VoIP.ms Wiki - User contributions [en]

File:IMG 9079.jpg

2022-07-29T05:01:45Z

Joseph:

User:Joseph/sandbox

2022-07-29T04:33:16Z

Joseph:

This is a dockerized Entry :)

[[File:ytttqrttquu.png]]

File:Ytttqrttquu.png

2022-07-29T04:32:43Z

Joseph:

User:Joseph/sandbox

2022-07-29T04:22:31Z

Joseph: Created page with "This is a dockerized Entry :)"

This is a dockerized Entry :)

Back to Basics - What is Elastic SIP Trunking

2020-12-11T17:38:51Z

Joseph: Undo revision 14986 by Johann (talk)

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[[File:elasticsiptrunking.jpg|center|What Is Elastic SIP Trunking?]]
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Businesses are looking for advanced communication technology to improve their customer experience and overall internal operations. This is the reason why SIP Trunking has become the go-to technology for companies.

Because of this, many businesses are opting to switch towards a more flexible communication technology known as Elastic SIP Trunking. In this article, we will explore how it works and its advantages.

__NOTOC__

''' So, What Is Elastic SIP Trunking? '''

As compared to traditional SIP Trunking, elastic SIP Trunking is more advanced and scalable. It makes use of a cloud-based virtual connection to include flexibility to the SIP Trunking experience. Because of the virtual connection, your business can scale up and down as needed during peak times. This flexibility is especially imperative for companies that have seasonal or punctual spikes in their call volumes.

Additionally, because of its cloud-based nature, businesses can also integrate elastic SIP Trunking with other cloud-based services. This functionality is useful for all types of corporations that have a unified communication system architecture.

'''Elastic SIP Trunking and How It Works'''

Elastic SIP trunks work similarly to traditional SIP trunks. It connects IP channels to the standard PSTN network or other VoIP-based communication channels. As a result, a business can make and receive calls to other companies and customers on a landline or a mobile. Elastic SIP Trunking specificity is the possibility to “burst over” the predetermined number of channels that is contained in a standard SIP trunk.

'''Elastic SIP Trunking at VoIP.ms'''

VoIP.ms offers two different billing methods for incoming calls: Pay per Minute or Pay per Channel (also called a Virtual PRI). The Virtual PRI allows a business to purchase a predetermined number of channels based on its average concurrent incoming calls (not the peak) then simply pay it monthly. To ensure the caller does not get a busy tone due to channel limits, here comes the “elastic” variable: on top of the predetermined number of channels, VoIP.ms allows its customers to establish a maximum “burst” level. This way, whenever a customer peaks its predetermined number of channels, calls still go through. A net advantage of VoIP.ms over competition is that, at VoIP.ms, the “elastic” portion of the virtual PRI channels is billed daily. This means that customers only pay for the daily prorated portion of a channel when bursting over the predetermined number of channels. Moreover, a business can assign one or all its local DIDs to its Virtual PRI to share the pool of channels (as long as in the same country).

'''Two Benefits of Elastic SIP Trunking'''

'''1.Call Concurrency'''

Call concurrency is one of the biggest advantages of Elastic SIP Trunking for businesses. There are no restrictions when working with an elastic framework. Therefore, when experiencing
growth, businesses can quickly access to new lines as they expand their operations.

'''2. Pay Only for What You Use'''

Another benefit of elastic SIP Trunking is that businesses pay only for what they are using. As a matter of fact, a traditional PRI limits business to pay for a full 23-channel PRI whether they use those channels or not. Standard SIP Trunking partly overcomes this limitation by allowing businesses to purchase additional connectivity, but it might not necessarily be used when the call volumes are low.

However, when a business decides to opt for an elastic SIP approach, it does not pay for useless calling capabilities when the call volume is somehow smaller than usual. It only pays for what the team is really using.

'''Final Toughts'''

SIP Trunking has become the go-to communication technology for most businesses. Elastic SIP Trunking only adds to the traditional SIP trunks. It provides companies more flexibility and overcomes some of the limitations of standard SIP trunks. That said, this communication technology is suitable for businesses that are looking for a more dynamic calling capacity.

You want to know more about VoIP.ms Virtual PRI?
Read VoIP.ms blog article (https://wiki.voip.ms/article/Virtual_PRI) or contact sales@voip.ms.

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______

For more information, visit us at <br/>
https://voip.ms or sign up now <br/>
to start making calls in under 5 minutes<br/>
at https://www.voip.ms/#Signup!<br/>
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SMPP (ES)

2020-11-20T15:17:02Z

Joseph: /* Configuración en su portal de VoIP.ms */

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[[File:Logo-VoIPms-light.png|left|400px]]
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Un servidor SMPP, acrónimo 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 a ésto.
En otras palabras, le permite tener un mejor control sobre la cantidad de mensajes que se enviarán a su público destinatario evitando así pasar por una aplicación de softphone o su portal VoIP.ms a fin de enviar / recibir estos SMS, eliminando así múltiples tareas repetitivas y ahorrando su tiempo.

Por ejemplo, si tiene una promoción en su negocio que ofrece un descuento por un período de tiempo determinado y sus clientes han dado su consentimiento de recibir este tipo de promociones vía SMS, usted podrá enviar la promoción a sus clientes más rápido que si lo hiciera desde su portal de clientes, aumentando así su eficiencia.

== Requisitos==

*Ésto requerirá que tenga 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 pueda comunicarse y autenticarse correctamente con el servidor SMPP de VoIP.ms, deberá configurar lo siguiente:
* Un '''nombre de usuario''' (username) y '''contraseña''' (password) de su elección para que el servidor SMPP de VoIP.ms lo autentique.
* Haga que todas sus comunicaciones se envíen a '''smpp.voip.ms''' a través del puerto '''2775''', para el SMPP normal, o al puerto '''3550''', que es nuestro puerto SMPP encriptado.
* Abra una conexión usando 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'''. Asegúrese de incluir el '''source_addr''' de su PDU como uno de los DID que tenga la función SMS habilitada desde su cuenta. Su número de SMS de destino debe establecerse como '''destination_addr''' mientras que su mensaje de texto se incluirá en '''short_message'''.

=== Configuración en su portal de VoIP.ms ===
Una vez que haya configurado lo anterior en su servidor, deberá habilitar y especificar qué nombre de usuario y contraseña utilizará VoIP.ms para autenticar sus solicitudes SMPP. Para ésto, diríjase a su portal de VoIP.ms, DID Numbers, Manage DIDs, edite el DID que soporte mensajes SMS y al dirigirse hacia abajo en la página, verá las siguientes opciones:

* SMPP Enabled: Si se selecciona, se permitirá enviar y recibir mensajes SMS mediante SMPP.

También puede especificar la URL que se utilizará para enviar una copia de los mensajes entrantes a su servidor SMPP. ''Déjelo en blanco si solo desea enviar mensajes'' Vea el ejemplo a continuación. La parte '''scheme''' es obligatoria y debe establecerse en '''smpp''' para SMPP no encriptado o en '''ssmpp''' para SMPP encriptado utilizando TLS. La parte '''user:password@''' es opcional y por defecto tendrá el mismo nombre de usuario y contraseña definidos para la autenticación en nuestro sistema 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 autenticarse en nuestro servidor SMPP para el envio mensajes.
* '''SMPP Password:''' La contraseña que utilizará para autenticarse en nuestro servidor SMPP para el envio de mensajes

'''Notas'''
- Para ayudar a evitar confusiones al configurar el nombre de usuario en su servidor, puede usar su ID de cuenta de VoIP.ms como nombre de usuario o username (los 6 dígitos de su cuenta).

- Asegúrese de tener habilitada la opción '''Message Service (SMS/MMS)'''

Habilite la opción '''SMPP Enabled''', complete el nombre de usuario y la contraseña según lo configurado en su servidor SMPP y una vez hecho ésto, aplique los cambios haciendo clic en '''Click here to apply changes'''.

[[File:SMPPconfigFR.png|900px]]

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

MediaWiki:Sidebar

2020-09-02T16:35:27Z

Joseph:

* VoIP.ms Wiki
** http://voip.ms| VoIP.ms Website
** mainpage|Welcome
** Features|Features
** FAQ|FAQ
** Porting FAQ|Porting FAQ

* VoIP.ms Blog
** Blog| Blog

* Configuration
** devices|Devices
** softphones|Softphones
** PBXs|PBXs
** Session Border Controllers|SBCs

* Guides (English)
** Getting Started|Getting Started
** Account Settings|Account Settings
** Add Articles|Add Articles
** Audio Conferencing|Audio Conferencing
** Calculating my expenses|Calculating my expenses
** Call Detail Records|Call Detail Records
** Call Encryption - TLS/SRTP|Call Encryption - TLS/SRTP
** Call Forwarding|Call Forwarding
** Call Hunting|Call Hunting
** Call quality issues|Call quality issues
** Call Recordings|Call Recordings
** Callback|Callback
** Caller ID|Caller ID
** CallerID Filtering|CallerID Filtering
** CallerID number spoofing|CallerID number spoofing
** Calling Queues|Calling Queue
** Calls Cost|Calls Cost
** Changing DID Billing Plan|Changing DID Billing Plan
** Choosing Server|Choosing Server
** Dial Plan for Linksys ATAs|Dial Plan for Linksys ATAs
** Dialing Codes|Dialing Codes
** Dialing Rules|Dialing Rules
** Dialing Rules and Patterns|Dialing Rules and Patterns
** DID Troubleshooting|DID Troubleshooting
** DigitalReceptionist_IVR|Digital Receptionist (IVR)
** DISA|DISA
** E911|E911
** Extra_Services_Costs|Extra Services Costs
** False_Answer_Supervision_FAS|FAS (False Answer Supervision)
** Finances|Finances
** Firewall|Firewall
** General_Security|General Security
** https://wiki.voip.ms/article/Finances#Generate_Invoice | Generate invoice
** International_Calls|International Calls
** Manage DID|Manage DID
** MMS | MMS
** Order_a_DID_Number|Order a DID Number
** PBX Security|PBX Security
** Phone book|Phone book
** Porting a Number|Porting a Number
** Problems logging in|Problems logging in
** Pulse_dial|Pulse Dial
** Recordings|Recordings
** Referral Program|Referral Program
** Registration issue|Registration Issue
** Registration status on desktop|Registration status on desktop
** Remove a DID|Remove a DID
** Reseller Basic Guide|Reseller Basic Guide
** Ring Groups|Ring Groups
** Service Cost|Service Cost
** SIP Requests|SIP Requests
** SIP_Responses|SIP Responses
** SIP URI|SIP URI
** Smartphone|Smartphone
** SMS|SMS
** Sub Accounts|Sub Accounts
** Time Conditions|Time Conditions
** Transaction_history|Transaction history
** Troubleshooting Outgoing Calls|Troubleshooting Outgoing Calls
** Two-step Verification|Two-step Verification
** TOTP Authentication|TOTP Authentication
** Unlocking All International Destinations|Unlocking All International Destinations
** Value vs Premium|Value vs Premium
** Virtual Fax|Virtual Fax
** Voicemail|Voicemail

* Guides (Français)
** Bienvenue|Bienvenue
** Mise en Marche|Mise en Marche
** Ajouter_un_Article| Ajouter un Article
**Audioconférence|Audioconférence
** Calculer_mes_dépenses|Calculer mes dépenses
** Changement du plan de facturation d'un Numéro|Changement du plan de facturation d'un Numéro
** Choisir un serveur | Choisir un serveur
** Codes de Composition|Codes de Composition
** Commander_un_numéro_DID|Commander un numéro DID
** Conditions Temporelles | Conditions Temporelles
** Coût des Services Rendus|Coût des Services Rendus
** Cryptage des appels - TLS/SRTP|Cryptage des appels - TLS/SRTP
** Débloquer les destinations internationales | Débloquer les destinations internationales
** Détails des appels|Détails des appels
** Accès direct en entrée au système - DISA | DISA
** Effacer un Numéro DID|Effacer un Numéro DID
** Enregistrements | Enregistrements
**Enregistrements d'appels|Enregistrements d'appels
** File d'attente | File d'attente
** Filtrage du numéro d'identification de l'appelant | Filtrage du numéro d'identification de l'appelant
** Finances_Fr|Finances
** Fonction de Rappel | Fonction de Rappel
** Gérer les numéros DID|Gérer les numéros DID
** Groupe de Sonnerie|Groupe de Sonnerie
** ID de l'appelant | ID de l'appelant
** Messagerie texte | Messagerie texte
** Messagerie multimédia | Messagerie multimédia
** Messagerie vocale | Messagerie vocale
** Paramètres du compte | Paramètres du compte
** Programme de référencement|Programme de référencement
** Problèmes_de_Connexion|Problèmes de Connexion
** Problème d'enregistrement|Problème d'enregistrement
** Questions Les Plus Fréquentes | Questions Les Plus Fréquentes
** Recherche d’Appel | Recherche d'Appel
** Renvoi d'appel | Renvoi d'appel
** Répertoire téléphonique | Répertoire téléphonique
** Réceptionniste virtuelle IVR | Réceptionniste virtuelle IVR
** Revendeur_Guide_Elementaire|Revendeur Guide Elementaire
** Service_E911|Service E911
** SIP_URI_FR|SIP URI
** Sous Comptes|Sous Comptes
** Télécopieur virtuel | Télécopieur virtuel
** Transférabilité des DID | Transférabilité des DID
** Value vs Premium (Français)|Value vs Premium

* Guías (Español)
** Bienvenido|Bienvenido
** Primeros Pasos|Primeros Pasos
** Preguntas_frecuentes_LNP|Preguntas frecuentes LNP
** Acceso directo al sistema interno (DISA)|Acceso directo al sistema interno
** Buzón de voz (Voicemail)|Buzón de voz
** Calculando mis gastos|Calculando mis gastos
** Cambiar Plan de DID (Changing DID Billing Plan)|Cambiar Plan de DID
** Cancelar/Borrar_un_DID| Cancelar/Borrar un DID
** Códigos de Marcado|Códigos de Marcado
** Condiciones de Tiempo (Time Conditions)|Condiciones de Tiempo
** Configuración de DID (Manage DID)|Configuración de DID
** Configuraciones de la Cuenta (Account Settings)|Configuraciones de la Cuenta
** Costo del servicio|Costo del servicio
** Costo por llamada|Costo por llamada
** Desbloqueo de todas las llamadas internacionales|Desbloqueo de todas las llamadas internacionales
** Desvío de Llamadas (Call Forwarding)|Desvio de Llamadas
** Dial Plan para ATA Linksys|Dial Plan para ATA Linksys
** Dirección URI (SIP URI)|Dirección URI
** Directorio (Phone Book)|Directorio
** E911_Spanish|E911 Espanol
** Elegir servidor|Elegir servidor
** Eliminar un DID|Eliminar un DID
** Estatus_de_registro_en_escritorio|Estatus de registro en escritorio
** Filtro de Llamadas (CallerID Filtering)|Filtro de Llamadas
** FAS_(supervisión_de_respuesta_falsa)|FAS (supervisión de respuesta falsa)
** Guía Básica de Reseller|Guía Básica de Reseller
** Grabaciones (Recordings)|Grabaciones
** Grupo de timbre / Ring Groups|Grupo de timbre
** Llamadas en Cola (Calling Queues)|Llamadas en Cola
** Numero Identificador (Caller ID)|Numero Identificador
** Porteo de numeros|Porteo de numeros
** Problemas de audio|Problemas de audio
** Problemas de registro|Problemas de registro
** Problemas Para Iniciar Sesión|Problemas Para Iniciar Sesión
** Recepcionista_Digital|Recepcionista_Digital (IVR)
** Registro de Llamadas (CDR)|Registro de Llamadas
** Reglas_y_Patrones_de_Marcado|Reglas y Patrones de Marcado
** Regreso de LLamada (Callback)|Regreso de LLamada
** Respuestas_SIP|Respuestas SIP
** Seguridad en PBX|Seguridad en PBX
** SIP_URI_español|SIP URI(Español)
** Solución de problemas de llamadas salientes|Solución de problemas de llamadas salientes
** Sub Cuentas (Sub Accounts)|Sub Cuentas
** Suplantación de identidad (CallerID spoofing)|Suplantación de identidad
** Value vs. Premium|Value vs. Premium

* SEARCH
* TOOLBOX
* LANGUAGES

Devices

2020-08-31T20:38:21Z

Joseph: /* SNOM professional D7XX */

== Blog Article ==

[https://wiki.voip.ms/article/VoIP_%E2%80%93_Bring_Your_Own_Device_(BYOD) VoIP – Bring Your Own Device (BYOD)]
<br />
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).

* '''ATA:''' An '''Analog Telephone Adapter''' is a device used to connect one or more standard analog telephones to a digital telephone system (such as Voice over IP) or a non-standard telephone system. Still not clear? Take a peek at our blog article : [https://wiki.voip.ms/article/Back_to_Basics_-_What_is_an_ATA%3F Back to Basics - What is an ATA?]

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

__TOC__

==ATAs==

===Atcom AG188N===

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

'''Product:''' Atcom AG188N

'''Company:''' Atcom

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

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

[[Atcom AG188N|See Configuration Details]]
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===Auerswald COMpact 5010===

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

'''Product:''' Auerswald COMpact 5010

'''Company:''' Auerswald

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

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

[[Auerswald COMpact 5010|See Configuration Details]]
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===Cisco===

====Cisco Linksys PAP2====

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

'''Product:''' Linksys PAP2

'''Company:''' Cisco

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

[[Cisco Linksys PAP2|See Configuration Details]]
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====Cisco Linksys PAP2T====

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

'''Product:''' Linksys PAP2T

'''Company:''' Cisco

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

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

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

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

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

'''Product:''' SPA112, SPA122

'''Company:''' Cisco

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

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

Introduced in late 2011, this box represents an inexpensive means to continue using existing analog hardware while migrating to voice over IP.

[[Cisco SPA112|See Configuration Details]]

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

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

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

'''Product:''' WRP400, WRP500

'''Company:''' Cisco

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

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

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

[[Cisco WRP400|See Configuration Details]]

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

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

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

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

'''Company:''' Cisco

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

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

[[Cisco SPA2100 Phone Adapter|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====Cisco SPA2102 Phone Adapter with Router====

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

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

'''Company:''' Cisco

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

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

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

====Grandstream HandyTone 486====

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

'''Product:''' Grandstream HandyTone 486

'''Company:''' Grandstream

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

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

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

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

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

'''Company:''' Grandstream

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

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

Enhanced security

Automated provisioning using symmetric and asymmetric voice

Support for a broad range of popular voice codec

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

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

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

'''Company:''' Grandstream

'''Overview:''' The HT702 is Grandstream´s Newest powerful VoIP router. The product's inclusion of an integrated high-performance NAT router and 10Mbps Ethernet WAN and LAN ports enables a shared broadband connection between multiple ethernet devices.

In addition to being SIP 2.0 standard compliant, the product supports Universal Plug-in-Play (UPnP), up to 2 SIP account profiles/Phones and advanced telephony features.

Enhanced Security

Automated provisioning using symmetric and asymmetric voice

Support for a broad range of popular voice codec

[[Grandstream_HandyTone_702_-_HT702|See Configuration Details]]
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====Grandstream HandyTone 802 - HT802====

[[File:HT802 Device.jpg|300px|thumb|left|Grandstream HandyTone 802 - HT802]]

'''Product:''' Grandstream HandyTone 802 - HT802

'''Company:''' Grandstream

'''Overview:''' The HT802 delivers powerful VoIP technology and routing capabilities to home and office environments, and allows users to successfully connect their analog devices to a manageable and powerful VoIP network. Built upon Grandstream’s market-leading SIP ATA/gateway technology, with millions of units successfully deployed worldwide, this powerful ATA features exceptional voice quality in various applications and environments.

The HT802 comes with 2 easy-to-use FXS ports, state-of-the-art encryption with a unique security certificate per unit, automated provisioning for volume deployment and device management and outstanding network performance.

TLS and SRTP security encryption technology to protect calls and accounts

Supports 2 SIP profiles through 2 FXS ports and a single 10/100Mbps port

Support for a broad range of popular voice codec

[[Grandstream HandyTone 802 - HT802|See Configuration Details]]
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===Mediatrix C7 and Mediatrix 4100 Series===

[[File:Mediatrix_C7.jpg|300px|thumb|left|Mediatrix C7]]
[[File:Mediatrix_4100_1_media5.jpg|300px|thumb|left|Mediatrix 4102]]

'''Product:''' : Mediatrix C7 and Mediatrix 4100 Series running DGW 2.0 firmware

'''Company:''' Media5 Corporation

'''Overview:''' The Mediatrix® C7 and Mediatrix® 4100 Series feature VoIP adaptors that interconnect up to 24 analog telephones, faxes, and modems ports into SIP-based systems. These series allow operators to shorten deployment time to enable cloud telephony services into branch offices and SMBs.

The DGW 2.0 firmware also offers security features such as SIP over TLS, SRTP, certificates management, and HTTPS designed to bring enhanced security for the network management, SIP signaling, and media transmission aspects.

[[Media5 Mediatrix C7 and 4100|See Configuration Details]]
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===Netgear WGR615V===

[[File:NetgearWGR615V.jpg|300px|thumb|left|Netgear WGR615V]]

'''Product:''' WGR615V

'''Company:''' Netgear

'''Overview:''' The WGR615V was never sold to directly to the public when it was new but since the model is discontinued, lots of them are being sold.
It is a wireless (802.11 G) router that happens to have a built-in ATA.
It is possible to use just the ATA part.

[[Netgear WGR615V|See Configuration Details]]
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=== OBiHAI (Polycom) ===

====OBi 100/110 & OBi 200====

[[File:OBi110-ATA.jpg|300px|thumb|left|OBi110]]

'''Product:''' OBi 100/110 & OBi 200

'''Company:''' Polycom (OBIHAI Technology Inc)

'''Overview:''' With OBi devices you are in control of your communications life. From the OBi's on-board connections as well as via the Internet, you have the power to bridge mobile, fixed line and Internet telephone services. The OBi100 & OBi110 are stand-alone, dedicated devices, built with a high-performance "system on a chip" platform to ensure high-quality voice conversations. Every OBi device has high availability and reliability. It is 'always-on' to make or receive a call. With an OBi device, a computer is not required and a computer does not need to be on to talk on the phone or use the OBi's powerful service bridging features. To get started, all you need is a phone, power and a connection to the Internet.

The OBi110 has one phone port and one line port. Use the OBi110 when you need an analog line connection to a traditional telephone network or service. Think of the OBi110 LINE port as a gateway to traditional phone service. If you do not have traditional phone service at home, then the OBi100 is probably the right product to get.

[[OBi 100/110 & OBi 200|See Configuration Details]]
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====Polycom OBi300====

[[File:obi300.png|300px|thumb|left|Polycom OBi300]]

'''Product:''' Polycom OBi300

'''Company:''' Polycom

'''Overview:''' Polycom OBi300 expands your service portfolio by enhancing the communication possibilities of home offices with flexibility in voice, fax and modem applications as they transition to the digital communications world.
* Flexible voice and fax applications
* Help home offices maximize current analog investment by keeping one analog phone or fax machine and connect them to VoIP services
* Supports T.38 fax standard so home office users can send or receive reliable facsimile calls over the Internet

[[OBi300|See Configuration Details]]
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===Telco AC-211===

[[File:Ac211n.jpg|300px|thumb|left|Telco AC-211]]

'''Product:''' Telco AC-211

'''Company:''' Telco

'''Overview:''' Telco AC-211 is a SIP ATA supporting two lines and numerous configuration options. This ATA is most commonly found as the AC-211-SR & AC-211N from the now-defunct SunRocket service. This device works well with VoIP.ms once configured.

[[Telco_AC-211|See Configuration Details]]

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===TP-Link TD-VG3631===

[[File:Tplink vg3631 front.png|300px|thumb|left|TPLink VG3631]]

'''Product:''' TP-Link TD-VG3631

'''Company:''' TP-Link

'''Overview:''' High-speed DSL modem, NAT router, and wireless access point in one device providing a one-stop networking solution Wireless N speed up to 300Mbps makes it ideal for heavy bandwidth consuming or interruption sensitive applications like online gaming, Internet calls and even the HD video streaming Supporting both traditional landlines and VoIP network offers you a multiple choice when making phone calls. Various call features such as caller ID, call waiting, call holding, call forwarding, 3-way conference calls and voicemail USB 2.0 port supports Storage Sharing (Samba or FTP), Media Server, Print Server and multi accounts Ethernet WAN (EWAN) offers another broadband connectivity option for connecting to Cable, VDSL or Fiber modems IP based Bandwidth Control makes it easier for you to manage the bandwidth of devices that connected to the modem router Wi-Fi On/Off Button allows users to simply turn their wireless radio on or off.

[[TP-Link TD-VG3631|See Configuration Details]]
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==IP Paging, Speakers, Stobe Lights==

===Algo Technologies SIP endpoints===

[[File:AlgoTechnologies.jpg|300px|thumb|left|Algo Technologies SIP endpoints]]

'''Product:''' IP speakers, paging adapters, strobe lights, clocks, push buttons, doorphones/intercoms, and more

'''Company:''' Algo Technologies SIP endpoints

'''Overview:''' Algo is a telecommunications manufacturer of endpoints and accessories including IP speakers, paging adapters, strobe lights, clocks, push buttons, doorphones / intercoms, and specialty handsets (PTT, PTM).

