP.ms Hosted Cloud PBX built directly into the customer portal. No hardware, no software to install, and no separate PBX subscription required.
* [[Asterisk SIP|'''Asterisk''']] — The world's most widely used open source telephony platform (Linux/UNIX, since 1999)
* [[FreePBX / PBX in a Flash|'''FreePBX''']] — Open source, web-based PBX built on Asterisk — easy to manage at any scale
* [[3CX Phone System|'''3CX Phone System''']] — Software-based IP PBX with complete Unified Communications out of the box
* [[VitalPBX|'''VitalPBX''']] — Free Linux/Asterisk-based communications system for on-premise or hosted deployments
* [[FusionPBX|'''FusionPBX''']] — Full-featured multi-tenant PBX and voice switch for FreeSWITCH
* [[Yeastar|'''Yeastar S/P-Series''']] — SMB-focused VoIP PBX with enterprise-grade UC capabilities

<div style="margin-top: 20px;">
[[PBXs|<span style="background: linear-gradient(135deg, #d0382d, #b8261a); color: white; padding: 12px 28px; border-radius: 20px; text-decoration: none; font-weight: 600; display: inline-block; box-shadow: 0 2px 10px rgba(208, 56, 45, 0.3);">Browse All PBX Systems →</span>]]
</div>
|}

{| cellspacing="0" style="width: 100%; background: linear-gradient(135deg, #f0f9ff 0%, #e0f2fe 100%); border: 1px solid #7dd3fc; border-radius: 8px; padding: 25px; margin: 25px 0; box-shadow: 0 2px 10px rgba(125, 211, 252, 0.15);"
|-
| style="text-align: left;" |
<div style="font-size: 1.1em; font-weight: 600; color: #0369a1; margin-bottom: 15px;">Not sure what to use?</div>

VoIP.ms supports many types of devices and phone systems. Choose the option that matches your setup:

* '''Using a mobile app or computer''' → [[Softphones|Softphones]]
* '''Using a desk phone in an office''' → [[IP_Phones|IP Phones]]
* '''Using a traditional analog phone''' → [[ATA_Devices|ATA Devices]]
* '''Running a business phone system''' → [[PBXs|PBX Systems]]
|}

[[Category:Devices]]

Devices

2026-07-30T19:06:37Z

Neocypher: /* PBXs */

{| cellspacing="0" style="width: 100%; background: linear-gradient(135deg, #f8fafc 0%, #f1f5f9 100%); border: 1px solid #e2e8f0; border-radius: 8px; padding: 35px; margin-bottom: 25px; box-shadow: 0 1px 3px rgba(0,0,0,0.05);"
|-
| style="text-align: center;" |

<div style="font-size: 2.8em; font-weight: 200; color: #1e293b; margin-bottom: 15px; letter-spacing: -2px;">Compatible Devices</div>

<div style="font-size: 1rem; color: #64748b; margin-bottom: 10px;">
Connect your existing equipment or choose something new to make calls with VoIP.ms.
</div>

|}

__TOC__

= Softphones =

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

A '''softphone''' is a software program for making telephone calls over the Internet using a general purpose computer, tablet or smartphone, rather than using dedicated hardware. Often designed to behave like a traditional telephone, with a display panel and buttons for user interaction.

'''Learn More:''' [https://voip.ms/blog/softphone-benefits-for-business/ Back to Basics, What is a Softphone and How to Leverage It?]
|}

{| cellspacing="0" style="width: 100%; background: white; border: 1px solid #e2e8f0; border-radius: 12px; padding: 25px; margin: 20px 0; box-shadow: 0 2px 10px rgba(0,0,0,0.06);"
|-
| style="vertical-align: top;" |
<div style="font-size: 1.1em; font-weight: 600; color: #1e293b; margin-bottom: 15px;">Popular Softphones</div>

* [[VoIP.ms_Softphone|'''VoIP.ms Softphone''']] — Official mobile app with push notifications and SMS support (Android & iOS)
* [[Zoiper_5|'''Zoiper''']] — Multi-platform SIP and IAX softphone (Windows, macOS, Android, iOS, Linux)
* [[Bria Solo|'''Bria Solo''']] / [[Bria Teams|'''Bria Teams''']] — Desktop and mobile softphones with call control and team messaging (CounterPath)
* [[Linphone|'''Linphone''']] — Open-source SIP softphone supporting voice, video, and messaging (all platforms)
* [[Microsip |'''MicroSIP''']] — Lightweight Windows/macOS/Linux SIP client
* [[Acrobits|'''Acrobits''']] — Certified mobile SIP client for iOS and Android

<div style="margin-top: 20px;">
[[Softphones|<span style="background: linear-gradient(135deg, #d0382d, #b8261a); color: white; padding: 12px 28px; border-radius: 20px; text-decoration: none; font-weight: 600; display: inline-block; box-shadow: 0 2px 10px rgba(208, 56, 45, 0.3);">Browse All Softphones →</span>]]
</div>
|}

= ATA Devices =

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

An '''Analog Telephone Adapter (ATA)''' is a device used to connect one or more standard analog telephones to a digital telephone system (such as Voice over IP) or a non-standard telephone system. ATAs allow you to keep using your existing analog phones on a VoIP network.

'''Learn More:''' [https://voip.ms/blog/what-is-an-analog-telephone-adapter/ Back to Basics - What is an ATA?]
|}

{| cellspacing="0" style="width: 100%; background: white; border: 1px solid #e2e8f0; border-radius: 12px; padding: 25px; margin: 20px 0; box-shadow: 0 2px 10px rgba(0,0,0,0.06);"
|-
| style="vertical-align: top;" |
<div style="font-size: 1.1em; font-weight: 600; color: #1e293b; margin-bottom: 15px;">Popular ATA Devices</div>

* [[Grandstream HandyTone 802 - HT802|'''Grandstream HT802''']] — 2-port ATA with T.38 faxing, SIP TLS, and SRTP encryption
* [[Grandstream_HT802v2|'''Grandstream HT802V2''']] — Updated HT802 with enhanced performance and security
* [[OBi300|'''Polycom OBi300''']] — Single-port ATA with T.38 fax support, ideal for home office use
* [[Media5 Mediatrix C7 and 4100|'''Mediatrix C7 / 4100 Series''']] — Enterprise-grade VoIP adapters, up to 24 analog ports, TLS/SRTP
* [[ReadyNet AC1000MS and AC1300MS|'''ReadyNet AC1000MS / AC1300MS''']] — Dual-band Wi-Fi router with integrated 2-line or 1-line SIP ATA

<div style="margin-top: 20px;">
[[ATA_Devices|<span style="background: linear-gradient(135deg, #d0382d, #b8261a); color: white; padding: 12px 28px; border-radius: 20px; text-decoration: none; font-weight: 600; display: inline-block; box-shadow: 0 2px 10px rgba(208, 56, 45, 0.3);">Browse All ATA Devices →</span>]]
</div>
|}

= IP Phones =

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

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. IP phones connect directly to your network and require no adapter.

'''Learn More:''' [https://wiki.voip.ms/article/Back_to_Basics:_What_is_an_IP_Phone%3F Back to Basics - What is an IP Phone?]
|}

{| cellspacing="0" style="width: 100%; background: white; border: 1px solid #e2e8f0; border-radius: 12px; padding: 25px; margin: 20px 0; box-shadow: 0 2px 10px rgba(0,0,0,0.06);"
|-
| style="vertical-align: top;" |
<div style="font-size: 1.1em; font-weight: 600; color: #1e293b; margin-bottom: 15px;">Popular IP Phones</div>

* [[Grandstream_GXP2170|'''Grandstream GXP2170''']] — High-end IP phone with color display and advanced telephony features
* [[Yealink_SIP-T46U|'''Yealink SIP-T46U''']] — Professional business phone with color display and advanced call management
* [[Fanvil_V66_Pro|'''Fanvil V66 Pro''']] — Premium phone with 7-inch rotatable touch screen and dual-band Wi-Fi 6
* [[Cisco IP Phone 68XX and 88XX|'''Cisco 68XX & 88XX Series''']] — Business phones ranging from affordable to HD video-capable models
* [[Fanvil_V67|'''Fanvil V67''']] — Executive phone with adjustable touch screen and colorful keypad lighting
* [[Grandstream_GRP2615|'''Grandstream GRP2615''']] — Carrier-grade enterprise phone with advanced security features

<div style="margin-top: 20px;">
[[IP_Phones|<span style="background: linear-gradient(135deg, #d0382d, #b8261a); color: white; padding: 12px 28px; border-radius: 20px; text-decoration: none; font-weight: 600; display: inline-block; box-shadow: 0 2px 10px rgba(208, 56, 45, 0.3);">Browse All IP Phones →</span>]]
</div>
|}

= PBXs =

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

An acronym for '''Private Branch eXchange'''. PBX telephone systems support incoming calls from the outside PSTN, placing calls between extensions, conferencing, voicemail, and a variety of other advanced telecommunication functions. PBX systems range from open-source software solutions to dedicated hardware appliances.

'''Learn More:''' [https://voip.ms/blog/back-to-basics-what-is-a-pbx/ Back to Basics - What is a PBX?]
|}

{| cellspacing="0" style="width: 100%; background: white; border: 1px solid #e2e8f0; border-radius: 12px; padding: 25px; margin: 20px 0; box-shadow: 0 2px 10px rgba(0,0,0,0.06);"
|-
| style="vertical-align: top;" |
<div style="font-size: 1.1em; font-weight: 600; color: #1e293b; margin-bottom: 15px;">Popular PBX Systems</div>

* [[VoIP.ms_PBX|'''VoIP.ms PBX''']] — VoIP.ms Hosted Cloud PBX built directly into the customer portal. No hardware, no software to install, and no separate PBX subscription required.
* [[Asterisk SIP|'''Asterisk''']] — The world's most widely used open source telephony platform (Linux/UNIX, since 1999)
* [[FreePBX / PBX in a Flash|'''FreePBX''']] — Open source, web-based PBX built on Asterisk — easy to manage at any scale
* [[3CX Phone System|'''3CX Phone System''']] — Software-based IP PBX with complete Unified Communications out of the box
* [[VitalPBX|'''VitalPBX''']] — Free Linux/Asterisk-based communications system for on-premise or hosted deployments
* [[FusionPBX|'''FusionPBX''']] — Full-featured multi-tenant PBX and voice switch for FreeSWITCH
* [[Yeastar|'''Yeastar S/P-Series''']] — SMB-focused VoIP PBX with enterprise-grade UC capabilities

<div style="margin-top: 20px;">
[[PBXs|<span style="background: linear-gradient(135deg, #d0382d, #b8261a); color: white; padding: 12px 28px; border-radius: 20px; text-decoration: none; font-weight: 600; display: inline-block; box-shadow: 0 2px 10px rgba(208, 56, 45, 0.3);">Browse All PBX Systems →</span>]]
</div>
|}

{| cellspacing="0" style="width: 100%; background: linear-gradient(135deg, #f0f9ff 0%, #e0f2fe 100%); border: 1px solid #7dd3fc; border-radius: 8px; padding: 25px; margin: 25px 0; box-shadow: 0 2px 10px rgba(125, 211, 252, 0.15);"
|-
| style="text-align: left;" |
<div style="font-size: 1.1em; font-weight: 600; color: #0369a1; margin-bottom: 15px;">Not sure what to use?</div>

VoIP.ms supports many types of devices and phone systems. Choose the option that matches your setup:

* '''Using a mobile app or computer''' → [[Softphones|Softphones]]
* '''Using a desk phone in an office''' → [[IP_Phones|IP Phones]]
* '''Using a traditional analog phone''' → [[ATA_Devices|ATA Devices]]
* '''Running a business phone system''' → [[PBXs|PBX Systems]]
|}

[[Category:Devices]]

Devices

2026-07-30T19:05:58Z

Neocypher: /* ATA Devices */

{| cellspacing="0" style="width: 100%; background: linear-gradient(135deg, #f8fafc 0%, #f1f5f9 100%); border: 1px solid #e2e8f0; border-radius: 8px; padding: 35px; margin-bottom: 25px; box-shadow: 0 1px 3px rgba(0,0,0,0.05);"
|-
| style="text-align: center;" |

<div style="font-size: 2.8em; font-weight: 200; color: #1e293b; margin-bottom: 15px; letter-spacing: -2px;">Compatible Devices</div>

<div style="font-size: 1rem; color: #64748b; margin-bottom: 10px;">
Connect your existing equipment or choose something new to make calls with VoIP.ms.
</div>

|}

__TOC__

= Softphones =

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

A '''softphone''' is a software program for making telephone calls over the Internet using a general purpose computer, tablet or smartphone, rather than using dedicated hardware. Often designed to behave like a traditional telephone, with a display panel and buttons for user interaction.

'''Learn More:''' [https://voip.ms/blog/softphone-benefits-for-business/ Back to Basics, What is a Softphone and How to Leverage It?]
|}

{| cellspacing="0" style="width: 100%; background: white; border: 1px solid #e2e8f0; border-radius: 12px; padding: 25px; margin: 20px 0; box-shadow: 0 2px 10px rgba(0,0,0,0.06);"
|-
| style="vertical-align: top;" |
<div style="font-size: 1.1em; font-weight: 600; color: #1e293b; margin-bottom: 15px;">Popular Softphones</div>

* [[VoIP.ms_Softphone|'''VoIP.ms Softphone''']] — Official mobile app with push notifications and SMS support (Android & iOS)
* [[Zoiper_5|'''Zoiper''']] — Multi-platform SIP and IAX softphone (Windows, macOS, Android, iOS, Linux)
* [[Bria Solo|'''Bria Solo''']] / [[Bria Teams|'''Bria Teams''']] — Desktop and mobile softphones with call control and team messaging (CounterPath)
* [[Linphone|'''Linphone''']] — Open-source SIP softphone supporting voice, video, and messaging (all platforms)
* [[Microsip |'''MicroSIP''']] — Lightweight Windows/macOS/Linux SIP client
* [[Acrobits|'''Acrobits''']] — Certified mobile SIP client for iOS and Android

<div style="margin-top: 20px;">
[[Softphones|<span style="background: linear-gradient(135deg, #d0382d, #b8261a); color: white; padding: 12px 28px; border-radius: 20px; text-decoration: none; font-weight: 600; display: inline-block; box-shadow: 0 2px 10px rgba(208, 56, 45, 0.3);">Browse All Softphones →</span>]]
</div>
|}

= ATA Devices =

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

An '''Analog Telephone Adapter (ATA)''' is a device used to connect one or more standard analog telephones to a digital telephone system (such as Voice over IP) or a non-standard telephone system. ATAs allow you to keep using your existing analog phones on a VoIP network.

'''Learn More:''' [https://voip.ms/blog/what-is-an-analog-telephone-adapter/ Back to Basics - What is an ATA?]
|}

{| cellspacing="0" style="width: 100%; background: white; border: 1px solid #e2e8f0; border-radius: 12px; padding: 25px; margin: 20px 0; box-shadow: 0 2px 10px rgba(0,0,0,0.06);"
|-
| style="vertical-align: top;" |
<div style="font-size: 1.1em; font-weight: 600; color: #1e293b; margin-bottom: 15px;">Popular ATA Devices</div>

* [[Grandstream HandyTone 802 - HT802|'''Grandstream HT802''']] — 2-port ATA with T.38 faxing, SIP TLS, and SRTP encryption
* [[Grandstream_HT802v2|'''Grandstream HT802V2''']] — Updated HT802 with enhanced performance and security
* [[OBi300|'''Polycom OBi300''']] — Single-port ATA with T.38 fax support, ideal for home office use
* [[Media5 Mediatrix C7 and 4100|'''Mediatrix C7 / 4100 Series''']] — Enterprise-grade VoIP adapters, up to 24 analog ports, TLS/SRTP
* [[ReadyNet AC1000MS and AC1300MS|'''ReadyNet AC1000MS / AC1300MS''']] — Dual-band Wi-Fi router with integrated 2-line or 1-line SIP ATA

<div style="margin-top: 20px;">
[[ATA_Devices|<span style="background: linear-gradient(135deg, #d0382d, #b8261a); color: white; padding: 12px 28px; border-radius: 20px; text-decoration: none; font-weight: 600; display: inline-block; box-shadow: 0 2px 10px rgba(208, 56, 45, 0.3);">Browse All ATA Devices →</span>]]
</div>
|}

= IP Phones =

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

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. IP phones connect directly to your network and require no adapter.

'''Learn More:''' [https://wiki.voip.ms/article/Back_to_Basics:_What_is_an_IP_Phone%3F Back to Basics - What is an IP Phone?]
|}

{| cellspacing="0" style="width: 100%; background: white; border: 1px solid #e2e8f0; border-radius: 12px; padding: 25px; margin: 20px 0; box-shadow: 0 2px 10px rgba(0,0,0,0.06);"
|-
| style="vertical-align: top;" |
<div style="font-size: 1.1em; font-weight: 600; color: #1e293b; margin-bottom: 15px;">Popular IP Phones</div>

* [[Grandstream_GXP2170|'''Grandstream GXP2170''']] — High-end IP phone with color display and advanced telephony features
* [[Yealink_SIP-T46U|'''Yealink SIP-T46U''']] — Professional business phone with color display and advanced call management
* [[Fanvil_V66_Pro|'''Fanvil V66 Pro''']] — Premium phone with 7-inch rotatable touch screen and dual-band Wi-Fi 6
* [[Cisco IP Phone 68XX and 88XX|'''Cisco 68XX & 88XX Series''']] — Business phones ranging from affordable to HD video-capable models
* [[Fanvil_V67|'''Fanvil V67''']] — Executive phone with adjustable touch screen and colorful keypad lighting
* [[Grandstream_GRP2615|'''Grandstream GRP2615''']] — Carrier-grade enterprise phone with advanced security features

<div style="margin-top: 20px;">
[[IP_Phones|<span style="background: linear-gradient(135deg, #d0382d, #b8261a); color: white; padding: 12px 28px; border-radius: 20px; text-decoration: none; font-weight: 600; display: inline-block; box-shadow: 0 2px 10px rgba(208, 56, 45, 0.3);">Browse All IP Phones →</span>]]
</div>
|}

= PBXs =

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

An acronym for '''Private Branch eXchange'''. PBX telephone systems support incoming calls from the outside PSTN, placing calls between extensions, conferencing, voicemail, and a variety of other advanced telecommunication functions. PBX systems range from open-source software solutions to dedicated hardware appliances.

'''Learn More:''' [https://wiki.voip.ms/article/Back_to_Basics_%E2%80%93_What_is_a_PBX%3F Back to Basics - What is a PBX?]
|}

{| cellspacing="0" style="width: 100%; background: white; border: 1px solid #e2e8f0; border-radius: 12px; padding: 25px; margin: 20px 0; box-shadow: 0 2px 10px rgba(0,0,0,0.06);"
|-
| style="vertical-align: top;" |
<div style="font-size: 1.1em; font-weight: 600; color: #1e293b; margin-bottom: 15px;">Popular PBX Systems</div>

* [[VoIP.ms_PBX|'''VoIP.ms PBX''']] — VoIP.ms Hosted Cloud PBX built directly into the customer portal. No hardware, no software to install, and no separate PBX subscription required.
* [[Asterisk SIP|'''Asterisk''']] — The world's most widely used open source telephony platform (Linux/UNIX, since 1999)
* [[FreePBX / PBX in a Flash|'''FreePBX''']] — Open source, web-based PBX built on Asterisk — easy to manage at any scale
* [[3CX Phone System|'''3CX Phone System''']] — Software-based IP PBX with complete Unified Communications out of the box
* [[VitalPBX|'''VitalPBX''']] — Free Linux/Asterisk-based communications system for on-premise or hosted deployments
* [[FusionPBX|'''FusionPBX''']] — Full-featured multi-tenant PBX and voice switch for FreeSWITCH
* [[Yeastar|'''Yeastar S/P-Series''']] — SMB-focused VoIP PBX with enterprise-grade UC capabilities

<div style="margin-top: 20px;">
[[PBXs|<span style="background: linear-gradient(135deg, #d0382d, #b8261a); color: white; padding: 12px 28px; border-radius: 20px; text-decoration: none; font-weight: 600; display: inline-block; box-shadow: 0 2px 10px rgba(208, 56, 45, 0.3);">Browse All PBX Systems →</span>]]
</div>
|}

{| cellspacing="0" style="width: 100%; background: linear-gradient(135deg, #f0f9ff 0%, #e0f2fe 100%); border: 1px solid #7dd3fc; border-radius: 8px; padding: 25px; margin: 25px 0; box-shadow: 0 2px 10px rgba(125, 211, 252, 0.15);"
|-
| style="text-align: left;" |
<div style="font-size: 1.1em; font-weight: 600; color: #0369a1; margin-bottom: 15px;">Not sure what to use?</div>

VoIP.ms supports many types of devices and phone systems. Choose the option that matches your setup:

* '''Using a mobile app or computer''' → [[Softphones|Softphones]]
* '''Using a desk phone in an office''' → [[IP_Phones|IP Phones]]
* '''Using a traditional analog phone''' → [[ATA_Devices|ATA Devices]]
* '''Running a business phone system''' → [[PBXs|PBX Systems]]
|}

[[Category:Devices]]

Devices

2026-07-30T19:05:19Z

Neocypher: /* Softphones */

{| cellspacing="0" style="width: 100%; background: linear-gradient(135deg, #f8fafc 0%, #f1f5f9 100%); border: 1px solid #e2e8f0; border-radius: 8px; padding: 35px; margin-bottom: 25px; box-shadow: 0 1px 3px rgba(0,0,0,0.05);"
|-
| style="text-align: center;" |

<div style="font-size: 2.8em; font-weight: 200; color: #1e293b; margin-bottom: 15px; letter-spacing: -2px;">Compatible Devices</div>

<div style="font-size: 1rem; color: #64748b; margin-bottom: 10px;">
Connect your existing equipment or choose something new to make calls with VoIP.ms.
</div>

|}

__TOC__

= Softphones =

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

A '''softphone''' is a software program for making telephone calls over the Internet using a general purpose computer, tablet or smartphone, rather than using dedicated hardware. Often designed to behave like a traditional telephone, with a display panel and buttons for user interaction.

'''Learn More:''' [https://voip.ms/blog/softphone-benefits-for-business/ Back to Basics, What is a Softphone and How to Leverage It?]
|}

{| cellspacing="0" style="width: 100%; background: white; border: 1px solid #e2e8f0; border-radius: 12px; padding: 25px; margin: 20px 0; box-shadow: 0 2px 10px rgba(0,0,0,0.06);"
|-
| style="vertical-align: top;" |
<div style="font-size: 1.1em; font-weight: 600; color: #1e293b; margin-bottom: 15px;">Popular Softphones</div>

* [[VoIP.ms_Softphone|'''VoIP.ms Softphone''']] — Official mobile app with push notifications and SMS support (Android & iOS)
* [[Zoiper_5|'''Zoiper''']] — Multi-platform SIP and IAX softphone (Windows, macOS, Android, iOS, Linux)
* [[Bria Solo|'''Bria Solo''']] / [[Bria Teams|'''Bria Teams''']] — Desktop and mobile softphones with call control and team messaging (CounterPath)
* [[Linphone|'''Linphone''']] — Open-source SIP softphone supporting voice, video, and messaging (all platforms)
* [[Microsip |'''MicroSIP''']] — Lightweight Windows/macOS/Linux SIP client
* [[Acrobits|'''Acrobits''']] — Certified mobile SIP client for iOS and Android

<div style="margin-top: 20px;">
[[Softphones|<span style="background: linear-gradient(135deg, #d0382d, #b8261a); color: white; padding: 12px 28px; border-radius: 20px; text-decoration: none; font-weight: 600; display: inline-block; box-shadow: 0 2px 10px rgba(208, 56, 45, 0.3);">Browse All Softphones →</span>]]
</div>
|}

= ATA Devices =

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

An '''Analog Telephone Adapter (ATA)''' is a device used to connect one or more standard analog telephones to a digital telephone system (such as Voice over IP) or a non-standard telephone system. ATAs allow you to keep using your existing analog phones on a VoIP network.

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

{| cellspacing="0" style="width: 100%; background: white; border: 1px solid #e2e8f0; border-radius: 12px; padding: 25px; margin: 20px 0; box-shadow: 0 2px 10px rgba(0,0,0,0.06);"
|-
| style="vertical-align: top;" |
<div style="font-size: 1.1em; font-weight: 600; color: #1e293b; margin-bottom: 15px;">Popular ATA Devices</div>

* [[Grandstream HandyTone 802 - HT802|'''Grandstream HT802''']] — 2-port ATA with T.38 faxing, SIP TLS, and SRTP encryption
* [[Grandstream_HT802v2|'''Grandstream HT802V2''']] — Updated HT802 with enhanced performance and security
* [[OBi300|'''Polycom OBi300''']] — Single-port ATA with T.38 fax support, ideal for home office use
* [[Media5 Mediatrix C7 and 4100|'''Mediatrix C7 / 4100 Series''']] — Enterprise-grade VoIP adapters, up to 24 analog ports, TLS/SRTP
* [[ReadyNet AC1000MS and AC1300MS|'''ReadyNet AC1000MS / AC1300MS''']] — Dual-band Wi-Fi router with integrated 2-line or 1-line SIP ATA

<div style="margin-top: 20px;">
[[ATA_Devices|<span style="background: linear-gradient(135deg, #d0382d, #b8261a); color: white; padding: 12px 28px; border-radius: 20px; text-decoration: none; font-weight: 600; display: inline-block; box-shadow: 0 2px 10px rgba(208, 56, 45, 0.3);">Browse All ATA Devices →</span>]]
</div>
|}

= IP Phones =

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

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. IP phones connect directly to your network and require no adapter.

'''Learn More:''' [https://wiki.voip.ms/article/Back_to_Basics:_What_is_an_IP_Phone%3F Back to Basics - What is an IP Phone?]
|}

{| cellspacing="0" style="width: 100%; background: white; border: 1px solid #e2e8f0; border-radius: 12px; padding: 25px; margin: 20px 0; box-shadow: 0 2px 10px rgba(0,0,0,0.06);"
|-
| style="vertical-align: top;" |
<div style="font-size: 1.1em; font-weight: 600; color: #1e293b; margin-bottom: 15px;">Popular IP Phones</div>

* [[Grandstream_GXP2170|'''Grandstream GXP2170''']] — High-end IP phone with color display and advanced telephony features
* [[Yealink_SIP-T46U|'''Yealink SIP-T46U''']] — Professional business phone with color display and advanced call management
* [[Fanvil_V66_Pro|'''Fanvil V66 Pro''']] — Premium phone with 7-inch rotatable touch screen and dual-band Wi-Fi 6
* [[Cisco IP Phone 68XX and 88XX|'''Cisco 68XX & 88XX Series''']] — Business phones ranging from affordable to HD video-capable models
* [[Fanvil_V67|'''Fanvil V67''']] — Executive phone with adjustable touch screen and colorful keypad lighting
* [[Grandstream_GRP2615|'''Grandstream GRP2615''']] — Carrier-grade enterprise phone with advanced security features

<div style="margin-top: 20px;">
[[IP_Phones|<span style="background: linear-gradient(135deg, #d0382d, #b8261a); color: white; padding: 12px 28px; border-radius: 20px; text-decoration: none; font-weight: 600; display: inline-block; box-shadow: 0 2px 10px rgba(208, 56, 45, 0.3);">Browse All IP Phones →</span>]]
</div>
|}

= PBXs =

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

An acronym for '''Private Branch eXchange'''. PBX telephone systems support incoming calls from the outside PSTN, placing calls between extensions, conferencing, voicemail, and a variety of other advanced telecommunication functions. PBX systems range from open-source software solutions to dedicated hardware appliances.

'''Learn More:''' [https://wiki.voip.ms/article/Back_to_Basics_%E2%80%93_What_is_a_PBX%3F Back to Basics - What is a PBX?]
|}

{| cellspacing="0" style="width: 100%; background: white; border: 1px solid #e2e8f0; border-radius: 12px; padding: 25px; margin: 20px 0; box-shadow: 0 2px 10px rgba(0,0,0,0.06);"
|-
| style="vertical-align: top;" |
<div style="font-size: 1.1em; font-weight: 600; color: #1e293b; margin-bottom: 15px;">Popular PBX Systems</div>

* [[VoIP.ms_PBX|'''VoIP.ms PBX''']] — VoIP.ms Hosted Cloud PBX built directly into the customer portal. No hardware, no software to install, and no separate PBX subscription required.
* [[Asterisk SIP|'''Asterisk''']] — The world's most widely used open source telephony platform (Linux/UNIX, since 1999)
* [[FreePBX / PBX in a Flash|'''FreePBX''']] — Open source, web-based PBX built on Asterisk — easy to manage at any scale
* [[3CX Phone System|'''3CX Phone System''']] — Software-based IP PBX with complete Unified Communications out of the box
* [[VitalPBX|'''VitalPBX''']] — Free Linux/Asterisk-based communications system for on-premise or hosted deployments
* [[FusionPBX|'''FusionPBX''']] — Full-featured multi-tenant PBX and voice switch for FreeSWITCH
* [[Yeastar|'''Yeastar S/P-Series''']] — SMB-focused VoIP PBX with enterprise-grade UC capabilities

<div style="margin-top: 20px;">
[[PBXs|<span style="background: linear-gradient(135deg, #d0382d, #b8261a); color: white; padding: 12px 28px; border-radius: 20px; text-decoration: none; font-weight: 600; display: inline-block; box-shadow: 0 2px 10px rgba(208, 56, 45, 0.3);">Browse All PBX Systems →</span>]]
</div>
|}

{| cellspacing="0" style="width: 100%; background: linear-gradient(135deg, #f0f9ff 0%, #e0f2fe 100%); border: 1px solid #7dd3fc; border-radius: 8px; padding: 25px; margin: 25px 0; box-shadow: 0 2px 10px rgba(125, 211, 252, 0.15);"
|-
| style="text-align: left;" |
<div style="font-size: 1.1em; font-weight: 600; color: #0369a1; margin-bottom: 15px;">Not sure what to use?</div>

VoIP.ms supports many types of devices and phone systems. Choose the option that matches your setup:

* '''Using a mobile app or computer''' → [[Softphones|Softphones]]
* '''Using a desk phone in an office''' → [[IP_Phones|IP Phones]]
* '''Using a traditional analog phone''' → [[ATA_Devices|ATA Devices]]
* '''Running a business phone system''' → [[PBXs|PBX Systems]]
|}

[[Category:Devices]]

SMS-MMS

2026-07-30T19:04:05Z

Neocypher: /* Blog Articles */

{| class="wikitable"
|+
|-
! Article en Français !! Artículo en Español
|-
| [https://wiki.voip.ms/article/SMS-MMS-FR Français] ||
[https://wiki.voip.ms/article/SMS-MMS-ES Español]
|}

SMS means '''S'''hort '''M'''essage '''S'''ervice ('''SMS'''). With the addition of '''M'''ultimedia '''M'''essage '''S'''ervice ('''MMS''') to the service, users are now capable of sending and receiving media files ('''photos, videos, and audio''') as well as sending text messages up to 160 characters (which is the limit for an SMS). The MMS service also allows the user to send a message to multiple recipients at the same time.

= Blog Articles =
:* [https://voip.ms/blog/benefits-of-sms-mms-campaigns-vs-email-marketing/ Benefits of SMS/MMS Campaigns vs. Email Marketing]
:* [https://voip.ms/blog/how-to-generate-leads-with-sms-mms/ How to Generate Leads with SMS/MMS?]
:* [https://voip.ms/blog/instantly-boost-your-customer-experience-allow-your-customers-to-text-you/ Instantly Boost Your Customer Experience, Allow Your Customers to Text You!]

= Service Limitations =

<div style="margin-left:50px;
padding:5px;
border-left: 6px solid rgb(208, 56, 45, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(208, 56, 45, 0.08);
width:70vw;">

🏗️ ''' Please notice that the MMS service is in the Beta phase.'''
: It is important to us that you report any issues with this service by sending an email to [mailto:support@voip.ms support] so that the developers can get involved if necessary.
</div>

<div style="margin-left:50px;
padding:5px;
border-left: 6px solid rgb(208, 56, 45, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(51, 51, 51, 0.08);
width:70vw;">

⚠️ '''LIMITATIONS'''
: ➜ '''The limit of the capacity to send SMS/MMS via the portal is unlimited.'''
: ➜ The limit of sending SMS/MMS via the API is limited to '''<u>100 per day</u>'''. For any modification to the limit, please contact support or open a ticket via the customer portal.
: ➜ At the moment, receiving and reading MMS is only possible through the message center or by using the SMS/MMS Forward option. '''MMS through SIP protocol is not available.'''

: ➜ The SMS/MMS Service is only available for <u>'''US and Canadian DID and Toll-Free Numbers'''</u> marked with the distinctive '''SMS/MMS Icon'''.
: ➜ The SMS/MMS Service can only be used to send SMS/MMS Messages to <u>'''Canadian and US'''</u> 10 digit numbers at this time.
: ➜ SIP SMS messages will only be delivered to and accepted from SIP trunks using username/password authentication.
: ➜ SIP SMS messages use the SIP MESSAGE method. See [https://tools.ietf.org/html/rfc3428 RFC 3428] for details.
: ➜ At this time we '''<u>cannot guarantee</u> that Short Code SMS Messages''', which are usually 6 digits or less ''(e.g.: Skype, Bank Codes, TV Commercials etc...)'' will work.
: ➜ At this time we '''<u>cannot guarantee</u> that 2FA SMS Messages''', which are usually to confirm your identity and log into your service portal ''(e.g.: Microsoft, among other providers offering this feature)'' will work.
: ➜ At this time SMS messages from/to short code numbers '''<u>aren't supported</u> for Toll-Free numbers'''.
: ➜ We cannot guarantee that '''accents or special characters''' including non-Latin letters will be properly delivered.

: ➜ Public URL shorteners such as '''''bit.ly, tinyurl.com''''' etc, <font style="color:red">'''are widely restricted'''</font> by mobile providers as a response from the industry to combat SPAM campaigns and protect the consumer. Including this type of links into your messages '''<u>may result in a blocked or undelivered message</u>'''.
: ➜ For '''Ported In Numbers''' compatible with the feature: The SMS/MMS Functionality will be available up to '''48 hours after the porting process is marked as Completed'''.

: ➜ VoIP.ms reserves the right, at its sole discretion, to limit or disable the SMS/MMS service from any account that may present marketing patterns/automation patterns/bulk sending or receiving.

</div>

<div style="margin-left:50px;
clear:left;
padding:5px;
border-left: 6px solid rgb(66, 192, 251, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(66, 192, 251, 0.05);
width:70vw;">

'''💬 NOTE ''' If you have further questions don't hesitate to contact the Support Staff on the Live Chat or Ticket System.
:➜ '''SMS Messages have a cost of $0.0075 per incoming and outgoing message.'''
:➜ '''MMS Messages have a cost of $0.02 per incoming and outgoing message.'''

: ➜ '''Additionally, please note of the following restrictions for sending MMS:'''
:: ➜ Permitted attachment types are JPG, GIF, JPEG, PNG, MP3, WAV, MIDI, MP4 and 3GP.
:: ➜ The maximum size for each attachment is 1300 KB. ''(1.3 MB)''
:: ➜ You can attach up to 3 files.
:: ➜ Maximum characters in a text message is 2048.

</div>

=Using the SMS/MMS=
==Identifying a DID Number SMS/MMS capable==

The first thing you need to know is that this feature is only available for local US '''(incl. Alaska, Hawaii and Puerto Rico)''' and Canadian numbers at this time. You will note that some numbers have a little [[File:Sms-mms-icon.png|15px]] icon, this indicates that the number supports this feature.

You can start ordering a DID Number from your portal >> DID Numbers >> Order DID >> Local Numbers >> (US or Canada) Numbers and you will be able to purchase your desired number, just make sure it supports the MMS feature.

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

==Configuring the SMS/MMS service==

Once purchased the SMS/MMS Service needs to be activated on the DID settings, from your portal >> DID Numbers >> Manage DID >> Edit Selection >> SMS/MMS configuration section.

In that section, you can activate and configure some forward options for the SMS/MMS service.

: [[File:SMSenable20240520.png|thumb|none|600px]]

:* '''Message Service (SMS/MMS)''': Check the box to enable the service.
:* '''VoIP.ms SMS/MMS Portal''': This is enabled by default and it can't be modified, that means that you will be able to create and send messages from the SMS Message Center.
:* '''SMS/MMS Email Address''': By activating this option all the SMS Messages will be sent to the email address you configure in this field. You can receive them to more than 1 email address by separating each emails with a comma. The advantage of this option is that you will receive your SMS Messages directly to your email and you will be able to Reply to these messages from your email too. You just need to click on '''reply''' to the email. Please note that e-mail responses are as well limited to 160 characters and if they exceed this limit, they'll be split in two (or more messages if applies) when sent.