[[Algo_Technologies_SIP_endpoints|See Configuration Details]]
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==IP Phones==

===3COM 3108 Wireless Phone===

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

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

'''Company:''' 3COM

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

[[3COM_3108_Wireless_Phone|See Configuration Details]]
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===Aastra 6730i/6731i VoIP Phone===

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

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

'''Company:''' Aastra

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

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

[[Aastra_6730i_VoIP_Phone|See Configuration Details]]
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===Audiocodes===

====400HD Series====

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

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

'''Company:''' Audiocodes

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

[[Audiocodes 400HD|See Configuration Details]]
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===Cisco===

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

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

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

'''Company:''' Cisco

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

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

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

[[Cisco IP Phone 68XX and 88XX|See Configuration Details]]
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====Cisco Linksys SPA942 NA====

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

'''Product:''' Linksys SPA942 NA

'''Company:''' Cisco

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

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

[[Cisco_Linksys_SPA942_NA|See Configuration Details]]
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====Cisco SPA525G====

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

'''Product:''' Cisco SPA525G

'''Company:''' Cisco

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

[[Cisco SPA525G|See Configuration Details]]
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====Cisco IP Phone 7940/7960====

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

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

'''Company:''' Cisco

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

[[Cisco_IP_Phone_7940/7960|See Configuration Details]]
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====Cisco SPA30x and SPA50x series IP phones====

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

'''Product:''' Cisco SPA504G Phone

'''Company:''' Cisco

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

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

[[Cisco_SPA504G_Phone|See Configuration Details]]
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===Fanvil ===

====Fanvil X4G====

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

'''Product:''' Fanvil X4G

'''Company:''' Fanvil

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

[[Fanvill X4G|See Configuration Details]]
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[[Category:Devices]]

===Fortinet===

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

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

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

'''Company:''' Fortinet

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

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

[[Fortinet_FortiFone_FON-570|See Configuration Details]]
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====FortiFone FON-375====

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

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

'''Company:''' Fortinet

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

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

[[Fortinet_FortiFone_FON-375|See Configuration Details]]
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====FortiFone FON-175====

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

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

'''Company:''' Fortinet

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

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

[[Fortinet_FortiFone_FON-175|See Configuration Details]]
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===Gigaset A510 IP===

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

'''Product:'''Gigaset A510 IP

'''Company:'''Gigaset

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

[[Gigaset_A510_IP| See Configuration Details]]
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===Grandstream===

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

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

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

'''Company:''' Grandstream

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

[[Grandstream_DP715/DP710| See Configuration Details]]
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====Grandstream DP750/DP720====

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

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

'''Company:''' Grandstream

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

[[Grandstream_DP750/DP720| See Configuration Details]]
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====Grandstream 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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===Konftel===
==== Konftel 300Wx IP ====
[[File:Konftel-300Wx-IP.png|300px|thumb|left|Konftel 300Wx IP]]

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

'''Company:''' Konftel

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

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

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

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

'''Product:''' Konftel 300IPx

'''Company:''' Konftel

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

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

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

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

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

'''Company:''' Panasonic

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

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

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

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

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

'''Company:''' Panasonic

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

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

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

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

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

'''Company:''' Panasonic

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

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

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

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

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

'''Company:''' Panasonic

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

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

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

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

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

'''Company:''' Pirelli

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

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

[[Pirelli DP-L10|See Configuration Details]]
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===Polycom===

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

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

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

'''Company:''' Polycom

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

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

[[Polycom_SoundStation_IP_4000_Conference_Phone|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====Polycom SoundPoint IP 501, 550, 650, etc.====

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

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

'''Company:''' Polycom

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

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

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

[[Polycom_SoundPoint_IP_501|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====Polycom SoundPoint IP 601====

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

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

'''Company:''' Polycom

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

[[Polycom_SoundPoint_IP_601|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====Polycom VVX 300, 400, etc====

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

'''Product:''' Polycom VVX Series

'''Company:''' Polycom

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

[[Polycom VVX 300, 400, etc|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

=== Positron IP phones ===

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

'''Product:''' Positron

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

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

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

==== Positron IP304 ====

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

'''Product:''' Positron IP304

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

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

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

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

[[Positron IP304 | View configuration for Positron IP304]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==== Positron IP304C ====

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

'''Product:''' Positron IP304C

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

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

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

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

[[Positron IP304C | View configuration for Positron IP304C]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==== Positron IP408 ====

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

'''Product:''' Positron IP408

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

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

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

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

[[Positron IP408 | View configuration for Positron IP408]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==== Positron IP410C ====

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

'''Product:''' Positron IP410C

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

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

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

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

[[Positron IP410C | View configuration for Positron IP410C]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==== Positron IP410G ====

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

'''Product:''' Positron IP410G

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

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

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

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

[[Positron IP410G | View configuration for Positron IP410G]]
<div style="width:100%;overflow:hidden;clear:both"></div>

===Siemens Gigaset C450-Ip===

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

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

'''Company:''' Siemens

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

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

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

[[Siemens Gigaset C450-Ip|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

===Snom===

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

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

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

'''Company:''' Snom

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

[[SNOM 320|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====Snom m3 VoIP Phone====

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

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

'''Company:''' Snom

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

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

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

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

====SNOM C520====

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

'''Product:''' SNOM C520 Conferencing

'''Company:''' Snom

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

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

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

====SNOM professional D7XX====

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

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

'''Company:''' Snom

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

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

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

====SNOM M100 KLE====

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

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

'''Company:''' Snom

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

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

[[M100_KLE_SIP_DECT_4-Line_Base_Station|See Configuration Details]]

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

=== Vtech ===

==== Vtech Conference Station ====

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

'''Product:''' VCSV752 & CS754

'''Company:''' Vtech

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

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

==== Vtech VSP Series ====

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

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

'''Company:''' Vtech

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

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

===Yealink===

====Yealink Voice Solutions====

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

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

'''Company:''' Yealink

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

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

====Yealink SIP-T28P (VSRF)====

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

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

'''Company:''' Yealink

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

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

[[Yealink_SIP-T28P|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>
[[Category:Devices]]

===Zycoo ZP502===

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

'''Product:''' Zycoo ZP502

'''Company:''' Zycoo ZP502

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

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

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

MediaWiki:Sidebar

2020-06-16T15:18:51Z

Joseph:

* VoIP.ms Wiki
** http://voip.ms| VoIP.ms Website
** mainpage|Welcome
** Features|Features
** FAQ|FAQ
** Porting FAQ|Porting FAQ

* VoIP.ms Blog
** Blog| Blog

* Configuration
** devices|Devices
** softphones|Softphones
** PBXs|PBXs
** Session Border Controllers|SBCs

* Guides (English)
** Getting Started|Getting Started
** Account Settings|Account Settings
** Add Articles|Add Articles
** Audio Conferencing|Audio Conferencing
** Calculating my expenses|Calculating my expenses
** Call Detail Records|Call Detail Records
** Call Encryption - TLS/SRTP|Call Encryption - TLS/SRTP
** Call Forwarding|Call Forwarding
** Call Hunting|Call Hunting
** Call quality issues|Call quality issues
** Call Recordings|Call Recordings
** Callback|Callback
** Caller ID|Caller ID
** CallerID Filtering|CallerID Filtering
** CallerID number spoofing|CallerID number spoofing
** Calling Queues|Calling Queue
** Calls Cost|Calls Cost
** Changing DID Billing Plan|Changing DID Billing Plan
** Choosing Server|Choosing Server
** Dial Plan for Linksys ATAs|Dial Plan for Linksys ATAs
** Dialing Codes|Dialing Codes
** Dialing Rules|Dialing Rules
** Dialing Rules and Patterns|Dialing Rules and Patterns
** DID Troubleshooting|DID Troubleshooting
** DigitalReceptionist_IVR|Digital Receptionist (IVR)
** DISA|DISA
** E911|E911
** Extra_Services_Costs|Extra Services Costs
** False_Answer_Supervision_FAS|FAS (False Answer Supervision)
** Finances|Finances
** Firewall|Firewall
** General_Security|General Security
** https://wiki.voip.ms/article/Finances#Generate_Invoice | Generate invoice
** International_Calls|International Calls
** Manage DID|Manage DID
** MMS | MMS
** Order_a_DID_Number|Order a DID Number
** PBX Security|PBX Security
** Phone book|Phone book
** Porting a Number|Porting a Number
** Problems logging in|Problems logging in
** Pulse_dial|Pulse Dial
** Recordings|Recordings
** Referral Program|Referral Program
** Registration issue|Registration Issue
** Registration status on desktop|Registration status on desktop
** Remove a DID|Remove a DID
** Reseller Basic Guide|Reseller Basic Guide
** Ring Groups|Ring Groups
** Service Cost|Service Cost
** SIP Requests|SIP Requests
** SIP_Responses|SIP Responses
** SIP URI|SIP URI
** Smartphone|Smartphone
** SMS|SMS
** Sub Accounts|Sub Accounts
** Time Conditions|Time Conditions
** Transaction_history|Transaction history
** Troubleshooting Outgoing Calls|Troubleshooting Outgoing Calls
** Two-step Verification|Two-step Verification
** TOTP Authentication|TOTP Authentication
** Unlocking All International Destinations|Unlocking All International Destinations
** Value vs Premium|Value vs Premium
** Virtual Fax|Virtual Fax
** Voicemail|Voicemail

* Guides (Français)
** Bienvenue|Bienvenue
** Mise en Marche|Mise en Marche
** Ajouter_un_Article| Ajouter un Article
**Audioconférence|Audioconférence
** Calculer_mes_dépenses|Calculer mes dépenses
** Changement du plan de facturation d'un Numéro|Changement du plan de facturation d'un Numéro
** Codes de Composition|Codes de Composition
** Commander_un_numéro_DID|Commander un numéro DID
** Conditions Temporelles | Conditions Temporelles
** Coût des Services Rendus|Coût des Services Rendus
** Cryptage des appels - TLS/SRTP|Cryptage des appels - TLS/SRTP
** Débloquer les destinations internationales | Débloquer les destinations internationales
** Détails des appels|Détails des appels
** Accès direct en entrée au système - DISA | DISA
** Effacer un Numéro DID|Effacer un Numéro DID
** Enregistrements | Enregistrements
**Enregistrements d'appels|Enregistrements d'appels
** File d'attente | File d'attente
** Filtrage du numéro d'identification de l'appelant | Filtrage du numéro d'identification de l'appelant
** Finances_Fr|Finances
** Fonction de Rappel | Fonction de Rappel
** Gérer les numéros DID|Gérer les numéros DID
** Groupe de Sonnerie|Groupe de Sonnerie
** ID de l'appelant | ID de l'appelant
** Messagerie texte | Messagerie texte
** Messagerie multimédia | Messagerie multimédia
** Messagerie vocale | Messagerie vocale
** Paramètres du compte | Paramètres du compte
** Programme de référencement|Programme de référencement
** Problèmes_de_Connexion|Problèmes de Connexion
** Problème d'enregistrement|Problème d'enregistrement
** Questions Les Plus Fréquentes | Questions Les Plus Fréquentes
** Recherche d’Appel | Recherche d'Appel
** Renvoi d'appel | Renvoi d'appel
** Répertoire téléphonique | Répertoire téléphonique
** Réceptionniste virtuelle IVR | Réceptionniste virtuelle IVR
** Revendeur_Guide_Elementaire|Revendeur Guide Elementaire
** Service_E911|Service E911
** SIP_URI_FR|SIP URI
** Sous Comptes|Sous Comptes
** Télécopieur virtuel | Télécopieur virtuel
** Transférabilité des DID | Transférabilité des DID
** Value vs Premium (Français)|Value vs Premium

* Guías (Español)
** Bienvenido|Bienvenido
** Primeros Pasos|Primeros Pasos
** Preguntas_frecuentes_LNP|Preguntas frecuentes LNP
** Acceso directo al sistema interno (DISA)|Acceso directo al sistema interno
** Buzón de voz (Voicemail)|Buzón de voz
** Calculando mis gastos|Calculando mis gastos
** Cambiar Plan de DID (Changing DID Billing Plan)|Cambiar Plan de DID
** Cancelar/Borrar_un_DID| Cancelar/Borrar un DID
** Códigos de Marcado|Códigos de Marcado
** Condiciones de Tiempo (Time Conditions)|Condiciones de Tiempo
** Configuración de DID (Manage DID)|Configuración de DID
** Configuraciones de la Cuenta (Account Settings)|Configuraciones de la Cuenta
** Costo del servicio|Costo del servicio
** Costo por llamada|Costo por llamada
** Desbloqueo de todas las llamadas internacionales|Desbloqueo de todas las llamadas internacionales
** Desvío de Llamadas (Call Forwarding)|Desvio de Llamadas
** Dial Plan para ATA Linksys|Dial Plan para ATA Linksys
** Dirección URI (SIP URI)|Dirección URI
** Directorio (Phone Book)|Directorio
** E911_Spanish|E911 Espanol
** Elegir servidor|Elegir servidor
** Eliminar un DID|Eliminar un DID
** Estatus_de_registro_en_escritorio|Estatus de registro en escritorio
** Filtro de Llamadas (CallerID Filtering)|Filtro de Llamadas
** FAS_(supervisión_de_respuesta_falsa)|FAS (supervisión de respuesta falsa)
** Guía Básica de Reseller|Guía Básica de Reseller
** Grabaciones (Recordings)|Grabaciones
** Grupo de timbre / Ring Groups|Grupo de timbre
** Llamadas en Cola (Calling Queues)|Llamadas en Cola
** Numero Identificador (Caller ID)|Numero Identificador
** Porteo de numeros|Porteo de numeros
** Problemas de audio|Problemas de audio
** Problemas de registro|Problemas de registro
** Problemas Para Iniciar Sesión|Problemas Para Iniciar Sesión
** Recepcionista_Digital|Recepcionista_Digital (IVR)
** Registro de Llamadas (CDR)|Registro de Llamadas
** Reglas_y_Patrones_de_Marcado|Reglas y Patrones de Marcado
** Regreso de LLamada (Callback)|Regreso de LLamada
** Respuestas_SIP|Respuestas SIP
** Seguridad en PBX|Seguridad en PBX
** SIP_URI_español|SIP URI(Español)
** Solución de problemas de llamadas salientes|Solución de problemas de llamadas salientes
** Sub Cuentas (Sub Accounts)|Sub Cuentas
** Suplantación de identidad (CallerID spoofing)|Suplantación de identidad
** Value vs. Premium|Value vs. Premium

* SEARCH
* TOOLBOX
* LANGUAGES

File:SMSSIPAccount.png

2020-06-04T15:32:04Z

Joseph: uploaded a new version of "File:SMSSIPAccount.png": Reverted to version as of 18:25, 12 July 2018

SMS SIP

Gérer les numéros DID

2020-05-21T16:19:49Z

Joseph:

Lorsque vous avez des numéros DIDs dans votre compte, il se pourrait que vous ayez à modifier leurs paramètres par défaut afin de recevoir / envoyer / router les appels entrants selon vos besoins. Pour ce faire, il suffit d'accéder au menu "Numéros DID" et choisir l'option "'''[https://voip.ms/m/managedid.php Gestion des DID]'''".

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

__TOC__

== Information des numéros DID ==

Voici comment vous allez trouver l'information général de votre (vos) numéro(s) (une fois que vous aurez acheté ou [[Transférabilité des DID | transféré]] un numéro DID )

Ensuite, suivra un écran général où vous pourrez voir tous les numéros DIDs de votre compte. Vous y apercevrez probablement différents mots ou icônes avec lesquels vous pourriez ne pas être familier, mais sachez que vous pourrez obtenir une brève description de chacun d'eux en cliquant sur "'''Assistance'''" (icône en haut à droite de la page).

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

== Exporter et sauvegarder la configuration des DIDs ==

À partir de la page ‘Gestion des DIDs’, vous pouvez exporter aussi un fichier CSV contenant une liste entière des numéros DIDs dans votre compte, avec ses respectifs paramètres de configuration et valeurs. Ceci sera particulièrement utile pour avoir une copie de securité de sa configuration, en cas d’un accident avec les paramètres lorsque vous faites des changements ou simplement pour avoir une liste complète des numéros dans le compte.

Cherchez l’option “Exporter la liste de DIDs du compte” et cliquez-la.

[[File:Exporter liste.png|thumb|none|700px]]

Ensuite, sélectionnez l’option "'''Nouvelle exportation'''". Un fichier CSV sera automatiquement généré et téléchargé sur votre ordinateur.

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

Si vous voulez téléverser un fichier avec la configuration des DID ultérieurement, vous pouvez le faire en cliquant le bouton "'''Mettre à jour les DIDs'''".

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

== Modifier un numéro DID ==

Si vous avez besoin d'accéder aux options permettant de modifier un seul DID, le plus simple est de cliquer sur le petit icône ''Papier et crayon'' "'''Modifier DID'''".

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

== Routage ==

Vous verrez alors apparaître la principale section d'un DID, appelée '''Routage''', qui, selon les fonctionnalités déjà activées dans votre portail, peut ou peut ne pas vous offrir toutes les options disponibles. Exemple, si vous n'avez pas encore créé de boîte vocale, vous ne serez pas en mesure de sélectionner cette option et l'appliquer à un DID.

Il s'agit essentiellement de configurer la route que l'appel prendra, lorsque quelqu'un compose votre DID. Notez que ce que vous sélectionnez ici, s'appliquera à tous les appels.

Pour sélectionner une option, vous devez cliquer sur le bouton radio, puis sélectionner le choix désiré dans la liste déroulante (dans le cas où vous avez plus d'une option pour le routage).

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

== Point de présence DID (PdP) ==

Ensuite, suivent les paramètres supplémentaires optionnels pour le DID (numéro).

* La première option est "'''Options de Basculement'''". Si vous cliquez sur le lien ''Afficher Les Options de Basculement'', vous verrez apparaître 3 sections de routage, qui offrent les mêmes options que dans la section précédente « routage », à la différence que celles-ci s'appliqueront aux appels si la destination est ''occupée'', ''non disponible'' ou ''aucune réponse''.

Cela vous permet de définir un routage personnalisé lorsque l'appel atteint l'un de ces 3 états, au lieu d'aller simplement à la boîte vocale.

*'''Point de présence DID''', ce paramètre est le serveur vers lequel vous allez connecter votre DID, afin d'acheminer les appels. Il doit correspondre au même serveur que vous utilisez sur votre dispositif [[http://wiki.voip.ms/article/Devices ATA]], [[http://wiki.voip.ms/article/PBXs PBX]] ou [[http://wiki.voip.ms/article/Softphones softphone]], afin de recevoir des appels.

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

== Configurations Additionnelles ==

* Ensuite, nous trouvons l'option '''[[Messagerie vocale | Messagerie Vocale associé avec le DID]]''', c'est là que vous devez assigner une messagerie vocale à votre DID.

*'''Temps de sonnerie en secondes''', est essentiellement le temps que l'appel sonnera avant qu'il ne bascule vers votre messagerie vocale.

*'''Recherche de nom d'identification de l'appelant''', lorsque vous activez cette option, le système effectue une recherche dans les bases de données du LIDB/CNAM (Line Information Database / Calling Name), afin de trouver le nom correspondant au numéro de votre appelant et ainsi l'afficher dans le ''Nom identification de l'appelant''.

*'''Préfixe du nom d'identification de l'appelant (facultatif)''', est un réglage optionnel qui consiste simplement à ajouter comme préfixe, le nom de votre choix, au ''Nom identification de l'appelant'' que vous recevez. Notez que cette option fonctionne même si vous n'avez pas activé option de l'[[ID de l'appelant | identification de l'appelant]] ou si vous ne recevez pas de "Nom identification de l'appelant". Cette option est particulièrement utile lorsque vous avez besoin de différencier les appels entrants provenant de différents numéros (DIDs), vers le même téléphone.

*'''Note''', c'est juste une description interne pour le DID, (optionnelle).

== Messagerie texte (SMS) ==

Dans cette section vous pouvez activer la Messagerie texte (Short Message Service). Lorsque cette option est sélectionnée, votre DID pourra recevoir des messages SMS et vous les transmettre en utilisant les méthodes sélectionnées.

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

Pour plusieurs renseignements, veuillez visiter notre page wiki [[Messagerie texte]]

'''Rappelez-vous qu'aucune de ces modifications ne sera enregistrée jusqu'à ce que vous
appuyiez sur le lien "Cliquer ici pour appliquer les changements".'''

= Modification de multiples DIDs en même temps (Editing Multiple DIDs at a time) =

Il existe 2 boutons supplémentaires que vous verrez lorsque vous accéderez à la page "Gestion des DID", du menu "Numéros DID", ce sont les options:

''"Modifier la sélection - tous les paramètres à la fois"''
<br />''"Modifier la sélection - un paramètre à la fois"''.

'''"Modifier la sélection - tous les paramètres à la fois"''', si vous choisissez cette option, vous entrez dans une page d'édition de DIDs, et la façon dont vous choisirez les paramètres de cette page, s'appliquera à tous les DIDs sélectionnés. Soyez prudent, car tel vous verrez vos dernières configurations dans cette page, même celles que vous n'aurez pas changées, tel elles s'appliqueront à tous les DIDs sélectionnés comme configuration finale.

Par exemple: Si vous sélectionnez une messagerie vocale, tous les DID vont utiliser la même messagerie vocale, si vous laissez un champ vide comme le "Préfixe du nom d'identification de l'appelant", tous les DID auront ce paramètre vide.
Cette option est utile si vous voulez que tous vos DID aient la même configuration, pour chacun des paramètres, y compris la note.

'''"Modifier la sélection - un paramètre à la fois"''', cette option vous permet d'accéder à la page d'édition de DIDs, mais cette fois-ci, chaque réglage a son propre bouton "Appliquer ", vous permettant donc de faire un changement à une section spécifique, sans affecter le reste. Cette option est utile si vous avez déjà vos DIDs configurés avec des paramètres différents, et vous devez seulement changer une option spécifique. En utilisant cette option, toutes les autres options des DIDs sélectionnés, ne seront pas affectées.

[[category:Guides en français]]

File:Mettre ajour liste.png

2020-05-21T16:19:00Z

Joseph:

File:Nouvelle export.png

2020-05-21T16:18:08Z

Joseph:

File:Exporter liste.png

2020-05-21T16:17:31Z

Joseph:

Konftel 300Wx IP

2020-05-18T19:43:35Z

Joseph:

[[File:Konftel-300Wx-IP.png|300px|thumb|left]]
<br clear="all" />

The [https://www.konftel.com/en/products/konftel-300wx 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.

<br clear="all" />

__TOC__

== Configuring a line ==

==='''Accessing The Web Interface'''===

In order to configure the 300WX device to be used along with our service, it is required to access it's web interface settings. For this, the IP address of the device must be acquired.

To perform this click on the menu and navigate to: " '''''Status''''' " , then select: " '''''Network''''' ". It should read back something as: " '''''192.168.0.1'''''

Once you have the IP address please open a web browser of your preference and at the URL bar enter the IP address you got by prepending: " '''''http://''''' " and access it. Once accessed, you'll be prompted to authenticate.

Profile: " '''''admin''''' "
Password: " '''''admin''''' "

[[File:kon300_main.png|thumb|none|600px]]

===Adding a SIP Server===

Once logged in, please navigate to: ''Server'' on the left menu and click ''Add Server''.

[[File:kon300_add_server.png|thumb|none|600px]]

Set up the account you will be using, with the following values

:*'''''Server Alias''''': An alias name for your server (It can be the server's name)
:*'''''NAT Adaption''''': Enabled
:*'''''Registrar''''': One of VoIP.ms multiple [[Choosing Server | servers]], you can choose the one closest to your location
:*'''''Outbound Proxy''''': Set the same server you set at ''Registrar''
:*'''''Reregistrarion time (s)''''': 300
:*'''''SIP Transport''''': TCP
:*'''''Keep Alive''''': Enabled
:*'''''Codec Priority''''': Here leave only the codecs you have enabled in your SIP account's settings

Then click "Save" and go to "Extensions"

Once done, click the ''Save'' and go to the "Extensions" menu on the left

[[File:kon300_server.png|thumb|none|600px]]

===Adding an Extension===

Here, click on "Add extension" and complete the fields with your sub account's information:

:*'''Extension''': 100000 (replace with your SIP main account or subaccount)
:*'''Authentication Username''': 100000 (replace with your SIP main account or subaccount)
:*'''Authentication password''': Your SIP account's password
:*'''Server''': The server you created on a [[#Adding_a_SIP_Server | previous step]] (Since you can have more than one server, be sure that is the one you need to use)

On the right, you will see listed the handsets available for your IP DECT, select the one you will use with the current extension (as marked within the image below)

Click on "Save" after you are done.

[[File:kon300_extension.png|thumb|none|600px]]

==Configuring with TLS / Encryption==

'''NOTE''': This section is optional. It explains in detail how to use encrypted traffic in the device. If you are not certain about how encrypted traffic works, or the benefits of encrypting SIP traffic, please contact technical support for more information.

The Konftel 300Wx is compatible with TLS, however, some settings need to be adjusted to have it working. This section will be assuming that your SIP account is already enabled to use TLS, if you have not enabled it yet, please follow these instructions before going further:

For more information on how to enable encrypted traffic for the main account, please click on [[Call_Encryption_-_TLS/SRTP#Activate_This_Option_on_Your_Main_Account | Main account]] or more information on how to enable encrypted traffic for the sub-account [[Call_Encryption_-_TLS/SRTP#Activate_This_Option_on_Your_Sub_Account | sub account]]

===Adding a SIP Server with TLS===

Once logged in, please navigate to: ''Server'' on the left menu and click ''Add Server''.

[[File:kon300_add_server.png|thumb|none|600px]]

Set up the account you will be using, with the following values

:*'''''Server Alias''''': An alias name for your server (It can be the server's name)
:*'''''NAT Adaption''''': Enabled
:*'''''Registrar''''': One of VoIP.ms multiple [[Choosing Server | servers]], you can choose the one closest to your location
:*'''''Outbound Proxy''''': Set the same server you set at ''Registrar''
:*'''''Reregistrarion time (s)''''': 300
:*'''''SIP Transport''''': TCP
:*'''''Keep Alive''''': Enabled
:*'''''Codec Priority''''': Here leave only the codecs you have enabled in your SIP account's settings
:*'''''Sercure RTP''''': Enabled
:*'''''Secure RTP Auth''''': Enabled

Then click "Save" and go to "Extensions"

Once done, click the ''Save'' and go to the "Extensions" menu on the left

[[File:kon300_serverTLS.png|thumb|none|600px]]

===Adding an Extension with a TLS server===

Here, click on "Add extension" and complete the fields with your sub account's information:

:*'''Extension''': 100000 (replace with your SIP main account or subaccount)
:*'''Authentication Username''': 100000 (replace with your SIP main account or subaccount)
:*'''Authentication password''': Your SIP account's password
:*'''Server''': The server you created on a [[#Adding_a_SIP_Server | previous step]] (Since you can have more than one server, be sure that is the one you need to use)

On the right, you will see listed the handsets available for your IP DECT, select the one you will use with the current extension (as marked within the image below)

Click on "Save" after you are done.

[[File:kon300_extensionTLS.png|thumb|none|600px]]

== Verifying its status ==

If all was properly set and your device has connectivity, you will be registered. You can confirm this on the ''Status'' section on your device (besides your VoIP.ms customer portal)

Go to ''''' Extensions''''' and you should see information similar to the one below

[[File:kon300_extensionsRegistered.png|thumb|none|600px]]

[[category:SIP phones]]

Konftel 300Wx

2020-05-18T19:42:05Z

Joseph: moved Konftel 300Wx to Konftel 300Wx IP: Using proper name

#REDIRECT [[Konftel 300Wx IP]]

Konftel 300Wx IP

2020-05-18T19:42:05Z

Joseph: moved Konftel 300Wx to Konftel 300Wx IP: Using proper name

[[File:Konftel-300Wx-IP.png|300px|thumb|left]]
<br clear="all" />

The wireless conference phone [https://www.konftel.com/en/products/konftel-300wx Konftel 300Wx] allows you to hold meetings wherever is convenient for you – without worrying about network and power outlets. The wireless DECT technology is both reliable and secure. Choose a base station to suit your company's telephony environment, SIP or analog, or connect to an installed DECT system.

The rechargeable battery ensures more than 60 call hours, so you can talk for a full work week without recharging!

<br clear="all" />

__TOC__

== Configuring a line ==

==='''Accessing The Web Interface'''===

In order to configure the 300WX device to be used along with our service, it is required to access it's web interface settings. For this, the IP address of the device must be acquired.

To perform this click on the menu and navigate to: " '''''Status''''' " , then select: " '''''Network''''' ". It should read back something as: " '''''192.168.0.1'''''

Once you have the IP address please open a web browser of your preference and at the URL bar enter the IP address you got by prepending: " '''''http://''''' " and access it. Once accessed, you'll be prompted to authenticate.

Profile: " '''''admin''''' "
Password: " '''''admin''''' "

[[File:kon300_main.png|thumb|none|600px]]

===Adding a SIP Server===

Once logged in, please navigate to: ''Server'' on the left menu and click ''Add Server''.