<div style="margin-left:50px;
padding:5px;
border-left: 6px solid rgb(208, 56, 45, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(51, 51, 51, 0.08);
width:70vw;">

'''NOTE''' 💬: ➜ To reply the SMS Message via your email it is important to click on '''Reply''' and '''DON'T change or modify''' the destination mail "TO" (sms@voip.ms) and the subject E.G ''[#USXXXXXX] Message sent to 5555555555''. Make sure that your email address inbox isn't full to avoid loops when receiving an SMS/MMS message via email from an automated service.
</div>

:*'''SMS/MMS Forward''': You can have your SMS/MMS Messages forwarded to your cellphone or any other number that supports SMS/MMS, with this option you will receive the SMS/MMS Messages from our system to the configured phone number.
<div style="margin-left:50px;
padding:5px;
border-left: 6px solid rgb(208, 56, 45, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(51, 51, 51, 0.08);
width:70vw;">

'''NOTE''' 💬: ➜ The CallerID sent to the Call Forwarded Cell Phone will be the DID Number that received the SMS Message. Is not possible to forward the original Caller ID at this time. When using SMS/MMS Forward there is a charge for the incoming message received by the DID and the outgoing message sent to the number set for the forward.
</div>

:*'''SMS/MMS SIP Account''': You can have your SMS Messages forwarded to your selected SIP account. They will be sent through our SMS/SIP gateway as a SIP MESSAGE.
<div style="margin-left:50px;
padding:5px;
border-left: 6px solid rgb(208, 56, 45, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(51, 51, 51, 0.08);
width:70vw;">

'''NOTE''' 💬: ➜ The destination ("To" header field) of the SIP MESSAGE will be your account name, instead of the destination number of the original SMS.
</div>

:'''SMS/MMS URL Callback''': By enabling this option you will be able to send the messages to another destination e.g. your own server. If selected SMS Messages received by your DID will Send a GET request to the URL Callback provided. Available variables for your URL:
<div style="margin-left:50px;
padding:5px;
font-family: Consolas, Monaco, Lucida Console, Liberation Mono, DejaVu Sans Mono, Bitstream Vera Sans Mono, Courier New, monospace;
color: silver;
border-left: 6px solid rgb(208, 56, 45, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: black;
width:58vw;">
:: '''Example''': <nowiki>https://YourDomainName.tld/sms.php?to={TO}&from={FROM}&message={MESSAGE}&id={ID}&date={TIMESTAMP}</nowiki>

: '''{ID}''' = The ID of the SMS message.
: '''{TIMESTAMP}''' = The date and time the message was received.
: '''{FROM}''' = The phone number that sent you the message.
: '''{TO}''' = The DID Number that received the message.
: '''{MESSAGE}''' = The content of the message.
: '''{MEDIA}''' = Comma-separated list of media files.

</div>

:*'''SMS/MMS Webhook URL (3CX)''': If you have a 3CX system, you will need to fill this information. For more information, [[3CX_Phone_System#Configuring_VoIP.ms_SMS.2FMMS_service_with_your_3CX | please review this article]].

:*'''Dialing Mode for SMS (3CX/API)''': If you are using a 3CX system or VoIP.ms API, you can choose to send and receive SMS messages with the NANPA or E164 format.
::* '''E164''': To send a SMS or MMS with this format, you will need to include '''+''' sign and the country code for every messages
::* '''NANPA''': To send a SMS or MMS with this format, you only need to enter the 10-digit number without the country code.

:*'''URL Callback Retry''': When selected, we will be expecting an "ok" output (without quotes) from your URL callback page as an indicator that you have received the message correctly. If we don't received the "ok" (without quotes) from your callback page, we will keep sending you the same message every 30 minutes.

Once you configure the desired option, click on Apply Changes.

==Sending a SMS/MMS (Web Interface)==

'''Step 1'''

: Click the "send New Message" button to get a list of contacts displayed or directly type in the number into the contact text field. If this number is registered in the phonebook, it will display its name instead.

:: [[File:MMS-Webportal1.png|border|400px]]

'''Step 2'''

:Click to display the list of available DID numbers. Choose the number to send the message from.

:: [[File:MMS-Webportal3.png|border|400px]]

'''Step 3'''

:Write your message on the text field and click the red arrow to send it. Text messages up to 160 characters will be considered SMS. Media messages or text messages over 160 characters will be considered MMS and you will see the icon turning from SMS to MMS indicating of this.
:: [[File:MMS-Webportal4.png|border|400px]]

'''Step 4'''

: In order to send a media file over an MMS, click the clip button to browse and choose a media file.

:: [[File:MMS-Webportal5.2.png|border|400px]]

: Click the browse button to search for the media file in your computer or drag and drop the file directly into the box.

:: [[File:MMS-Webportal6.PNG|border|400px]]

: The attached files will show below the text box.

:: [[File:MMS-Webportal5.PNG|border|400px]]

: Once the message is sent, a new conversation will be opened on the left. Selecting this conversation will show the messages history on the box.

:: [[File:MMS_WINDOW_CONVO.png|border|400px]]

==Receiving an SMS/MMS (Web Interface)==

: While SMS messages can be received directly in an App through the SIP Protocol and correct configuration, at the moment, '''receiving and reading MMS is only possible through the message center.'''

: You can access the communication center from your customer portal at DID numbers > SMS/MMS Message Center. Any message received will appear here on the left side of the panel.

: Use the DID Filter to choose and display messages for a specific DID number or for all DID numbers.

: You can also filter the messages by date and time zone by clicking on the down arrow, which will show the search parameters

:: [[File:SMSMMSfilter1.png|border|300px]]

:: [[File:MMS-Inbound-1.png|border|500px]]

: Choose a conversation from the left panel. An option to delete the conversation is also available. Choose the DID number from the text box to reply.

:: [[File:MMS-Inbound_DELETE.png|border|500px]]

== SMS/MMS History ==

You can now also gain access to a general history of SMS/MMS that was sent and received for all of your DIDs, or for a specific one by going to Customer Portal >> DID Numbers >> SMS/MMS Message Center >> History.

:: [[File:SMSMMSHistoryLoc.png|border|500px]]

:: [[File:History.png|border|500px]]

= Email to SMS=

As of January 14th 2025, it is now possible to initiate a SMS conversation via email. To enable the function, simply head into your VoIP.ms portal, DID Numbers, SMS/MMS Message Center and then click on '''Email to SMS'''.

: [[File:EmailtoSMS1.png|border|500px]]

Then proceed on clicking on '''Add New Email to SMS''' and fill the following information:

* Enabled: Yes
* Email Address: Your email address
* Security Code Enabled: If you want to enter a security code for the SMS/MMS to be sent.
* Security Code: The security code to be used.
* Default From Number: The DID Number you want to use to send the SMS/MMS Message

: [[File:EmailtoSMS2.png|border|500px]]

Once all the information has been filled, press '''Save''' and you are good to go!

==How to iniate the SMS/MMS Conversation via email==

1. Use the email account you provided when enabling the Email to SMS service.

2. Send the email to sms@voip.ms.

3. In the subject field, enter the recipient's telephone number. (example: 5552340000).

4. Attach a document to the email message if you wish to send multimedia.

5. Permitted attachment types: JPG, GIF, JPEG, PNG, MP3, WAV, MIDI, MP4 and 3GP.

6. Maximum size for each attachment is 1300 KB.

7. You can attach up to 3 files.

8. Send the email.

===Security Code and From Number:===

1. If Security Code is enabled, you need to add a dot (.) and the Security Code after the Destination SMS Number (Example: 5552340000.Az09).

2. If you have more than one DID number, you could change the From Number by adding a dot (.) and the From Number you'd like to use after the Destination SMS Number and the Security Code (Example: 5552340000.Az09.5557890000).

3. If you have not activated the security code and wish to use another of your DIDs as the 'Sender number'. You must add a dot (.) after the phone number of the recipient of your SMS followed by the DID number you wish to use (example: 5552340000.5557890000).

= SMS using a SMPP Server =

An SMPP server, short of '''Short Message Peer to Peer''' is a method designed to send and receive high volumes of SMS messages through a dedicated server.

In other words, this allows you to better control the quantity of messages that will be sent to your target audience while avoiding to pass via a softphone application or your VoIP.ms portal to send/receive these SMS, thus eliminating multiple repetitive tasks and saving you time.

For example, if you have a promotion for your business that offers a discount for a certain lapse of time and that your customers have provided their agreement to receive these types of promotions via SMS, you could send your promotion to your customers in a faster manner than if you would do it from your customer portal, thus increasing efficiency.
<div style="margin-left:50px;
padding:5px;
border-left: 6px solid rgb(208, 56, 45, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(51, 51, 51, 0.08);
width:70vw;">

'''NOTES''' 💬: ➜ Note that sending and receiving SMS messages will be charged our usual rate at $0.0075 per messages'''
</div>

==Requirements==

: '''This will require you to have your own SMPP server.'''
: Once configured properly, you will then be able to send and receive SMS messages to your own server.

== Configuration on your SMPP Server ==

: In order for your server to communicate and authenticate accordingly with VoIP.ms SMPP server, you will need to configure the following:

::* A '''username''' and '''password''' of your choice in order for VoIP.ms SMPP server to authenticate you.
::* Have all your communications be sent to '''smpp.voip.ms''' via port '''2775''', for regular SMPP, or port '''3550''' which is our encrypted SMPP port.
::* Open a connection using the '''bind_transceiver''' command. You can keep this connection open for as long as you need to and inquire about it using the '''enquire_link''' command.
::* Send your messages with either '''deliver_sm''' or '''submit_sm''' commands. Make sure to include your PDU's '''source_addr''' as one of your SMS enabled DIDs from your account. Your destination SMS number should be set as '''destination_addr''' while your text message will be included in '''short_message'''.

== Configuration SMPP on your VoIP.ms Portal ==

Once you have set up the above on your server, you will need to enable and specify which username and password will be used by VoIP.ms to authenticate your SMPP requests. In order to do so, head into your VoIP.ms portal, DID Numbers, Manage DIDs, edit your DID that supports SMS messages and by heading down the page, you will see the following options:

:* '''SMPP Enabled:''' If selected, SMS Messages will be allowed to be sent and received using SMPP.
:: You may specify the URL to be used to send a copy of inbound messages to your SMPP server. Leave empty if you only wish to send messages. See the example below. The '''scheme''' part is mandatory and must be set to either of '''smpp''' for unencrypted SMPP, or '''ssmpp''' for encrypted SMPP using TLS. The '''user:password@''' part is optional and will default to the same username and password defined for authentication on our side as defined below. The ''':port''' part is also optional and will default to 2775 for '''smpp''' and to '''3550''' for ssmpp.

:* '''SMPP Username:''' The username that you will use to authenticate to our SMPP server for sending messages
:* '''SMPP Password:''' The password that you will use to authenticate to our SMPP server for sending messages

<div style="margin-left:50px;
padding:5px;
border-left: 6px solid rgb(208, 56, 45, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(51, 51, 51, 0.08);
width:70vw;">

'''NOTES''' 💬
: ➜ In order to help avoid any confusion while configuring the username on your server, you can use your VoIP.ms account ID as a username (the 6 digits of your account).
: ➜ Make sure to have the option '''Message Service (SMS/MMS)''' enabled.
</div>

Enable the option '''SMPP Enabled''', fill the username and password per what was configured on your SMPP server and once done, apply the changes by pressing '''Click here to apply changes'''.

: [[File:SMPPconfig.png|border|750px]]

You may now start sending and receiving SMS messages from your SMPP server.

= Send and Receive Text Messages (Web Interface)=

==SMS Message Center for mobile==

To start using the service you will need to use the SMS Message Center, from your portal >> DID Numbers >> [https://www.voip.ms/m/sms.php SMS Message Center].

For Mobile Devices please use [https://sms.voip.ms/ sms.voip.ms] as a complete all in one solution.

From the SMS Message Center you can check your SMS History, send and receive SMS messages.

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

:* '''Search Range''': Select a Date Range of your SMS History. You can select a time period by marking the "Show Details" field and the text messages will be displayed.
:* '''Search Filter''': You can filter your Search by DID, Contact and Type (Sent, Received and Both).
:* '''Send New SMS Messages''': Another window will open by clicking here in order to create a new SMS Message.
:* '''Delete Selected Messages''': This action will delete any messages that you have selected, by clicking on the box next to them. '''This action cannot be undone.'''
:* '''Delete All Messages''': Delete ALL your SMS Messages, both sent and received. '''This action cannot be undone.'''
:* '''Checking a Message''': To check a received SMS Message just click it directly in order to display the information.

==Create a New SMS Message (Legacy)==

: [[File:SMS41.png|thumb|none|600px]]

You can create a new '''SMS Message''' we will explain you all the parameters you can find on this section:

* '''DID''': If you have more than one DID Number you can select any of the numbers available to send your message.
* '''Contact''': The destination phone number or if you have a [[Phone book|Phone book entry]] start typing the name and the system will display the information if available.
* '''Message''': You can create your message on that field, please note that you have up to 160 characters.

Then click on 'Send Message' to send the SMS.

= Send and Receive Messages (via SIP MESSAGE Protocol) =
<div style="margin-left:50px;
padding:5px;
border-left: 6px solid rgb(208, 56, 45, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(51, 51, 51, 0.08);
width:70vw;">

'''NOTES''' 💬
: ➜ '''SMS via SIP''' service is only available through '''SIP Protocol and user/password authentication'''. '''IP Auth and IAX2 are not available at this time.'''
: ➜ '''Please refer to our guide [[SIP/SMS with FreePBX]] for instructions on how to configure the service with a FreePBX'''
</div>

== Sending ==

:* DID must have the '''"Short Message Service (SMS)"''' service option enabled. This option can be enabled from the Manage DIDs > Edit DID page.

:* To send an SMS from your SIP account, it is required that you set your Caller ID number to one of your SMS enabled DIDs. This will be the number you will be sending the message from. You can configure the Caller ID number from your customer portal for the specific subaccount if you are using a softphone or directly from your extension or trunk if you are using an Asterisk or PBX server.

:* It's important to note that if you configure your caller ID name in your SIP client to be a 10 digits number, this will override your caller ID number. If the caller ID name is anything different than a 10 digits number this will be discarded and the Caller ID number will be used.

:* If you are an asterisk or PBX user, please make sure to use the latest version of '''Asterisk (v12 or higher)''' and use '''chan_pjsip''' for the trunk. This is an asterisk limitation.

:* The Desktop version of Zoiper requires the user to have the '''PRO version''' activated to be able to send SMS messages as well as to enable '''SIP Presence'''. Please contact Technical Support to request SIP presence to be enabled for your account for the VoIP.ms POP you are using. While normally not required, there may be other cases not documented of softphones or apps requiring SIP Presence to be enabled. Please contact technical support if you encounter any issue with your specific software.

:* The '''free Desktop version''' of Zoiper cannot send SMS messages and is only capable of receiving SMS.

:* The '''mobile version''' of Zoiper is completely free of these limitations. Free version already works for sending and receiving without the need of SIP presence.

:* When sending an SMS message, the recipient number must be 10 digits. Avoid using the prefix '"1"' since is not supported at the moment. The system will recognize the numbers as in 10 digits.

:* The current characters limit for a '''single SMS is 160'''. If a message is bigger than 160 characters, it will not be sent.

== Receiving ==
<div style="margin-left:50px;
padding:5px;
border-left: 6px solid rgb(208, 56, 45, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(51, 51, 51, 0.08);
width:70vw;">

'''NOTES''' 💬: ➜ AT THE MOMENT SENDING MMS USING SIP ACCOUNT IS NOT POSSIBLE AT THIS TIME.'''
</div>

:*DID must have the '''"Message Service (SMS/MMS)"''' service option enabled. This option can be enabled from the Manage DIDs > Edit DID page.
:*Enable the option '''"SMS SIP Account"''' and select the account destination for your SMS messages to this number.
:*The account receiving SMS must be registered successfully in any of our POPs.
:*To ensure the best results receiving the SMS messages, do not register the same SIP account on more than one softphone or app at the same time, this can lead to unexpected results.
:*For SIP devices that require this information, such as a trunk or PBX, we encode the original destination number in a custom SIP header field named "X-SMS-To".
:*3CX softphone currently does not support any third party SMS service. It will only work by having a direct connection to the 3CX PBX system. '''Please contact 3CX for more details'''.

== Send SIP/SMS with FreePBX ==

<div style="margin-left:50px;
padding:5px;
border-left: 6px solid rgb(208, 56, 45, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(51, 51, 51, 0.08);
width:70vw;">

'''NOTES''' 💬: ➜ AT THE MOMENT SENDING MMS USING SIP ACCOUNT IS NOT POSSIBLE AT THIS TIME.'''
</div>

Yes, it is possible to configure a FreePBX system to send SMS, please refer [[SIP/SMS with FreePBX | here]] for detailed instructions.

=SMS/MMS 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 your customers the ability to leverage this.

Go under the navigation bar on '''[Reseller]''' then click on '''[Manage Rates & Packages]'''
: [[File:CallForwarding_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:CallForwarding_Reseller_2.png|thumb|none|700px]]

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

: [[File:CallForwarding_Reseller_3.png|thumb|none|700px]]

Then scroll down and find the feature "’’’SMS/MMS’’’", and enable it.

: [[File:SMSMMSreseller4.png|thumb|none|500px]]

= Enabling/Disabling SMS/MMS on a DID via the client's interface =
For your client to enable or disable SMS/MMS from his portal, your customer will need to head under Services, My Services.
: [[File:SMSMMSreseller5.png|thumb|none|500px]]

Then proceed by heading into ‘’’DID Lists’’’ tab and edit DID.

: [[File:SMSMMSreseller6.png|thumb|none|500px]]
You will then be able to enable SMS/MMS, receive via email and more.

: [[File:SMSMMSreseller7.png|thumb|none|500px]]

= Accessing their SMS/MMS via the clients interface =
1) To gain access to their SMS/MMS messages, head under Services and then SMS/MMS.
: [[File:SMSMMSreseller8.png|thumb|none|500px]]

= Configuring VoIP.ms SMS/MMS service with your 3CX =

==Prerequisites==
: 1. Ensure that you have the latest version of your 3CX PBX, including the updated version of the VoIP.ms trunk template.
: 2. Have a DID that supports SMS/MMS with the service enabled. ''(Navigate to Main Menu > DID Number > Edit DID)''
: 3. Generate a copy of your API Key, known as the "Bearer Token." [[#Generate_a_Bearer_Token|See this section]]
: 4. Follow the rest of the guide [[3CX_Phone_System#Configuring_VoIP.ms_SMS.2FMMS_service_with_your_3CX | Here]].

Softphones

2026-07-30T15:00:25Z

Neocypher:

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

A '''softphone''' is a software program for making telephone calls over the Internet using a general purpose computer, tablet or smartphone, rather than using dedicated hardware. Often designed to behave like a traditional telephone with a display panel and buttons for user interaction.

'''Learn More:''' [https://voip.ms/blog/softphone-benefits-for-business/ Back to Basics, What is a Softphone and How to Leverage It?]
|}

__TOC__

= Softphone Compatibility =

{| class="wikitable" style="width: 100%; margin: 20px 0; border-collapse: collapse; border: none; border-radius: 8px; overflow: hidden; box-shadow: 0 2px 8px rgba(0,0,0,0.06);"
! style="background: linear-gradient(135deg, #d0382d 0%, #b8261a 100%); color: white; padding: 18px 16px; font-weight: 700; text-align: center;" | '''Softphone'''
! style="background: linear-gradient(135deg, #d0382d 0%, #b8261a 100%); color: white; padding: 18px 16px; font-weight: 700; text-align: center;" | '''Android'''
! style="background: linear-gradient(135deg, #d0382d 0%, #b8261a 100%); color: white; padding: 18px 16px; font-weight: 700; text-align: center;" | '''iOS'''
! style="background: linear-gradient(135deg, #d0382d 0%, #b8261a 100%); color: white; padding: 18px 16px; font-weight: 700; text-align: center;" | '''Windows'''
! style="background: linear-gradient(135deg, #d0382d 0%, #b8261a 100%); color: white; padding: 18px 16px; font-weight: 700; text-align: center;" | '''macOS'''
! style="background: linear-gradient(135deg, #d0382d 0%, #b8261a 100%); color: white; padding: 18px 16px; font-weight: 700; text-align: center;" | '''Linux'''
! style="background: linear-gradient(135deg, #d0382d 0%, #b8261a 100%); color: white; padding: 18px 16px; font-weight: 700; text-align: center;" | '''SMS'''
! style="background: linear-gradient(135deg, #d0382d 0%, #b8261a 100%); color: white; padding: 18px 16px; font-weight: 700; text-align: center;" | '''TLS'''
|-
| style="padding: 16px; background: #ffffff; border-bottom: 1px solid #f1f5f9; font-weight: 600;" | [[VoIP.ms_Softphone|VoIP.ms Softphone]]
| style="padding: 16px; background: #ffffff; border-bottom: 1px solid #f1f5f9; text-align: center;" | ✅
| style="padding: 16px; background: #ffffff; border-bottom: 1px solid #f1f5f9; text-align: center;" | ✅
| style="padding: 16px; background: #ffffff; border-bottom: 1px solid #f1f5f9; text-align: center;" | ❌
| style="padding: 16px; background: #ffffff; border-bottom: 1px solid #f1f5f9; text-align: center;" | ❌
| style="padding: 16px; background: #ffffff; border-bottom: 1px solid #f1f5f9; text-align: center;" | ❌
| style="padding: 16px; background: #ffffff; border-bottom: 1px solid #f1f5f9; text-align: center;" | ✅
| style="padding: 16px; background: #ffffff; border-bottom: 1px solid #f1f5f9; text-align: center;" | ✅
|-
| style="padding: 16px; background: #fafbfc; border-bottom: 1px solid #f1f5f9;" | [[Acrobits|Acrobits]]
| style="padding: 16px; background: #fafbfc; border-bottom: 1px solid #f1f5f9; text-align: center;" | ✅
| style="padding: 16px; background: #fafbfc; border-bottom: 1px solid #f1f5f9; text-align: center;" | ✅
| style="padding: 16px; background: #fafbfc; border-bottom: 1px solid #f1f5f9; text-align: center;" | ❌
| style="padding: 16px; background: #fafbfc; border-bottom: 1px solid #f1f5f9; text-align: center;" | ❌
| style="padding: 16px; background: #fafbfc; border-bottom: 1px solid #f1f5f9; text-align: center;" | ❌
| style="padding: 16px; background: #fafbfc; border-bottom: 1px solid #f1f5f9; text-align: center;" | ✅
| style="padding: 16px; background: #fafbfc; border-bottom: 1px solid #f1f5f9; text-align: center;" | ✅
|-
| style="padding: 16px; background: #ffffff; border-bottom: 1px solid #f1f5f9;" | [[Bria Teams|Bria Teams]]
| style="padding: 16px; background: #ffffff; border-bottom: 1px solid #f1f5f9; text-align: center;" | ✅
| style="padding: 16px; background: #ffffff; border-bottom: 1px solid #f1f5f9; text-align: center;" | ✅
| style="padding: 16px; background: #ffffff; border-bottom: 1px solid #f1f5f9; text-align: center;" | ✅
| style="padding: 16px; background: #ffffff; border-bottom: 1px solid #f1f5f9; text-align: center;" | ✅
| style="padding: 16px; background: #ffffff; border-bottom: 1px solid #f1f5f9; text-align: center;" | ❌
| style="padding: 16px; background: #ffffff; border-bottom: 1px solid #f1f5f9; text-align: center;" | ✅
| style="padding: 16px; background: #ffffff; border-bottom: 1px solid #f1f5f9; text-align: center;" | ✅
|-
| style="padding: 16px; background: #fafbfc; border-bottom: 1px solid #f1f5f9;" | [[Bria Solo|Bria Solo]]
| style="padding: 16px; background: #fafbfc; border-bottom: 1px solid #f1f5f9; text-align: center;" | ✅
| style="padding: 16px; background: #fafbfc; border-bottom: 1px solid #f1f5f9; text-align: center;" | ✅
| style="padding: 16px; background: #fafbfc; border-bottom: 1px solid #f1f5f9; text-align: center;" | ✅
| style="padding: 16px; background: #fafbfc; border-bottom: 1px solid #f1f5f9; text-align: center;" | ✅
| style="padding: 16px; background: #fafbfc; border-bottom: 1px solid #f1f5f9; text-align: center;" | ❌
| style="padding: 16px; background: #fafbfc; border-bottom: 1px solid #f1f5f9; text-align: center;" | ✅
| style="padding: 16px; background: #fafbfc; border-bottom: 1px solid #f1f5f9; text-align: center;" | ✅
|-
| style="padding: 16px; background: #ffffff; border-bottom: 1px solid #f1f5f9;" | [[Jami|Jami]]
| style="padding: 16px; background: #ffffff; border-bottom: 1px solid #f1f5f9; text-align: center;" | ✅
| style="padding: 16px; background: #ffffff; border-bottom: 1px solid #f1f5f9; text-align: center;" | ✅
| style="padding: 16px; background: #ffffff; border-bottom: 1px solid #f1f5f9; text-align: center;" | ✅
| style="padding: 16px; background: #ffffff; border-bottom: 1px solid #f1f5f9; text-align: center;" | ✅
| style="padding: 16px; background: #ffffff; border-bottom: 1px solid #f1f5f9; text-align: center;" | ✅
| style="padding: 16px; background: #ffffff; border-bottom: 1px solid #f1f5f9; text-align: center;" | ✅
| style="padding: 16px; background: #ffffff; border-bottom: 1px solid #f1f5f9; text-align: center;" | ✅
|-
| style="padding: 16px; background: #fafbfc; border-bottom: 1px solid #f1f5f9;" | [[Linphone|Linphone]]
| style="padding: 16px; background: #fafbfc; border-bottom: 1px solid #f1f5f9; text-align: center;" | ✅
| style="padding: 16px; background: #fafbfc; border-bottom: 1px solid #f1f5f9; text-align: center;" | ✅
| style="padding: 16px; background: #fafbfc; border-bottom: 1px solid #f1f5f9; text-align: center;" | ✅
| style="padding: 16px; background: #fafbfc; border-bottom: 1px solid #f1f5f9; text-align: center;" | ✅
| style="padding: 16px; background: #fafbfc; border-bottom: 1px solid #f1f5f9; text-align: center;" | ✅
| style="padding: 16px; background: #fafbfc; border-bottom: 1px solid #f1f5f9; text-align: center;" | ✅
| style="padding: 16px; background: #fafbfc; border-bottom: 1px solid #f1f5f9; text-align: center;" | ✅
|-
| style="padding: 16px; background: #ffffff; border-bottom: 1px solid #f1f5f9;" | [[Microsip|MicroSIP]]
| style="padding: 16px; background: #ffffff; border-bottom: 1px solid #f1f5f9; text-align: center;" | ❌
| style="padding: 16px; background: #ffffff; border-bottom: 1px solid #f1f5f9; text-align: center;" | ❌
| style="padding: 16px; background: #ffffff; border-bottom: 1px solid #f1f5f9; text-align: center;" | ✅
| style="padding: 16px; background: #ffffff; border-bottom: 1px solid #f1f5f9; text-align: center;" | ✅
| style="padding: 16px; background: #ffffff; border-bottom: 1px solid #f1f5f9; text-align: center;" | ✅
| style="padding: 16px; background: #ffffff; border-bottom: 1px solid #f1f5f9; text-align: center;" | ✅
| style="padding: 16px; background: #ffffff; border-bottom: 1px solid #f1f5f9; text-align: center;" | ✅
|-
| style="padding: 16px; background: #fafbfc; border-bottom: 1px solid #f1f5f9;" | [[Softphone.Pro|Softphone.Pro]]
| style="padding: 16px; background: #fafbfc; border-bottom: 1px solid #f1f5f9; text-align: center;" | ❌
| style="padding: 16px; background: #fafbfc; border-bottom: 1px solid #f1f5f9; text-align: center;" | ❌
| style="padding: 16px; background: #fafbfc; border-bottom: 1px solid #f1f5f9; text-align: center;" | ✅
| style="padding: 16px; background: #fafbfc; border-bottom: 1px solid #f1f5f9; text-align: center;" | ✅
| style="padding: 16px; background: #fafbfc; border-bottom: 1px solid #f1f5f9; text-align: center;" | ❌
| style="padding: 16px; background: #fafbfc; border-bottom: 1px solid #f1f5f9; text-align: center;" | ✅
| style="padding: 16px; background: #fafbfc; border-bottom: 1px solid #f1f5f9; text-align: center;" | ❌
|-
| style="padding: 16px; background: #ffffff; border-bottom: 1px solid #f1f5f9;" | [[Zoiper_5|Zoiper]]
| style="padding: 16px; background: #fafbfc; border-bottom: 1px solid #f1f5f9; text-align: center;" | ✅
| style="padding: 16px; background: #fafbfc; border-bottom: 1px solid #f1f5f9; text-align: center;" | ✅
| style="padding: 16px; background: #fafbfc; border-bottom: 1px solid #f1f5f9; text-align: center;" | ✅
| style="padding: 16px; background: #fafbfc; border-bottom: 1px solid #f1f5f9; text-align: center;" | ✅
| style="padding: 16px; background: #fafbfc; border-bottom: 1px solid #f1f5f9; text-align: center;" | ✅
| style="padding: 16px; background: #fafbfc; border-bottom: 1px solid #f1f5f9; text-align: center;" | ✅
| style="padding: 16px; background: #fafbfc; border-bottom: 1px solid #f1f5f9; text-align: center;" | ✅
|}

= Popular Softphones =

{| cellspacing="0" style="width: 100%; background: white; border: 2px solid #d0382d; border-radius: 12px; padding: 25px; margin: 20px 0; box-shadow: 0 2px 10px rgba(208, 56, 45, 0.1);"
|-
| style="width: 300px; vertical-align: top; padding-right: 25px;" |
[[File:VoipmsLogoSymbolSquare.png|300px|thumb|center|VoIP.ms Softphone]]

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

'''Company:''' [https://www.voip.ms VoIP.ms]<br>
'''Platform:''' Android and iOS<br>
'''SMS Support:''' Yes<br>
'''TLS Support:''' Yes<br>
'''Overview:''' Official VoIP.ms mobile application with push notifications and SMS support.

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

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

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

'''Company:''' [https://www.acrobits.net//?utm_campaign=itsp-partners&utm_medium=cp-certified-logo&utm_source=voipms Acrobits]<br>
'''Platform:''' iOS, Android<br>
'''Overview:''' Mobile SIP client for iOS and Android devices.

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

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

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

'''Company:''' [https://www.acrobits.net/ Acrobits]<br>
'''Platform:''' iOS, Android<br>
'''Overview:''' Feature-rich SIP softphone from Acrobits with push notifications, call recording, background mode, and advanced codec support. Groundwire is the flagship consumer app from Acrobits, offering more configuration options than the standard Acrobits client.

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

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

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

'''Company:''' [https://www.counterpath.com/?utm_campaign=itsp-partners&utm_medium=cp-certified-logo&utm_source=voipms CounterPath]<br>
'''Platform:''' Windows, macOS, iOS, Android<br>
'''Overview:''' Desktop and mobile softphone application with call control features.

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

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

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

'''Company:''' [https://www.counterpath.com/?utm_campaign=itsp-partners&utm_medium=cp-certified-logo&utm_source=voipms CounterPath]<br>
'''Platform:''' Windows, macOS, iOS, Android<br>
'''Overview:''' Team communication softphone with unified messaging across multiple devices.

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

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

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

'''Company:''' Linphone<br>
'''Platform:''' Windows, Linux, macOS, iPhone, Android<br>
'''Overview:''' Open-source SIP softphone supporting voice, video, and instant messaging.

<div style="margin-top: 20px;">
[[Linphone|<span style="background: linear-gradient(135deg, #d0382d, #b8261a); color: white; padding: 10px 20px; border-radius: 20px; text-decoration: none; font-weight: 600; display: inline-block; box-shadow: 0 2px 10px rgba(208, 56, 45, 0.3); margin: 5px;">PC Config</span>]]
[[Linphone_iOS|<span style="background: linear-gradient(135deg, #d0382d, #b8261a); color: white; padding: 10px 20px; border-radius: 20px; text-decoration: none; font-weight: 600; display: inline-block; box-shadow: 0 2px 10px rgba(208, 56, 45, 0.3); margin: 5px;">iOS Config</span>]]
[[Linphone_Android|<span style="background: linear-gradient(135deg, #d0382d, #b8261a); color: white; padding: 10px 20px; border-radius: 20px; text-decoration: none; font-weight: 600; display: inline-block; box-shadow: 0 2px 10px rgba(208, 56, 45, 0.3); margin: 5px;">Android Config</span>]]
</div>
|}

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

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

'''Company:''' MicroSIP<br>
'''Platform:''' Windows, Linux, macOS<br>
'''Overview:''' Lightweight SIP client for Windows, Linux, and macOS.

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

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

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

'''Company:''' Vedisoft<br>
'''Platform:''' Windows, macOS<br>
'''Overview:''' Professional solution for call center agents, sales teams, and support staff with CRM integration.

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

{| cellspacing="0" style="width: 100%; background: white; border: 1px solid #e2e8f0; border-radius: 12px; padding: 25px; margin: 20px 0; box-shadow: 0 2px 10px rgba(0,0,0,0.06);"
|-
| style="width: 300px; vertical-align: top; padding-right: 25px;" |
[[File:Zoiper0.png|300px|thumb|center|Zoiper]]

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

'''Company:''' [https://www.zoiper.com/en Zoiper]<br>
'''Platform:''' Windows, macOS, Android, iOS, Linux<br>
'''Overview:''' Multi-platform SIP and IAX softphone with support for multiple languages.

<div style="margin-top: 20px;">
[[Zoiper_Classic|<span style="background: linear-gradient(135deg, #d0382d, #b8261a); color: white; padding: 10px 20px; border-radius: 20px; text-decoration: none; font-weight: 600; display: inline-block; box-shadow: 0 2px 10px rgba(208, 56, 45, 0.3); margin: 5px;">Classic</span>]]
[[Zoiper_5|<span style="background: linear-gradient(135deg, #d0382d, #b8261a); color: white; padding: 10px 20px; border-radius: 20px; text-decoration: none; font-weight: 600; display: inline-block; box-shadow: 0 2px 10px rgba(208, 56, 45, 0.3); margin: 5px;">Zoiper 5</span>]]
[[Zoiper_5_Pro|<span style="background: linear-gradient(135deg, #d0382d, #b8261a); color: white; padding: 10px 20px; border-radius: 20px; text-decoration: none; font-weight: 600; display: inline-block; box-shadow: 0 2px 10px rgba(208, 56, 45, 0.3); margin: 5px;">Zoiper 5 Pro</span>]]
[[ZoIPer_for_Android|<span style="background: linear-gradient(135deg, #d0382d, #b8261a); color: white; padding: 10px 20px; border-radius: 20px; text-decoration: none; font-weight: 600; display: inline-block; box-shadow: 0 2px 10px rgba(208, 56, 45, 0.3); margin: 5px;">Android</span>]]
[[ZoIPer_for_iOS|<span style="background: linear-gradient(135deg, #d0382d, #b8261a); color: white; padding: 10px 20px; border-radius: 20px; text-decoration: none; font-weight: 600; display: inline-block; box-shadow: 0 2px 10px rgba(208, 56, 45, 0.3); margin: 5px;">iOS</span>]]
</div>
|}

= Legacy Softphones =

{| cellspacing="0" style="width: 100%; background: linear-gradient(135deg, #fef3c7 0%, #fde68a 100%); border: 1px solid #f59e0b; border-radius: 8px; padding: 25px; margin: 25px 0; box-shadow: 0 2px 10px rgba(245, 158, 11, 0.1);"
|-
| style="text-align: left;" |
<div style="font-size: 1.1em; font-weight: 600; color: #92400e; margin-bottom: 10px;">⚠️ The following softphones are included for reference but may have limited support, discontinued development, or platform compatibility issues:</div>

* '''[[VoIP_by_Antisip|AntiSIP]]''' - Android-only SIP client with limited recent updates
* '''[[Blink|Blink]]''' - Desktop softphone for macOS and Linux with basic features
* '''[[Dialer+|Dialer+]]''' - iOS-only softphone with limited modern features
* '''[[Ekiga_Softphone|Ekiga]]''' - Windows/Linux softphone with dated interface
* '''[[Express_Talk|Express Talk]]''' - Softphone with dated interface and limited updates
* '''[[FgVoIP|FgVoIP]]''' - BlackBerry-only client (BlackBerry platform discontinued)
* '''[[Fring|Fring]]''' - Multi-platform client with reduced VoIP focus
* '''[[Jitsi|Jitsi]]''' - Desktop softphone with focus shifted to web conferencing
* '''[[Kiax|Kiax]]''' - IAX2-only client for Asterisk systems
* '''[[NetDial_Sip_Phone|NetDial]]''' - Mobile SIP client with limited features
* '''[[MizuDroid|MizuDroid]]''' - Softphone with dated interface and limited updates
* '''[[MizuPhone|MizuPhone]]''' - Softphone with dated interface and limited updates
* '''[[MizuPhone_Classic|MizuPhone Classic]]''' - Softphone with dated interface and limited updates
* '''[[MizuPhone_Modern_Edition|MizuPhone Modern]]''' - Softphone with dated interface and limited updates
* '''[[NinjaLite|NinjaLite]]''' - Windows-only softphone with dated interface
* '''[[PhonerLite|PhonerLite]]''' - Windows-only basic SIP client
* '''[[PortGo|PortGo]]''' - Windows desktop and mobile client
* '''[[QuteCom|QuteCom]]''' - Multi-platform client with limited development
* '''[[SessionTalk|SessionTalk]]''' - Mobile SIP client for Android/iOS
* '''[[SIP_Droid|SIP Droid]]''' - Android-only open source client
* '''[[Sipnetic|SIPnetic]]''' - Android-only VoIP client
* '''[[SJPhone|SJPhone]]''' - Windows/Windows Mobile client (Windows Mobile discontinued)
* '''[[Taki|Taki]]''' - BlackBerry-only client (BlackBerry platform discontinued)
* '''[[Telephone|Telephone]]''' - macOS-only minimalist SIP client
* '''[[TSIP|tSIP]]''' - Windows-only lightweight client
* '''[[X-Lite|X-Lite]]''' - Multi-platform client with limited free features
* '''[[Zoiper_Communicator|Zoiper Communicator]]''' - Desktop client superseded by Zoiper 5

For new deployments, we recommend choosing from the popular softphones listed above.
|}

[[Category:Softphones]]

ATA Devices

2026-07-30T14:59:14Z

Neocypher:

Caller ID

2026-07-30T14:52:55Z

Neocypher: /* Blog Article */

{| class="wikitable"
|+
|-
! Article en Français !! Artículo en Español
|-
| [https://wiki.voip.ms/article/ID_de_l%27appelant Français] ||
[https://wiki.voip.ms/article/Numero_Identificador_(Caller_ID) Español]
|}

== Blog Article ==

<div style="position: relative; width: 90%; height: 20px;
background: linear-gradient(to right, #f8f9fa, rgba(255, 0, 0, 0));
border: 1px solid transparent;
border-image: linear-gradient(to right, #dee2e6, rgba(255, 0, 0, 0));
border-radius: 8px; padding: 12px; overflow: hidden; display: flex; align-items: left; justify-content: left; text-align: left; font-family: sans-serif; color: #333;">
<div style="position: absolute; font-size: 100px; opacity: 0.15; z-index: 1; pointer-events: none; user-select: none; transform: rotate(-10deg);">📰</div>
<p style="position: relative; z-index: 2; margin: 0; line-height: 1.5; background: none !important;">

📃 [https://voip.ms/blog/what-is-cnam-and-how-to-leverage-it-for-your-business/ What Is CNAM and How to Leverage It for Your Business?]