[[File:kon300_add_server.png|thumb|none|600px]]

Set up the account you will be using, with the following values

:*'''''Server Alias''''': An alias name for your server (It can be the server's name)
:*'''''NAT Adaption''''': Enabled
:*'''''Registrar''''': One of VoIP.ms multiple [[Choosing Server | servers]], you can choose the one closest to your location
:*'''''Outbound Proxy''''': Set the same server you set at ''Registrar''
:*'''''Reregistrarion time (s)''''': 300
:*'''''SIP Transport''''': TCP
:*'''''Keep Alive''''': Enabled
:*'''''Codec Priority''''': Here leave only the codecs you have enabled in your SIP account's settings

Then click "Save" and go to "Extensions"

Once done, click the ''Save'' and go to the "Extensions" menu on the left

[[File:kon300_server.png|thumb|none|600px]]

===Adding an Extension===

Here, click on "Add extension" and complete the fields with your sub account's information:

:*'''Extension''': 100000 (replace with your SIP main account or subaccount)
:*'''Authentication Username''': 100000 (replace with your SIP main account or subaccount)
:*'''Authentication password''': Your SIP account's password
:*'''Server''': The server you created on a [[#Adding_a_SIP_Server | previous step]] (Since you can have more than one server, be sure that is the one you need to use)

On the right, you will see listed the handsets available for your IP DECT, select the one you will use with the current extension (as marked within the image below)

Click on "Save" after you are done.

[[File:kon300_extension.png|thumb|none|600px]]

==Configuring with TLS / Encryption==

'''NOTE''': This section is optional. It explains in detail how to use encrypted traffic in the device. If you are not certain about how encrypted traffic works, or the benefits of encrypting SIP traffic, please contact technical support for more information.

The Konftel 300Wx is compatible with TLS, however, some settings need to be adjusted to have it working. This section will be assuming that your SIP account is already enabled to use TLS, if you have not enabled it yet, please follow these instructions before going further:

For more information on how to enable encrypted traffic for the main account, please click on [[Call_Encryption_-_TLS/SRTP#Activate_This_Option_on_Your_Main_Account | Main account]] or more information on how to enable encrypted traffic for the sub-account [[Call_Encryption_-_TLS/SRTP#Activate_This_Option_on_Your_Sub_Account | sub account]]

===Adding a SIP Server with TLS===

Once logged in, please navigate to: ''Server'' on the left menu and click ''Add Server''.

[[File:kon300_add_server.png|thumb|none|600px]]

Set up the account you will be using, with the following values

:*'''''Server Alias''''': An alias name for your server (It can be the server's name)
:*'''''NAT Adaption''''': Enabled
:*'''''Registrar''''': One of VoIP.ms multiple [[Choosing Server | servers]], you can choose the one closest to your location
:*'''''Outbound Proxy''''': Set the same server you set at ''Registrar''
:*'''''Reregistrarion time (s)''''': 300
:*'''''SIP Transport''''': TCP
:*'''''Keep Alive''''': Enabled
:*'''''Codec Priority''''': Here leave only the codecs you have enabled in your SIP account's settings
:*'''''Sercure RTP''''': Enabled
:*'''''Secure RTP Auth''''': Enabled

Then click "Save" and go to "Extensions"

Once done, click the ''Save'' and go to the "Extensions" menu on the left

[[File:kon300_serverTLS.png|thumb|none|600px]]

===Adding an Extension with a TLS server===

Here, click on "Add extension" and complete the fields with your sub account's information:

:*'''Extension''': 100000 (replace with your SIP main account or subaccount)
:*'''Authentication Username''': 100000 (replace with your SIP main account or subaccount)
:*'''Authentication password''': Your SIP account's password
:*'''Server''': The server you created on a [[#Adding_a_SIP_Server | previous step]] (Since you can have more than one server, be sure that is the one you need to use)

On the right, you will see listed the handsets available for your IP DECT, select the one you will use with the current extension (as marked within the image below)

Click on "Save" after you are done.

[[File:kon300_extensionTLS.png|thumb|none|600px]]

== Verifying its status ==

If all was properly set and your device has connectivity, you will be registered. You can confirm this on the ''Status'' section on your device (besides your VoIP.ms customer portal)

Go to ''''' Extensions''''' and you should see information similar to the one below

[[File:kon300_extensionsRegistered.png|thumb|none|600px]]

[[category:SIP phones]]

Devices

2020-05-18T19:40:45Z

Joseph: /* Konftel 300Wx-IP */

== Blog Article ==

[https://wiki.voip.ms/article/VoIP_%E2%80%93_Bring_Your_Own_Device_(BYOD) VoIP – Bring Your Own Device (BYOD)]
<br />
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).

* '''ATA:''' An '''Analog Telephone Adapter''' is a device used to connect one or more standard analog telephones to a digital telephone system (such as Voice over IP) or a non-standard telephone system. Still not clear? Take a peek at our blog article : [https://wiki.voip.ms/article/Back_to_Basics_-_What_is_an_ATA%3F Back to Basics - What is an ATA?]

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

__TOC__

==ATAs==

===Atcom AG188N===

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

'''Product:''' Atcom AG188N

'''Company:''' Atcom

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

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

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

===Auerswald COMpact 5010===

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

'''Product:''' Auerswald COMpact 5010

'''Company:''' Auerswald

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

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

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

===Cisco===

====Cisco Linksys PAP2====

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

'''Product:''' Linksys PAP2

'''Company:''' Cisco

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

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

====Cisco Linksys PAP2T====

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

'''Product:''' Linksys PAP2T

'''Company:''' Cisco

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

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

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

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

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

'''Product:''' SPA112, SPA122

'''Company:''' Cisco

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

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

Introduced in late 2011, this box represents an inexpensive means to continue using existing analog hardware while migrating to voice over IP.

[[Cisco SPA112|See Configuration Details]]

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

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

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

'''Product:''' WRP400, WRP500

'''Company:''' Cisco

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

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

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

[[Cisco WRP400|See Configuration Details]]

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

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

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

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

'''Company:''' Cisco

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

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

[[Cisco SPA2100 Phone Adapter|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====Cisco SPA2102 Phone Adapter with Router====

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

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

'''Company:''' Cisco

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

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

[[Cisco_SPA2102_Phone_Adapter_with_Router|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

===Grandstream===

====Grandstream HandyTone 486====

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

'''Product:''' Grandstream HandyTone 486

'''Company:''' Grandstream

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

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

[[Grandstream_HandyTone_486|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====Grandstream HandyTone 502 - HT502====

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

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

'''Company:''' Grandstream

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

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

Enhanced security

Automated provisioning using symmetric and asymmetric voice

Support for a broad range of popular voice codec

[[Grandstream_HandyTone_502_-_HT502|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====Grandstream HandyTone 702 - HT702====

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

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

'''Company:''' Grandstream

'''Overview:''' The HT702 is Grandstream´s Newest powerful VoIP router. The product's inclusion of an integrated high-performance NAT router and 10Mbps Ethernet WAN and LAN ports enables a shared broadband connection between multiple ethernet devices.

In addition to being SIP 2.0 standard compliant, the product supports Universal Plug-in-Play (UPnP), up to 2 SIP account profiles/Phones and advanced telephony features.

Enhanced Security

Automated provisioning using symmetric and asymmetric voice

Support for a broad range of popular voice codec

[[Grandstream_HandyTone_702_-_HT702|See Configuration Details]]
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====Grandstream HandyTone 802 - HT802====

[[File:HT802 Device.jpg|300px|thumb|left|Grandstream HandyTone 802 - HT802]]

'''Product:''' Grandstream HandyTone 802 - HT802

'''Company:''' Grandstream

'''Overview:''' The HT802 delivers powerful VoIP technology and routing capabilities to home and office environments, and allows users to successfully connect their analog devices to a manageable and powerful VoIP network. Built upon Grandstream’s market-leading SIP ATA/gateway technology, with millions of units successfully deployed worldwide, this powerful ATA features exceptional voice quality in various applications and environments.

The HT802 comes with 2 easy-to-use FXS ports, state-of-the-art encryption with a unique security certificate per unit, automated provisioning for volume deployment and device management and outstanding network performance.

TLS and SRTP security encryption technology to protect calls and accounts

Supports 2 SIP profiles through 2 FXS ports and a single 10/100Mbps port

Support for a broad range of popular voice codec

[[Grandstream HandyTone 802 - HT802|See Configuration Details]]
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===Mediatrix C7 and Mediatrix 4100 Series===

[[File:Mediatrix_C7.jpg|300px|thumb|left|Mediatrix C7]]
[[File:Mediatrix_4100_1_media5.jpg|300px|thumb|left|Mediatrix 4102]]

'''Product:''' : Mediatrix C7 and Mediatrix 4100 Series running DGW 2.0 firmware

'''Company:''' Media5 Corporation

'''Overview:''' The Mediatrix® C7 and Mediatrix® 4100 Series feature VoIP adaptors that interconnect up to 24 analog telephones, faxes, and modems ports into SIP-based systems. These series allow operators to shorten deployment time to enable cloud telephony services into branch offices and SMBs.

The DGW 2.0 firmware also offers security features such as SIP over TLS, SRTP, certificates management, and HTTPS designed to bring enhanced security for the network management, SIP signaling, and media transmission aspects.

[[Media5 Mediatrix C7 and 4100|See Configuration Details]]
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===Netgear WGR615V===

[[File:NetgearWGR615V.jpg|300px|thumb|left|Netgear WGR615V]]

'''Product:''' WGR615V

'''Company:''' Netgear

'''Overview:''' The WGR615V was never sold to directly to the public when it was new but since the model is discontinued, lots of them are being sold.
It is a wireless (802.11 G) router that happens to have a built-in ATA.
It is possible to use just the ATA part.

[[Netgear WGR615V|See Configuration Details]]
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=== OBiHAI (Polycom) ===

====OBi 100/110 & OBi 200====

[[File:OBi110-ATA.jpg|300px|thumb|left|OBi110]]

'''Product:''' OBi 100/110 & OBi 200

'''Company:''' Polycom (OBIHAI Technology Inc)

'''Overview:''' With OBi devices you are in control of your communications life. From the OBi's on-board connections as well as via the Internet, you have the power to bridge mobile, fixed line and Internet telephone services. The OBi100 & OBi110 are stand-alone, dedicated devices, built with a high-performance "system on a chip" platform to ensure high-quality voice conversations. Every OBi device has high availability and reliability. It is 'always-on' to make or receive a call. With an OBi device, a computer is not required and a computer does not need to be on to talk on the phone or use the OBi's powerful service bridging features. To get started, all you need is a phone, power and a connection to the Internet.

The OBi110 has one phone port and one line port. Use the OBi110 when you need an analog line connection to a traditional telephone network or service. Think of the OBi110 LINE port as a gateway to traditional phone service. If you do not have traditional phone service at home, then the OBi100 is probably the right product to get.

[[OBi 100/110 & OBi 200|See Configuration Details]]
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====Polycom OBi300====

[[File:obi300.png|300px|thumb|left|Polycom OBi300]]

'''Product:''' Polycom OBi300

'''Company:''' Polycom

'''Overview:''' Polycom OBi300 expands your service portfolio by enhancing the communication possibilities of home offices with flexibility in voice, fax and modem applications as they transition to the digital communications world.
* Flexible voice and fax applications
* Help home offices maximize current analog investment by keeping one analog phone or fax machine and connect them to VoIP services
* Supports T.38 fax standard so home office users can send or receive reliable facsimile calls over the Internet

[[OBi300|See Configuration Details]]
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===Telco AC-211===

[[File:Ac211n.jpg|300px|thumb|left|Telco AC-211]]

'''Product:''' Telco AC-211

'''Company:''' Telco

'''Overview:''' Telco AC-211 is a SIP ATA supporting two lines and numerous configuration options. This ATA is most commonly found as the AC-211-SR & AC-211N from the now-defunct SunRocket service. This device works well with VoIP.ms once configured.

[[Telco_AC-211|See Configuration Details]]

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===TP-Link TD-VG3631===

[[File:Tplink vg3631 front.png|300px|thumb|left|TPLink VG3631]]

'''Product:''' TP-Link TD-VG3631

'''Company:''' TP-Link

'''Overview:''' High-speed DSL modem, NAT router, and wireless access point in one device providing a one-stop networking solution Wireless N speed up to 300Mbps makes it ideal for heavy bandwidth consuming or interruption sensitive applications like online gaming, Internet calls and even the HD video streaming Supporting both traditional landlines and VoIP network offers you a multiple choice when making phone calls. Various call features such as caller ID, call waiting, call holding, call forwarding, 3-way conference calls and voicemail USB 2.0 port supports Storage Sharing (Samba or FTP), Media Server, Print Server and multi accounts Ethernet WAN (EWAN) offers another broadband connectivity option for connecting to Cable, VDSL or Fiber modems IP based Bandwidth Control makes it easier for you to manage the bandwidth of devices that connected to the modem router Wi-Fi On/Off Button allows users to simply turn their wireless radio on or off.

[[TP-Link TD-VG3631|See Configuration Details]]
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==IP Phones==

===3COM 3108 Wireless Phone===

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

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

'''Company:''' 3COM

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

[[3COM_3108_Wireless_Phone|See Configuration Details]]
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===Aastra 6730i/6731i VoIP Phone===

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

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

'''Company:''' Aastra

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

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

[[Aastra_6730i_VoIP_Phone|See Configuration Details]]
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===Audiocodes===

====400HD Series====

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

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

'''Company:''' Audiocodes

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

[[Audiocodes 400HD|See Configuration Details]]
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===Cisco===

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

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

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

'''Company:''' Cisco

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

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

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

[[Cisco IP Phone 68XX and 88XX|See Configuration Details]]
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====Cisco Linksys SPA942 NA====

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

'''Product:''' Linksys SPA942 NA

'''Company:''' Cisco

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

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

[[Cisco_Linksys_SPA942_NA|See Configuration Details]]
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====Cisco SPA525G====

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

'''Product:''' Cisco SPA525G

'''Company:''' Cisco

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

[[Cisco SPA525G|See Configuration Details]]
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====Cisco IP Phone 7940/7960====

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

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

'''Company:''' Cisco

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

[[Cisco_IP_Phone_7940/7960|See Configuration Details]]
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====Cisco SPA30x and SPA50x series IP phones====

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

'''Product:''' Cisco SPA504G Phone

'''Company:''' Cisco

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

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

[[Cisco_SPA504G_Phone|See Configuration Details]]
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===Fanvil ===

====Fanvil X4G====

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

'''Product:''' Fanvil X4G

'''Company:''' Fanvil

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

[[Fanvill X4G|See Configuration Details]]
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[[Category:Devices]]

===Fortinet===

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

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

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

'''Company:''' Fortinet

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

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

[[Fortinet_FortiFone_FON-570|See Configuration Details]]
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====FortiFone FON-375====

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

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

'''Company:''' Fortinet

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

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

[[Fortinet_FortiFone_FON-375|See Configuration Details]]
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====FortiFone FON-175====

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

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

'''Company:''' Fortinet

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

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

[[Fortinet_FortiFone_FON-175|See Configuration Details]]
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===Gigaset A510 IP===

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

'''Product:'''Gigaset A510 IP

'''Company:'''Gigaset

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

[[Gigaset_A510_IP| See Configuration Details]]
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===Grandstream===

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

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

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

'''Company:''' Grandstream

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

[[Grandstream_DP715/DP710| See Configuration Details]]
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====Grandstream DP750/DP720====

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

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

'''Company:''' Grandstream

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

[[Grandstream_DP750/DP720| See Configuration Details]]
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====Grandstream 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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===Konftel===
==== Konftel 300Wx IP ====
[[File:Konftel-300Wx-IP.png|300px|thumb|left|Konftel 300Wx IP]]

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

'''Company:''' Konftel

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

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

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

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

'''Product:''' Konftel 300IPx

'''Company:''' Konftel

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

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

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

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

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

'''Company:''' Panasonic

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

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

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

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

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

'''Company:''' Panasonic

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

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

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

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

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

'''Company:''' Panasonic

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

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

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

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

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

'''Company:''' Panasonic

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

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

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

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

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

'''Company:''' Pirelli

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

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

[[Pirelli DP-L10|See Configuration Details]]
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===Polycom===

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

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

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

'''Company:''' Polycom

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

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

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

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

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

'''Company:''' Polycom

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

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

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

[[Polycom_SoundPoint_IP_501|See Configuration Details]]
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====Polycom SoundPoint IP 601====

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

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

'''Company:''' Polycom

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

[[Polycom_SoundPoint_IP_601|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====Polycom VVX 300, 400, etc====

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

'''Product:''' Polycom VVX Series

'''Company:''' Polycom

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

[[Polycom VVX 300, 400, etc|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

=== Positron IP phones ===

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

'''Product:''' Positron

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

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

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

==== Positron IP304 ====

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

'''Product:''' Positron IP304

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

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

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

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

[[Positron IP304 | View configuration for Positron IP304]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==== Positron IP304C ====

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

'''Product:''' Positron IP304C

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

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

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

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

[[Positron IP304C | View configuration for Positron IP304C]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==== Positron IP408 ====

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

'''Product:''' Positron IP408

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

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

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

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

[[Positron IP408 | View configuration for Positron IP408]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==== Positron IP410C ====

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

'''Product:''' Positron IP410C

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

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

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

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

[[Positron IP410C | View configuration for Positron IP410C]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==== Positron IP410G ====

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

'''Product:''' Positron IP410G

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

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

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

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

[[Positron IP410G | View configuration for Positron IP410G]]
<div style="width:100%;overflow:hidden;clear:both"></div>

===Siemens Gigaset C450-Ip===

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

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

'''Company:''' Siemens

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

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

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

[[Siemens Gigaset C450-Ip|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

===Snom===

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

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

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

'''Company:''' Snom

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

[[SNOM 320|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====Snom m3 VoIP Phone====

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

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

'''Company:''' Snom

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

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

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

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

====SNOM C520====

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

'''Product:''' SNOM C520 Conferencing

'''Company:''' Snom

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

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

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

====SNOM professional D7XX====

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

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

'''Company:''' Snom

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

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

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

=== Vtech ===

==== Vtech Conference Station ====

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

'''Product:''' VCSV752 & CS754

'''Company:''' Vtech

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

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

==== Vtech VSP Series ====

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

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

'''Company:''' Vtech

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

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

===Yealink===

====Yealink Voice Solutions====

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

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

'''Company:''' Yealink

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

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

====Yealink SIP-T28P (VSRF)====

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

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

'''Company:''' Yealink

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

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

[[Yealink_SIP-T28P|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>
[[Category:Devices]]

===Zycoo ZP502===

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

'''Product:''' Zycoo ZP502

'''Company:''' Zycoo ZP502

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

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

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

File:Sms-mms-icon.png

2020-05-07T17:36:21Z

Joseph:

File:Guide-001.png

2020-04-27T20:19:03Z

Joseph:

VoIP.ms

2020-04-22T18:38:51Z

Joseph:

{| cellspacing="0" style="width: 100%; border: 0; padding: 0 0 0 0; background: transparent;"
|-
| style="vertical-align:top;" |
{| cellspacing="0" style="height: {{#if:{{{1|}}}|115pt|100pt}}; border: 2px {{{border|gainsboro}}} solid; background: {{{bgcolor|white}}}; width: 100%; margin-bottom: 0.5em;"
|-
| style="padding: 0.5em; width: 100pt;" |
<span style="font-size:400%; line-height: 1.3em;"><center>VoIP.ms</center></span>

<div style="text-align:center; font-family:Arial; word-spacing: 60px"><ins>'''[[Bienvenue|Français]]'''</ins> <ins>'''[[Bienvenido|Español]]'''</ins> </div>

__TOC__

<table style="font-family:Arial; width:60%; border: 1px solid black; border-collapse:collapse" align="center">
<tr style="background-color:darkgray; color:white">
<th><ins>'''Popular Articles'''</ins> </th>
<th><ins>'''Latest Articles'''</ins> </th>
<th><ins>'''Frequently Asked Questions'''</ins> </th>
<th><ins>'''Popular Devices'''</ins></th>
</tr>
<tr style="align:left">
<td>[[Getting Started|Getting Started]] </td>
<td>Cool Stuff</td>
<td>[[FAQ#Can_I_port_my_existing_number_from_another_provider_to_VoIP.ms.3F|Can I Port My Number...?]] </td>
<td>[[Cisco_SPA112|Cisco SPA112]]</td>
</tr>
<tr>
<td>[[Voicemail|Voicemail]] </td>
<td>Cool Stuff</td>
<td>[[FAQ#Can_I_use_my_existing_device_with_VoIP.ms.3F|Can I Use My Existing device with VoIP.ms?]]</td>
<td>[[OBi_100/110|OBi 100/110]]</td>
</tr>
<tr>
<td>[[Choosing Server|Choosing Server]] </td>
<td>Cool Stuff</td>
<td> [[FAQ#Can_I_register_2_or_more_different_devices_with_the_same_account_username.3F|Can I Register 2 Different Devices...?]] </td>
<td>[[Grandstream_HandyTone_702_-_HT702|Grandstream HandyTone 702]]</td>
</tr>
<tr >
<td>[[Manage DID|Manage DID]] </td>
<td>Cool Stuff</td>
<td>[[Choosing_Server#IPs|What are the IP addresses of VoIP.ms' servers ?]]</td>
<td>[[Cisco_Linksys_PAP2T|Cisco Linksys PAP2T]]</td>
</tr>
<tr>
<td><span style="color:blue">More in The Guides Section</span> </td>
<td>Cool Stuff</td>
<td>[[FAQ|See all>]] </td>
<td> [[Devices|See all>]] </td>
</tr>
</table>
<br clear="all" />

== VoIP.ms Wiki ==

Welcome to the VoIP.ms Community Wiki! VoIP.ms is the most reliable, affordable and customizable VoIP experience on the market and this Wiki will help navigate the features.

In this wiki you can find guides, information and configuration examples for configuring your PBX, SBC, Device or Softphone.

We encourage users from the community to collaborate along with VoIP.ms staff to add guides, information and other articles pertinent to the service and products.

If you want to help by contributing with a new article or adding information to existing articles, please write to [mailto:support@voip.ms support@voip.ms] requesting for edit permission for the username you use in this wiki.

==Configuration Samples==
<span style="font-size:200%; line-height: 1.3em;"><center>[[Devices]] - [[Softphones]] - [[PBXs]] - [[Session Border Controllers | SBCs]]</center></span>

==Contact Us==

* Technical Support - [mailto:support@voip.ms support@voip.ms]
* Feedback - [mailto:feedback@voip.ms feedback@voip.ms]
* Porting - [mailto:ports@voip.ms ports@voip.ms]
* Sales and General Information - [mailto:sales@voip.ms sales@voip.ms]
** Toll Free, USA/Canada - 1.877.7.VOIP.MS (1.877.786.4767)
** Worldwide - 1.214.615.8599
* FAX
** Toll Free, USA - 1.888.311.7782
** Dallas, Texas, USA - 1.214.723.7555

==Social Media==

* [https://www.facebook.com/VoIP.ms Facebook]
* [https://twitter.com/voipms Twitter]
* [https://www.instagram.com/voip.ms Instagram]
* [https://www.linkedin.com/company/voip-ms LinkedIn]
* [https://www.youtube.com/channel/UCB6Mbg1XUoM5DTKeQ5oegZA/VoIP.ms YouTube Channel]

Calling Queue

2020-04-14T13:24:16Z

Joseph: Redirected page to Calling Queues

#REDIRECT [[Calling Queues]]

VoIP.ms

2020-04-10T16:40:05Z

Joseph: /* Configuration Samples */

{| cellspacing="0" style="width: 100%; border: 0; padding: 0 0 0 0; background: transparent;"
|-
| style="vertical-align:top;" |
{| cellspacing="0" style="height: {{#if:{{{1|}}}|115pt|100pt}}; border: 2px {{{border|gainsboro}}} solid; background: {{{bgcolor|white}}}; width: 100%; margin-bottom: 0.5em;"
|-
| style="padding: 0.5em; width: 100pt;" |
<span style="font-size:400%; line-height: 1.3em;"><center>VoIP.ms</center></span>

<div style="text-align:center; font-family:Arial; word-spacing: 60px"><ins>'''[[Bienvenue|Français]]'''</ins> <ins>'''[[Bienvenido|Español]]'''</ins> </div>

__TOC__

<table style="font-family:Arial; width:60%; border: 1px solid black; border-collapse:collapse" align="center">
<tr style="background-color:darkgray; color:white">
<th><ins>'''Popular Articles'''</ins> </th>
<th><ins>'''Frequently Asked Questions'''</ins> </th>
<th><ins>'''Popular Devices'''</ins></th>
</tr>
<tr>
<td>[[Getting Started|Getting Started]] </td>
<td>[[FAQ#Can_I_port_my_existing_number_from_another_provider_to_VoIP.ms.3F|Can I Port My Number...?]] </td>
<td>[[Cisco_SPA112|Cisco SPA112]]</td>
</tr>
<tr>
<td>[[Voicemail|Voicemail]] </td>
<td>[[FAQ#Can_I_use_my_existing_device_with_VoIP.ms.3F|Can I Use My Existing device with VoIP.ms?]]</td>
<td>[[OBi_100/110|OBi 100/110]]</td>
</tr>
<tr>
<td>[[Choosing Server|Choosing Server]] </td>
<td> [[FAQ#Can_I_register_2_or_more_different_devices_with_the_same_account_username.3F|Can I Register 2 Different Devices...?]] </td>
<td>[[Grandstream_HandyTone_702_-_HT702|Grandstream HandyTone 702]]</td>
</tr>
<tr >
<td>[[Manage DID|Manage DID]] </td>
<td>[[Choosing_Server#IPs|What are the IP addresses of VoIP.ms' servers ?]]</td>
<td>[[Cisco_Linksys_PAP2T|Cisco Linksys PAP2T]]</td>
</tr>
<tr>
<td><span style="color:blue">More in The Guides Section</span> </td>
<td>[[FAQ|See all>]] </td>
<td> [[Devices|See all>]] </td>
</tr>
</table>

== VoIP.ms Wiki ==

Welcome to the VoIP.ms Community Wiki! VoIP.ms is the most reliable, affordable and customizable VoIP experience on the market and this Wiki will help navigate the features.

In this wiki you can find guides, information and configuration examples for configuring your PBX, SBC, Device or Softphone.

We encourage users from the community to collaborate along with VoIP.ms staff to add guides, information and other articles pertinent to the service and products.

If you want to help by contributing with a new article or adding information to existing articles, please write to [mailto:support@voip.ms support@voip.ms] requesting for edit permission for the username you use in this wiki.