</p>
</div>


= Caller ID =
: Caller ID is a telephone service that transmits the calling party´s number to the called party´s telephone . When available, the Caller ID number can be complemented with Caller ID name (e.g., John Smith).
: If you are placing outgoing calls, you will need to pass a Caller ID to ensure proper termination of your calls.
: There are two types of caller ID and it is important to differentiate them: Caller ID Number (CID) and Caller ID Name (CNAM).
: Please note that the Caller ID is only guaranteed when using Premium routes, and only for US48 and Canadian calls. In Canada you may find Caller ID working on some Value routes.
: Incoming Caller ID (from people calling you) is addressed below.

__TOC__

= Caller ID Rules =
: In line with all the rulings in favor of protecting and empowering consumers, VoIP.ms enforces valid Caller IDs as per the NANPA and ITU-T E.164 standards.
: Therefore, calls displaying the following Caller ID information will be rejected:

: * International calls with Caller ID containing '''less than 7 digits''' and '''greater than 15 digits''';
: * North America calls with Caller ID not containing exactly '''10 digits''' ''(exception will be made for 7 digits for 310-xxx numbers)'';
: * North America calls with Caller ID containing '''10 digits''', but with an unassigned NPA (first 3 digits of the number);
: * North America calls with Caller ID containing '''10 digits''', but where an unassignable NXX is used (i.e. the second block of 3 digits where the first digit is either zero (0) or one (1)).

== Outgoing Caller ID Name ==
: The Caller ID name is an additional information you can pass along with your Caller ID number. This will be received on the callee's end and it could be your given name or the name of your business.
:: For example: ''"John Smith" <9145551234>''

: The sample above is a Caller ID that includes both Caller ID name and Caller ID number, commonly abbreviated as CID and CNAM among other variations.

: It is not possible to set a Caller ID name from the VoIP.ms portal.

: If you plan to make calls to Canadian numbers, you can simply pass the Caller ID name from your device or system as most of them support this.'''*''' <br/>
: You will need to check for a field on the interface from the device to enter this setting, and in case you are using a more advanced system, get assistance to set the outgoing Caller ID name set up.

: The Caller ID name on US calls works differently, this is controlled by a national CNAM database with records of numbers and names matching each number.
: When you make a call to a US number, you will send a Caller ID number and the system will check the CNAM database for a name matching that same Caller ID number in order to display both name and number to the final phone.

: CNAM is only available for some USA numbers. In order to update your Caller ID'''*''' on the CNAM database for your US calls, there is a process to follow which has a cost of $10 USD (one time only fee).
: Please contact the VoIP.ms support staff to get further details on what information you need to submit and to confirm if your local US DID is available for a CNAM update.

:: '''CNAM update is only available for some Local US DIDs. Toll frees cannot have their Caller ID name updated'''.

<div style="border-left: 6px solid blue; background-color: lightgrey; width: 85%;">
'''IMPORTANT''':
- 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.
- Do not exceed '''15 characters''' max! Some of regular Canadian providers will not show more than '''15 characters'''. We suggest shrinking or adapt your caller ID.
- Spaces are allowed in a caller id name.
</div>

== Outgoing Caller ID Number ==
: Caller ID number is the most common Caller ID type passed. If a more complex system capable of passing its own Caller ID is being used, such as a [[Welcome#PBX|PBX]], the Caller ID field is likely set from the trunk, or one of its extensions.
: If you are using devices like [[Welcome#Devices|Analogue Telephone Adapters]], [[Welcome#Devices|IP phones]], or [[Welcome#Softphones|softphones]], the Caller ID number is available to be set from your VoIP.ms account via the customer portal.

== Outbound Caller ID - Main account ==
: To set the Caller ID number for your Main account, access:

:: '''1'''. Form your VoIP.ms Portal, go to "[[Account Settings]]" from '''Main Menu''' -> '''Account settings'''
::: [[File:AccountSettingsMenu.png|thumb|none|400px|Account settings]]
:: '''2'''. Go to '''General''' Tab and there set the '''CallerID number'''
::: [[File:VoIPms_AccountSettings_General_CallerID_Number.png|thumb|none|400px|General]]

== Outbound Caller ID - Sub Accounts ==
:The Caller ID number for a sub account can be set during the creation process, or later by clicking Edit on the [[Sub Accounts|subaccount]].
::[[File:CallerIDSubAcc.png|thumb|none|700px|CallerID field in a sub account]]

: * '''For calls to US and Canada, a 10-digit Caller ID number is required to ensure proper call termination. The portal's Caller ID field only supports numerical digits.'''
: * '''Toll free Caller IDs are not recommended, especially when Calling Toll free numbers, to reduce potential connection issues.'''

: If you are using an analog telephone adapter (ATA), IP phone, or softphone, this is where you need to adjust the settings.
: It is important to ensure that you transmit a valid caller ID to ensure proper termination.

: Here you will have the option of having one of your incomings DID numbers displayed as an outgoing call identification number.
: You can also use a personalized number that does not appear in your lists of DID numbers.

: However, you must respect the standards in force by the regulations to this effect.
: '''Always use a number that belongs to you''' or to that '''you have permission to use''' as the outgoing call display number.

: '''I use a system of capable of passing its own CallerID Number''': If you have a PBX that manages the outgoing call identification number, you will have an option for this.

:: ''Note: Using a Toll free outbound caller id is not recommended, especially when calling Toll-Free numbers.''

=Different methods of passing your Caller ID Number=
'''CallerID Number''': You can set the [[Caller ID|callerID Number]] or your DID Number or proceed with a verification process to pass the callerID of a number that is not within VoIP.ms network. If you have a a device capable of passing its own [[Caller ID|CallerID number]] such as a soft switch or [[PBXs|PBX]], make sure that the callerID matches a DID or a verified number from within your account to successfully make outgoing calls.

*'''Use one of my DIDs''': If you want to pass one of your DID Numbers as the callerID Number, select this option and choose the DID, if more than one, that will display on the called party.

* '''Use a Verified CallerID''': This is a quick verification process to ensure that the callerID you will pass is validated and authorized to be used. To start the process:

: '''1'''. Simply click on '''Click here to get started''' above the CallerID Number setting
::: [[File:VoIPms_AccountSettings_General_Number_Verification_Section.png|none|650px]]
: '''2'''. Follow the CallerID Verification Process
:: '''1'''. Select the country of your DID and Enter the number you want to verify.
:: '''2'''. You can also add a reference/note for your own use.
:: '''3'''. Choose the verification to be done by receiving a call or by receiving a SMS (only available for Canada/United States).
::: [[File:VoIPms_Non-VoIPms_Number_Verification.png|none|650px]]
: '''3'''. Once you press '''Send verification Code''', you will receive a phone call / SMS message with a validation code.
:: Enter that code in the '''Validation Code''' field and press '''Verify'''.
::: [[File:VoIPms_Non-VoIPms_Number_Verification_Code.png|none|650px]]

:: > If for any reasons you have not received the code, you can ask the system to send it again or press cancel to change the method you would like to receive the code.

: * '''I use a system capable of passing its own CallerID''': If you are using a PBX, you will need to pass the callerID directly from your system.
:: Our system will verify if the callerID you are passing matches one of your DID or one of your verified callerID.
:: If you are passing an invalid callerID and attempt to make an outgoing call, our system will send you an email advising that a call was attempted with an invalid callerID.

==Outbound caller ID number using Call Forwarding==
: When you chose to forward all your incoming calls from your VoIP.ms number to an external phone number, the original's caller ID number will be transmitted to this external number.
: However, if the incoming caller is hiding their phone number this call will fail due to the new regulations for CID, on where sending a valid caller ID number is mandatory.

: A workaround is to create a filter for '''"Anonymous"''' calls along with another call forwarding entry but this one using the '''"Caller ID number override"'''.
: This way, incoming calls using a valid CID will be transmitted to your external number showing the original's CID and incoming calls hiding their number will use the filter and the forward with caller ID Override, showing the CID override number and completing the call.

: '''To receive "Anonymous" calls when using call forwarding, follow these steps:'''
:: '''1'''. Go to '''DID Numbers>> Call forwarding''' section and create a new call forwarding entry to the same number you're forwarding to, but this new entry will contain the '''"Caller ID override"'''. For the caller ID override, you can type your VoIP.ms number or any valid number you want to see when you receive an anonymous call. You might want to fill the '''"Description field"''' so you can identify this entry on the next step.
:: '''2'''. Thereafter, go to '''DID Numbers>> CallerID filtering''' section and create a new filter.
::: * For step 1, mark '''"Anonymous CallerID number"''' checkbox
::: * For step 2, this is optional you can choose to apply this filter to all your numbers or just to a specific one.
::: * For Step 3, mark the '''"Call forwarding"''' checkbox and make sure to use the call forwarding entry with the caller ID override number.
::: * Save your filter.

== Incoming Caller ID number and name ==

: You will receive the Caller ID number and Caller ID name that the VoIP.ms server receives from the caller, this is exactly what will be sent to you on Incoming calls.
: You can always check what Caller ID number VoIP.ms receives by going into your [[Call Detail Records]] to check the incoming calls.

: The incoming Caller ID name works almost the same way, except that this is an optional setting that you need to enable per DID number on the [[Manage DID|DID settings]] page. This option is called "CallerID Name Lookup". : When enabled, the system will perform a query on the LIBD/CNAM Database, for callers with Canadian or US CID number, in order to find a name matching that CID number. The system then will display the result of this query in the Caller ID name portion of the '''Caller ID''', leading to a "Caller ID name"<5551231234> when people call your number.

: If the calling number is already in your [[phone book]], the name will be taken from there instead of doing a CNAM lookup on an external database.

:: '''IMPORTANT NOTE FOR CANADIAN DIDs'''
:: The majority of the Canadian DID numbers support CNAM Pass-Through.
:: This means that for your incoming calls the system '''won't do a CNAM query''' (and also not charge you) if the incoming call already has a Caller ID name, even if the DID receiving the call has the CNAM queries enabled in your VoIP.ms customer portal.
----

== Listen to the current account's Caller ID Number ==
: To listen to the caller ID set in the account you're dialing from, please dial '''822''' and the Caller ID number will be read back.
: Notice that it will depend on your device's dial plan or dial pattern for this to work.
: These test calls won't have any cost for you and will appear in your [[Call_Detail_Records | CDR]] as "CallerID Testing".

==Configuring your CallerID with the Reseller Portal==
: Once within the reseller portal, head into Services, My Services to see the callerID options.
: The options available are the same as your main account or sub accounts.
:: [[File: CallerIDverif_Reseller.png |thumb|none|600px]]

=Notes=

<div style="position: relative; width: 90%; /* height: 20px; */ background-color: #f0fbff; border: 2px solid #2f6185; border-radius: 8px; padding: 12px; overflow: hidden; display: flex; align-items: left; justify-content: left; flex-wrap: wrap; text-align: stretch; font-family: sans-serif; color: #333;">
<div style="position: absolute; margin-top:35px; margin-left: 20px; font-size: 150px; opacity: 0.1; z-index: 1; pointer-events: none; user-select: none; transform: rotate(-20deg);">ℹ️</div>
<p style="position: relative; z-index: 2; margin: 0; line-height: 1.5;">

:'''Outgoing Caller ID is not guaranteed on calls to Canadian cellular numbers, even when using the Premium route.'''
:'''This is due to the way Canadian carriers work - they sometimes pass a random Caller ID that they have on record, changing the original.'''
:'''This is out of our control as it is the way Canadian carriers handle calls to cellular numbers.'''
:'''There are also issues with incorrect Caller ID being sent on outbound [[toll-free number]] calls from Bell Mobility (and its resellers);'''
:'''the display shows a disconnected Bell number in an area code corresponding to the caller's location instead of the caller's mobile number.'''
</p></div>

[[category:guides]]

Sub Accounts

2026-07-30T14:51:19Z

Neocypher: /* Blog Article */

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

Having a Sub Account allows you to register more than one device to make or receive calls simultaneously, you can also use it as an internal extension for your office or even your house. Many of the features within voip.ms make use of sub accounts. With this guide we are going to learn how to create and use this feature properly.

__TOC__

== Blog Article ==

[https://voip.ms/blog/what-is-ip-authentication-and-how-it-works/ How to Use IP Authentication with a Sub Account]

== How to Create a Sub Account ==

You can create as many Sub Accounts as required and there's no extra charge for doing this. First you need to enter your Customer Portal and select "Create Sub Account" under the Sub Account Menu.

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

===Basic Options ===

These are the basic options you need to configure in order to properly create a Sub Account.

'''Protocol''': Here you can select either SIP or IAX2 protocol. This will depend on the device you want to use with this Sub Account. SIP is the recommended protocol.

'''Authentication type''': This is an option that you don't have with your main account. You can choose between '''User/Password Authentication''' or '''IP Authentication'''. Again this setting will depend on the device you're using.

-- '''User/Password Authentication''': This is the recommended setting and most devices only support this option.

-- '''IP Authentication''': Recommended for advanced users. Mostly this is used for PBX or Asterisk-based servers. When you select this option you can set the IP of your device/server.

'''Note:''' When you use IP Authentication the registration status will not be displayed in your Customer Portal: Home Page.

'''Username''': This is the username of the sub-account. You can use alphanumeric characters for the username (make sure your device supports the use of alphanumeric characters in the user field). The format of the username will be as follow: '''{Main Account}_{username}'''.

For example, let say that your account is ''100000'' and you set the username as ''home'' the username that you're going to use in your device will be ''100000_home''.

'''Password/IP Address''': Here you can set the password to use for this sub account. The minimum is 6 characters, although it's strongly suggested that you use more characters and also use alphanumeric characters to create a strong password.

If you're using IP Authentication here you can set the IP address of your device/server. Format e.g. 201.202.203.204.

[[File:IP AUTH.jpg|thumb|none|700px]]

'''Trunk''': This option only becomes available if IAX2 is the chosen protocol. You can leave the default to send mini voice packets, or select "Send trunk packets" to group outgoing media frames into trunk packets, each with their own timestamps.

[[File:Trunk.jpg|thumb|none|700px]]

'''Device type''': Make sure to select the correct type of device you're going to use with this subaccount to properly receive incoming calls.

'''CallerID Number''': If you are using an analog telephone adapter (ATA), IP phone, or softphone, this is where you need to adjust the settings. It is important to ensure that you transmit a valid caller ID to ensure proper termination. Here you will have the option of having one of your incoming DID numbers displayed as an outgoing call identification number. You can also use a personalized number that does not appear in your lists of DID numbers. However ''you must respect the standards in force by the regulations to this effect. Always use a number that belongs to you or to that you have permission to use as the outgoing call display number.'' If you have a PBX that manages the outgoing call identification number, you will have an option for this. In order to be able to define a name in output display, generally this will have to be set from your ATA / PBX / Softphone device, etc. ''Nota: Using a Toll free outbound caller id is not recommended, especially when calling Toll-Free numbers.''

'''Canada Routing''': This defines the route the system will use when you place a call to Canada with your Sub Account.

'''International Route''': You can select the route that is going to be used for International Calls from this sub-account.

'''Allow International Calls''': With this setting, you can enable or disable the International Calls from the sub-account.

'''Allow *225 for Balance''': When Enabled, calls placed to *225 will provide the Current Balance of the VoIP.ms account. When Disabled calls placed to *225 will be rejected.

'''Music on Hold''': This setting allows you to select whether or not the person calling your DID number or extension will hear Music while the call is on hold.<br/>
The following options for music are available:<br/>

{| class="wikitable" border: none;"
! colspan="3" | Music on Hold
|-
! scope="col" | Test Code
! scope="col" | Name
! scope="col" | Description
|-
| '''***89''' || Away in the Tropics || From Hawaii to the Caribbean, these tracks deliver sounds of ukulele, steel drums, and steel guitars.
|-
| '''***90''' || Coffee and Sunrise || Uplifting without being perky, and positive without being too smiley.
|-
| '''***91''' || Coffee Shop Acoustic || Soothing, acoustic guitar tracks makes for a relaxed feel and sets a great atmosphere.
|-
| '''***92''' || Easy Listening || Smooth, casual tunes.
|-
| '''***93''' || Guitar Alchemy || Clever harmonics and progressive chord sequences to create a joyful and warming musical experience.
|-
| '''***94''' || Happy Endings || Uplifting, commercial style. Guitar, drums, ukulele, harmonica and bells.
|-
| '''***95''' || Light and Casual || Soothing and peaceful songs with a light, positive feeling.
|-
| '''***96''' || Orchestral Moods || Emotional and dramatic tales spun by violins, pianos and full orchestras.
|-
| '''***97''' || Piano Mix || Smooth Piano
|-
| '''***98''' || Rock me Easy || Feel-good music to create a relaxing atmosphere.
|-
| '''***99''' || Spa Sounds || Soft, slow and serene instrumentals.
|-
| '''***100''' || Aucune musique, bip intermittent || Silent, but with a subtle intermittent bleep sound to let your callers know that they are still on the line while they hold.
|}

'''Account name or description''': This setting can help you to easily identify each sub-account.

=== Advanced Options ===

These options allow you to set the allowed codecs, DTMF modes and NAT for each subaccount. These settings are recommended for advanced users only.

[[File:Advsettingssub.png|800px]]

*'''Allowed Codecs:''' This setting allows you to select the codecs that you're going to use, you can choose to allow one or more codecs at the time.

'''G.722 Codec'''

The '''codec G.722 or VoIP HD audio''' is a wideband audio codec that operates at a high data sampling rate. The higher sampling rate allows the G.722 codec to provide higher clarity of audio signals than G.711.<br>
Codec G.722 and HD voice is '''available''' and fully supported when performing SIP calls '''where both ends of the calls have codec G.722 configured''' and will also work for internal calls between customers within VoIP.ms '''as long as they are in the same POP server''', system recordings or messages.<br>

Codec G.722 is currently not supported for communication between different POP/servers.

Due to the nature of its technology and the current limitations with the existing regular phone providers (not supporting G.722), '''HD voice is currently not available''' when calling external regular phone lines or mobiles.

*'''DTMF Mode:''' This allows you to select the DTMF mode that is going to be used with this sub account. If you set this to ''AUTO'' the ''RFC2833 (AVT)'' is going to be used and automatically switch to ''INBAND'' if the other end doesn't support ''RFC2833''.
'''Note:''' Its recommended that you select the same DTMF mode in your device.

*'''NAT (Network Address Translation):''' This setting should be set to ''Yes'' if you're behind a NAT, if not set to ''No''. If you're unsure what this setting means, is highly recommended that you leave it to ''Yes''.

*'''Encrypted SIP Traffic''': This setting allows you to encrypt the communication between your device and our server, by using the SIP-TLS ''(Transport Layer Security)'' and SRTP ''(Secure Real-Time Transport Protocol)'' protocol.

Before using this feature, please consult the article about the [[Call Encryption - TLS/SRTP]]

'''Note:''' If enabled, all SIP traffic calls will be encrypted for this sub account.
''' ''' Please note that if encrypted calls are enabled then you need to configure your device to make and receive encrypted calls.

*'''Max Expiry:''' Sets the '''maximum''' amount of time (in seconds) until a device or phone system registration expires. The default value is 3600 but the range can go from 60 to 3600 seconds. If your device or phone system requests a lower registration expiry time to the server, this will be respected. MaxExpiry also applies to subscriptions, Message-Waiting Indicators (MWI) as well as Presence (BLF).

*'''RTP Time Out:''' Sets the amount of time (in seconds) of '''no RTP activity (silence)''' before the call is terminated when the call is '''NOT''' on Hold. If the field is left empty the default value of 60 seconds will be used. RTP time out can range from 1 to 3600 seconds but it must always be equal or lower than RTP Hold Time Out.

*'''RTP Hold Time Out:''' Sets the amount of time (in seconds) of '''no RTP activity (silence)''' before terminating a call that is '''On-Hold'''. If the field is left empty the default value of 600 seconds will be used. RTP Hold Time Out can range from 1 to 3600 seconds but it must always be equal or greater than RTP Time Out.

Note: '''Max Expiry, RTP Time Out, and RTP Hold Time Out''' settings are available for the main account and subaccounts independently.

=== Optional Settings ===

Although these are optional settings, you can use these settings to assign an internal extension number, voicemail and Ring Time for your sub account.

[[File:subaccountoptional2.png|800px]]

'''Internal Extension Number''': You set an internal extension number in order to call between Sub Accounts under the same Main Account. The extension number you enter in this field will have a leading 10, you can enter from 1 to 10 digits. For example if you set ''55'' the extension number will be ''1055''.
Note: Make sure that the sub accounts are registered to the same server in order to make an internal call. The call to an internal extension is FREE.

'''Enable Internal CallerID / Internal CallerID Name:''' Allows you to set a specific callerID Name that will be seen only on internal calls between you and your sub account users. Additionally, your extension number will show on your internal calls instead of your 10digit callerID number.

==== Sub Account as an External SIP URI ====

To use a sub account as an external [[SIP URI]], you only need to enable it as an Internal Extension first. For example, let's say your Internal Extension is set to 2 (102 with the leading 10), you can be reached directly via SIP from another network with a URI that is going to look like this: ''1000002@server.voip.ms'' (Replace server.voip.ms by the server you are registered to, and the 2 by your internal extension).

'''Internal Extension VoiceMail''': Here you can select which [[voicemail]] is going to be associated with this sub account. Please set this to be able to access a particular Voice Mailbox using *97 from the device registered with this sub account's credentials, even if you are not using Extensions.

'''Note:''' If you associate a [[voicemail]] with this subaccount, you are going to receive a '''Message Waiting Indicator''' when you have new messages in your mailbox.
This will lead to different results depending on your type of adapter, soft phone or IP phone.
For example, when using a Linksys ATA adapter, it's usually a stutter tone with periodic ring, IP phones usually use a stutter dial tone
when you pick up the line and are equipped with a blinking light, soft phones usually show a Voicemail Icon in the dial display.

'''Internal Extension Ringing Time''': This is the amount of time the phone will stay ringing when you call this internal extension directly. 5 sec = 1 ring.

==== call pick-up ====

Call Pick-up is a feature that allows a device registered under one sub-account to answer an incoming call intended for a device registered under a different sub-account.

'''For example:'''
If both Alex and Bob are receiving calls on their respective devices and Bob needs to step away, Alex can answer Bob’s incoming call directly from his own device, ensuring no calls are missed.

This functionality is particularly useful in shared work environments, such as call centers or reception areas, where team members may need to support one another by handling calls on each other’s behalf.

For more information on how to use Call Pickup, consult this [[Call_Pickup | article]].

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

=== Security ===

We provide some additional settings to further enhance the security of your systems with our service.

[[File:IP_Restrict_En.jpg|thumb|none|700px]]

'''IP Restriction''': Enabling this option will provide you with an extra security layer where outgoing calls will be authorized only from the IP address or IP range specified.

'''POP Restriction''': Enabling this option will provide you with an extra security layer where outgoing calls will be authorized only from the Points of Presence (POPs) selected.

=== Reseller Configuration ===

Also if you're using the [[Reseller Basic Guide|Reseller Interface]], you can associate each sub account with one of your customers.

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

'''Reseller Client''': Here you can select the customer that you want to associate with this subaccount. You need first to create the account of your customer using the [[Reseller Basic Guide|Reseller section]] in your Customer Portal.

'''Reseller client package''': Here you can select the package that you want to assign to your customer. You need first to create a package in the [[Reseller Basic Guide|Reseller section]].

'''next billing date''': Here you can set the next billing date. Usually the system sets this automatically, but if you manually change the package associated to this subaccount, you can change the next billing date manually.

'''Charge setup fees now''': Once you check this option, the monthly fee for the package is going to be charged. You can use this option when you have applied a change in the package of the customer.

== Sub Account Bulk Creation ==

You can also create in bulk sub accounts from your customer portal. To do so, head to your customer portal, Sub Account, Manage Sub Accounts.
Once there, on the top right of the page there is the option '''Import Sub Account'''. Click on it, download the template, fill the template and upload it. This will automatically create the sub accounts for you based on the template uploaded.

== Sub Account Registration ==

You can check your sub accounts registration at the Main Page of the Customer Portal. Your sub accounts will appear listed under '''Sub Account Registration Status''' and you will be able to search or filter them using the '''Sub-Account Filter''' searchbox.

[[File:SubaccountRegistration.png|800px]]

== Sub Account Reports ==

You can see the amount of minutes, number of calls and even the amount spent for each sub account. This information is useful even if you're not using the [[Reseller Basic Guide|Reseller Interface]]. You can access this from your Customer Portal>>Sub Accounts>>Sub Accounts Reports menu.

[[File:Subacc report.jpg|800px]]

'''Report Range''': You can display a report with a range of 92 days only (3 months).

'''Minutes''': The number of minutes used by this sub account in the given range. Expressed using decimal time.

'''Calls''': The number of calls made for this sub account in the given range.

'''Amount Spent''': This is the amount spent for this sub account in the given range.

== Use of the subaccount ==

Once you have created one subaccount, you can use it with most of the features available within your Customer Portal, for example with the [[Digital Receptionist (IVR)]], as an agent to receive calls from [[Calling Queues]], Routed directly to your DID numbers, as an external [[SIP URI]] to receive incoming calls from another networks, etc.

[[category:guides]]

Emergency Services

2026-07-30T14:19:45Z

Neocypher: /* Article blog */

{| class="wikitable"
|+
|-
! Article en Français !! Artículo en Español
|-
| [https://wiki.voip.ms/article/Service_E911 Français] ||
[https://wiki.voip.ms/article/Servicios_de_Emergencia Español]
|}

== Article blog ==
[https://voip.ms/blog/enhanced-911-and-voip/ Enhanced 911 and VoIP]

== Emergency Services ==

For customers residing in <b>United States</b>, when you make an emergency call, VoIP.ms will attempt to automatically route your call through a third-party service provider to the Public Safety Answering Point ("PSAP") corresponding to your address of record on your account.

For customers residing in <b>Canada</b>, your emergency call will be directly sent to an emergency call center confirming your identity and location, and then immediately transferred to the local PSAP.

For customers residing <b>outside United States or Canada</b>, telecommunication regulators generally provide that all Customers who are using VoIP services as their primary residential or business telephone carrier must activate Emergency Services on at least one of their DIDs.

For emergency services address information to be passed to your local PSAP dispatcher, you must set your outbound CallerID value to the specific DID you are purchasing emergency service for.

Due to the various dependencies of VoIP networks, VoIP.ms cannot and does not guarantee your emergency call will go through. Many conditions such as loss of power, Internet access or connectivity and/or several other conditions may cause emergency services to be inoperable.

Emergency Services have a one-time fee <b>starting at $ 1.50 and a monthly fee starting at $ 1.50 per month per DID number activated per month</b>. For complete pricing, visit: https://www.voip.ms/residential/pricing

VoIP.ms has added an extension to its network where all VoIP.ms users may call to test their CallerID. At any time, you may test your outbound CallerID by dialing ’1-555-555-0911’ through VoIP.ms’ network and the system will playback your CallerID, then make a short pause, and play the test result. <b>As a reminder, if it is not correctly configured, your call will not go through</b>.

For all the details, refer to our Terms of Service: https://www.voip.ms/terms-of-service



You can activate this service at your Customer Portal >> DID Numbers >> Emergency Services.

'''Important Note:'''
<span style="color:red">'''Note that if your DID from Voip.ms is not subscribed to the Emergency Services, attempting to make a call to 911 will result in a
'''busy signal, since your callerID (which should be a valid DID or number) won't be in Voip.ms Database. This will result by not'''
'''reaching the desired destination. Also make sure to read the Terms and Condition on this page.'''</span>

''' <span style="color:red"> Update as of October 2021 </span> '''

Due to new regulations for Emergency Services, softphones on mobile devices will start using the native mobile dialer to place calls to the emergency services. '''It is possible that not all softphone devices will provide this functionality and it is highly suggested to come into close communication with the softphone provider for further questions.'''

[[File:Emergency Services option.png|thumb|none|600px]]

Use of Emergency Services costs a recovery setup fee of $ 1.50 on activation and a regulatory recovery fee of $ 1.50 per DID number activated per month. VoIP.MS does not make a cent on this charge, it is simply what must be paid to provide this service.

Please note you can enable this service only for Canadian or US numbers (including USA or Canadian toll free numbers).
You just have to click on the checkbox button in order to start the process to enable the service for this number.

[[File:Enable emergency services.png|thumb|none|600px]]

After this you can verify the Terms & conditions regarding of this service. Just please read them carefully and type "I agree" if you wish to enable the emergency services.

Click here to read Terms & conditions regarding this service. [[E911#VoIP.ms_911.2Fe911_Emergency_Service:_Terms_of_Service|Emergency services TOS]]

A new window will request some required information that will be used for the emergency services. Just fill the required fields and be sure to use the correct address. After this, click on the Validate button and if everything is fine, you will receive an email confirmation.

[[File:required.JPG|thumb|none|600px]]

Once your emergency service is active, you have to set your DID number as your outgoing [[Caller ID]] number for your main account or [[Sub Accounts]] you are going to use to dial out.

You can set the [[Caller ID]] number for your main account at your Customer Portal >> Main Menu >> [[Account Settings]] >> General tab >> "[[Caller ID]] number" option.

If you are going to use a [[Sub Accounts|sub account]] to dial out, you can set the [[Caller ID]] for it inside your Customer Portal >> [[Sub Accounts|sub account]] >> Manage [[Sub Accounts|sub account]] >> Edit >> [[Caller ID]] number.

To test that your [[Caller ID]] is working properly, you can Dial 1-555-555-0911 from VoIP.ms Network. The system will playback your [[Caller ID]], then make a short pause, and play the test result. To make sure your device/switch is correctly configured for emergency services, you must ensure that your [[Caller ID]] number matches exactly the DID number that is activated for emergency services. That is the only way to identify you correctly. Otherwise your 911 call will not go through. Correctly formatted [[Caller ID]] consist of a 10 digits number, identical to your DID. For example, if your DID is 555.555.1234, your [[Caller ID]] number should be 5555551234.

If at any time you need to change your address, you just need to select the "Modify" icon (The one with the pencil and paper). After this you will be able to change your emergency services information and after the information has been approved, you will be confirmed by email and the emergency services will be updated.

[[File:E911Enable.png|thumb|none|600px]]

== Unverified Emergency Calls (E911) ==

We’ve introduced an important update to help keep you safe. From now on, 911 calls will connect from any number within the U.S. or Canada, even if E911 service isn’t configured.

This change ensures that you can reach emergency services in critical situations, no matter your setup.

Please note that a $75 fee applies per 911 call attempts if E911 is not active on your number. We strongly recommend enabling E911 to avoid this fee and ensure your location is automatically shared with emergency responders.

== VoIP.ms 911/e911 Emergency Service: Terms of Service ==

EMERGENCY COMMUNICATIONS

The Company wants to make sure that Customers are aware of important differences in the way Emergency Services operate when using VoIP services when compared with traditional telephone service. Please find below what Customers need to keep in mind.  

'''Routing of Emergency Calls'''
For United States residents, when a Customer makes an emergency call, the Company will attempt to automatically route its call through a third-party service provider to the Public Safety Answering Point (“PSAP”) corresponding to the Customer’s address of record on its account. However, the delivery of the Customer’s physical location to its local PSAP is not guaranteed. It is possible that the Customer’s location will not be provided to the PSAP dispatcher On such occasions, it will be the Customer’s sole responsibility to give the dispatcher its name, location (or location of the emergency) and contact information to receive emergency service assistance. For Canadian residents, a Customer’s emergency call will be directly sent to an emergency call center confirming its identity and location, and then immediately transferred to the local PSAP.

'''Limitations Due to VoIP Networks'''
Due to the various dependencies of VoIP networks, the Company cannot and does not guarantee a Customer’s emergency call will go through. Many conditions such as loss of power, Internet access or connectivity and/or several other conditions may cause emergency services to be inoperable. The Company does not have control over those types of situations and therefore cannot be held liable of such inoperability. The Company will take commercially reasonable measures to prevent service outages within its network.  

'''Outbound CallerID'''
For emergency services address information to be passed to a Cusomter’s local PSAP dispatcher, the Customer’s outbound CallerID value must be set to the specific DID it is purchasing emergency service for. Therefore, by agreeing to these Terms, a Customer is deemed to have set the outbound CallerID number to the DID it has enabled emergency services for when making an outbound emergency call.  The Company has added an extension to its network where all Customers may call to test their CallerID value. At any time, a Customer may test its outbound CallerID value by dialing ‘1-555-555-0911’ through the Company’s network.  

'''Limitations on Emergency Services'''
CUSTOMERS UNDERSTAND THE LIMITATIONS OF THE COMPANY’S EMERGENCY SERVICES AND ASSUMES ALL LIABILITY AND RESPONSIBILITY, AND RELEASES THE COMPANY TO SUCH EXTENT, FOR THE USE OF EMERGENCY SERVICES,AND FURTHER AGREES TO HOLD THE COMPANY, ITS OFFICERS, DIRECTORS, EMPLOYEES AND AGENTS HARMLESS FOR ANY DAMAGE, WHETHER DIRECT OR INDIRECT THAT MAY RESULT FROM : (1) THE EMERGENCY SERVICES PROVIDED BY THE COMPANY (INCLUDING BUT NOT LIMITED TO SITUATIONS OF UNAVAILABILITY OF EMERGENCY SERVICES AS DESCRIBED IN THESE TERMS AND INCOMPLETE OR INCORRECT LOCATION INFORMATION PROVIDED BY THE CUSTOMER); (2) CUSTOMER’S FAILURE TO OBTAIN ACCESS TO CONVENTIONAL EMERGENCY SERVICE AS PART OF A TELEPHONE LINE SUBSCRIPTION FROM ANOTHER COMPANY UNDER SEPARATE AGREEMENT; (3) CUSTOMER’S FAILURE OR DELAY IN UTILIZING CONVENTIONAL EMERGENCY SERVICE. CUSTOMERS WHO RESELL THE SERVICES FURTHER AGREE THAT THEY ARE RESPONSIBLE FOR NOTIFYING, AND AGREE TO NOTIFY, THEIR CUSTOMERS, CONTRACTORS, AGENTS, EMPLOYEES, ASSOCIATES, SHAREHOLDERS, PARTNERS, AND ANY OTHER POTENTIAL USER OF THE COMPANY’S SERVICES OF THE NATURE AND LIMITATIONS OF THE EMERGENCY SERVICES.  IF A CUSTOMER IS NOT COMFORTABLE WITH THE LIMITATIONS OF EMERGENCY CALLS, THE CUSTOMER MUST CONSIDER AN ALTERNATE MEANS FOR ACCESSING TRADITIONAL EMERGENCY SERVICES, AS REGISTRATION TO EMERGENCY SERVICES IS MANDATORY IN MOST COUNTRIES. FURTHERMORE, THE CUSTOMER AGREES THAT THE COMPANY HAS NO LIABILITY IN RELATION TO THE QUALITY OF THE ADVICE AND SERVICES PROVIDED BY A PSAP.