==Configuration Samples==
<span style="font-size:200%; line-height: 1.3em;"><center>[[Devices]] - [[Softphones]] - [[PBXs]] - [[Session Border Controllers | SBCs]]</center></span>

==Contact Us==

* Technical Support - [mailto:support@voip.ms support@voip.ms]
* Feedback - [mailto:feedback@voip.ms feedback@voip.ms]
* Porting - [mailto:ports@voip.ms ports@voip.ms]
* Sales and General Information - [mailto:sales@voip.ms sales@voip.ms]
** Toll Free, USA/Canada - 1.877.7.VOIP.MS (1.877.786.4767)
** Worldwide - 1.214.615.8599
* FAX
** Toll Free, USA - 1.888.311.7782
** Dallas, Texas, USA - 1.214.723.7555

==Social Media==

* [https://www.facebook.com/VoIP.ms Facebook]
* [https://twitter.com/voipms Twitter]
* [https://www.instagram.com/voip.ms Instagram]
* [https://www.linkedin.com/company/voip-ms LinkedIn]
* [https://www.youtube.com/channel/UCB6Mbg1XUoM5DTKeQ5oegZA/VoIP.ms YouTube Channel]

Call Recording

2020-04-10T16:30:01Z

Joseph: Redirected page to Call Recordings

#REDIRECT [[Call Recordings]]

MediaWiki:Sidebar

2020-04-09T17:40:54Z

Joseph:

* VoIP.ms Wiki
** http://voip.ms| VoIP.ms Website
** mainpage|Welcome
** Features|Features
** FAQ|FAQ
** Porting FAQ|Porting FAQ

* VoIP.ms Blog
** Blog| Blog

* Configuration
** devices|Devices
** softphones|Softphones
** PBXs|PBXs
** Session Border Controllers|Session Border Controllers

* Guides (English)
** Getting Started|Getting Started
** Account Settings|Account Settings
** Add Articles|Add Articles
** Audio Conferencing|Audio Conferencing
** Calculating my expenses|Calculating my expenses
** Call Detail Records|Call Detail Records
** Call Encryption - TLS/SRTP|Call Encryption - TLS/SRTP
** Call Forwarding|Call Forwarding
** Call Hunting|Call Hunting
** Call quality issues|Call quality issues
** Call Recordings|Call Recordings
** Callback|Callback
** Caller ID|Caller ID
** CallerID Filtering|CallerID Filtering
** CallerID number spoofing|CallerID number spoofing
** Calling Queues|Calling Queue
** Calls Cost|Calls Cost
** Changing DID Billing Plan|Changing DID Billing Plan
** Choosing Server|Choosing Server
** Dial Plan for Linksys ATAs|Dial Plan for Linksys ATAs
** Dialing Codes|Dialing Codes
** Dialing Rules|Dialing Rules
** Dialing Rules and Patterns|Dialing Rules and Patterns
** DID Troubleshooting|DID Troubleshooting
** DigitalReceptionist_IVR|Digital Receptionist (IVR)
** DISA|DISA
** E911|E911
** Extra_Services_Costs|Extra Services Costs
** False_Answer_Supervision_FAS|FAS (False Answer Supervision)
** Finances|Finances
** Firewall|Firewall
** General_Security|General Security
** https://wiki.voip.ms/article/Finances#Generate_Invoice | Generate invoice
** International_Calls|International Calls
** Manage DID|Manage DID
** Order_a_DID_Number|Order a DID Number
** PBX Security|PBX Security
** Phone book|Phone book
** Porting a Number|Porting a Number
** Problems logging in|Problems logging in
** Pulse_dial|Pulse Dial
** Recordings|Recordings
** Referral Program|Referral Program
** Registration issue|Registration Issue
** Registration status on desktop|Registration status on desktop
** Remove a DID|Remove a DID
** Reseller Basic Guide|Reseller Basic Guide
** Ring Groups|Ring Groups
** Service Cost|Service Cost
** SIP Requests|SIP Requests
** SIP_Responses|SIP Responses
** SIP URI|SIP URI
** Smartphone|Smartphone
** SMS|SMS
** Sub Accounts|Sub Accounts
** Time Conditions|Time Conditions
** Transaction_history|Transaction history
** Troubleshooting Outgoing Calls|Troubleshooting Outgoing Calls
** Two-step Verification|Two-step Verification
** TOTP Authentication|TOTP Authentication
** Unlocking All International Destinations|Unlocking All International Destinations
** Value vs Premium|Value vs Premium
** Virtual Fax|Virtual Fax
** Voicemail|Voicemail

* Guides (Français)
** Bienvenue|Bienvenue
** Mise en Marche|Mise en Marche
** Ajouter_un_Article| Ajouter un Article
**Audioconférence|Audioconférence
** Calculer_mes_dépenses|Calculer mes dépenses
** Changement du plan de facturation d'un Numéro|Changement du plan de facturation d'un Numéro
** Codes de Composition|Codes de Composition
** Commander_un_numéro_DID|Commander un numéro DID
** Conditions Temporelles | Conditions Temporelles
** Coût des Services Rendus|Coût des Services Rendus
** Cryptage des appels - TLS/SRTP|Cryptage des appels - TLS/SRTP
** Débloquer les destinations internationales | Débloquer les destinations internationales
** Détails des appels|Détails des appels
** Accès direct en entrée au système - DISA | DISA
** Effacer un Numéro DID|Effacer un Numéro DID
** Enregistrements | Enregistrements
**Enregistrements d'appels|Enregistrements d'appels
** File d'attente | File d'attente
** Filtrage du numéro d'identification de l'appelant | Filtrage du numéro d'identification de l'appelant
** Finances_Fr|Finances
** Fonction de Rappel | Fonction de Rappel
** Gérer les numéros DID|Gérer les numéros DID
** Groupe de Sonnerie|Groupe de Sonnerie
** ID de l'appelant | ID de l'appelant
** Messagerie texte | Messagerie texte
** Messagerie vocale | Messagerie vocale
** Paramètres du compte | Paramètres du compte
** Programme de référencement|Programme de référencement
** Problèmes_de_Connexion|Problèmes de Connexion
** Problème d'enregistrement|Problème d'enregistrement
** Questions Les Plus Fréquentes | Questions Les Plus Fréquentes
** Recherche d’Appel | Recherche d'Appel
** Renvoi d'appel | Renvoi d'appel
** Répertoire téléphonique | Répertoire téléphonique
** Réceptionniste virtuelle IVR | Réceptionniste virtuelle IVR
** Revendeur_Guide_Elementaire|Revendeur Guide Elementaire
** Service_E911|Service E911
** SIP_URI_FR|SIP URI
** Sous Comptes|Sous Comptes
** Télécopieur virtuel | Télécopieur virtuel
** Transférabilité des DID | Transférabilité des DID
** Value vs Premium (Français)|Value vs Premium

* Guías (Español)
** Bienvenido|Bienvenido
** Primeros Pasos|Primeros Pasos
** Preguntas_frecuentes_LNP|Preguntas frecuentes LNP
** Acceso directo al sistema interno (DISA)|Acceso directo al sistema interno
** Buzón de voz (Voicemail)|Buzón de voz
** Calculando mis gastos|Calculando mis gastos
** Cambiar Plan de DID (Changing DID Billing Plan)|Cambiar Plan de DID
** Cancelar/Borrar_un_DID| Cancelar/Borrar un DID
** Códigos de Marcado|Códigos de Marcado
** Condiciones de Tiempo (Time Conditions)|Condiciones de Tiempo
** Configuración de DID (Manage DID)|Configuración de DID
** Configuraciones de la Cuenta (Account Settings)|Configuraciones de la Cuenta
** Costo del servicio|Costo del servicio
** Costo por llamada|Costo por llamada
** Desbloqueo de todas las llamadas internacionales|Desbloqueo de todas las llamadas internacionales
** Desvío de Llamadas (Call Forwarding)|Desvio de Llamadas
** Dial Plan para ATA Linksys|Dial Plan para ATA Linksys
** Dirección URI (SIP URI)|Dirección URI
** Directorio (Phone Book)|Directorio
** E911_Spanish|E911 Espanol
** Elegir servidor|Elegir servidor
** Eliminar un DID|Eliminar un DID
** Estatus_de_registro_en_escritorio|Estatus de registro en escritorio
** Filtro de Llamadas (CallerID Filtering)|Filtro de Llamadas
** FAS_(supervisión_de_respuesta_falsa)|FAS (supervisión de respuesta falsa)
** Guía Básica de Reseller|Guía Básica de Reseller
** Grabaciones (Recordings)|Grabaciones
** Grupo de timbre / Ring Groups|Grupo de timbre
** Llamadas en Cola (Calling Queues)|Llamadas en Cola
** Numero Identificador (Caller ID)|Numero Identificador
** Porteo de numeros|Porteo de numeros
** Problemas de audio|Problemas de audio
** Problemas de registro|Problemas de registro
** Problemas Para Iniciar Sesión|Problemas Para Iniciar Sesión
** Recepcionista_Digital|Recepcionista_Digital (IVR)
** Registro de Llamadas (CDR)|Registro de Llamadas
** Reglas_y_Patrones_de_Marcado|Reglas y Patrones de Marcado
** Regreso de LLamada (Callback)|Regreso de LLamada
** Respuestas_SIP|Respuestas SIP
** Seguridad en PBX|Seguridad en PBX
** SIP_URI_español|SIP URI(Español)
** Solución de problemas de llamadas salientes|Solución de problemas de llamadas salientes
** Sub Cuentas (Sub Accounts)|Sub Cuentas
** Suplantación de identidad (CallerID spoofing)|Suplantación de identidad
** Value vs. Premium|Value vs. Premium

* SEARCH
* TOOLBOX
* LANGUAGES

MediaWiki:Sidebar

2020-04-09T17:38:25Z

Joseph:

* VoIP.ms Wiki
** http://voip.ms| VoIP.ms Website
** mainpage|Welcome
** Features|Features
** FAQ|FAQ
** Porting FAQ|Porting FAQ

* VoIP.ms Blog
** Blog| Blog

* Configuration
** devices|Devices
** PBXs|PBXs
** Session Border Controllers|Session Border Controllers
** softphones|Softphones

* Guides (English)
** Getting Started|Getting Started
** Account Settings|Account Settings
** Add Articles|Add Articles
** Audio Conferencing|Audio Conferencing
** Calculating my expenses|Calculating my expenses
** Call Detail Records|Call Detail Records
** Call Encryption - TLS/SRTP|Call Encryption - TLS/SRTP
** Call Forwarding|Call Forwarding
** Call Hunting|Call Hunting
** Call quality issues|Call quality issues
** Call Recordings|Call Recordings
** Callback|Callback
** Caller ID|Caller ID
** CallerID Filtering|CallerID Filtering
** CallerID number spoofing|CallerID number spoofing
** Calling Queues|Calling Queue
** Calls Cost|Calls Cost
** Changing DID Billing Plan|Changing DID Billing Plan
** Choosing Server|Choosing Server
** Dial Plan for Linksys ATAs|Dial Plan for Linksys ATAs
** Dialing Codes|Dialing Codes
** Dialing Rules|Dialing Rules
** Dialing Rules and Patterns|Dialing Rules and Patterns
** DID Troubleshooting|DID Troubleshooting
** DigitalReceptionist_IVR|Digital Receptionist (IVR)
** DISA|DISA
** E911|E911
** Extra_Services_Costs|Extra Services Costs
** False_Answer_Supervision_FAS|FAS (False Answer Supervision)
** Finances|Finances
** Firewall|Firewall
** General_Security|General Security
** https://wiki.voip.ms/article/Finances#Generate_Invoice | Generate invoice
** International_Calls|International Calls
** Manage DID|Manage DID
** Order_a_DID_Number|Order a DID Number
** PBX Security|PBX Security
** Phone book|Phone book
** Porting a Number|Porting a Number
** Problems logging in|Problems logging in
** Pulse_dial|Pulse Dial
** Recordings|Recordings
** Referral Program|Referral Program
** Registration issue|Registration Issue
** Registration status on desktop|Registration status on desktop
** Remove a DID|Remove a DID
** Reseller Basic Guide|Reseller Basic Guide
** Ring Groups|Ring Groups
** Service Cost|Service Cost
** SIP Requests|SIP Requests
** SIP_Responses|SIP Responses
** SIP URI|SIP URI
** Smartphone|Smartphone
** SMS|SMS
** Sub Accounts|Sub Accounts
** Time Conditions|Time Conditions
** Transaction_history|Transaction history
** Troubleshooting Outgoing Calls|Troubleshooting Outgoing Calls
** Two-step Verification|Two-step Verification
** TOTP Authentication|TOTP Authentication
** Unlocking All International Destinations|Unlocking All International Destinations
** Value vs Premium|Value vs Premium
** Virtual Fax|Virtual Fax
** Voicemail|Voicemail

* Guides (Français)
** Bienvenue|Bienvenue
** Mise en Marche|Mise en Marche
** Ajouter_un_Article| Ajouter un Article
**Audioconférence|Audioconférence
** Calculer_mes_dépenses|Calculer mes dépenses
** Changement du plan de facturation d'un Numéro|Changement du plan de facturation d'un Numéro
** Codes de Composition|Codes de Composition
** Commander_un_numéro_DID|Commander un numéro DID
** Conditions Temporelles | Conditions Temporelles
** Coût des Services Rendus|Coût des Services Rendus
** Cryptage des appels - TLS/SRTP|Cryptage des appels - TLS/SRTP
** Débloquer les destinations internationales | Débloquer les destinations internationales
** Détails des appels|Détails des appels
** Accès direct en entrée au système - DISA | DISA
** Effacer un Numéro DID|Effacer un Numéro DID
** Enregistrements | Enregistrements
**Enregistrements d'appels|Enregistrements d'appels
** File d'attente | File d'attente
** Filtrage du numéro d'identification de l'appelant | Filtrage du numéro d'identification de l'appelant
** Finances_Fr|Finances
** Fonction de Rappel | Fonction de Rappel
** Gérer les numéros DID|Gérer les numéros DID
** Groupe de Sonnerie|Groupe de Sonnerie
** ID de l'appelant | ID de l'appelant
** Messagerie texte | Messagerie texte
** Messagerie vocale | Messagerie vocale
** Paramètres du compte | Paramètres du compte
** Programme de référencement|Programme de référencement
** Problèmes_de_Connexion|Problèmes de Connexion
** Problème d'enregistrement|Problème d'enregistrement
** Questions Les Plus Fréquentes | Questions Les Plus Fréquentes
** Recherche d’Appel | Recherche d'Appel
** Renvoi d'appel | Renvoi d'appel
** Répertoire téléphonique | Répertoire téléphonique
** Réceptionniste virtuelle IVR | Réceptionniste virtuelle IVR
** Revendeur_Guide_Elementaire|Revendeur Guide Elementaire
** Service_E911|Service E911
** SIP_URI_FR|SIP URI
** Sous Comptes|Sous Comptes
** Télécopieur virtuel | Télécopieur virtuel
** Transférabilité des DID | Transférabilité des DID
** Value vs Premium (Français)|Value vs Premium

* Guías (Español)
** Bienvenido|Bienvenido
** Primeros Pasos|Primeros Pasos
** Preguntas_frecuentes_LNP|Preguntas frecuentes LNP
** Acceso directo al sistema interno (DISA)|Acceso directo al sistema interno
** Buzón de voz (Voicemail)|Buzón de voz
** Calculando mis gastos|Calculando mis gastos
** Cambiar Plan de DID (Changing DID Billing Plan)|Cambiar Plan de DID
** Cancelar/Borrar_un_DID| Cancelar/Borrar un DID
** Códigos de Marcado|Códigos de Marcado
** Condiciones de Tiempo (Time Conditions)|Condiciones de Tiempo
** Configuración de DID (Manage DID)|Configuración de DID
** Configuraciones de la Cuenta (Account Settings)|Configuraciones de la Cuenta
** Costo del servicio|Costo del servicio
** Costo por llamada|Costo por llamada
** Desbloqueo de todas las llamadas internacionales|Desbloqueo de todas las llamadas internacionales
** Desvío de Llamadas (Call Forwarding)|Desvio de Llamadas
** Dial Plan para ATA Linksys|Dial Plan para ATA Linksys
** Dirección URI (SIP URI)|Dirección URI
** Directorio (Phone Book)|Directorio
** E911_Spanish|E911 Espanol
** Elegir servidor|Elegir servidor
** Eliminar un DID|Eliminar un DID
** Estatus_de_registro_en_escritorio|Estatus de registro en escritorio
** Filtro de Llamadas (CallerID Filtering)|Filtro de Llamadas
** FAS_(supervisión_de_respuesta_falsa)|FAS (supervisión de respuesta falsa)
** Guía Básica de Reseller|Guía Básica de Reseller
** Grabaciones (Recordings)|Grabaciones
** Grupo de timbre / Ring Groups|Grupo de timbre
** Llamadas en Cola (Calling Queues)|Llamadas en Cola
** Numero Identificador (Caller ID)|Numero Identificador
** Porteo de numeros|Porteo de numeros
** Problemas de audio|Problemas de audio
** Problemas de registro|Problemas de registro
** Problemas Para Iniciar Sesión|Problemas Para Iniciar Sesión
** Recepcionista_Digital|Recepcionista_Digital (IVR)
** Registro de Llamadas (CDR)|Registro de Llamadas
** Reglas_y_Patrones_de_Marcado|Reglas y Patrones de Marcado
** Regreso de LLamada (Callback)|Regreso de LLamada
** Respuestas_SIP|Respuestas SIP
** Seguridad en PBX|Seguridad en PBX
** SIP_URI_español|SIP URI(Español)
** Solución de problemas de llamadas salientes|Solución de problemas de llamadas salientes
** Sub Cuentas (Sub Accounts)|Sub Cuentas
** Suplantación de identidad (CallerID spoofing)|Suplantación de identidad
** Value vs. Premium|Value vs. Premium

* SEARCH
* TOOLBOX
* LANGUAGES

Session Border Controllers

2020-04-09T17:35:22Z

Joseph: Created page with "A session border controller (SBC) is a network element deployed to protect SIP-based voice over Internet Protocol (VoIP) networks. Early deployments of SBCs were focused on the b..."

A session border controller (SBC) is a network element deployed to protect SIP-based voice over Internet Protocol (VoIP) networks. Early deployments of SBCs were focused on the borders between two service provider networks in a peering environment.

MediaWiki:Sidebar

2020-04-09T17:34:17Z

Joseph:

* VoIP.ms Wiki
** http://voip.ms| VoIP.ms Website
** mainpage|Welcome
** Features|Features
** FAQ|FAQ
** Porting FAQ|Porting FAQ

* VoIP.ms Blog
** Blog| Blog

* Configuration
** Session Border Controllers|Session Border Controllers
** devices|Devices
** PBXs|PBXs
** softphones|Softphones

* Guides (English)
** Getting Started|Getting Started
** Account Settings|Account Settings
** Add Articles|Add Articles
** Audio Conferencing|Audio Conferencing
** Calculating my expenses|Calculating my expenses
** Call Detail Records|Call Detail Records
** Call Encryption - TLS/SRTP|Call Encryption - TLS/SRTP
** Call Forwarding|Call Forwarding
** Call Hunting|Call Hunting
** Call quality issues|Call quality issues
** Call Recordings|Call Recordings
** Callback|Callback
** Caller ID|Caller ID
** CallerID Filtering|CallerID Filtering
** CallerID number spoofing|CallerID number spoofing
** Calling Queues|Calling Queue
** Calls Cost|Calls Cost
** Changing DID Billing Plan|Changing DID Billing Plan
** Choosing Server|Choosing Server
** Dial Plan for Linksys ATAs|Dial Plan for Linksys ATAs
** Dialing Codes|Dialing Codes
** Dialing Rules|Dialing Rules
** Dialing Rules and Patterns|Dialing Rules and Patterns
** DID Troubleshooting|DID Troubleshooting
** DigitalReceptionist_IVR|Digital Receptionist (IVR)
** DISA|DISA
** E911|E911
** Extra_Services_Costs|Extra Services Costs
** False_Answer_Supervision_FAS|FAS (False Answer Supervision)
** Finances|Finances
** Firewall|Firewall
** General_Security|General Security
** https://wiki.voip.ms/article/Finances#Generate_Invoice | Generate invoice
** International_Calls|International Calls
** Manage DID|Manage DID
** Order_a_DID_Number|Order a DID Number
** PBX Security|PBX Security
** Phone book|Phone book
** Porting a Number|Porting a Number
** Problems logging in|Problems logging in
** Pulse_dial|Pulse Dial
** Recordings|Recordings
** Referral Program|Referral Program
** Registration issue|Registration Issue
** Registration status on desktop|Registration status on desktop
** Remove a DID|Remove a DID
** Reseller Basic Guide|Reseller Basic Guide
** Ring Groups|Ring Groups
** Service Cost|Service Cost
** SIP Requests|SIP Requests
** SIP_Responses|SIP Responses
** SIP URI|SIP URI
** Smartphone|Smartphone
** SMS|SMS
** Sub Accounts|Sub Accounts
** Time Conditions|Time Conditions
** Transaction_history|Transaction history
** Troubleshooting Outgoing Calls|Troubleshooting Outgoing Calls
** Two-step Verification|Two-step Verification
** TOTP Authentication|TOTP Authentication
** Unlocking All International Destinations|Unlocking All International Destinations
** Value vs Premium|Value vs Premium
** Virtual Fax|Virtual Fax
** Voicemail|Voicemail

* Guides (Français)
** Bienvenue|Bienvenue
** Mise en Marche|Mise en Marche
** Ajouter_un_Article| Ajouter un Article
**Audioconférence|Audioconférence
** Calculer_mes_dépenses|Calculer mes dépenses
** Changement du plan de facturation d'un Numéro|Changement du plan de facturation d'un Numéro
** Codes de Composition|Codes de Composition
** Commander_un_numéro_DID|Commander un numéro DID
** Conditions Temporelles | Conditions Temporelles
** Coût des Services Rendus|Coût des Services Rendus
** Cryptage des appels - TLS/SRTP|Cryptage des appels - TLS/SRTP
** Débloquer les destinations internationales | Débloquer les destinations internationales
** Détails des appels|Détails des appels
** Accès direct en entrée au système - DISA | DISA
** Effacer un Numéro DID|Effacer un Numéro DID
** Enregistrements | Enregistrements
**Enregistrements d'appels|Enregistrements d'appels
** File d'attente | File d'attente
** Filtrage du numéro d'identification de l'appelant | Filtrage du numéro d'identification de l'appelant
** Finances_Fr|Finances
** Fonction de Rappel | Fonction de Rappel
** Gérer les numéros DID|Gérer les numéros DID
** Groupe de Sonnerie|Groupe de Sonnerie
** ID de l'appelant | ID de l'appelant
** Messagerie texte | Messagerie texte
** Messagerie vocale | Messagerie vocale
** Paramètres du compte | Paramètres du compte
** Programme de référencement|Programme de référencement
** Problèmes_de_Connexion|Problèmes de Connexion
** Problème d'enregistrement|Problème d'enregistrement
** Questions Les Plus Fréquentes | Questions Les Plus Fréquentes
** Recherche d’Appel | Recherche d'Appel
** Renvoi d'appel | Renvoi d'appel
** Répertoire téléphonique | Répertoire téléphonique
** Réceptionniste virtuelle IVR | Réceptionniste virtuelle IVR
** Revendeur_Guide_Elementaire|Revendeur Guide Elementaire
** Service_E911|Service E911
** SIP_URI_FR|SIP URI
** Sous Comptes|Sous Comptes
** Télécopieur virtuel | Télécopieur virtuel
** Transférabilité des DID | Transférabilité des DID
** Value vs Premium (Français)|Value vs Premium

* Guías (Español)
** Bienvenido|Bienvenido
** Primeros Pasos|Primeros Pasos
** Preguntas_frecuentes_LNP|Preguntas frecuentes LNP
** Acceso directo al sistema interno (DISA)|Acceso directo al sistema interno
** Buzón de voz (Voicemail)|Buzón de voz
** Calculando mis gastos|Calculando mis gastos
** Cambiar Plan de DID (Changing DID Billing Plan)|Cambiar Plan de DID
** Cancelar/Borrar_un_DID| Cancelar/Borrar un DID
** Códigos de Marcado|Códigos de Marcado
** Condiciones de Tiempo (Time Conditions)|Condiciones de Tiempo
** Configuración de DID (Manage DID)|Configuración de DID
** Configuraciones de la Cuenta (Account Settings)|Configuraciones de la Cuenta
** Costo del servicio|Costo del servicio
** Costo por llamada|Costo por llamada
** Desbloqueo de todas las llamadas internacionales|Desbloqueo de todas las llamadas internacionales
** Desvío de Llamadas (Call Forwarding)|Desvio de Llamadas
** Dial Plan para ATA Linksys|Dial Plan para ATA Linksys
** Dirección URI (SIP URI)|Dirección URI
** Directorio (Phone Book)|Directorio
** E911_Spanish|E911 Espanol
** Elegir servidor|Elegir servidor
** Eliminar un DID|Eliminar un DID
** Estatus_de_registro_en_escritorio|Estatus de registro en escritorio
** Filtro de Llamadas (CallerID Filtering)|Filtro de Llamadas
** FAS_(supervisión_de_respuesta_falsa)|FAS (supervisión de respuesta falsa)
** Guía Básica de Reseller|Guía Básica de Reseller
** Grabaciones (Recordings)|Grabaciones
** Grupo de timbre / Ring Groups|Grupo de timbre
** Llamadas en Cola (Calling Queues)|Llamadas en Cola
** Numero Identificador (Caller ID)|Numero Identificador
** Porteo de numeros|Porteo de numeros
** Problemas de audio|Problemas de audio
** Problemas de registro|Problemas de registro
** Problemas Para Iniciar Sesión|Problemas Para Iniciar Sesión
** Recepcionista_Digital|Recepcionista_Digital (IVR)
** Registro de Llamadas (CDR)|Registro de Llamadas
** Reglas_y_Patrones_de_Marcado|Reglas y Patrones de Marcado
** Regreso de LLamada (Callback)|Regreso de LLamada
** Respuestas_SIP|Respuestas SIP
** Seguridad en PBX|Seguridad en PBX
** SIP_URI_español|SIP URI(Español)
** Solución de problemas de llamadas salientes|Solución de problemas de llamadas salientes
** Sub Cuentas (Sub Accounts)|Sub Cuentas
** Suplantación de identidad (CallerID spoofing)|Suplantación de identidad
** Value vs. Premium|Value vs. Premium

* SEARCH
* TOOLBOX
* LANGUAGES

PRI Virtuels

2020-03-20T18:06:24Z

Joseph: Created page with "Tout d'abord, un VPRI est un accès primaire RNIS virtuel, qui n’a pas de quantité spécifique de "canaux" comme un PRI traditionnel, qui est généralement composé de 23 can..."

Tout d'abord, un VPRI est un accès primaire RNIS virtuel, qui n’a pas de quantité spécifique de "canaux" comme un PRI traditionnel, qui est généralement composé de 23 canaux et 1 canal de données gérant l'échange d'informations.

Avec un VPRI, le nombre de canaux est déterminé par la quantité d'appels entrants simultanés dont vous avez besoin. De plus, vous avez la possibilité d'ajouter plus d'un DID dans votre VPRI. De cette façon, ils partageront le même bassin de canaux entrants.

L’avantage d'un VPRI est qu’il vous permettra de payer par canal, au lieu d'une facturation entrante de paiement à l'utilisation / par minute.

''De plus, la différence entre un PRI et un VPRI avec VoIP.ms concerne uniquement les appels entrants. Cela signifie que pour tous les appels sortants, ils seront à votre taux par minute. Si vous avez un volume élevé d'appels sortants, nous pouvons certainement offrir un meilleur taux par minute. ''

=== Flexibilité ===
* Vous avez la possibilité d'ajouter plus de DID que le nombre de canaux dont vous disposez.
* Le nombre de canaux peut être changé à tout moment et au moment où vous le souhaitez, il n'y a pas de contrat.
* Nous proposons une solution élastique.