==Pricing for Emergency Services==

{| class="wikitable"
|- style="font-weight:bold; vertical-align:bottom;"
! Country
! One-time fee
! Recurring Fee
|-
| style="vertical-align:bottom;" | Australia
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | Austria
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | Belgium
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | Bulgaria
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | Canada
| $ 1.50
| $ 1.50
|-
| style="vertical-align:bottom;" | Croatia
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | Cyprus
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | Czech Republic
| $ 4.50
| $ 2.50
|-
| style="vertical-align:bottom;" | Denmark
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | Estonia
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | Finland
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | France
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | Germany
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | Greece
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | Hungary
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | Ireland
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | Italy
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | Latvia
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | Lithuania
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | Netherlands
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | New Zealand
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | Norway
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | Poland
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | Portugal
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | Romania
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | Slovakia
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | Slovenia
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | Spain
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | Sweden
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | United Kingdom
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | United States
| $ 1.50
| $ 1.50
|}

=Using 911 using the Reseller Interface=

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

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

Go under the navigation bar on '''[Reseller]''' then click on '''[Manage Rates & Packages]'''
: [[File:e911_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:e911_Reseller_2.png|thumb|none|700px]]

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

: [[File:e911_Reseller_3.png|thumb|none|700px]]

Then scroll down and find the feature "'''e911'''", and enable it.

: [[File:e911_Reseller_4.png|thumb|none|500px]]

1) To add a e911 entry for your client, or to help your client adding one. Go under the '''[Services]''' at the left navigation bar, then on '''[e911]'''
: [[File:e911_Add.png|thumb|none|400px]]

2) In the row where the wished DID, click on the [ENABLED] button.
: [[File:e911_Add2.png|thumb|none|600px]]

3) Your client '''must''' read and '''Agreed the Terms and Conditions'''.

4) Complete the form and click the '''[Validate]''' button.

[[category:guides]]

Porting a Number

2026-07-30T14:17:10Z

Neocypher: /* Blogs */

{| class="wikitable"
|+
|-
! Article en Français !! Artículo en Español
|-
| [https://wiki.voip.ms/article/Transf%C3%A9rabilit%C3%A9_des_DID Français] ||
[https://wiki.voip.ms/article/Porteo_de_numeros Español]
|}

== Tutorial Video ==

[[Image:PortThumbnail.png|200px|link=https://www.youtube.com/watch?v=vnVXPP1S9l0]]

== Blogs ==

[https://voip.ms/blog/porting-phone-number-voip-ms/ How To Successfully Port Your Number to VoIP.ms and Avoid Common Pitfalls?]

== Port a number ==

Porting a number is the process of transfer a number from one provider to another. If you are a [http://voip.ms/ VoIP.ms] customer and have an existing number with a different provider you can optionally bring it (port it in) to [http://voip.ms/ VoIP.ms] network. Toll free numbers, local USA/Canada and even International numbers can be ported to [http://voip.ms/ VoIP.ms]. As of January 15th, 2020, all portings are free of charge across Canada and US48.

Our LNP (Local Number Portability) Department handles the porting process and guides the customer through it. They are also in charge of updating the customer via email with any relevant information about the port every time there is a change of the port request status. (Such as when the information has been submitted to the carrier, when a porting date -known as FOC- has been received, when the port has completed, etc)

To port your number over, please verify with your existing service provider that the address listed in your Customer Service Record (CSR)/Account information/Customer portal is a physical address. '''PO BOX addresses are not valid for portability.'''

'''Note:''' For Wireless Ports in Canada, some Carriers may send either an SMS or email to the owner regarding the approval for the current port request. Once you receive this notice, you have 90 minutes to reply and approve the port, otherwise, the losing carrier may reject the port request and cancel the process.

== Numbers portable ==

We can port numbers from most countries all over the world, even if we do not offer DID numbers from those countries. If you wish to verify the portability of an International number please contact our LNP staff and provide the Country code, Area code and phone number, it is important to also mention if the number is a National, Geographical or Toll-Free number.
You may email us at ports@voip.ms
For Faxes, we can port from Canada, United States local numbers as well as toll free numbers. The porting fee per number is $15 USD.

For more information about prices you can contact [http://voip.ms/ VoIP.ms] staff, since the rate can vary depending on the country your number belongs to.

== International Port IN Fees ==

Please contact VoIP.ms LNP Department at [mailto:ports@voip.ms ports@voip.ms] to confirm availability and final price for your International numbers. Remember to provide the full number along with the city name and city code. The upstream carrier name is also needed.

Prices are subject to change without notice at any time.

== Check Availability ==

Not all numbers from a specific country can be ported in. The most important factor to find out if a number is portable or not is the number´s city (also called ratecenter). In your customer portal, refer to DID Numbers section, you will find a link for DID Portability. In this link you can verify if your number is portable to our network, just click on the link that says "Check availability".

[[File:Port.jpg|thumb|none|600px]]

Important: For Canadian, US and Toll-free Numbers the porting fee is free per number in order to port to our network, the process can takes around 5 working days before everything is completed.

== Start Procedure ==

Once you have verified your number can be ported, on the same link (DID Portability) click on "Start procedure" you will be presented with a process of 5 steps to initiate the port of your number.

[[File:Steps.jpg|thumb|none|600px]]

: Important: Make sure you have a scan of your latest invoice/statement in PDF, MS Word or as an Image file, SIGNED on a blank part.
: Do not sign it over any lines or graphics.

After following all the steps in the process and confirmed your order, you will receive an email notification from our LNP Department, that will guide you during the entire process and provide all the necessary updates. If any additional information is required they will contact you, so you can be informed all the time.

Here you can find a detailed explanation of each step you have to follow in order to start a port request:

* [[Porting a US Number]]
* [[Porting a Canadian Number]]
* [[Porting a Toll Free Number]]

== Notification on Portal ==

Once you have placed a porting order, you will notice a notification next to DID Numbers Menu and DID Portability. This is to help you remember an ongoing port is happening and to pay attention for any updates coming from VoIP.ms LNP team.

[[File:PortNotification2024.png|thumb|none|600px]]

== Port Rejection ==

If there is a problem with your porting process, the LNP department will notify you via email on the following steps to take to address this problem. You can read more about port rejections and the most common reasons here, [[Port Rejection]].

== Internal Port (VoIP.ms to VoIP.ms) ==

Do you currently have a phone number (DID) in your account that you want to move to a different account? You can request this transfer through our portal's ticket system. If you need further help, please get in touch with our support team.

== Important Notes to Remember ==

Some important requirements you need to know before starting a port request:

*We are not responsible for any Port Out Fee your carrier may charge when leaving them, as it is beyond our control. Prior to starting a porting order with us, please contact the losing carrier and ask them about any port out fee they may charge you.

*Make sure to provide the exact information as it is on the losing carrier's '''Customer Service Record'''. This is usually your Service Address. Please note that the service address CAN be different from your billing address. Billing addresses will usually get rejected if it's not the same as your Service Address. If you are not sure of what is the correct service information, please contact your current service provider to obtain the Customer Service Record before proceeding.

*During a regular porting process your number must be working with the losing provider, which means you need to also make sure your number is active and not scheduled for cancellation, until the port gets completed. Disconnected numbers are not portable.

*If you decide to port in your number, you will need to make sure that you have a balance above $0. Otherwise, our system will not allow you to start the process until the balance is in positive standing.

*It is the customer's responsibility to contact the losing provider '''after''' the number has been completely ported to VoIP.ms in order to have the number and/or other services disconnected from them.

*It is important to remember that, if the initial request comes back rejected due to needing additional information or some error, the process time starts again after it has been resubmitted.

*Once you receive a FOC, for a number being ported, that number is now treated as if it´s already on our network. The Number will show in your available DID list and you can configure it. You must configure the DID to be able to reach it via the VOIP.ms network.

''Canadian and USA numbers are activated on per minute plan once ported. If you would rather have it activated on flat rate, please mention it in the Notes/Comments field, on the step 5 of the port request.''

== What about Porting Out? ==

If you wish to port out a number from our service you can do so at any time by starting a port request with the new provider you will use. As for the information you will need, you can request it via email at [mailto:ports@voip.ms ports@voip.ms] from the email address associated with your account, just let us know the number you will port.

Canadian local and Toll-Free port out numbers are authorized as long as the information matches with what is in your Contact Information section and you have a positive balance.
Please avoid deleting the number since in most cases the number is not ported yet and if you delete it, you may lose it. Once the number is ported out, the LNP staff will remove it from your account.
We are only notified once the number has ported away so we can remove it. We have no insight into an order prior to that. Please let us know if there is a specific rejection that you have received and we may be able to help further.

This is the same for US port outs since we are not a CLEC, we are not advised when a number is either rejected or ported out. Please open up a ticket to our LNP Department if there is a specific rejection that you have received. Also, make sure you delete the number after the new carrier confirms it has been properly ported and tested with them.

Please open up a ticket to our LNP Department if you get a rejection when porting out your number.
[[category:guides]]

Porting a Number

2026-07-30T14:16:35Z

Neocypher: /* Blogs */

{| class="wikitable"
|+
|-
! Article en Français !! Artículo en Español
|-
| [https://wiki.voip.ms/article/Transf%C3%A9rabilit%C3%A9_des_DID Français] ||
[https://wiki.voip.ms/article/Porteo_de_numeros Español]
|}

== Tutorial Video ==

[[Image:PortThumbnail.png|200px|link=https://www.youtube.com/watch?v=vnVXPP1S9l0]]

== Blogs ==

[[https://voip.ms/blog/porting-phone-number-voip-ms/ How To Successfully Port Your Number to VoIP.ms and Avoid Common Pitfalls?]]

== Port a number ==

Porting a number is the process of transfer a number from one provider to another. If you are a [http://voip.ms/ VoIP.ms] customer and have an existing number with a different provider you can optionally bring it (port it in) to [http://voip.ms/ VoIP.ms] network. Toll free numbers, local USA/Canada and even International numbers can be ported to [http://voip.ms/ VoIP.ms]. As of January 15th, 2020, all portings are free of charge across Canada and US48.

Our LNP (Local Number Portability) Department handles the porting process and guides the customer through it. They are also in charge of updating the customer via email with any relevant information about the port every time there is a change of the port request status. (Such as when the information has been submitted to the carrier, when a porting date -known as FOC- has been received, when the port has completed, etc)

To port your number over, please verify with your existing service provider that the address listed in your Customer Service Record (CSR)/Account information/Customer portal is a physical address. '''PO BOX addresses are not valid for portability.'''

'''Note:''' For Wireless Ports in Canada, some Carriers may send either an SMS or email to the owner regarding the approval for the current port request. Once you receive this notice, you have 90 minutes to reply and approve the port, otherwise, the losing carrier may reject the port request and cancel the process.

== Numbers portable ==

We can port numbers from most countries all over the world, even if we do not offer DID numbers from those countries. If you wish to verify the portability of an International number please contact our LNP staff and provide the Country code, Area code and phone number, it is important to also mention if the number is a National, Geographical or Toll-Free number.
You may email us at ports@voip.ms
For Faxes, we can port from Canada, United States local numbers as well as toll free numbers. The porting fee per number is $15 USD.

For more information about prices you can contact [http://voip.ms/ VoIP.ms] staff, since the rate can vary depending on the country your number belongs to.

== International Port IN Fees ==

Please contact VoIP.ms LNP Department at [mailto:ports@voip.ms ports@voip.ms] to confirm availability and final price for your International numbers. Remember to provide the full number along with the city name and city code. The upstream carrier name is also needed.

Prices are subject to change without notice at any time.

== Check Availability ==

Not all numbers from a specific country can be ported in. The most important factor to find out if a number is portable or not is the number´s city (also called ratecenter). In your customer portal, refer to DID Numbers section, you will find a link for DID Portability. In this link you can verify if your number is portable to our network, just click on the link that says "Check availability".

[[File:Port.jpg|thumb|none|600px]]

Important: For Canadian, US and Toll-free Numbers the porting fee is free per number in order to port to our network, the process can takes around 5 working days before everything is completed.

== Start Procedure ==

Once you have verified your number can be ported, on the same link (DID Portability) click on "Start procedure" you will be presented with a process of 5 steps to initiate the port of your number.

[[File:Steps.jpg|thumb|none|600px]]

: Important: Make sure you have a scan of your latest invoice/statement in PDF, MS Word or as an Image file, SIGNED on a blank part.
: Do not sign it over any lines or graphics.

After following all the steps in the process and confirmed your order, you will receive an email notification from our LNP Department, that will guide you during the entire process and provide all the necessary updates. If any additional information is required they will contact you, so you can be informed all the time.

Here you can find a detailed explanation of each step you have to follow in order to start a port request:

* [[Porting a US Number]]
* [[Porting a Canadian Number]]
* [[Porting a Toll Free Number]]

== Notification on Portal ==

Once you have placed a porting order, you will notice a notification next to DID Numbers Menu and DID Portability. This is to help you remember an ongoing port is happening and to pay attention for any updates coming from VoIP.ms LNP team.

[[File:PortNotification2024.png|thumb|none|600px]]

== Port Rejection ==

If there is a problem with your porting process, the LNP department will notify you via email on the following steps to take to address this problem. You can read more about port rejections and the most common reasons here, [[Port Rejection]].

== Internal Port (VoIP.ms to VoIP.ms) ==

Do you currently have a phone number (DID) in your account that you want to move to a different account? You can request this transfer through our portal's ticket system. If you need further help, please get in touch with our support team.

== Important Notes to Remember ==

Some important requirements you need to know before starting a port request:

*We are not responsible for any Port Out Fee your carrier may charge when leaving them, as it is beyond our control. Prior to starting a porting order with us, please contact the losing carrier and ask them about any port out fee they may charge you.

*Make sure to provide the exact information as it is on the losing carrier's '''Customer Service Record'''. This is usually your Service Address. Please note that the service address CAN be different from your billing address. Billing addresses will usually get rejected if it's not the same as your Service Address. If you are not sure of what is the correct service information, please contact your current service provider to obtain the Customer Service Record before proceeding.

*During a regular porting process your number must be working with the losing provider, which means you need to also make sure your number is active and not scheduled for cancellation, until the port gets completed. Disconnected numbers are not portable.

*If you decide to port in your number, you will need to make sure that you have a balance above $0. Otherwise, our system will not allow you to start the process until the balance is in positive standing.

*It is the customer's responsibility to contact the losing provider '''after''' the number has been completely ported to VoIP.ms in order to have the number and/or other services disconnected from them.

*It is important to remember that, if the initial request comes back rejected due to needing additional information or some error, the process time starts again after it has been resubmitted.

*Once you receive a FOC, for a number being ported, that number is now treated as if it´s already on our network. The Number will show in your available DID list and you can configure it. You must configure the DID to be able to reach it via the VOIP.ms network.

''Canadian and USA numbers are activated on per minute plan once ported. If you would rather have it activated on flat rate, please mention it in the Notes/Comments field, on the step 5 of the port request.''

== What about Porting Out? ==

If you wish to port out a number from our service you can do so at any time by starting a port request with the new provider you will use. As for the information you will need, you can request it via email at [mailto:ports@voip.ms ports@voip.ms] from the email address associated with your account, just let us know the number you will port.

Canadian local and Toll-Free port out numbers are authorized as long as the information matches with what is in your Contact Information section and you have a positive balance.
Please avoid deleting the number since in most cases the number is not ported yet and if you delete it, you may lose it. Once the number is ported out, the LNP staff will remove it from your account.
We are only notified once the number has ported away so we can remove it. We have no insight into an order prior to that. Please let us know if there is a specific rejection that you have received and we may be able to help further.

This is the same for US port outs since we are not a CLEC, we are not advised when a number is either rejected or ported out. Please open up a ticket to our LNP Department if there is a specific rejection that you have received. Also, make sure you delete the number after the new carrier confirms it has been properly ported and tested with them.

Please open up a ticket to our LNP Department if you get a rejection when porting out your number.
[[category:guides]]

Porting a Number

2026-07-30T14:09:03Z

Neocypher:

File:Time destination.jpg

2026-06-17T16:22:42Z

Neocypher: uploaded a new version of "File:Time destination.jpg"

Destination for the Time conditions.

File:Time destination.jpg

2026-06-17T16:16:07Z

Neocypher: uploaded a new version of "File:Time destination.jpg": Reverted to version as of 16:09, 17 June 2026

Destination for the Time conditions.

File:Time destination.jpg

2026-06-17T16:14:04Z

Neocypher: uploaded a new version of "File:Time destination.jpg"

Destination for the Time conditions.

File:Time destination.jpg

2026-06-17T16:09:30Z

Neocypher: uploaded a new version of "File:Time destination.jpg"

Destination for the Time conditions.

File:Routing.png

2026-06-17T15:39:14Z

Neocypher: uploaded a new version of "File:Routing.png": Reverted to version as of 16:13, 24 June 2015

Manage DID

2026-06-17T15:37:25Z

Neocypher: /* Edit a single DID number */

__TOC__

{| class="wikitable"
|+
|-
! Article en Français !! Artículo en Español
|-
| [https://wiki.voip.ms/article/Gestion_des_DID Français] ||
[https://wiki.voip.ms/article/Configuraci%C3%B3n_de_DID_(Manage_DID) Español]
|}

The '''Manage DID''' section is where you can see listed your DID numbers, check and edit their routes (where the call will be routed) and other options as well like enable/disable [[SMS | SMS]], [[Voicemail#Assigning_your_Voicemail_to_your_DID | assign Voicemail]]'s to them, set their fail over options, etc.

Once you have DID numbers in your account, you will have to edit their default settings in order to receive/forward/route the incoming calls the way you need to. To do this you need to access the "'''Manage DID(s)'''" option from the "'''DID Numbers'''" menu tab.

[[File:MenuOption.jpg|thumb|none|600px]]

== DID's information ==

At this part you can see the general information for the DID numbers (once you have [[Order a DID Number | ordered]] or [[Porting a Number | ported ]] a DID number in)

Here you will see different keywords and icons that you may not be familiar with, however you can get a quick description of each one of them by clicking on the "'''Help'''" icon at the top right of the page. There is also the option to ''Export Account DIDs'', this will create a .txt file of your DID numbers and their rate centers.