=== Limitations ===
Le VPRI est limité aux DID locaux uniquement, ce qui signifie qu'un numéro sans frais ne peut pas être ajouté dans un VPRI.

Si vous avez des DID américains (États-Unis) et des DID canadiens par exemple, ils ne peuvent pas être fusionnés.

Dans ce cas, deux VPRI différents seront nécessaires, un pour vos DID canadiens et un autre pour vos DID américains. Vous ne pouvez avoir qu'un seul VPRI par pays, tant qu'il y a des DID locaux disponibles dans ce pays.

Le tarif des canaux sera appliqué au montant total des canaux par pays (VPRI).

== Le concept de VPRI Élastique ==

Nous savons tous qu'avec un PRI régulier, si vous atteignez le nombre de canaux simultanés alloué, vous serez injoignable et votre client entendra une tonalité de congestion '' (tonalité occupée)''.

Grâce au concept d’élasticité, il est rassurant de savoir que nous vous offrons la possibilité de dépasser votre limite!

Votre client n'entendra plus jamais la tonalité occupée en raison de l'atteinte de votre limite d'appels simultanés.

=== Comment cela fonctionne ===
Supposons que vous disposiez à l'origine d'un VPRI à 50 canaux. Si à un moment vous avez reçu plus de 50 appels en même temps, l'appelant recevrait un signal occupé.

Pour éviter cette situation, vous pourriez décider d'autoriser votre VPRI à dépasser jusqu'à 100 appels, et vous paieriez seulement un léger supplément pour les canaux additionnels utilisés au-delà de votre limite initiale.

Contrairement à la plupart des produits PRI Virtuels élastiques de l'industrie, nous avons gardé à l'esprit notre approche de paiement à l'utilisation et sans tracas.

Par conséquent, nous facturons sur un modèle de paiement à l'utilisation.

Donc, si vous utilisez 10 canaux supplémentaires uniquement pour une journée dans un mois donné, nous ne vous facturerons pas les canaux pour tout le mois, mais uniquement pour cette journée très achalandée.

Si vous ne dépassez pas votre limite de base, le tarif mensuel pour votre VPRI sera le même qu'à l'habitude.

=== Comment puis-je avoir un VPRI Élastique? ===
Il vous suffit de nous contacter et de nous fournir le nombre de canaux dont vous avez besoin ainsi que les DID que vous souhaitez leur
associer.

De cette façon, nous serons en mesure de fournir un meilleur tarif.

Si vous utilisez déjà notre service et que vous souhaitez passer à un VPRI, la transition est fluide et sans interruption.

Aucune reconfiguration de votre côté n'est nécessaire.

Virtual PRI

2020-03-20T18:04:57Z

Joseph: Created page with "First, a vPRI is a virtual PRI, that's not including a specific amount of "channel" like traditional PRI, that’s usually composed of 23 channels and 1 data channel managing the..."

First, a vPRI is a virtual PRI, that's not including a specific amount of "channel" like traditional PRI, that’s usually composed of 23 channels and 1 data channel managing the exchange of information.

With a vPRI the amount is determined by how much concurrent incoming calls you need. You can add more than one DID, in your vPRI, in that way, they will share the same pool of incoming channels.

The value of a vPRI will allow you to pay per channels, instead of an incoming pay-per-use/per minute billings.

''Also, the difference between a PRI and a vPRI with VoIP.ms is only for the incoming calls traffic, it means that for all outbound calls will be at your rates/minutes. If you have a high volume of outgoing calls, we can surely explore with you the better rates per minute.''

=== Flexibility ===
*You can add more DIDs than the number of channels you have.
* The number of channels can be changed whenever you want at any time, there’s no contract.
* We offer burst solution

=== Limitations ===
The vPRI is limited to the local DIDs only, which means a toll-free number cannot be added to a vPRI.

Moreover, if you have some DIDs from the US and some DIDs from CDN for example, they cannot be merged together.

In this case, two different vPRI will be needed, one for all your CDN DIDs, another for all US DIDs. You can only have one vPRI per country as long as there are locals DID is available.

The price of the channels, considering volume discounts, will be applied to the total amount of channels per country.

== The Concept of “Bursting”, or On Demand. ==
We all know with a regular PRI, that if you reach the number of your concurrent channels, you will be unreachable, and your customer will hear a congestion tone ''(the famous busy tone)''.

It is now that the concept of “bursting” comes into place. It’s interesting that we offer you the opportunity to go beyond your limit!

Your customer will never hear the famous busy tone due to reaching your concurrent call limits.

=== How it's work - Scenario ===
Let’s pretend that in normal time, your organization requires an average of 50 inbound concurrent calls for a couple of DID. Instead of having a per minute incoming rate for each DID, you can have a fixed monthly price for those channels.

When the eventuality of receiving a peak of calls at the same time occurs, instead of having a busy signal and risking missing an important call, we give you the flexibility to receive more than 50 inbound channels at the same time. For safety, you can choose up to how many channels you want to be able to burst. Let’s say you decide to allow your vPRI to burst until 100 calls, and you would simply get charged a small premium for the additional channels used over your original limit.

We charge on a pay-per-use model where if you use 10 extra channels only for 1 day in a given month, we won’t charge you the channels for the entire month, but rather only for this specific busy day.

=== How can I have a vPRI? ===
You just need to contact us and provide us the number of channels you need, and which DIDs you want to link to it.

In this way, we will be able to provide a better price.

If you already use our service and you want to switch to a vPRI, the transition is smooth and without downtime.

No reconfiguration on your side is needed.

Softphones

2020-03-18T15:22:36Z

Joseph: /* Bria */

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 a softphone is designed to behave like a traditional telephone, sometimes appearing as an image of a phone with a display panel and buttons with which the user can interact. A softphone is usually used with a headset connected to the sound card of the PC, or as an installed app on your phone. See also: [[Smartphone]]

Take a peek at VoIP.ms Blog Article : [https://facebook.com/notes/voipms/back-to-basics-what-is-a-softphone-and-how-to-leverage-it/2799391566799404/?__tn__=HH-R Back to Basics, What is a Softphone and How to Leverage It?]

==Acrobits==

[[File:AcrobitsLogo.png|left|Acrobits]]

[[File:Acrobitscert.png|180px|right|Acrobits|link=https://www.acrobits.net//?utm_campaign=itsp-partners&utm_medium=cp-certified-logo&utm_source=voipms]]

'''Product:''' Acrobits

'''Company:''' [https://www.acrobits.net//?utm_campaign=itsp-partners&utm_medium=cp-certified-logo&utm_source=voipms Acrobits]

'''OS:''' IOS, Android

'''Overview:'''
Acrobits is a mobile software development company with a focus on SIP Clients for mobile devices; their SIP client, Acrobits Softphone, is the leading app of it's type on the iTunes App Store.

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

==Bria==

[[File:Softphone Clients - Smartphones.jpg|350px|left|Bria|link=https://www.counterpath.com/?utm_campaign=itsp-partners&utm_medium=cp-certified-logo&utm_source=voipms]]

[[File:CounterPath ITSP Certified Partner 2017.png|250px|right|CounterPath|link=https://www.counterpath.com/?utm_campaign=itsp-partners&utm_medium=cp-certified-logo&utm_source=voipms]]

'''Product:''' Bria

'''Company:''' [https://www.counterpath.com/?utm_campaign=itsp-partners&utm_medium=cp-certified-logo&utm_source=voipms CounterPath]

'''OS:''' Windows, MacOS, Linux (Ubuntu), iOS, Android

'''Overview:'''
Bria 5 is a carrier-grade next generation softphone application that enables you to manage your communications easily and efficiently
– all from your computer desktop. Replacing or complementing your hard phone, the Bria softphone allows you to make VoIP and Video calls over IP, see when your contacts are available, send Instant Messages and transfer files with ease and efficiency.

Bria Android Edition is a highly secure, standards-based mobile VoIP softphone that works over both 3G and Wi-Fi networks.

Using the device’s existing contact list, Bria Android Edition facilitates easy and effective communication management with an intuitive interface. Deskphone-class calling functionality includes the ability to swap between two calls, merge calls and perform attended and unattended transfers.

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

==Bria Teams==

[[File:Bria-teams-thumb.png|350px|left|Bria Teams|link=https://www.counterpath.com/?utm_campaign=itsp-partners&utm_medium=cp-certified-logo&utm_source=voipms]]

[[File:CounterPath ITSP Certified Partner 2017.png|250px|right|CounterPath|link=https://www.counterpath.com/?utm_campaign=itsp-partners&utm_medium=cp-certified-logo&utm_source=voipms]]

'''Product:''' [https://www.softphone.com/ Bria Teams]

'''Company:''' [https://www.counterpath.com/?utm_campaign=itsp-partners&utm_medium=cp-certified-logo&utm_source=voipms CounterPath]

'''OS:''' Windows, MacOS, iOS, Android

'''Overview:'''
Team Communication made Easy
Bria Teams helps you regain productivity by streamlining all team communications into one application across your devices. With Bria's provisioning, it's easy to get your team talking.

It offers a softphone that helps you gain productivity by streamlining all team communications into one application. It has all the tools you need in one interface, across your devices and can be managed by one simple dashboard.

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

==Dialer+==

[[File:Dialerplus.png|200px|thumb|left|Dialer+]]

'''Product:''' cd26

'''Company:''' cd26

'''OS:''' IOS

'''Overview:'''
Dialer+ is a SIP-based softphone for iPhone , iPod touch and iPad

Standard Phone Features
* Fast Contact Search – Use phone pad to find contacts by names, initials or any part of the numbers.
• iOS 4.0+ Background Support
• Supports GSM calls when the network is unavailable or slowly
• Multiple Account Support
• Auto switch to best service provider when use multiple account
• Pre-configured VoIP providers list
• Call display
• Speakerphone, Mute and Hold
• Call history
• Contact List and Contact Favorites - leveraging the iPhone Contacts
• Ringtones and contact avatars
• Multiple incoming Call Support - swap between two active calls; merge and split calls
• Audio codecs include G.711,G.722, iLBC ,Speex and GSM, Make an in app purchase to add G.729 for great performance over 3G networks
• Support for RFC 2833 DTMF
• VPN Support
• uPNP Support
• iPhone 4 Retina Display Support

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

==Ekiga==

[[File:Ekigaw0.png|200px|thumb|left|Ekiga Softphone]]

'''Product:''' Ekiga

'''Company:''' Ekiga

'''OS:''' Windows, Linux

'''Overview:'''Ekiga is a software phone and video conferencing application. It allows you to hear and see your friends for free using your computer and Internet. You can also chat with them and see if they are online or not. It works on GNU/Linux and Windows.

More specifically, Ekiga is a VoIP, IP Telephony, and Video Conferencing application that allows you to make audio and video calls to remote users with SIP or H.323 compatible hardware and software.

It supports many audio and video codecs and all modern VoIP features for both SIP and H.323. Ekiga is the first Open Source application to support both H.323 and SIP, as well as audio and video. Ekiga was formerly known as GnomeMeeting.

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

==Express Talk==

[[File:Express0.png|200px|thumb|left|Express Talk]]

'''Product:''' Express Talk

'''Company:''' NCH Software

'''OS:''' Windows, MacOS

'''Overview:''' Express Talk works like a telephone to let you make calls through your computer. Call anyone via the internet who also has a softphone installed and if you sign up with a VoIP gateway service company you call regular telephone numbers as well.

With Express Talk you could potentially call someone using their internet IP (eg. "bob@1.2.3.4") but this is not usually practical. It is much easier to use a friendly address like "bob@myphoneco.com". To have this you need to sign up with a SIP proxy service. This is almost always free to signup for and usually requires nothing more than an email address. See our Recommended SIP Service Providers List for a list of options.

Some of the companies also provide gateway services. This means connecting phone calls to ordinary (non internet or analog PSTN) phone lines. These calls tend to be much less expensive particularly for international calls compared to regular long distance charges since they can offer local call rates.

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

==FgVoIP==

[[File:Fgvoip0.jpeg|200px|thumb|left|FgVoIP]]

'''Product:''' FgVoIP

'''Company:''' FG Microtec

'''OS:''' BlackberryOS

'''Overview:''' fg's BlackBerry SIP VoIP Client is a feature rich soft phone application for BlackBerry devices. It can register to any SIP VoIP provider or IP/SIP capable PBX using a WiFi connection in office, campus, home, hotel rooms, public hot-spots etc.

Its much like having a BlackBerry desktop phone in your pocket, providing significantly higher overall productivity while at the same time lowering telephony costs.

Experience the freedom of choosing your own VoIP provider whenever you want, wherever you want. All you need is a WiFi network and a VoIP account.

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

==Fring==

[[File:Fring00.jpg|left|Fring]]

'''Product:''' Fring

'''Company:''' Fring

'''OS:''' Windows, MacOS, IOS, Linux, Android, Symbian

'''Overview:'''
Fring mobile communication service that gives users internet-rich communication from their mobile phones.

Fring members make FREE video calls, voice calls, live chat to other fringsters and to friends on other social networks.

You can now use your SIP account even from non-SIP enabled phones.

Choose whichever SIP provider you have and in just few clicks, start using this account to make calls to landlines, GSM phones, or to other SIP providers.

Supports Symbian S60, iPhone/ iPod touch, Android, Windows Mobile, MeeGo, Linux, J2MEand proprietary OS.

Fring allows you to easily switch between them, without having to reconfigure your handset through a nightmarish process.

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

==Grandstream Wave==

[[File:Grandstream Wave.png|200px|Thumb|left|Grandstream Wave]]

'''Product:''' Grandstream Wave

'''Company:''' Grandstream

'''OS:''' Android

'''Overview:'''
Grandstream Wave is a FREE softphone application that allows users to make and receive voice calls through their business or residential SIP accounts on any Android™ device (version 4.0+) from anywhere in the world.

This application supports integration of up to 6 SIP accounts, 6-way voice conferencing, and allows users to monitor their PBX (such as Grandstream's UCM6100 series IP PBX & UCM6510 IP PBX) while utilizing speed dial with up to 24 virtual BLF keys.

Grandstream Wave also supports advanced SIP telephony features including call transfer, LDAP phonebook integration and more.

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

==Jitsi==

[[File:Logo2.png|200px|left|Jitsi]]

'''Product:''' Jitsi

'''Company:''' Jitsi

'''OS:''' Windows, MacOS, Linux

'''Overview:''' Jitsi (formerly SIP Communicator) is an audio/video and chat communicator that supports protocols such as SIP, XMPP/Jabber, AIM/ICQ, Windows Live, Yahoo! and many other useful features.

Jitsi is an Open Source / Free Software, and is available under the terms of the LGPL.

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

==Kiax==

[[File:Kiax3.png|left|200px|Kiax]]

'''Product:''' Kiax

'''Company:''' Kiax

'''OS:''' Windows, MacOS, Linux

'''Overview:'''
Kiax is a softphone (soft phone, VoIP client) with a simple and comfortable user interface for making VoIP calls to Asterisk PBX. It depends on the iaxclient library to use Asterisk's IAX2 protocol for easy call configuration and audio setup.

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

==Linphone==

[[File:Linphone.jpg|left|200px|Linphone]]

'''Product''': Linphone

'''Company''': Linphone

'''OS:''' Windows, Linux, MacOSX, IPhone, Android.

'''App Storage Consumption:''' 8.09mb (ability to move app to SD card. Data uses an additional ~0.89mb approx.)

'''Overview:''' Linphone is an internet phone or Voice Over IP phone (VoIP). With linphone you can communicate freely with people over the internet, with voice, video, and text instant messaging. Linphone makes use of the SIP protocol, an open standard for internet telephony. You can use Linphone with any SIP VoIP operator.

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

==NetDial Sip Phone==

[[File:NetDial.jpg|left|200px|Linphone]]

'''Product''': NetDial Sip Phone

'''Company''': NeoMecca

'''OS:''' iOS, Android.

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

==Media5-fone==

[[File:Media5Android.jpg|left|Media5Android]]

'''Product:''' Media5-fone

'''Company:''' Media5 Corporation

'''OS:''' Android, iOS

'''Overview:'''
The Media5-fone is a softphone application that runs on the Android and the iOS Operating System. It is a SIP Client (softphone) that enables users to make and receive VoIP calls, enabling them to use their devices as an IP-PBX phone extension in their office or anywhere else in the world.

[[Media5-fone|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==MizuPhone==

[[File:Mizuphone3.png|left|200px|MizuPhone]]

'''Product:''' MizuPhone

'''Company:''' Mizutech

'''OS:''' Windows

'''Overview:'''
Mizu SoftPhone is our award winner professional VoIP client application based on the open standard SIP protocol with an easy to use modern interface. With Mizu SoftPhone you can connect to any SIP server on the public internet or on your local area network.

Multiple accounts and multiple SIP server registrations combined with a powerful dial plan can minimize your telecom bills while using the greatest features from the VoIP industry.

Features include built-in encryption, IM, presence, HD Video, history with voice records, skype like voice quality using ultra-wideband codecs and much more.

A free edition is also available based on the same engine as our corporate sip softphone but it has some features disabled (like G.729 and wideband codecs, multiple accounts and file transfer over SIP).

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

==NinjaLite==

[[File:Ninja6.png|left|200px|NinjaLite]]

'''Product:''' NinjaLite

'''Company:''' Global IP Telecommunications

'''OS:''' Windows

'''Overview:'''
Ninja Lite offers Audio- and video telephony for everybody, Ninja features individual skins and ringtones!, Ninja is natively Vista 32bit /64 bit and Windows7 compatible.

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

==PhonerLite==

[[File:PhonerLite.png|left|300px|PhonerLite]]

'''Product:''' PhonerLite

'''Company:''' PhonerLite

'''OS:''' Windows

'''Overview:'''
PhonerLite is a clearly arranged application for Windows. PhonerLite enables your PC to use it for internet telephony (VoIP , Voice over IP ). Pre-conditions are a full-duplex sound card , a microphone and speakers (alternativelya headset), an internet connection and a registration at a provider supporting the protocol SIP .PhonerLite supports several SIP profiles, each configurable independently. In thesame way the integrated phone book and call log are easy to use.

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

==PortGo==

[[File:Portgo3.png|left|200px|PortGo]]

'''Product:''' PortGo

'''Company:''' PortSIP Solutions, Inc

'''OS:''' Windows (desktop and mobile)

'''Overview:'''
PortGo is the newest SIP softphone from PortSIP, it's built base on PortSIP VoIP SDK, allowing users to enjoy multimedia communications in a dynamic way.

Featuring an intuitive interface, PortGo is expanding the softphone experience by making it even easier to make VoIP and Video over IP calls, see when your contacts are available and send Instant Messages. PortGo features an IM interface which focuses on your contacts and friends.

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

==QuteCom==

[[File:Qutecom3.png|left|200px|QuteCom]]

'''Product:''' QuteCom

'''Company:''' QuteCom

'''OS:''' Windows,MacOS,Linux

'''Overview:'''
QuteCom is a community of enthusiasts and developers, creating free software products related to communication over IP. The flagship product of the QuteCom project is a softphone which allows you to make free PC to PC video and voice calls, and to integrate all your IM contacts in one place.

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

==SIP Droid==

[[File:Sipdroid_00.png|200px|thumb|left|SIP Droid]]

'''Product:''' SIP Droid

'''Company:''' SIP Droid

'''OS:''' Android

'''App Storage Consumption:''' 1.85mb (ability to move app to SD card. Data uses an additional ~4kb approx.)

'''Overview:''' SIPDroid is a java based, open source SIP client that has recently been developed for use with mobile devices based on Google’s Android platform.

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

==SJPhone==

[[File:Sjphone0.png|200px|thumb|left|SJPhone]]

'''Product:''' SJPhone

'''Company:''' SJ Labs, Inc.

'''OS:''' Windows,Windows Mobile

'''Overview:'''
SJphone is a VOIP softphone that allows you to speak with any other softphone, any stand-alone IP-phone, or using ITSP with any traditional wired or mobile phone.

It supports both SIP and H.323 standards and is fully inter-operable with most major VOIP vendors and ITSP.

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

==Taki==

[[File:blackberry_z30_taki_oncall.png|200px|thumb|left|Taki]]

'''Product:''' Taki

'''Company:''' Taki

'''OS:''' BlackBerry 10, BlackBerry PlayBook

'''Project site:''' http://taki.sourceforge.net/

'''App in the AppWorld:''' http://appworld.blackberry.com/webstore/content/130295

'''Overview:''' Taki is a native SIP softphone for BlackBerry® PlayBook™ and
BlackBerry® 10 platforms.

'''Main Features:'''
- Multiple SIP accounts support
- Multiple simultaneous calls
- Call recording
- Conference calls
- Advanced call control: Transfer, Hold, Mute, Reject, Redial, switch
between multiple active calls
- Speaker phone support
- Address book. Integration with local contacts on BlackBerry 10 phones
- Comprehensive SIP settings (port binding, outbound proxy, registrar,
codecs priority, NAT traversal, STUN, Presence, and more)
- Transports: UDP/TCP
- Audio codecs: G.711a/u, GSM, iLBC, G.722 (HD), Speex (NB, WB, UWB)
- Built for BlackBerry

[[Taki|See Configuration Details]]

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

==Telephone==

[[File:telephone-user-interface.png|200px|thumb|left|Telephone]]

'''Product:''' Telephone

'''Company:''' 64 Characters

'''OS:''' OS X 10.9 or later

'''Project site:''' http://www.64characters.com/telephone/

'''Mac App Store:''' https://itunes.apple.com/us/app/telephone/id406825478

'''Overview:''' Telephone is a fully-featured SIP client with a minimalist user interface.

<br>

'''Main Features:'''

* Multiple SIP accounts
* Multiple simultaneous calls w/conferencing
* Call control: Transfer, Hold, Mute, Reject, Redial, Call Waiting, multiple active calls
* Speaker phone/headset support (via Mac sound device)
* Integration with Contacts app, dial by name, autocomplete, click to dial
* Send DTMF tones
* Languages: English, German, Russian

[[Telephone|See Configuration Details]]

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

==X-Lite==

[[File:Xlite_00.png|200px|thumb|left|X-Lite]]

'''Product:''' X-Lite

'''Company:''' CounterPath

'''OS:''' Windows, MacOS, Linux

'''Overview:''' CounterPath's X-Lite helps you seamlessly transition from a traditional phone environment into the world of Voice over IP.

The latest release of X-Lite provides a completely redesigned interface that allows for a contact-centric or dialpad-centric user experience, or a combination of the two. It also provides you with some of the most popular features of our fully loaded Bria and eyeBeam softphones so you can take them for a test drive before you make your purchase.

Having a simple voice conversation, you’ll soon see why having a softphone on your desktop or laptop is the ultimate communications experience.

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

==ZoiPer==

[[File:Zoiper0.png|300px|thumb|left|Zoiper Classic]]
[[File:zlogo.jpg|300px|thumb|right|https://www.zoiper.com/en]]

'''Product:''' ZoiPer

'''Company:''' ZoiPer

'''OS:''' Windows, Mac OS, Android, iOS

'''Overview:''' IAX & SIP multilanguage softphone is a VoIP soft client, meant to work with any IP-based communications systems and infrastructure.

Zoiper is available for Windows, Mac OS X and Linux and supports the following languages: English, German, Spanish, French, Dutch, Portuguese, Russian, Chinese, Japanese, Italian, Polish, Magyar.

*[[Zoiper_Classic|See Configuration Details for PC/MAC]]<br />
*[[Zoiper_5| See Configuration Details for ZoiPer 5]]<br />
*[[ZoIPer_for_Android|See Configuration Details for android]]<br/>
*[[ZoIPer_for_iOS|See Configuration Details for iOS]]<br/>
<div style="width:100%;overflow:hidden;clear:both"></div>
[[category:softphones]]

==ZoiPer Communicator==

[[File:Zoiperco1.png|200px|thumb|left|Zoiper Communicator]]
[[File:zlogo.jpg|300px|thumb|right|https://www.zoiper.com/en]]

'''Product:''' Zoiper Communicator

'''Company:''' Zoiper

'''OS:''' Windows, MacOS, Linux, Solaris

'''Overview:''' Zoiper Communicator IAX & SIP softphone is a converged Internet communication tool combining high-quality voice and video calls, fax, instant messaging and presence through a contact-centric intuitive interface.

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

File:Bria-teams-thumb.png

2020-03-18T15:21:35Z

Joseph:

Bria Teams

2020-03-18T15:12:28Z

Joseph: /* Service Settings using SIP TLS/SRTP */

[[File:BriaTeamsLogo.png|right|BriaTeams|300px|border]]
Bria Teams is a softphone that helps you gain productivity by streamlining all team communications into one application. It has all the tools you need in one interface, across your devices and can be managed by one simple dashboard.

Bria Team's App client is available on Windows 10+, Mac 10.13+, Android 5.0+ and IOS 11+

__TOC__

In order to allow each member of your team to make outgoing calls and receive calls you need to create a sub account for each of them, then link each one to a specific member of your team.

When you complete your Bria Teams registration. You will have access to your Bria Teams's dashboard. You will be invited to create an access for each members of your team at the first part.

For this wiki, we skip this part, and we will not create any other team member, you will be able to create them later.

== Configure your team ==
=== Set your codecs ===

The first of the configuration, we will need to specify the default codec, for all your devices. When you use the softphone on your Desktop, on your Mobile (by using your WiFi internet or mobile data) a specific codec may be use.

The allows codec needs to reflect your VoIP.ms sub-accounts parameters that you plan to use with your team.
This settings can be found when you edit or create a sub account and expend the "Advanced Options" section. This will be "Allowed Codecs".

----

In your Bria Teams dashboard, click on "'''Settings and Preferences'''" where the upper navigation bar.

:[[File:BriaTeams_SettingsPref.png|border]]

Edit the codec part by clicking on the [[File:BriaTeams_SetGear.png]] "Gear" button. Then edit the '''AUDIO CODECS''' part.

We suggest using G.711u as the primary and uncheck unwanted codecs. ''If you want to choose a different codec, such as G.729 for mobile cellular, you need to activate it into each sub-accounts.''
You can select multiple codec, simply drag each codec by priority.
Since is done, apply the change for each type of use. ''(Upper right devices, Desktop, Mobile Cellular, Mobile Wifi or the button NEXT at the bottom.)''

When is finish, press the button close [[File:BriaTeams_closebtn.png]].

:[[File:BriaTeams_SettingsCodecs.png|border|700px]]

== Create a Voice Configuration ==
=== Add a new Voice Service ===

Now you will need to add VoIP.ms as your Voice service provider.

Click on "'''Voice and Video'''" where the upper navigation bar.

:[[File:BriaTeams_VoiceVideo.png|border]]

Now click on [[File:BriaTeams_AddVoicebtn.png|120px]] or [[File:BriaTeams_AddPhonebtn.png|30px]]

:[[File:BriaTeams_BriaTeams_AddVoiceConfig.png|border|700px]]

Click on the right option, "'''Select From Providers'''".

:[[File:BriaTeams_VoiceVideo_SelectProviders.png|border|700px]]

You will find "'''VoIP.ms'''" in the list.

:[[File:BriaTeams_VoiceVideo_SelectProvidersVoIPms.png|border|700px]]

=== Configure your service settings ===
In this panel, you will need to specify some general information that will be used for every team member profile.

:[[File:BriaTeams_VoiceVideo_SelectProvidersVoIPmsSIP.png|border|700px]]

:* '''Service Label''': Only for your own purposes to be able to associate this profile to your team. ''(Per instance: MyTeamName)''
:* '''Domain''': This is your VoIP.ms PoP server, that you will use to register every user of your team. (This needs to match with your DID to be able to receive incoming calls.)
:* '''Port''': 5060 ''(alternative port may be: 5080 or 42872)''
:* '''Register with domain and receive calls''': [CHECKED]
:* '''Transport''': Automatic
:* '''Keep Alive''': Enabled
:* '''Voicemail''': Voicemail Number: *97
:* '''Service Options''': [CHECKED] This voice service requires an authorization username for each voice account.

==== Service Settings using SIP TLS/SRTP ====
----

'''IMPORTANT''': This section is optional. It explains in detail how to use encrypted traffic in the device. If you are not certain on how encrypted traffic works, or the benefits of encrypting SIP traffic, please contact technical support for more information.

These devices are compatible with TLS, however, some settings need to be adjusted to have it working. This section will be assuming that your SIP account is already enabled to use TLS, if you have not enabled it yet, please follow these instructions before going further:

For more information on how to enable encrypted traffic for the main account, please click on [[Call_Encryption_-_TLS/SRTP#Activate_This_Option_on_Your_Main_Account | Main account]] or more information on how to enable encrypted traffic for the sub-account [[Call_Encryption_-_TLS/SRTP#Activate_This_Option_on_Your_Sub_Account | sub account]]

----

In order to use the SIP TLS/SRTP call encryption, you need to activate this option into all sub-accounts of your team members, under '''advanced options''': '''Encrypt sip traffic''' when you edit or create a sub-account.

If you have already configured your voice and wish to activate traffic encryption, activate the option into each sub-account the SIP Traffic encryption and only select TLS in transport and activate SRTP into the Bria portal.

:[[File:BriaTeams_VoiceVideo_SelectProvidersVoIPmsSIPTLS.png|border|700px]]

:* '''Service Label''': ''(only for your own purposes to be able to associate this profile to your team.) (Per instance: MyTeamName)''
:* '''Domain''': Is your VoIP.ms PoP server, that you will use to register every user of your team. (This needs to match with your DID to be able to receive incoming calls)
:* '''Port''': 5061 ''(alternative port may be: 5081 or 42873)''
:* '''Register with domain and receive calls''': [CHECKED]
:* '''Transport''': TLS
:* '''SRTP''': Enabled
:* '''Keep Alive''': Enabled
:* '''Voicemail''': Voicemail Number: *97
:* '''Service Options''': [CHECKED] This voice service requires an authorization username for each voice account

=== Set Service Compatibility Options ===

Click on the tab "'''Service Compatibility Options'''" and under "'''Port ranges'''" [CHECK] the box of '''Restrict Port ranges used for RTP audio'''. Then indicate these RTP audio ports range.
:[[File:BriaTeams_VoiceVideo_SelectProvidersVoIPmsSIPSrvOpts.png|border|700px]]
:* '''Min''': 10001
:* '''Max''': 20000

Then click on [[File:BriaTeams_SaveAndCloseebtn.png|80px]]

== Team members ==
=== Link a sub-account and profile to a team member ===

In this part you will need to associate your team members, to your profile and configure each of them.

Go to "'''Team members'''" tab, in the upper navigation menu.

:[[File:BriaTeams_TeamMembers.png|border]]

Click on one of your team members and then on the left, "'''Add Voice Account'''" [[File:BriaTeams_AddVoiceAccount.png|100px]].

Select the Voice service that you've just created. ''(Per instance: MyTeamName)''

:[[File:BriaTeams_TeamMembersEdit.png|border]]

:* '''SIP USERNAME''': Your extension sub-account username
:* '''AUTHORIZATION USERNAME''': Your extension sub-account username
:* '''SIP/VOICE PASSWORD''': This is your sub-account password.
:* '''CALL DISPLAY''': Is your Outbound Caller ID Name. ''('''Note''': We suggest entering it in capital letters, and no longer than 15 characters, without special characters such as dots, commas or apostrophe.)''

Then click on [[File:BriaTeams_SaveAndCloseebtn.png|80px]]

'''Note:''' You need to have one sub account PER members/seats.

=== Install your softphone to your team member ===

When you create a new member with his email address, they will receive a link in order to download and install the softphone.
Then they will need to follow the instructions. A new password will need to be created by them using the email address that you used to create the member.

Bria Teams

2020-03-18T15:11:48Z

Joseph: /* Service Settings using SIP TLS/SRTP */

[[File:BriaTeamsLogo.png|right|BriaTeams|300px|border]]
Bria Teams is a softphone that helps you gain productivity by streamlining all team communications into one application. It has all the tools you need in one interface, across your devices and can be managed by one simple dashboard.

Bria Team's App client is available on Windows 10+, Mac 10.13+, Android 5.0+ and IOS 11+

__TOC__

In order to allow each member of your team to make outgoing calls and receive calls you need to create a sub account for each of them, then link each one to a specific member of your team.

When you complete your Bria Teams registration. You will have access to your Bria Teams's dashboard. You will be invited to create an access for each members of your team at the first part.

For this wiki, we skip this part, and we will not create any other team member, you will be able to create them later.

== Configure your team ==
=== Set your codecs ===

The first of the configuration, we will need to specify the default codec, for all your devices. When you use the softphone on your Desktop, on your Mobile (by using your WiFi internet or mobile data) a specific codec may be use.

The allows codec needs to reflect your VoIP.ms sub-accounts parameters that you plan to use with your team.
This settings can be found when you edit or create a sub account and expend the "Advanced Options" section. This will be "Allowed Codecs".

----

In your Bria Teams dashboard, click on "'''Settings and Preferences'''" where the upper navigation bar.

:[[File:BriaTeams_SettingsPref.png|border]]

Edit the codec part by clicking on the [[File:BriaTeams_SetGear.png]] "Gear" button. Then edit the '''AUDIO CODECS''' part.

We suggest using G.711u as the primary and uncheck unwanted codecs. ''If you want to choose a different codec, such as G.729 for mobile cellular, you need to activate it into each sub-accounts.''
You can select multiple codec, simply drag each codec by priority.
Since is done, apply the change for each type of use. ''(Upper right devices, Desktop, Mobile Cellular, Mobile Wifi or the button NEXT at the bottom.)''

When is finish, press the button close [[File:BriaTeams_closebtn.png]].

:[[File:BriaTeams_SettingsCodecs.png|border|700px]]

== Create a Voice Configuration ==
=== Add a new Voice Service ===

Now you will need to add VoIP.ms as your Voice service provider.

Click on "'''Voice and Video'''" where the upper navigation bar.

:[[File:BriaTeams_VoiceVideo.png|border]]

Now click on [[File:BriaTeams_AddVoicebtn.png|120px]] or [[File:BriaTeams_AddPhonebtn.png|30px]]

:[[File:BriaTeams_BriaTeams_AddVoiceConfig.png|border|700px]]

Click on the right option, "'''Select From Providers'''".

:[[File:BriaTeams_VoiceVideo_SelectProviders.png|border|700px]]

You will find "'''VoIP.ms'''" in the list.

:[[File:BriaTeams_VoiceVideo_SelectProvidersVoIPms.png|border|700px]]

=== Configure your service settings ===
In this panel, you will need to specify some general information that will be used for every team member profile.

:[[File:BriaTeams_VoiceVideo_SelectProvidersVoIPmsSIP.png|border|700px]]

:* '''Service Label''': Only for your own purposes to be able to associate this profile to your team. ''(Per instance: MyTeamName)''
:* '''Domain''': This is your VoIP.ms PoP server, that you will use to register every user of your team. (This needs to match with your DID to be able to receive incoming calls.)
:* '''Port''': 5060 ''(alternative port may be: 5080 or 42872)''
:* '''Register with domain and receive calls''': [CHECKED]
:* '''Transport''': Automatic
:* '''Keep Alive''': Enabled
:* '''Voicemail''': Voicemail Number: *97
:* '''Service Options''': [CHECKED] This voice service requires an authorization username for each voice account.

==== Service Settings using SIP TLS/SRTP ====

'''IMPORTANT''': This section is optional. It explains in detail how to use encrypted traffic in the device. If you are not certain on how encrypted traffic works, or the benefits of encrypting SIP traffic, please contact technical support for more information.

These devices are compatible with TLS, however, some settings need to be adjusted to have it working. This section will be assuming that your SIP account is already enabled to use TLS, if you have not enabled it yet, please follow these instructions before going further:

For more information on how to enable encrypted traffic for the main account, please click on [[Call_Encryption_-_TLS/SRTP#Activate_This_Option_on_Your_Main_Account | Main account]] or more information on how to enable encrypted traffic for the sub-account [[Call_Encryption_-_TLS/SRTP#Activate_This_Option_on_Your_Sub_Account | sub account]]

In order to use the SIP TLS/SRTP call encryption, you need to activate this option into all sub-accounts of your team members, under '''advanced options''': '''Encrypt sip traffic''' when you edit or create a sub-account.

If you have already configured your voice and wish to activate traffic encryption, activate the option into each sub-account the SIP Traffic encryption and only select TLS in transport and activate SRTP into the Bria portal.

:[[File:BriaTeams_VoiceVideo_SelectProvidersVoIPmsSIPTLS.png|border|700px]]

:* '''Service Label''': ''(only for your own purposes to be able to associate this profile to your team.) (Per instance: MyTeamName)''
:* '''Domain''': Is your VoIP.ms PoP server, that you will use to register every user of your team. (This needs to match with your DID to be able to receive incoming calls)
:* '''Port''': 5061 ''(alternative port may be: 5081 or 42873)''
:* '''Register with domain and receive calls''': [CHECKED]
:* '''Transport''': TLS
:* '''SRTP''': Enabled
:* '''Keep Alive''': Enabled
:* '''Voicemail''': Voicemail Number: *97
:* '''Service Options''': [CHECKED] This voice service requires an authorization username for each voice account

=== Set Service Compatibility Options ===

Click on the tab "'''Service Compatibility Options'''" and under "'''Port ranges'''" [CHECK] the box of '''Restrict Port ranges used for RTP audio'''. Then indicate these RTP audio ports range.
:[[File:BriaTeams_VoiceVideo_SelectProvidersVoIPmsSIPSrvOpts.png|border|700px]]
:* '''Min''': 10001
:* '''Max''': 20000

Then click on [[File:BriaTeams_SaveAndCloseebtn.png|80px]]

== Team members ==
=== Link a sub-account and profile to a team member ===

In this part you will need to associate your team members, to your profile and configure each of them.

Go to "'''Team members'''" tab, in the upper navigation menu.

:[[File:BriaTeams_TeamMembers.png|border]]

Click on one of your team members and then on the left, "'''Add Voice Account'''" [[File:BriaTeams_AddVoiceAccount.png|100px]].

Select the Voice service that you've just created. ''(Per instance: MyTeamName)''

:[[File:BriaTeams_TeamMembersEdit.png|border]]

:* '''SIP USERNAME''': Your extension sub-account username
:* '''AUTHORIZATION USERNAME''': Your extension sub-account username
:* '''SIP/VOICE PASSWORD''': This is your sub-account password.
:* '''CALL DISPLAY''': Is your Outbound Caller ID Name. ''('''Note''': We suggest entering it in capital letters, and no longer than 15 characters, without special characters such as dots, commas or apostrophe.)''

Then click on [[File:BriaTeams_SaveAndCloseebtn.png|80px]]

'''Note:''' You need to have one sub account PER members/seats.

=== Install your softphone to your team member ===

When you create a new member with his email address, they will receive a link in order to download and install the softphone.
Then they will need to follow the instructions. A new password will need to be created by them using the email address that you used to create the member.

VoIP.ms Essentials to Work from Home

2020-03-17T14:56:24Z

Joseph:

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[[File:16 03 20 blog workfromehome BLOG.jpg|center|VoIP.ms Essentials to Work from Home]]
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=== VoIP.ms Essentials to Work from Home ===
<br/>

Remote work has been part of our reality for quite a few years, but the recent outbreak of Coronavirus (COVID-19) has made it mandatory for most businesses in order to maintain business continuity. However, some businesses have never fully tested their work-from-home contingency plan.
This on-going crisis has forced many organizations to nearly adopt an overnight strategy. The team at VoIP.ms has gathered some of the key essentials to maintain business continuity while working from home.

=== 1. When it comes to communications, use the softphone of your choice. There is a plurality of options and it is set up in a matter of minutes. ===
<br/>

To start working from home, the first thing to take into consideration is to remain in touch with the same landline and extension that you have at work.
A softphone is a software application that can be added to any existing computer or mobile device.
As a result, there is no need for a business to invest in any additional hardware equipment. Softphones are amongst the most scalable option available on the market.
You or your IT department can install the software of your choice and get started right away. As long as you are connected to the internet, there is theoretically no place where you cannot make or receive a call.

To know more about softphones, please refer to our Wiki article here: https://wiki.voip.ms/article/Softphones

=== 2. An easy one, set up a Call Forward and increase your mobility. ===
<br/>

One common and most used features is Call Forwarding. This way, you can stay in touch with your customers, no matter where you are.
This feature allows an incoming call to be redirected to a mobile telephone or any other telephone number where the desired called party is able to answer.
You can set a Call Forwarding to any number, even international numbers.

Learn how to set it up right here: https://wiki.voip.ms/article/Call_Forwarding

=== 3. To bring your mobility to the next level, set up Call Hunting! ===
<br/>

The Call Hunting feature is commonly known as the “find me” feature. It simply means that an incoming call can be configured to ring multiple phones or devices, one after the other until it gets answered or gets transferred to the voicemail if the person is not available.

Here is the link to our step by step guide to set it up: https://wiki.voip.ms/article/Call_Hunting

=== 4. Got teams? Distribute your calls properly, set up some Ring Groups! ===
<br/>

Using the ring groups feature, you can easily distribute the incoming calls among the employees of a particular department like sales, customer support or tech.
In other words, the ring group feature allows you to have 8 phones to ring at the same time when one number or extension is dialed.

Here is the direct link to our step-by-step guide to set up yours: https://wiki.voip.ms/article/Ring_Group

=== 5. Stay in touch with everybody over phone, set up your conference bridge. ===
<br/>

Conference bridge provides a virtual room to simultaneously get in touch with a group of clients and/or your team!
Setting up your conference bridge is easy and can be done within a few seconds.

Get started right here: https://wiki.voip.ms/article/Audio_Conferencing

<br/>

We hope these few tips will help you and your team quickly build intuitive and powerful remote office workspaces. All these features can be used together in order to create an effective phone system to reflect your business reality even in a remote situation.
If this is still not clear to you or you need any assistance with your business continuity plan, reach out to our support team over live chat right here (https://voip.ms/?startchat=1) or send us an e-mail at support@voip.ms.

Your VoIP.ms Team

</div>

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______

For more information, visit us at <br/>
https://voip.ms or sign up now <br/>
to start making calls in under 5 minutes<br/>
at https://www.voip.ms/#Signup!<br/>

</div>

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__NOTOC__

SIP URI

2020-03-02T18:31:37Z

Joseph: /* Using your DID number */

A SIP URI is the SIP addressing schema to call another person via SIP. In other words, a SIP URI is a user's SIP phone number. The SIP URI resembles an e-mail address and is written in the following format: x@y:port (x=Username, y=host|domain|IP)

: Example: johnsmith@my-uri.com

A general description of SIP addressing is at [[wikipedia:SIP address]]. The addresses, which use the same user@domain... format as e-mail addresses, allow an individual Internet telephony user to be reached directly online without passing via the public switched telephone network or incurring the corresponding tolls.

A SIP address may be used as a destination to which to forward a voip.ms DID number, as a target for an individual speed dial entry (*75xx) in a voip.ms user address book or as a means to transfer incoming calls into your voip.ms extensions or numbers from outside Internet servers.

# One is to send calls to an external SIP URI, via your DID number,
# A second option is to receive calls via SIP URI, we can achieve this using our DID number or an internal extension from a [[Sub Accounts|sub account]].
# The third option is to use a Virtual number.

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

__TOC__

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

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

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

=== Creating a new SIP URI ===
To make a SIP URI that can be used with multiple DIDs, use the {DID} tag. The {DID} expression will be replaced automatically by the DID number as it is listed in the "Manage DID section".

[[Image:Forward.jpg]]

; Examples:
: 1{DID}@128.144.122.12
: 12143221234@128.144.122.12
: some_extension_name@128.144.122.12:5080
: other_extension_name@voip.example.com

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

=== Creating a phone book entry ===
A SIP URI may be associated with a [[phone book]] or speed dial entry in the same manner as any other telephone number.

See [[Phone book#Create a Phone Book Entry with a SIP URI]].

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

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

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

=== Codec Negotation ===

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

: disallow=all
: allow=ulaw

== Receiving incoming calls from a SIP URI ==

=== Using your DID number ===
You can receive SIP URI calls using the following format {Number}@sip.voip.ms, this can be used with your local US and Canada numbers, so they can be reached from outside.

[[Image:Did.jpg]]

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

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

:sip:4166471234@toronto.voip.ms

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

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

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

Important: no call flow or filtering can be applied to calls make to the external SIP URI. Calls will immediately ring the device registered to this sub-account.

[[Image:Extension.jpg]]

=== Using iNum ===
Any iNum number (from any provider) is a SIP URI; just append @sip.inum.net

For example, iNum 883510009999999 becomes:
:883510009999999@sip.inum.net

=== Using a Virtual number ===

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

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

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

[[File:Virtualsip.jpg]]

[[category:guides]]

Devices

2020-02-25T18:19:32Z

Joseph: Added Konftel devices

== Blog Article ==

[https://wiki.voip.ms/article/VoIP_%E2%80%93_Bring_Your_Own_Device_(BYOD) VoIP – Bring Your Own Device (BYOD)]
<br />
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).

* '''ATA:''' An '''Analog Telephone Adapter''' is a device used to connect one or more standard analog telephones to a digital telephone system (such as Voice over IP) or a non-standard telephone system. Still not clear? Take a peek at our blog article : [https://wiki.voip.ms/article/Back_to_Basics_-_What_is_an_ATA%3F Back to Basics - What is an ATA?]

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

__TOC__

==ATAs==

===Atcom AG188N===

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

'''Product:''' Atcom AG188N

'''Company:''' Atcom

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

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

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

===Auerswald COMpact 5010===

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

'''Product:''' Auerswald COMpact 5010

'''Company:''' Auerswald

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

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

[[Auerswald COMpact 5010|See Configuration Details]]
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===Cisco===

====Cisco Linksys PAP2====

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

'''Product:''' Linksys PAP2

'''Company:''' Cisco

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

[[Cisco Linksys PAP2|See Configuration Details]]
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====Cisco Linksys PAP2T====

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

'''Product:''' Linksys PAP2T

'''Company:''' Cisco

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

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

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

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

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

'''Product:''' SPA112, SPA122

'''Company:''' Cisco

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

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

Introduced in late 2011, this box represents an inexpensive means to continue using existing analog hardware while migrating to voice over IP.

[[Cisco SPA112|See Configuration Details]]

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

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

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

'''Product:''' WRP400, WRP500

'''Company:''' Cisco

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

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

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

[[Cisco WRP400|See Configuration Details]]

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

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

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

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

'''Company:''' Cisco

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

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

[[Cisco SPA2100 Phone Adapter|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====Cisco SPA2102 Phone Adapter with Router====

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

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

'''Company:''' Cisco

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

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

[[Cisco_SPA2102_Phone_Adapter_with_Router|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

===Grandstream===

====Grandstream HandyTone 486====

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

'''Product:''' Grandstream HandyTone 486

'''Company:''' Grandstream

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

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

[[Grandstream_HandyTone_486|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====Grandstream HandyTone 502 - HT502====

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

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

'''Company:''' Grandstream

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

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

Enhanced security

Automated provisioning using symmetric and asymmetric voice

Support for a broad range of popular voice codec

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

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

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

'''Company:''' Grandstream

'''Overview:''' The HT702 is Grandstream´s Newest powerful VoIP router. The product's inclusion of an integrated high-performance NAT router and 10Mbps Ethernet WAN and LAN ports enables a shared broadband connection between multiple ethernet devices.

In addition to being SIP 2.0 standard compliant, the product supports Universal Plug-in-Play (UPnP), up to 2 SIP account profiles/Phones and advanced telephony features.

Enhanced Security

Automated provisioning using symmetric and asymmetric voice

Support for a broad range of popular voice codec

[[Grandstream_HandyTone_702_-_HT702|See Configuration Details]]
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====Grandstream HandyTone 802 - HT802====

[[File:HT802 Device.jpg|300px|thumb|left|Grandstream HandyTone 802 - HT802]]

'''Product:''' Grandstream HandyTone 802 - HT802

'''Company:''' Grandstream

'''Overview:''' The HT802 delivers powerful VoIP technology and routing capabilities to home and office environments, and allows users to successfully connect their analog devices to a manageable and powerful VoIP network. Built upon Grandstream’s market-leading SIP ATA/gateway technology, with millions of units successfully deployed worldwide, this powerful ATA features exceptional voice quality in various applications and environments.

The HT802 comes with 2 easy-to-use FXS ports, state-of-the-art encryption with a unique security certificate per unit, automated provisioning for volume deployment and device management and outstanding network performance.

TLS and SRTP security encryption technology to protect calls and accounts

Supports 2 SIP profiles through 2 FXS ports and a single 10/100Mbps port

Support for a broad range of popular voice codec

[[Grandstream HandyTone 802 - HT802|See Configuration Details]]
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===Mediatrix C7 and Mediatrix 4100 Series===

[[File:Mediatrix_C7.jpg|300px|thumb|left|Mediatrix C7]]
[[File:Mediatrix_4100_1_media5.jpg|300px|thumb|left|Mediatrix 4102]]

'''Product:''' : Mediatrix C7 and Mediatrix 4100 Series running DGW 2.0 firmware

'''Company:''' Media5 Corporation

'''Overview:''' The Mediatrix® C7 and Mediatrix® 4100 Series feature VoIP adaptors that interconnect up to 24 analog telephones, faxes, and modems ports into SIP-based systems. These series allow operators to shorten deployment time to enable cloud telephony services into branch offices and SMBs.

The DGW 2.0 firmware also offers security features such as SIP over TLS, SRTP, certificates management, and HTTPS designed to bring enhanced security for the network management, SIP signaling, and media transmission aspects.

[[Media5 Mediatrix C7 and 4100|See Configuration Details]]
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===Netgear WGR615V===

[[File:NetgearWGR615V.jpg|300px|thumb|left|Netgear WGR615V]]

'''Product:''' WGR615V

'''Company:''' Netgear

'''Overview:''' The WGR615V was never sold to directly to the public when it was new but since the model is discontinued, lots of them are being sold.
It is a wireless (802.11 G) router that happens to have a built-in ATA.
It is possible to use just the ATA part.

[[Netgear WGR615V|See Configuration Details]]
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=== OBiHAI (Polycom) ===

====OBi 100/110 & OBi 200====

[[File:OBi110-ATA.jpg|300px|thumb|left|OBi110]]

'''Product:''' OBi 100/110 & OBi 200

'''Company:''' Polycom (OBIHAI Technology Inc)

'''Overview:''' With OBi devices you are in control of your communications life. From the OBi's on-board connections as well as via the Internet, you have the power to bridge mobile, fixed line and Internet telephone services. The OBi100 & OBi110 are stand-alone, dedicated devices, built with a high-performance "system on a chip" platform to ensure high-quality voice conversations. Every OBi device has high availability and reliability. It is 'always-on' to make or receive a call. With an OBi device, a computer is not required and a computer does not need to be on to talk on the phone or use the OBi's powerful service bridging features. To get started, all you need is a phone, power and a connection to the Internet.

The OBi110 has one phone port and one line port. Use the OBi110 when you need an analog line connection to a traditional telephone network or service. Think of the OBi110 LINE port as a gateway to traditional phone service. If you do not have traditional phone service at home, then the OBi100 is probably the right product to get.

[[OBi 100/110 & OBi 200|See Configuration Details]]
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====Polycom OBi300====

[[File:obi300.png|300px|thumb|left|Polycom OBi300]]

'''Product:''' Polycom OBi300

'''Company:''' Polycom

'''Overview:''' Polycom OBi300 expands your service portfolio by enhancing the communication possibilities of home offices with flexibility in voice, fax and modem applications as they transition to the digital communications world.
* Flexible voice and fax applications
* Help home offices maximize current analog investment by keeping one analog phone or fax machine and connect them to VoIP services
* Supports T.38 fax standard so home office users can send or receive reliable facsimile calls over the Internet

[[OBi300|See Configuration Details]]
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===Telco AC-211===

[[File:Ac211n.jpg|300px|thumb|left|Telco AC-211]]

'''Product:''' Telco AC-211

'''Company:''' Telco

'''Overview:''' Telco AC-211 is a SIP ATA supporting two lines and numerous configuration options. This ATA is most commonly found as the AC-211-SR & AC-211N from the now-defunct SunRocket service. This device works well with VoIP.ms once configured.

[[Telco_AC-211|See Configuration Details]]

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===TP-Link TD-VG3631===

[[File:Tplink vg3631 front.png|300px|thumb|left|TPLink VG3631]]

'''Product:''' TP-Link TD-VG3631

'''Company:''' TP-Link

'''Overview:''' High-speed DSL modem, NAT router, and wireless access point in one device providing a one-stop networking solution Wireless N speed up to 300Mbps makes it ideal for heavy bandwidth consuming or interruption sensitive applications like online gaming, Internet calls and even the HD video streaming Supporting both traditional landlines and VoIP network offers you a multiple choice when making phone calls. Various call features such as caller ID, call waiting, call holding, call forwarding, 3-way conference calls and voicemail USB 2.0 port supports Storage Sharing (Samba or FTP), Media Server, Print Server and multi accounts Ethernet WAN (EWAN) offers another broadband connectivity option for connecting to Cable, VDSL or Fiber modems IP based Bandwidth Control makes it easier for you to manage the bandwidth of devices that connected to the modem router Wi-Fi On/Off Button allows users to simply turn their wireless radio on or off.