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

== Export and Back-Up DID Configuration ==

From Manage DID numbers page, you can also export a CSV file containing the entire list of the DIDs in the account and all their respective configuration parameters and values. This is particularly useful to always have a back-up of the configuration in case of an accident with the settings while configuring the numbers or if you simply want a complete list of all the numbers in your account.

Look for the '''"Export Account DIDs"''' option and click on it.

[[File:Exportdid.png|600px]]

Next, select '''"New Export"''' option. This will automatically generate and download a CSV file to your computer.

[[File:Newexport.png|600px]]

If you want to upload a the file with the DID configuration at a later time, you can do so by clicking on the '''"Update DIDs"''' button.

[[File:Uploaddid.png|600px]]

== Edit a single DID number ==

If you need to access the options to edit a single DID, the easiest way is to click on the little "'''Paper and pencil'''" icon.

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

The first and Main section you will see, is the "'''DID Routing'''" section, which may or may not have all the options available, this simply depends on which features you have already enabled from your portal, that is, if you have not created a [[Voicemail]], you wont be able to select this option and apply it to a DID.

This is basically the route the call will take, when someone calls to the DID. Whatever you select here, is the route the call will follow and will apply for all the calls.

In order to select an option, you need to click on the radio button, and then select the desired choice from the drop-down list (in case you have more than one option for that routing).

[[File:Routing.png|thumb|none|800px]]

== Failover Options ==

[[File:Failover.png|800px]]

*You can also click the '''Show Failover Options''' button, which will display 4 more DID routing sections. These Failover options will trigger an alternative routing option for your incoming calls depending on the state of the DID number.

'''Busy''': Choose a new routing for your calls in case your phone device is returning a busy signal.

'''Unreachable''': Choose a new routing for your calls in case your phone device is unreachable (Unregistered, registration issue, firewall and any other cause that would make the device to be unreachable).

'''No Answer''': Choose a new routing for your calls in case you are not answering the incoming call.

'''Max Capacity''': When enabled, you can choose a custom amount of maximum incoming callers for your number and send any call over that limit to an alternative route.

== DID's Point of Presence (POP) ==

*'''DID point of presence''', this setting is the server where you will locate your DID, in order to route the calls, this setting must match the same [[Choosing Server|server or proxy]] you are using on your [[Welcome#Devices|ATA device]], [[Welcome#PBX|PBX]] trunk or [[Welcome#Softphones|softphone]] registration in order to receive calls.

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

== Additional settings ==

*'''Voicemail associated with DID''', This is where you can assign a Mailbox to your DID. '''[[Voicemail]]'''
*'''Ring Time in seconds''', is basically the time the call will ring before it reaches one of the Failover states or [[Voicemail]].

*'''Caller ID name lookup''', when you enable this option, the system will perform a Query on the Databases, looking for a name matching the number of your caller, and will display the name on the [[Caller ID]] name section of the [[Caller ID]].

*'''Caller ID name prefix''', this setting will simply add any word you set, as a prefix to the [[Caller ID]] name you receive. Will also work even if you don't have "Caller ID name lookup" enabled or if you don't receive a [[Caller ID]] name. This option is specially useful when you need to differentiate incoming calls from different DIDs going to the same phone.
'''Note''', this is just an Internal description for the DID, so you can manage them.

*'''Dialing Mode for Inbound Calls''', When someone calls your number, the system sends a SIP INVITE to your device. This is like a digital message saying “you have an incoming call.” The way the phone number is sent can follow one of two formats:
**'''NANPA Format:''' The number is sent with just 10 digits, like this: 5551234567.This is common in the U.S. and Canada and does not include the country code.
**'''E.164 Format:''' The number is sent with a + and the country code first, like this: +15551234567. This is the international standard and works globally.
'''Make sure your device or phone system is set up to understand the format you’re using so calls come through correctly.'''

'''Remember that none of these changes will be saved until you press the "Click here to apply changes settings" button.'''

== SMS message service ==

In this section, you will be able to configure the SMS service. Please notice that it has some restrictions. You can find details about in our [[SMS#Configuring_the_SMS_service | wiki entry for SMS]]

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

== Editing Multiple DIDs at a time ==

There are 2 additional buttons you will see when you access the Manage DID option from the menu, those are ''"Edit Selection - All settings at Once"'' and ''"Edit Selection - One Setting at a Time"''

'''"Edit Selection - All settings at Once"''', if you choose this option, you will enter an Edition page, the way the settings on this page end, is the exact way the settings will be applied to all the DIDs selected for edition. Be careful, as the final settings from the Edition page will be applied exactly as you see them, even those settings you did not change, will be replaced for the final configuration of the Edition page. That is if you select a [[Voicemail]], all the DIDs will use the same [[Voicemail]], if you leave a blank field like the "Caller ID prefix", all of the DIDs will have this setting blank.
This option is useful if you want all your DIDs with the exact same configuration for each one of the settings, including the NOTE.

'''"Edit Selection - One Setting at a Time"''', this option allows you to access the Edition page, but this time, every setting will have its own independent "Apply" button, so you can make a change to a specific setting, without affecting the rest.
This option is useful if you already have your DIDs configured with different settings, and you need to set only a specific setting for all of them. By doing it this way, the rest of the settings from the DIDs will remain untouched.

= Reseller Configuration =

Also if you're using the [[Reseller Basic Guide|Reseller Interface]], you can associate a DID to one of your reseller client.

'''Reseller Client''': Here you can select your reseller client that you want to associate this DID.
:'''NOTE:''' You need first to create the account of your customer using the [[Reseller Basic Guide|Reseller section]] in your Customer Portal.

: [[File:ResellerClient_SelectClient.png|500px]]

'''Sub Account''': It allows you to associate this feature with a '''Reseller Client''' by selecting a '''Sub Account''' that is already linked to that client.

:'''NOTE''': The Reseller Client must already exist and have at least one Sub Account linked to it.
Once linked, the DID will be able to be managed in the Reseller Portal and inbound calls.
:'''NOTE''': In your CDR, all incoming calls received on that DID after being linked will be showing the selected Sub‑Account in the “Account” field. This remains the case even if the DID’s routing is later changed—for example, to an IVR or something else.
The CDR will always reflect the Sub‑Account to which the DID was originally linked for billing purposes.

: [[File:ResellerClient_SelectClient_Subaccount.png|500px]]

[[category:guides]]

File:Routing.png

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File:Failover.png

2026-06-17T15:30:56Z

Neocypher: uploaded a new version of "File:Failover.png": Reverted to version as of 15:26, 17 June 2026

File:Failover.png

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Neocypher: uploaded a new version of "File:Failover.png"

File:Failover.png

2026-06-17T15:26:52Z

Neocypher: uploaded a new version of "File:Failover.png"

Porting from Skype to VoIP.ms

2025-03-25T17:16:21Z

Neocypher:

<br clear="all" />
[[File:Logo-VoIPms-light.png|left|400px]]
<br clear="all" />

{| class="wikitable"
|+
|-
! Article en Français !! Artículo en Español
|-
| [] || []
|}

This article provides general guidelines on how to start your porting process from Skype to VoIP.ms

== Documentation Requirements ==

* Service address associated with the number. Note due to industry standards, all US numbers must provide a US address when doing a port.
* Account number. This is typically the username of the skype account
* PIN if applicable.
* List of numbers to port.
* Proof of ownership. A screenshot taken directly from the skype portal, showing the numbers to be ported. This screenshot must be e.signed or handsigned.

== How to Start The Process ==

'''1. Create an Account and Add Funds''' <br>
Before proceeding with the port request, you will need to create an account on our platform and add funds. This ensures that your service is ready once the number is ported. https://voip.ms/SignUp

'''2. Request Port-Out Information from Skype''' <br>
You will need to contact Skype and request the port-out details for your phone number. This typically includes:

* Service address associated with the number (US addresses only)
* Account number: you can use the user name with Skype
* PIN (if applicable)
* List of phone numbers to port (in case there are other numbers on the account)

'''3. Prepare Proof of Ownership of the numbers'''<br>
Take a screenshot from the Skype portal where the phone number and account details are visible. Please add your signature to these screenshots before uploading the documentation.

'''4. Start the Process'''<br>
From the customer portal, go to DID numbers menu, select the option DID Porting to start the process. Please refer to our general [[Porting FAQ]] for more details and information about porting.

== Important Notes ==

* '''The phone number must remain active throughout the porting process.'''
* '''The porting process typically takes 5-7 business days, depending on the response time from Skype.'''
* '''If your Skype account has an international or Canadian address, please request Skype a valid physical U.S. address before submitting the porting request.'''

Porting from Skype to VoIP.ms

2025-03-25T17:03:39Z

Neocypher:

<br clear="all" />
[[File:Logo-VoIPms-light.png|left|400px]]
<br clear="all" />

{| class="wikitable"
|+
|-
! Article en Français !! Artículo en Español
|-
| [] || []
|}

This article provides general guidelines on how to start your porting process from Skype to VoIP.ms

== Documentation Requirements ==

* Service address associated with the number. *Skype only accepts US addresses.
* Account number. This is typically the username of the skype account
* PIN if applicable.
* List of numbers to port.
* Proof of ownership. A screenshot taken directly from the skype portal, showing the numbers to be ported. This screenshot must be e.signed or handsigned.

== How to Start The Process ==

'''1. Create an Account and Add Funds''' <br>
Before proceeding with the port request, you will need to create an account on our platform and add funds. This ensures that your service is ready once the number is ported. https://voip.ms/SignUp

'''2. Request Port-Out Information from Skype''' <br>
You will need to contact Skype and request the port-out details for your phone number. This typically includes:

* Service address associated with the number (US addresses only)
* Account number: you can use the user name with Skype
* PIN (if applicable)
* List of phone numbers to port (in case there are other numbers on the account)

'''3. Prepare Proof of Ownership of the numbers'''<br>
Take a screenshot from the Skype portal where the phone number and account details are visible. Please add your signature to these screenshots before uploading the documentation.

'''4. Start the Process'''<br>
From the customer portal, go to DID numbers menu, select the option DID Porting to start the process. Please refer to our general [[Porting FAQ]] for more details and information about porting.

== Important Notes ==

* '''The phone number must remain active throughout the porting process.'''
* '''The porting process typically takes 5-7 business days, depending on the response time from Skype.'''
* '''If your Skype account has an international or Canadian address, update it to a valid physical U.S. address before submitting the porting request.'''

Porting from Skype to VoIP.ms

2025-03-25T17:02:05Z

Neocypher: /* How to Start The Process */

<br clear="all" />
[[File:Logo-VoIPms-light.png|left|400px]]
<br clear="all" />

{| class="wikitable"
|+
|-
! Article en Français !! Artículo en Español
|-
| [] || []
|}

This article provides general guidelines on how to start your porting process from Skype to VoIP.ms

== Documentation Requirements ==

* Service address associated with the number. *Skype only accepts US addresses.
* Account number. This is typically the username of the skype account
* PIN if applicable.
* List of numbers to port.
* Proof of ownership. A screenshot taken directly from the skype portal, showing the numbers to be ported. This screenshot must be e.signed or handsigned.

== How to Start The Process ==

'''1. Create an Account and Add Funds''' <br>
Before proceeding with the port request, you will need to create an account on our platform and add funds. This ensures that your service is ready once the number is ported. https://voip.ms/SignUp

'''2. Request Port-Out Information from Skype''' <br>
You will need to contact Skype and request the port-out details for your phone number. This typically includes:

* Service address associated with the number (US addresses only)
* Account number: you can use the user name with Skype
* PIN (if applicable)
* List of phone numbers to port (in case there are other numbers on the account)

'''3. Prepare Proof of Ownership of the numbers'''<br>
Take a screenshot from the Skype portal where the phone number and account details are visible. Please add your signature to these screenshots before uploading the documentation.

'''4. Start the Process'''<br>
From the customer portal, go to DID numbers menu, select the option DID Porting to start the process. Follow this guide for more details.

== Important Notes ==

* '''The phone number must remain active throughout the porting process.'''
* '''The porting process typically takes 5-7 business days, depending on the response time from Skype.'''
* '''If your Skype account has an international or Canadian address, update it to a valid physical U.S. address before submitting the porting request.'''

Porting from Skype to VoIP.ms

2025-03-25T16:58:51Z

Neocypher:

<br clear="all" />
[[File:Logo-VoIPms-light.png|left|400px]]
<br clear="all" />

{| class="wikitable"
|+
|-
! Article en Français !! Artículo en Español
|-
| [] || []
|}

This article provides general guidelines on how to start your porting process from Skype to VoIP.ms

== Documentation Requirements ==

* Service address associated with the number. *Skype only accepts US addresses.
* Account number. This is typically the username of the skype account
* PIN if applicable.
* List of numbers to port.
* Proof of ownership. A screenshot taken directly from the skype portal, showing the numbers to be ported. This screenshot must be e.signed or handsigned.

== How to Start The Process ==

1. Create an Account and Add Funds
Before proceeding with the port request, you will need to create an account on our platform and add funds. This ensures that your service is ready once the number is ported. https://voip.ms/SignUp

2. Request Port-Out Information from Skype
You will need to contact Skype and request the port-out details for your phone number. This typically includes:

* Service address associated with the number (US addresses only)
* Account number: you can use the user name with Skype
* PIN (if applicable)
* List of phone numbers to port (in case there are other numbers on the account)

3. Prepare Proof of Ownership of the numbers

Take a screenshot from the Skype portal where the phone number and account details are visible. Please add your signature to these screenshots before uploading the documentation.

== Important Notes ==

* '''The phone number must remain active throughout the porting process.'''
* '''The porting process typically takes 5-7 business days, depending on the response time from Skype.'''
* '''If your Skype account has an international or Canadian address, update it to a valid physical U.S. address before submitting the porting request.'''

Porting from Skype to VoIP.ms

2025-03-25T16:58:11Z

Neocypher:

<br clear="all" />
[[File:Logo-VoIPms-light.png|left|400px]]
<br clear="all" />

{| class="wikitable"
|+
|-
! Article en Français !! Artículo en Español
|-
| [] || []
|}

This article provides general guidelines on how to start your porting process from Skype to VoIP.ms

== Documentation Requirements ==

* Service address associated with the number. *Skype only accepts US addresses.
* Account number. This is typically the username of the skype account
* PIN if applicable.
* List of numbers to port.
* Proof of ownership. A screenshot taken directly from the skype portal, showing the numbers to be ported. This screenshot must be e.signed or handsigned.

== How to Start The Process ==

1. Create an Account and Add Funds
Before proceeding with the port request, you will need to create an account on our platform and add funds. This ensures that your service is ready once the number is ported. https://voip.ms/SignUp

2. Request Port-Out Information from Skype
You will need to contact Skype and request the port-out details for your phone number. This typically includes:

* Service address associated with the number (US addresses only)
* Account number: you can use the user name with Skype
* PIN (if applicable)
* List of phone numbers to port (in case there are other numbers on the account)

3. Prepare Proof of Ownership of the numbers

Take a screenshot from the Skype portal where the phone number and account details are visible. Please add your signature to these screenshots before uploading the documentation.

== Important Notes ==

'''* The phone number must remain active throughout the porting process.'''
'''* The porting process typically takes 5-7 business days, depending on the response time from Skype.'''
'''* If your Skype account has an international or Canadian address, update it to a valid physical U.S. address before submitting the porting request.'''

Porting from Skype to VoIP.ms

2025-03-25T16:57:30Z

Neocypher:

<br clear="all" />
[[File:Logo-VoIPms-light.png|left|400px]]
<br clear="all" />

{| class="wikitable"
|+
|-
! Article en Français !! Artículo en Español
|-
| [] || []
|}

This article provides general guidelines on how to start your porting process from Skype to VoIP.ms

== Documentation Requirements ==

* Service address associated with the number. *Skype only accepts US addresses.
* Account number. This is typically the username of the skype account
* PIN if applicable.
* List of numbers to port.
* Proof of ownership. A screenshot taken directly from the skype portal, showing the numbers to be ported. This screenshot must be e.signed or handsigned.

== How to Start The Process ==

1. Create an Account and Add Funds
Before proceeding with the port request, you will need to create an account on our platform and add funds. This ensures that your service is ready once the number is ported. https://voip.ms/SignUp

2. Request Port-Out Information from Skype
You will need to contact Skype and request the port-out details for your phone number. This typically includes:

* Service address associated with the number (US addresses only)
* Account number: you can use the user name with Skype
* PIN (if applicable)
* List of phone numbers to port (in case there are other numbers on the account)

3. Prepare Proof of Ownership of the numbers

Take a screenshot from the Skype portal where the phone number and account details are visible. Please add your signature to these screenshots before uploading the documentation.

== Important Notes ==

'''* The phone number must remain active throughout the porting process.
* The porting process typically takes 5-7 business days, depending on the response time from Skype.
* If your Skype account has an international or Canadian address, update it to a valid physical U.S. address before submitting the porting request.'''

Porting from Skype to VoIP.ms

2025-03-25T16:56:39Z

Neocypher:

<br clear="all" />
[[File:Logo-VoIPms-light.png|left|400px]]
<br clear="all" />

{| class="wikitable"
|+
|-
! Article en Français !! Artículo en Español
|-
| [] || []
|}

This article provides general guidelines on how to start your porting process from Skype to VoIP.ms

== Documentation Requirements ==

* Service address associated with the number. *Skype only accepts US addresses.
* Account number. This is typically the username of the skype account
* PIN if applicable.
* List of numbers to port.
* Proof of ownership. A screenshot taken directly from the skype portal, showing the numbers to be ported. This screenshot must be e.signed or handsigned.

== How to Start The Process ==

1. Create an Account and Add Funds
Before proceeding with the port request, you will need to create an account on our platform and add funds. This ensures that your service is ready once the number is ported. https://voip.ms/SignUp

2. Request Port-Out Information from Skype
You will need to contact Skype and request the port-out details for your phone number. This typically includes:

* Service address associated with the number (US addresses only)

* Account number: you can use the user name with Skype

PIN (if applicable)
List of phone numbers to port (in case there are other numbers on the account)

3. Prepare Proof of Ownership of the numbers

A screenshot from the Skype portal where the phone number and account details are visible. Please add your signature to these screenshots before uploading the documentation.

== Important Notes ==

'''* The phone number must remain active throughout the porting process.
* The porting process typically takes 5-7 business days, depending on the response time from Skype.
* If your Skype account has an international or Canadian address, update it to a valid physical U.S. address before submitting the porting request.'''