[[TP-Link TD-VG3631|See Configuration Details]]
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==IP Phones==

===3COM 3108 Wireless Phone===

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

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

'''Company:''' 3COM

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

[[3COM_3108_Wireless_Phone|See Configuration Details]]
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===Aastra 6730i/6731i VoIP Phone===

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

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

'''Company:''' Aastra

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

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

[[Aastra_6730i_VoIP_Phone|See Configuration Details]]
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===Audiocodes===

====400HD Series====

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

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

'''Company:''' Audiocodes

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

[[Audiocodes 400HD|See Configuration Details]]
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===Cisco===

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

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

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

'''Company:''' Cisco

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

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

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

[[Cisco IP Phone 68XX and 88XX|See Configuration Details]]
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====Cisco Linksys SPA942 NA====

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

'''Product:''' Linksys SPA942 NA

'''Company:''' Cisco

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

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

[[Cisco_Linksys_SPA942_NA|See Configuration Details]]
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====Cisco SPA525G====

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

'''Product:''' Cisco SPA525G

'''Company:''' Cisco

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

[[Cisco SPA525G|See Configuration Details]]
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====Cisco IP Phone 7940/7960====

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

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

'''Company:''' Cisco

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

[[Cisco_IP_Phone_7940/7960|See Configuration Details]]
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====Cisco SPA30x and SPA50x series IP phones====

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

'''Product:''' Cisco SPA504G Phone

'''Company:''' Cisco

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

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

[[Cisco_SPA504G_Phone|See Configuration Details]]
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===Fanvil ===

====Fanvil X4G====

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

'''Product:''' Fanvil X4G

'''Company:''' Fanvil

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

[[Fanvill X4G|See Configuration Details]]
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[[Category:Devices]]

===Fortinet===

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

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

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

'''Company:''' Fortinet

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

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

[[Fortinet_FortiFone_FON-570|See Configuration Details]]
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====FortiFone FON-375====

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

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

'''Company:''' Fortinet

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

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

[[Fortinet_FortiFone_FON-375|See Configuration Details]]
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====FortiFone FON-175====

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

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

'''Company:''' Fortinet

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

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

[[Fortinet_FortiFone_FON-175|See Configuration Details]]
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===Gigaset A510 IP===

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

'''Product:'''Gigaset A510 IP

'''Company:'''Gigaset

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

[[Gigaset_A510_IP| See Configuration Details]]
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===Grandstream===

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

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

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

'''Company:''' Grandstream

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

[[Grandstream_DP715/DP710| See Configuration Details]]
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====Grandstream DP750/DP720====

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

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

'''Company:''' Grandstream

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

[[Grandstream_DP750/DP720| See Configuration Details]]
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====Grandstream 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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===Konftel===
==== Konftel 300Wx-IP ====
[[File:Konftel-300Wx-IP.png|300px|thumb|left|Konftel 300Wx-IP]]

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

'''Company:''' Konftel

'''Overview:''' The wireless conference phone Konftel 300Wx allows you to hold meetings wherever is convenient for you – without worrying about network and power outlets. The wireless DECT technology is both reliable and secure. Choose a base station to suit your company's telephony environment, SIP or analog, or connect to an installed DECT system.

The rechargeable battery ensures more than 60 call hours, so you can talk for a full work week without recharging!

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

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

'''Product:''' Konftel 300IPx

'''Company:''' Konftel

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

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

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

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

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

'''Company:''' Panasonic

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

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

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

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

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

'''Company:''' Panasonic

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

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

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

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

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

'''Company:''' Panasonic

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

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

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

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

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

'''Company:''' Panasonic

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

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

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

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

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

'''Company:''' Pirelli

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

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

[[Pirelli DP-L10|See Configuration Details]]
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===Polycom===

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

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

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

'''Company:''' Polycom

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

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

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

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

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

'''Company:''' Polycom

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

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

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

[[Polycom_SoundPoint_IP_501|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====Polycom SoundPoint IP 601====

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

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

'''Company:''' Polycom

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

[[Polycom_SoundPoint_IP_601|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====Polycom VVX 300, 400, etc====

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

'''Product:''' Polycom VVX Series

'''Company:''' Polycom

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

[[Polycom VVX 300, 400, etc|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

=== Positron IP phones ===

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

'''Product:''' Positron

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

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

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

==== Positron IP304 ====

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

'''Product:''' Positron IP304

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

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

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

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

[[Positron IP304 | View configuration for Positron IP304]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==== Positron IP304C ====

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

'''Product:''' Positron IP304C

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

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

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

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

[[Positron IP304C | View configuration for Positron IP304C]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==== Positron IP408 ====

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

'''Product:''' Positron IP408

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

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

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

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

[[Positron IP408 | View configuration for Positron IP408]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==== Positron IP410C ====

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

'''Product:''' Positron IP410C

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

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

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

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

[[Positron IP410C | View configuration for Positron IP410C]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==== Positron IP410G ====

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

'''Product:''' Positron IP410G

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

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

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

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

[[Positron IP410G | View configuration for Positron IP410G]]
<div style="width:100%;overflow:hidden;clear:both"></div>

===Siemens Gigaset C450-Ip===

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

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

'''Company:''' Siemens

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

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

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

[[Siemens Gigaset C450-Ip|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

===Snom===

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

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

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

'''Company:''' Snom

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

[[SNOM 320|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====Snom m3 VoIP Phone====

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

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

'''Company:''' Snom

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

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

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

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

====SNOM C520====

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

'''Product:''' SNOM C520 Conferencing

'''Company:''' Snom

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

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

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

====SNOM professional D7XX====

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

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

'''Company:''' Snom

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

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

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

=== Vtech ===

==== Vtech Conference Station ====

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

'''Product:''' VCSV752 & CS754

'''Company:''' Vtech

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

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

==== Vtech VSP Series ====

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

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

'''Company:''' Vtech

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

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

===Yealink===

====Yealink Voice Solutions====

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

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

'''Company:''' Yealink

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

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

====Yealink SIP-T28P (VSRF)====

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

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

'''Company:''' Yealink

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

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

[[Yealink_SIP-T28P|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>
[[Category:Devices]]

===Zycoo ZP502===

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

'''Product:''' Zycoo ZP502

'''Company:''' Zycoo ZP502

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

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

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

Devices

2020-02-25T18:12:04Z

Joseph: /* OBi 100/110 & OBi 200 */

== Blog Article ==

[https://wiki.voip.ms/article/VoIP_%E2%80%93_Bring_Your_Own_Device_(BYOD) VoIP – Bring Your Own Device (BYOD)]
<br />
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).

* '''ATA:''' An '''Analog Telephone Adapter''' is a device used to connect one or more standard analog telephones to a digital telephone system (such as Voice over IP) or a non-standard telephone system. Still not clear? Take a peek at our blog article : [https://wiki.voip.ms/article/Back_to_Basics_-_What_is_an_ATA%3F Back to Basics - What is an ATA?]

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

__TOC__

==ATAs==

===Atcom AG188N===

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

'''Product:''' Atcom AG188N

'''Company:''' Atcom

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

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

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

===Auerswald COMpact 5010===

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

'''Product:''' Auerswald COMpact 5010

'''Company:''' Auerswald

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

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

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

===Cisco===

====Cisco Linksys PAP2====

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

'''Product:''' Linksys PAP2

'''Company:''' Cisco

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

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

====Cisco Linksys PAP2T====

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

'''Product:''' Linksys PAP2T

'''Company:''' Cisco

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

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

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

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

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

'''Product:''' SPA112, SPA122

'''Company:''' Cisco

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

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

Introduced in late 2011, this box represents an inexpensive means to continue using existing analog hardware while migrating to voice over IP.

[[Cisco SPA112|See Configuration Details]]

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

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

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

'''Product:''' WRP400, WRP500

'''Company:''' Cisco

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

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

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

[[Cisco WRP400|See Configuration Details]]

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

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

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

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

'''Company:''' Cisco

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

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

[[Cisco SPA2100 Phone Adapter|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====Cisco SPA2102 Phone Adapter with Router====

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

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

'''Company:''' Cisco

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

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

[[Cisco_SPA2102_Phone_Adapter_with_Router|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

===Grandstream===

====Grandstream HandyTone 486====

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

'''Product:''' Grandstream HandyTone 486

'''Company:''' Grandstream

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

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

[[Grandstream_HandyTone_486|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====Grandstream HandyTone 502 - HT502====

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

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

'''Company:''' Grandstream

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

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

Enhanced security

Automated provisioning using symmetric and asymmetric voice

Support for a broad range of popular voice codec

[[Grandstream_HandyTone_502_-_HT502|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====Grandstream HandyTone 702 - HT702====

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

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

'''Company:''' Grandstream

'''Overview:''' The HT702 is Grandstream´s Newest powerful VoIP router. The product's inclusion of an integrated high-performance NAT router and 10Mbps Ethernet WAN and LAN ports enables a shared broadband connection between multiple ethernet devices.

In addition to being SIP 2.0 standard compliant, the product supports Universal Plug-in-Play (UPnP), up to 2 SIP account profiles/Phones and advanced telephony features.

Enhanced Security

Automated provisioning using symmetric and asymmetric voice

Support for a broad range of popular voice codec

[[Grandstream_HandyTone_702_-_HT702|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====Grandstream HandyTone 802 - HT802====

[[File:HT802 Device.jpg|300px|thumb|left|Grandstream HandyTone 802 - HT802]]

'''Product:''' Grandstream HandyTone 802 - HT802

'''Company:''' Grandstream

'''Overview:''' The HT802 delivers powerful VoIP technology and routing capabilities to home and office environments, and allows users to successfully connect their analog devices to a manageable and powerful VoIP network. Built upon Grandstream’s market-leading SIP ATA/gateway technology, with millions of units successfully deployed worldwide, this powerful ATA features exceptional voice quality in various applications and environments.

The HT802 comes with 2 easy-to-use FXS ports, state-of-the-art encryption with a unique security certificate per unit, automated provisioning for volume deployment and device management and outstanding network performance.

TLS and SRTP security encryption technology to protect calls and accounts

Supports 2 SIP profiles through 2 FXS ports and a single 10/100Mbps port

Support for a broad range of popular voice codec

[[Grandstream HandyTone 802 - HT802|See Configuration Details]]
<br clear="all" />

===Mediatrix C7 and Mediatrix 4100 Series===

[[File:Mediatrix_C7.jpg|300px|thumb|left|Mediatrix C7]]
[[File:Mediatrix_4100_1_media5.jpg|300px|thumb|left|Mediatrix 4102]]

'''Product:''' : Mediatrix C7 and Mediatrix 4100 Series running DGW 2.0 firmware

'''Company:''' Media5 Corporation

'''Overview:''' The Mediatrix® C7 and Mediatrix® 4100 Series feature VoIP adaptors that interconnect up to 24 analog telephones, faxes, and modems ports into SIP-based systems. These series allow operators to shorten deployment time to enable cloud telephony services into branch offices and SMBs.

The DGW 2.0 firmware also offers security features such as SIP over TLS, SRTP, certificates management, and HTTPS designed to bring enhanced security for the network management, SIP signaling, and media transmission aspects.

[[Media5 Mediatrix C7 and 4100|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

===Netgear WGR615V===

[[File:NetgearWGR615V.jpg|300px|thumb|left|Netgear WGR615V]]

'''Product:''' WGR615V

'''Company:''' Netgear

'''Overview:''' The WGR615V was never sold to directly to the public when it was new but since the model is discontinued, lots of them are being sold.
It is a wireless (802.11 G) router that happens to have a built-in ATA.
It is possible to use just the ATA part.

[[Netgear WGR615V|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

=== OBiHAI (Polycom) ===

====OBi 100/110 & OBi 200====

[[File:OBi110-ATA.jpg|300px|thumb|left|OBi110]]

'''Product:''' OBi 100/110 & OBi 200

'''Company:''' Polycom (OBIHAI Technology Inc)

'''Overview:''' With OBi devices you are in control of your communications life. From the OBi's on-board connections as well as via the Internet, you have the power to bridge mobile, fixed line and Internet telephone services. The OBi100 & OBi110 are stand-alone, dedicated devices, built with a high-performance "system on a chip" platform to ensure high-quality voice conversations. Every OBi device has high availability and reliability. It is 'always-on' to make or receive a call. With an OBi device, a computer is not required and a computer does not need to be on to talk on the phone or use the OBi's powerful service bridging features. To get started, all you need is a phone, power and a connection to the Internet.

The OBi110 has one phone port and one line port. Use the OBi110 when you need an analog line connection to a traditional telephone network or service. Think of the OBi110 LINE port as a gateway to traditional phone service. If you do not have traditional phone service at home, then the OBi100 is probably the right product to get.

[[OBi 100/110 & OBi 200|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====Polycom OBi300====

[[File:obi300.png|300px|thumb|left|Polycom OBi300]]

'''Product:''' Polycom OBi300

'''Company:''' Polycom

'''Overview:''' Polycom OBi300 expands your service portfolio by enhancing the communication possibilities of home offices with flexibility in voice, fax and modem applications as they transition to the digital communications world.
* Flexible voice and fax applications
* Help home offices maximize current analog investment by keeping one analog phone or fax machine and connect them to VoIP services
* Supports T.38 fax standard so home office users can send or receive reliable facsimile calls over the Internet

[[OBi300|See Configuration Details]]
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===Telco AC-211===

[[File:Ac211n.jpg|300px|thumb|left|Telco AC-211]]

'''Product:''' Telco AC-211

'''Company:''' Telco

'''Overview:''' Telco AC-211 is a SIP ATA supporting two lines and numerous configuration options. This ATA is most commonly found as the AC-211-SR & AC-211N from the now-defunct SunRocket service. This device works well with VoIP.ms once configured.

[[Telco_AC-211|See Configuration Details]]

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===TP-Link TD-VG3631===

[[File:Tplink vg3631 front.png|300px|thumb|left|TPLink VG3631]]

'''Product:''' TP-Link TD-VG3631

'''Company:''' TP-Link

'''Overview:''' High-speed DSL modem, NAT router, and wireless access point in one device providing a one-stop networking solution Wireless N speed up to 300Mbps makes it ideal for heavy bandwidth consuming or interruption sensitive applications like online gaming, Internet calls and even the HD video streaming Supporting both traditional landlines and VoIP network offers you a multiple choice when making phone calls. Various call features such as caller ID, call waiting, call holding, call forwarding, 3-way conference calls and voicemail USB 2.0 port supports Storage Sharing (Samba or FTP), Media Server, Print Server and multi accounts Ethernet WAN (EWAN) offers another broadband connectivity option for connecting to Cable, VDSL or Fiber modems IP based Bandwidth Control makes it easier for you to manage the bandwidth of devices that connected to the modem router Wi-Fi On/Off Button allows users to simply turn their wireless radio on or off.

[[TP-Link TD-VG3631|See Configuration Details]]
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==IP Phones==

===3COM 3108 Wireless Phone===

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

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

'''Company:''' 3COM

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

[[3COM_3108_Wireless_Phone|See Configuration Details]]
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===Aastra 6730i/6731i VoIP Phone===

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

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

'''Company:''' Aastra

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

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

[[Aastra_6730i_VoIP_Phone|See Configuration Details]]
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===Audiocodes===

====400HD Series====

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

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

'''Company:''' Audiocodes

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

[[Audiocodes 400HD|See Configuration Details]]
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===Cisco===

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

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

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

'''Company:''' Cisco

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

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

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

[[Cisco IP Phone 68XX and 88XX|See Configuration Details]]
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====Cisco Linksys SPA942 NA====

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

'''Product:''' Linksys SPA942 NA

'''Company:''' Cisco

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

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

[[Cisco_Linksys_SPA942_NA|See Configuration Details]]
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====Cisco SPA525G====

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

'''Product:''' Cisco SPA525G

'''Company:''' Cisco

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

[[Cisco SPA525G|See Configuration Details]]
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====Cisco IP Phone 7940/7960====

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

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

'''Company:''' Cisco

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

[[Cisco_IP_Phone_7940/7960|See Configuration Details]]
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====Cisco SPA30x and SPA50x series IP phones====

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

'''Product:''' Cisco SPA504G Phone

'''Company:''' Cisco

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

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

[[Cisco_SPA504G_Phone|See Configuration Details]]
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===Fanvil ===

====Fanvil X4G====

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

'''Product:''' Fanvil X4G

'''Company:''' Fanvil

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

[[Fanvill X4G|See Configuration Details]]
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[[Category:Devices]]

===Fortinet===

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

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

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

'''Company:''' Fortinet

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

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

[[Fortinet_FortiFone_FON-570|See Configuration Details]]
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====FortiFone FON-375====

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

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

'''Company:''' Fortinet

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

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

[[Fortinet_FortiFone_FON-375|See Configuration Details]]
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====FortiFone FON-175====

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

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

'''Company:''' Fortinet

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

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

[[Fortinet_FortiFone_FON-175|See Configuration Details]]
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===Gigaset A510 IP===

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

'''Product:'''Gigaset A510 IP

'''Company:'''Gigaset

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

[[Gigaset_A510_IP| See Configuration Details]]
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===Grandstream===

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

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

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

'''Company:''' Grandstream

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

[[Grandstream_DP715/DP710| See Configuration Details]]
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====Grandstream DP750/DP720====

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

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

'''Company:''' Grandstream

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

[[Grandstream_DP750/DP720| See Configuration Details]]
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====Grandstream 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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===Panasonic===

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

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

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

'''Company:''' Panasonic

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

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

[[Panasonic KX-TGP 550|See Configuration Details]]
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====Panasonic KX-HDV130C====

[[File:Panasonic_KXHDV130_01.jpg|300px|thumb|left|Panasonic KX-HDV130C]]

'''Product:''' Panasonic KX-HDV130C

'''Company:''' Panasonic

'''Overview:''' The Panasonic KX-HDV130 SIP desk phone delivers the ideal balance of low cost and high quality, along a range of value added features.

*Support for up to 2 SIP registrations (e.g. up to 2 DID lines or extensions)
*Support for 3 simultaneous network conversations (3-way conferencing)*
*2 Programmable keys / Line keys
*Hands-free speakerphone on a cordless handset

[[Panasonic KX-HDV130C|See Configuration Details]]
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====Panasonic KX-HDV230====

[[File:Panasonic_KXHDV230_01.jpg|300px|thumb|left|Panasonic KX-HDV230]]

'''Product:''' Panasonic KX-HDV230

'''Company:''' Panasonic

'''Overview:''' The Panasonic KX-HDV230 IP phone offers streamlined functions and the high definition voice quality that's essential for effective communication.

*Support for up to 6 SIP registrations (e.g. up to 6 DID lines or extensions)
*24 Flexible Function Keys
*2 ethernet ports 10/100/1000 Base -T
*Large LCD with Backlight

[[Panasonic KX-HDV130C|See Configuration Details]]
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====Panasonic KX-HDV330====

[[File:Panasonic_KXHDV330_01.jpg|300px|thumb|left|Panasonic KX-HDV330]]

'''Product:''' Panasonic KX-HDV330

'''Company:''' Panasonic

'''Overview:''' The Panasonic KX-HDV330 is a multi-functional business SIP phone equipped with a colour touch panel for intuitive operation.

*Large LCD with Backlight
*Built-in Bluetooth®
*Support for up to 12 SIP registrations (e.g. up to 12 DID lines or extensions)
*24 Flexible Function Keys
*Hands-free speakerphone on a cordless handset

[[Panasonic KX-HDV130C|See Configuration Details]]
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===Pirelli DP-L10===

[[File:Pirelli1.jpg|300px|thumb|left|Pirelli DP-L10]]

'''Product:''' Pirelli DP-L10

'''Company:''' Pirelli

'''Overview:''' Pirelli's DualPhone DP-L10 phone combines GSM tri-band capabilities with SIP-based Wireless VoIP via an integrated WLAN 802.11b/g interface, combining cell phone technology with the added advantage of transporting phone calls over the Internet.

The Dual Phone enables mobile phone features such as Internet browsing, e-mail, SMS, and MMS, which can also be used through a fixed-broadband connection, providing the same mobile services with the added benefits of greater bandwidth and cost savings.

[[Pirelli DP-L10|See Configuration Details]]
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===Polycom===

====Polycom SoundStation IP 4000 Conference Phone====

[[File:Polycom-4000-0.gif|300px|thumb|left|Polycom SoundStation IP 4000 Conference Phone]]

'''Product:''' Polycom SoundStation IP 4000 Conference Phone

'''Company:''' Polycom

'''Overview:''' the SoundStation IP 4000 SIP voice conferencing unit is the answer for organizations that are ready for the
benefits and versatility of a SIP enabled business. Designed for offices or small to medium-sized conference rooms, the SoundStation IP 4000 SIP provides remarkable room coverage. You can speak naturally from up to 10-feet away from a microphone and still be heard clearly on the far end of the call. Need coverage for a larger room? The optional extension microphones offer an increased pickup for larger rooms. Plus, with gated microphone technology, echo and background noise is almost entirely eliminated.

The SoundStation IP 4000 SIP also provides familiar features that are easy to use. The menu-driven user interface, viewable on a high-resolution backlit LCD, offers convenient access to business telephony functions you use most including transfer, hold, redial, and conference.

[[Polycom_SoundStation_IP_4000_Conference_Phone|See Configuration Details]]
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====Polycom SoundPoint IP 501, 550, 650, etc.====

[[File:Polycom501.jpg|258px|thumb|left|Polycom SoundPoint IP 501]]

'''Product:''' Polycom SoundPoint IP 501, 550, 650, etc.

'''Company:''' Polycom

'''Overview:''' The SoundPoint IP series are multi-line (3-6 lines depending on model) Voice over IP telephones that seamlessly integrate with IP PBX and softswitch vendors’ IP solutions.

As protocols develop and standards evolve, it’s easy to update the phone in the field via a software download, thus enabling new features and functionality for the phone and protecting your investment.

Whether you deploy MGCP or SIP standards, Polycom offers a solution that fits all your business communication needs.

[[Polycom_SoundPoint_IP_501|See Configuration Details]]
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====Polycom SoundPoint IP 601====

[[File:Voipms-polycom601.jpg|258px|thumb|left|Polycom SoundPoint IP 601]]

'''Product:''' Polycom SoundPoint IP 601

'''Company:''' Polycom

'''Overview:''' The 6-line Polycom® SoundPoint IP™ 601 offers industry-leading functionality and call handling unmatched voice quality an intuitive user interface & expandability to 12 lines!

[[Polycom_SoundPoint_IP_601|See Configuration Details]]
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====Polycom VVX 300, 400, etc====

[[File:Vvx300.png|250px|thumb|left|Polycom VVX 300 Series]]

'''Product:''' Polycom VVX Series

'''Company:''' Polycom

'''Overview:''' The VVX series provides high-quality audio (HD Voice) and video communications from 6 lines and up.

[[Polycom VVX 300, 400, etc|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

=== Positron IP phones ===

[[File:PositronLogo.jpeg|250px|thumb|left|Positron IP phones]]

'''Product:''' Positron

'''Company:''' [http://www.positrontelecom.com/products.html Positron]

'''Overview:''' Positron IP Phones is an affordable next-generation SIP phone including wideband audio support, ethernet ports and integrated Power over Ethernet.

All the IP Phones are optimized for executives, administrative assistants and those working with bandwidth-intensive applications on collocated PCs. Line keys can also be configured as IP PBX features such as BLF, SCA, Intercom, Call Pickup, Call Park, and many others. The high-resolution screen and the HD voice features provide a high quality visual and authentic audio experience.

==== Positron IP304 ====

[[File:IP304.png |250px|thumb|left|Positron IP304]]

'''Product:''' Positron IP304

'''Company:''' [http://www.positrontelecom.com/products.html Positron]

'''Overview:''' Positron IP304 is an affordable next-generation SIP phone with wideband audio support, dual Ethernet port and integrated Power over Ethernet.

The IP304 enterprise VoIP phone is Positron’s entry-level phone with 3 VoIP accounts. Its functionalities make it easy to place calls on hold, to transfer calls and to initiate a conference call.

With HD audio for both headset and speakerphone, IP304 provides a clear authentic audio experience

[[Positron IP304 | View configuration for Positron IP304]]
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==== Positron IP304C ====

[[File:IP304C.png |250px|thumb|left|Positron IP304C]]

'''Product:''' Positron IP304C

'''Company:''' [http://www.positrontelecom.com/products.html Positron]

'''Overview:''' Positron IP304C is an innovative enterprise-level IP Phone that features 4 line keys, color display, 3.5” TFT-LCD with 480 x 320 pixel. It supports up to a 5-way conference.

IP304C is optimized for executives, administrative assistants and those working with bandwidth-intensive applications on collocated PCs. Line keys can also be configured as IP PBX features such as BLF, SCA, Intercom, Call Pickup, Call Park, and many others.

The high-resolution TFT-LCD screen and the HD voice features provide a high quality visual and authentic audio experience.

[[Positron IP304C | View configuration for Positron IP304C]]
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==== Positron IP408 ====

[[File:IP408.png |250px|thumb|left|Positron IP408]]

'''Product:''' Positron IP408

'''Company:''' [http://www.positrontelecom.com/products.html Positron]

'''Overview:''' Positron IP408 is an affordable next-generation SIP 2.0 phone including wideband audio support and WAN/LAN Ethernet ports with route and bridge mode.

The IP408 enterprise VoIP phone supports 4 SIP lines. Its functionalities make it easy to place calls on hold, to transfer calls and to initiate a conference call.

With HD audio for both headset and speakerphone, IP408 provides a clear authentic audio experience

[[Positron IP408 | View configuration for Positron IP408]]
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==== Positron IP410C ====

[[File:IP410C.png |250px|thumb|left|Positron IP410C]]

'''Product:''' Positron IP410C

'''Company:''' [http://www.positrontelecom.com/products.html Positron]

'''Overview:''' Positron IP410C is an affordable next-generation SIP Phone that features 4 line keys, 10 programmable extension keys, color display, wideband audio support and dual Ethernet port with integrated Power over Ethernet.

IP410C is optimized for executives, administrative assistants and those working with bandwidth-intensive applications on collocated PCs. Line keys can also be configured as IP PBX features such as BLF, SCA, Intercom, Call Pickup, Call Park, and many others.

The high-resolution TFT-LCD screen and the HD voice features provide a high quality visual and authentic audio experience.

[[Positron IP410C | View configuration for Positron IP410C]]
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==== Positron IP410G ====

[[File:IP410G.png |250px|thumb|left|Positron IP410G]]

'''Product:''' Positron IP410G

'''Company:''' [http://www.positrontelecom.com/products.html Positron]

'''Overview:''' Positron IP410G is an innovative enterprise-level color IP Phone that features 4 line keys, 10 programmable extension keys, color display, 3.5” TFT-LCD with 480*320 pixel, wideband audio support and dual Gigabit Ethernet port with integrated Power over Ethernet.

IP410G is optimized for executives, administrative assistants and those working with bandwidth-intensive applications on collocated PCs. Ten programmable keys can also be configured as IP PBX features such as BLF, SCA, Intercom, Call Pickup, Call Park, and many others.

The high-resolution TFT-LCD screen and the HD voice features provide a high quality visual and authentic audio experience.

[[Positron IP410G | View configuration for Positron IP410G]]
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===Siemens Gigaset C450-Ip===

[[File:Siemens_gigaset_c450_IP.jpg|300px|thumb|left|Siemens Gigaset C450-Ip]]

'''Product:''' Siemens Gigaset C450-Ip

'''Company:''' Siemens

'''Overview:''' The Gigaset C450 is an entry phone in the Siemens Gigaset product line, the addition of IP connectivity via a standard Ethernet 10/100 BaseT interface and SIP protocol within a reasonable price bracket, opens large applications where wireless operation and SIP are required.
The Ethernet connectivity allows the unit to work without a PC.

The dual mode (SIP/PSTN) enables incoming call on a legacy interface with existing directory number and safe emergency outgoing calls.

The presentation of the unit with an independent base and a dedicated phone charger/support will simplify cabling.

[[Siemens Gigaset C450-Ip|See Configuration Details]]
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===Snom===

====Snom 320 VoIP Phone====

[[File:Snom320.png|300px|frame|left|Snom 320 VoIP Phone]]

'''Product:''' Snom 320 VoIP Phone

'''Company:''' Snom

'''Overview:'''Ideal for the office and everyone who spends a lot of time on the phone, the snom 320 is an affordable, yet powerful SIP business phone with a built-in, full-duplex speakerphone and three-party conference bridging.

[[SNOM 320|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====Snom m3 VoIP Phone====

[[File:Snom_m3.png|300px|thumb|left|Snom m3 VoIP Phone]]

'''Product:''' Snom m3 VoIP Phone

'''Company:''' Snom

'''Overview:''' The snom m9 is the next generation DECT handheld (CAT-iq) that empowers users with the convenience of wireless communication along with the widely accepted benefits and feature richness of Voice-over-IP telephony.

The snom m9 promises to deliver excellent speech quality through digital and wideband audio (klarVOICE).

By combining professional functions of versatile business communication with the intuitive features of the mobile-carrier world, the snom m9 is ideally suited for professional and private use alike. With features such as hands-free mode, calling line identification (CLI) by displaying name, number, and image of the caller as well as typical mobile-phone features such as Address Book, Calendar, Calculator and Alarm function, the snom m9 provides the perfect blend of mobility and accessibility.

[[Snom_m3_VoIP_Phone|See Configuration Details]]
<br clear="all" />

====SNOM C520====

[[File:snom_c520.png|300px|thumb|left|C520]]

'''Product:''' SNOM C520 Conferencing

'''Company:''' Snom

'''Overview:''' With its modern and sleek design, the C520 fits seamlessly into your working day. Two detachable DECT microphones can be positioned freely or carried in the room as required to ensure the best sound and voice quality.

Built-in charging stations with magnetic bays directly on the base station mean both microphones are always charged and ready for use in the next meeting. The conference phone also features automatic volume control and digital noise reduction so that all call participants can be understood in best sound quality.

[[Snom C520|See Configuration Details]]
<br clear="all" />

====SNOM professional D7XX====

[[File:Snom.jpg|300px|thumb|left|Snom D7XX]]

'''Products:''' D120, D717, D735, D785

'''Company:''' Snom

'''Overview:''' The [https://www.snom.com/en/ip-phones/desk-phones/d7xx-series/ professional D7XX] Series telephones are both aesthetically appealing and highly practical, meeting business requirements when a telephone is a key tool in daily work.

These high-performance devices are future-proofed and provide the best in Wideband HD audio, ensuring crystal clear sound quality. They are Bluetooth compatible to meet the connectivity requirements of today’s offices.

[[Snom IP Phones|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

=== Vtech ===

==== Vtech Conference Station ====

[[File:VCS754-thumb.PNG|300px|thumb|left|Vtech VCS Series]]

'''Product:''' VCSV752 & CS754

'''Company:''' Vtech

'''Overview:''' The [https://businessphones.vtech.com/pd/3439/VCS754-ErisStation-SIP-Conference-Phone-with-Four-Wireless-Mics Vtech VCS754 ErisStation] conference phone features a compact, all-in-one design makes it easy to keep everything together—no clutter, no hassle. Built-in charging stations with magnetic bays ensure the microphones are charged and available for the next meeting.

[[Vtech Conference Station|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==== Vtech VSP Series ====

[[File:VSP736 ErisTerminal.jpg|300px|thumb|left|VSP Series]]

'''Products:''' VSP600 - VSP715 - VSP725 - VSP726 - VSP736

'''Company:''' Vtech

'''Overview:''' The [https://businessphones.vtech.com/products/sip-phones/vsp700 Vtech VSP700 Series] comes with all the essential features you need to keep pace with your business and your budget. Depending on the model, support two to six SIP accounts with these easy-to-use phones.

[[Vtech VSP Series|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

===Yealink===

====Yealink Voice Solutions====

[[File:Yealink_easyVoip.png|300px|thumb|left|Yealink SIP-T28P]]

'''Product:''' Yealink W60B, Yealink T21, Yealink T42S

'''Company:''' Yealink

'''Overview:''' Yealink offers solutions for each customer's needs, starting from basic to more complex ones.

[[Yealink Voice Solutions|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

====Yealink SIP-T28P (VSRF)====

[[File:Yealink-sip-t28p.jpg|300px|thumb|left|Yealink SIP-T28P]]

'''Product:''' Yealink SIP-T28P

'''Company:''' Yealink

'''Overview:''' Yealink SIP-T28P represents the next generation VoIP phone which designed for the business user who needs rich telephony features, friendly UI and super voice quality. It is equipped with the TI TITAN chipset, offers high definition voice quality through TI voice engine, HD handset, HD speaker and HD codec (G.722).

By the large, high-resolution graphical display, and together with all the 48 keys, SIP-T28P offers an excellent user experience to configure, make calls, express XML browser, etc. Moreover, to avoid problems with unwanted violations of your audio data, Yealink SIP-T28P supports the security standards TLS, SRTP, HTTPS, 802.1x, Open VPN and AES encryption which are necessary to protect against electronic eavesdropping and data theft.

[[Yealink_SIP-T28P|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>
[[Category:Devices]]

===Zycoo ZP502===

[[File:ZP502.jpg|left|Zycoo ZP502]]

'''Product:''' Zycoo ZP502

'''Company:''' Zycoo ZP502

'''Overview:''' ZP502 VoIP Phone is ZYCOO business IP phone terminal which adopts SIP and IAX2 protocols.

With Broadcom solution, compatible with various Platforms such as Asterisk, FreePBX, Broadsoft, Cisco call manager, etc.

[[Zycoo ZP502|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

OBi300

2020-02-25T18:00:47Z

Joseph: /* Configuration for usage with call encryption TLS/SRTP */

[[File:obi300.png|none|200px|center|link=https://www.polycom.com/voice-conferencing-solutions/voip-adapters/obi300.html|OBi 300]]

'' The OBi100 & OBi200 are perfect for customers who do not have a traditional phone service, yet need a similar solution and want the savings and simplicity of using a VoIP service for all their calls.

To start configuring your device you will need to plug it in to your router/modem via its Internet port with an Ethernet cable and connect a regular handset phone to it's Phone port, then follow the next steps.''<br/>

== Manual Configuration Details ==

Start by dialing ''' * * * ''' from the connected phone, then press '''1''' to confirm your choice, this will return the IP address of your device being a number similar to '''192.168.xxx.xxx'''.

Once you get the IP address, enter it in the URL address bar '''"http://"''' of your Internet Browser to get access to the Graphic User Interface of the OBi100.

If done properly, the following window should appear on your screen:

[[File:Obi300_Login.png|300px|thumb|left]]
<br clear="all" />

Once you get the Authentication Required window you will be prompted for a User Name and Password, the default credentials are:

'''User Name:''' admin

'''Password:''' admin

After this, you should now be able to see the OBi Web interface.

Now on the left side of the screen please find the next options and follow the next steps:

===Disabling auto-provisioning===

'''**NOTE :''' You may use this guide to configure an OBi110 as well. This is the VoIP.ms recommended configuration versus using the Obihai configuration dashboard (more on this later on this page) and you may also not find all new VoIP.ms servers on the Obihai Dahsboard. In order to make sure there will be no conflicts between this Manual configuration and the Obihai dashboard, please perform the following steps to disable auto-provisioning:

*System Management -> Auto Provisioning -> Auto Firmware Update -> Method : Disabled
*System Management -> Auto Provisioning -> ITSP Provisioning -> Method : Disabled
*System Management -> Auto Provisioning -> OBiTALK Provisioning -> Method : Disabled
*Voice Services -> OBiTALK Service -> Enable : Unchecked

Please note you must remove the check mark from the "default" column, then under "Method" please use the ''''Drop Down Selection'''' and choose '''Disabled'''.

[[File:Obi300_AutoProv.png|450px|thumb|left|Disabling Auto Provisioning]]
<br clear="all" />
After this, save all changes and you are ready to move on to the actual configuration.
<br />
===Configuring the ITSP Profile===

====General Section====
In this section you will set the name and the DigiMap you will use in the profile you configure. By default you will configure the profile A, unless you use the same device with another provider.

:'''Name''': 100000 (Replace with your 6 digit Main SIP account UserID or Sub Account user name, i.e. 123456 or 123456_sub)<br/>
:'''DigitMap''': Copy the line, including parenthesis, in the Digitmap field in the ITSP Profile and replace the "555" digits in the following lines by the area code of your choice:

::Dial Plan (recommended):
(1xxxxxxxxxx|<1555>[2-9]xxxxxx|<1>[2-9]xxxxxxxxx|911|011xx.|xx.|*xx.|***xxx|4xxx|(Mipd)|[^*]@@.)

Also, some clients have been successful by dialing '''**1 and *97''' for line 1 or '''**2 and *97''' for line 2.

:*If you need to set the dial plan back to Default, you can use this:

(1xxxxxxxxxx|<1>[2-9]xxxxxx|<1>[2-9]xxxxxxxxx|011xx.|***xxx|xx.|(Mipd)|[^*]@@.)

[[File:Step2.png|550px|thumb|left|ITSP profile, General - click to enlarge]]
<br clear="all" />

====SIP Section====
In this section you can set the server and the port that you wish to register to.

Please note that in order to change the settings, you need to uncheck the Default box on the right hand side.

*ProxyServer: denver.voip.ms (one of VoIP.ms multiple [http://wiki.voip.ms/article/Choosing_Server servers], you can choose the one closest to your location.)
*ProxyServerPort: 5060
*RegistrarServer: denver.voip.ms (one of VoIP.ms multiple servers [http://wiki.voip.ms/article/Choosing_Server servers], you can choose the one closest to your location.)
*RegistrarServerPort: 5060

[[File:Step3.png|550px|thumb|left|ITSP profile, SIP - click to enlarge]]
<br clear="all" />

Additionally, you may want to change the RegisterExpires value to 300, scroll down, deselect the default box and set the value there from 3600 to 300.

[[File:Step4.png|550px|thumb|left|ITSP profile, SIP (Register Expires)- click to enlarge]]
<br clear="all" />

===Configuring Voice Services===
<br/>
In this section you can set your Main account/sub_account credentials like User name and Password. The Main account password by default is the same password as the Customer Portal.

*AuthUserName: 100000 (Replace with your 6 digits Main SIP account UserID or Sub Account user name, i.e. 123456 or 123456_sub)
*AuthPassword: ****** (''Your SIP Account Password'')

[[File:Step5.png|550px|thumb|left|Voice Services (SIP Credentials) - click to enlarge]]
<br clear="all" />

Once you have finished changing all those settings, click on the button ''Submit'' to save the changes and ''reboot your OBi device'', your device should now be registered.

===Configuring a Voice line using TLS===

'''NOTE''': This section is optional. It explains in detail how to use encrypted traffic in the device. If you are not certain about how encrypted traffic works, or the benefits of encrypting SIP traffic, please contact technical support for more information.

These devices are compatible with TLS, however, some settings need to be adjusted to have it working. This section will be assuming that your SIP account is already enabled to use TLS, if you have not enabled it yet, please follow these instructions before going further:

For more information on how to enable encrypted traffic for the main account, please click on [[Call_Encryption_-_TLS/SRTP#Activate_This_Option_on_Your_Main_Account | Main account]] or more information on how to enable encrypted traffic for the sub-account [[Call_Encryption_-_TLS/SRTP#Activate_This_Option_on_Your_Sub_Account | sub account]]

=== Configuring your OBi device ===

In order to use TLS/call encryption with OBi devices, you'll need to modify the following parameters accordingly. Please note that this is only available for some OBi devices, and the screenshots are from the most recent firmware version.

Under Service Providers > ITSP Profile > SIP use the following values:
:'''ProxySeverPort''': 5061
:'''ProxyServerTransport''': TLS
:'''RegistrarServerPort''': 5061
:'''OutboundProxyPort''': 5061
:'''X_OutboundProxyTransport''': TLS

[[File:TLSOBi1.png|550px|thumb|left|ITSP profile, SIP (TLS)- click to enlarge]]
<br clear="all" />

Under Voice Services > SP Service use the following values:
:'''X_KeepAliveServerPort''': 5061
:'''X_SRTP''': Use SRTP Only

[[File:TLSOBi2.png|550px|thumb|left|Voice Services, SP Service (SRTP)- click to enlarge]]
<br clear="all" />

OBi 100/110 & OBi 200

2020-02-25T17:59:53Z

Joseph: /* Configuration for usage with call encryption TLS/SRTP */

[[File:OBi110-ATA.jpg|none|200px|center|link=http://www.polycom.com/voice-conferencing-solutions/voip-adapters.html|Obi]]

<br/>''The OBi100 & OBi200 are single phone port ATA adapters that support SIP VoIP services. The OBi100 & OBi200 are perfect for customers who do not have a traditional phone service, yet need a similar solution and want the savings and simplicity of using a VoIP service for all their calls. To start configuring your OBi100 or OBi200 device you will need to plug it in to your router/modem via its Internet port with an Ethernet cable and connect a regular handset phone to it's Phone port, then follow the next steps.''<br/>
<br/>
== Manual Configuration Details ==
<br/>
Start by dialing ''' * * * ''' from the connected phone, then press '''1''' to confirm your choice, this will return the IP address of your device being a number similar to '''192.168.xxx.xxx'''.<br/>
Once you get the IP address, enter it in the URL address bar '''"http://"''' of your Internet Browser to get access to the Graphic User Interface of the OBi100.

For an OBi202 please do the following to enable the GUI Web Interface:

Dial *** from the phone connected to the OBi202
Enter 0 For Advanced
Enter 30# Check Mark from
Press 1 to Enter a New Value
Press 1# to Enable
Press 1 to Save
Hang up

If done properly, the following window should appear on your screen:
[[File:ObiLogin.png|300px|thumb|left|Authentication Window - click to enlarge]]
<br clear="all" />

Once you get the Authentication Required window you will be prompted for a User Name and Password, the default credentials are:

'''User Name:''' admin

'''Password:''' admin

After this, you should now be able to see the OBi Web interface.

Now on the left side of the screen please find the next options and follow the next steps:

===Disabling auto-provisioning===

'''**NOTE :''' You may use this guide to configure an OBi110 as well. This is the VoIP.ms recommended configuration versus using the Obihai configuration dashboard (more on this later on this page) and you may also not find all new VoIP.ms servers on the Obihai Dahsboard. In order to make sure there will be no conflicts between this Manual configuration and the Obihai dashboard, please perform the following steps to disable auto-provisioning:

*System Management -> Auto Provisioning -> Auto Firmware Update -> Method : Disabled
*System Management -> Auto Provisioning -> ITSP Provisioning -> Method : Disabled
*System Management -> Auto Provisioning -> OBiTALK Provisioning -> Method : Disabled
*Voice Services -> OBiTALK Service -> Enable : Unchecked

Please note you must remove the check mark from the "default" column, then under "Method" please use the ''''Drop Down Selection'''' and choose '''Disabled'''.

[[File:Step1.png|450px|thumb|left|Disabling Auto Provision - click to enlarge]]
<br clear="all" />
After this, save all changes and you are ready to move on to the actual configuration.
<br />
===Configuring the ITSP Profile===

====General Section====
In this section you will set the name and the DigiMap you will use in the profile you configure. By default you will configure the profile A, unless you use the same device with another provider.

:'''Name''': 100000 (Replace with your 6 digit Main SIP account UserID or Sub Account user name, i.e. 123456 or 123456_sub)<br/>
:'''DigitMap''': Copy the line, including parenthesis, in the Digitmap field in the ITSP Profile and replace the "555" digits in the following lines by the area code of your choice:

::Dial Plan (recommended):
(1xxxxxxxxxx|<1555>[2-9]xxxxxx|<1>[2-9]xxxxxxxxx|911|011xx.|xx.|*xx.|***xxx|4xxx|(Mipd)|[^*]@@.)

Also, some clients have been successful by dialing '''**1 and *97''' for line 1 or '''**2 and *97''' for line 2.

:*If you need to set the dial plan back to Default, you can use this:

(1xxxxxxxxxx|<1>[2-9]xxxxxx|<1>[2-9]xxxxxxxxx|011xx.|***xxx|xx.|(Mipd)|[^*]@@.)

[[File:Step2.png|550px|thumb|left|ITSP profile, General - click to enlarge]]
<br clear="all" />

====SIP Section====
In this section you can set the server and the port that you wish to register to.

Please note that in order to change the settings, you need to uncheck the Default box on the right hand side.

*ProxyServer: denver.voip.ms (one of VoIP.ms multiple [http://wiki.voip.ms/article/Choosing_Server servers], you can choose the one closest to your location.)
*ProxyServerPort: 5060
*RegistrarServer: denver.voip.ms (one of VoIP.ms multiple servers [http://wiki.voip.ms/article/Choosing_Server servers], you can choose the one closest to your location.)
*RegistrarServerPort: 5060

[[File:Step3.png|550px|thumb|left|ITSP profile, SIP - click to enlarge]]
<br clear="all" />

Additionally, you may want to change the RegisterExpires value to 300, scroll down, deselect the default box and set the value there from 3600 to 300.

[[File:Step4.png|550px|thumb|left|ITSP profile, SIP (Register Expires)- click to enlarge]]
<br clear="all" />

===Configuring Voice Services===
<br/>
In this section you can set your Main account/sub_account credentials like User name and Password. The Main account password by default is the same password as the Customer Portal.

*AuthUserName: 100000 (Replace with your 6 digits Main SIP account UserID or Sub Account user name, i.e. 123456 or 123456_sub)
*AuthPassword: ****** (''Your SIP Account Password'')

[[File:Step5.png|550px|thumb|left|Voice Services (SIP Credentials) - click to enlarge]]
<br clear="all" />

Once you have finished changing all those settings, click on the button ''Submit'' to save the changes and ''reboot your OBi device'', your device should now be registered.

== Configuration for usage with call encryption TLS/SRTP ==

===Configuring a Voice line using TLS===

'''NOTE''': This section is optional. It explains in detail how to use encrypted traffic in the device. If you are not certain about how encrypted traffic works, or the benefits of encrypting SIP traffic, please contact technical support for more information.

These devices are compatible with TLS, however, some settings need to be adjusted to have it working. This section will be assuming that your SIP account is already enabled to use TLS, if you have not enabled it yet, please follow these instructions before going further:

For more information on how to enable encrypted traffic for the main account, please click on [[Call_Encryption_-_TLS/SRTP#Activate_This_Option_on_Your_Main_Account | Main account]] or more information on how to enable encrypted traffic for the sub-account [[Call_Encryption_-_TLS/SRTP#Activate_This_Option_on_Your_Sub_Account | sub account]]

=== Configuring your OBi device ===

In order to use TLS/call encryption with OBi devices, you'll need to modify the following parameters accordingly. Please note that this is only available for some OBi devices, and the screenshots are from the most recent firmware version.

Under Service Providers > ITSP Profile > SIP use the following values:
:'''ProxySeverPort''': 5061
:'''ProxyServerTransport''': TLS
:'''RegistrarServerPort''': 5061
:'''OutboundProxyPort''': 5061
:'''X_OutboundProxyTransport''': TLS

[[File:TLSOBi1.png|550px|thumb|left|ITSP profile, SIP (TLS)- click to enlarge]]
<br clear="all" />

Under Voice Services > SP Service use the following values:
:'''X_KeepAliveServerPort''': 5061
:'''X_SRTP''': Use SRTP Only

[[File:TLSOBi2.png|550px|thumb|left|Voice Services, SP Service (SRTP)- click to enlarge]]
<br clear="all" />

== Configuration Using OBi Dashboard ==

Besides the Manual Configuration previously explained, Obihai also provides us with their own API dashboard where you can add your device, to complete the configuration in easy steps.
Add your device to the OBiTALK service in the OBi Dashboard [http://www.obitalk.com/obinet/]. Instructions for this are included with the OBi110 and are not discussed here.

After the OBi110 is added, edit the device. You can select '''Service Provider 1''' or '''Service Provider 2''' under the '''Configure Voice Services''' heading. This will take you to a page where you can select ''voip.ms''. Follow the instructions and once you are done the configuration will be downloaded to your Obi110.

==Known Issues and Resolutions==
=== Ghost Calls ===
<br/>
Some customers have reported receiving constant calls from "100" or "101" as callerID. These calls are going directly to your device and do not pass through our servers, so we cannot filter them. You can read more about these calls, including how to prevent them, here: [[Sip Scanner Ghost Calls]]

You can also follow these suggestions, specific for the OBi brand devices:
*You can just disable (by unchecking Enable) for SP2 and OBiTALK under your Voice Tab (If you are using our service as SP1).

*You can restrict which IP addresses that can connect to your OBi. Going to "Service Providers -> ITSP Profile A -> SIP -> X_AccessList" : voip.ms_ip_address. You can see the IP address of the server you are currently using from this link: [http://wiki.voip.ms/article/Choosing_Server#IPs Server's IPs]

*You can also change your Obi Firewall Setting X_InboundCallRoute to : {(?|x|xx|xxx|xxxx|xxxxx|xxxxxx|un@@.|anon@@.):}, ph
This will only allow 7 digit or greater numbers through.

*Another alternative: OBi Interface>> Voice Services>> SP1 Service>> X_InboundCallRoute: {>('Insert your AuthUserName here'):ph}, example:

{>('100000'):ph} where 100000 is replaced with your own six digit SIP account UserID or the sub-account registered with your device.

By default, OBi devices accept calls destined for any username. The above syntax rejects calls that are not intended for whatever you have configured as AuthUserName.

<br/>

===10 Second Delay Reaching voip.ms Voicemail Attendant when dialing *97 or *98===

The Obi 100, 110 and 202 devices have non-configurable 'short' and 'long' delays if a dialed sequence does not match a digitmap. So you may have a 10 second delay when you dial into your voip.ms voicemail because of the built-in 'long' delay. This can be resolved in a couple of ways. Simply dial a # sign after you dial *97 or *98. Or include literals in your digitmap under the Service Provider / ITSP profile A or B / General / digitmap. Here is an example digitmap with a *97 literal included:

(1xxxxxxxxxx|<1>[2-9]xxxxxxxxx|011xx.|xx.|*xx.|'*97'|(Mipd)|[^*#]@@.)

The literal in the example is '*97'. You could also add a literal for '*98'.

Then when you dial *97, the device immediately sends it instead of waiting 10 seconds.

Read more on digitmaps under the topic Digit Map Configuration in the Obi Device Admin Guide.

=== Call Drops ===

If you experience random call drops while in the middle of a call or if the person you talk to remains silent for over a minute (60 seconds by default), OBi will hang up the call. Please go here and check and increase the following setting (Physical interface -> LINE port -> DetectFarEndLongSilence / SilenceTimeThreshold)

=== Enable Message Waiting Indicator MWI ===

To enable MWI please refer to the following section of the OBI web page:

Voice Services -> SP1 Service -> Calling Features ->

'''MWIEnable''' - Uncheck the box at the far right, to be able to check the box at the left, this enables the option.

'''X_VMWIEnable''' - Uncheck the box at the far right, to be able to check the box at the left, this enables the option.

'''MessageWaiting''' - Mark the checkbox

After these steps, the MWI should be active and working.

'''If you are trying to place an outbound call and get a recorded message ¨There is no service to complete your call¨ Please do the following to resolve this.'''
In Your OBi Device please go to Physical Interfaces >> PHONE Port which by default it is PSTN and it needs to be changed to Trunk Group 1

=== Using the OBi Network ===

You can use your OBi device to make calls directly to other OBi devices "''The OBi comes out of the box ready to make FREE calls to other OBi endpoints using the OBiTALK network. Dialing **9 + obi account number will use the OBiTALK feature and does not place calls to regular numbers nor use our network. ''" (you can get more information about [http://www.obihai.com/features-and-set-up here]), be aware that those calls will not pass through our network. If you need assistance with that feature, please contact [http://www.obihai.com/request-support OBI's support].
<br/>
=== Can not dial *98 even if is on your DigiMap ===
By default your OBi device uses *98 as Blind transfer code. If you want to be able to dial *98 from your device, you should change this code. You can achieve this in the settings of your device at: ''Star Codes >> Star code profile (A/B)'', unmark the "default" box and change *98 for something else (like *99)

[[File:Step6.png|550px|thumb|left|Changing *98 default code - click to enlarge]]
<br clear="all" />

=== Using the Phone book of your customer portal ===

If you plan on using the Phone Book in your Customer Portal and Speed Dial *75. Please log into your OBi and change the built-in speed dial code from *75 in the device to something else.

=== An additional note regarding outgoing calls===

In at least one instance it was necessary to specify a non-default outbound calling route in the OBi110 to be able to place calls using the voip.ms service. The default setting had the OBi110 attempting to place calls using the PSTN port on the device. The relevant setting is:

'''Physical Interfaces >> PHONE Port '''
*PrimaryLine: (Select from drop-down)

[[File:ObiPhoneport.JPG|550px|thumb|left|Changing Phone Port - click to enlarge]]
<br clear="all" />

The default is PSTN. Select SP1 Service if you only have one SIP account configured on the device. Select Trunk Group 1 to have it attempt to place calls using SP1 first, then SP2. Additional Trunk groups can be configured under Voice Services >> Gateways and Trunk Groups.

Portions of this article have been taken from [http://www.toao.net/500-mangos-guide-to-configuring-an-obi100-obi110-and-obi202-ata Mango's Guide to Configuring an OBi ATA]. Used with permission.

==Guide Links==
In the event where you need the guide directly from OBIHAI Technology Inc, you may find the admin manual guide below:

Admin Manual : [https://www.obitalk.com/info/documents/admin_guide/OBiDeviceAdminGuide.pdf Download PDF]

[[Category:Analog Telephone Adapters]]

== OBI 100/110 OBI 200/202 star codes ==

Here's a list of the most common used Star codes available for the OBI. <br>
You may also consult the following link in order to see the full list of star codes available for the OBI devices:
<br>
[http://www.obihai.com/docs/OBiFeatureStarCodes.pdf OBI feature star codes]
<br>

* *81 Block Caller ID (Persistent Mode)
* *82 Unblock Caller ID (Persistent Mode)
* *67 Block Caller ID (One Time)
* *68 Unblock Caller ID (One Time)
* *72 Call Forward All (Enter Number + #)
* *73 Disable Call Forward All
* *60 Call Forward on Busy (Enter Number + #)
* *61 Disable Call Forward in Busy
* *62 Call Forward on No Answer (Enter Number + #)
* *63 Disable Call Forward No Answer
* *56 Enable Call Waiting
* *57 Disable Call Waiting
* *78 Do Not Disturb - Turn On
* *79 Do Not Disturb – Disable

OBi300

2020-02-25T17:58:43Z

Joseph: /* Configuration for usage with call encryption TLS/SRTP */

[[File:obi300.png|none|200px|center|link=https://www.polycom.com/voice-conferencing-solutions/voip-adapters/obi300.html|OBi 300]]

'' The OBi100 & OBi200 are perfect for customers who do not have a traditional phone service, yet need a similar solution and want the savings and simplicity of using a VoIP service for all their calls.

To start configuring your device you will need to plug it in to your router/modem via its Internet port with an Ethernet cable and connect a regular handset phone to it's Phone port, then follow the next steps.''<br/>

== Manual Configuration Details ==

Start by dialing ''' * * * ''' from the connected phone, then press '''1''' to confirm your choice, this will return the IP address of your device being a number similar to '''192.168.xxx.xxx'''.

Once you get the IP address, enter it in the URL address bar '''"http://"''' of your Internet Browser to get access to the Graphic User Interface of the OBi100.

If done properly, the following window should appear on your screen:

[[File:Obi300_Login.png|300px|thumb|left]]
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Once you get the Authentication Required window you will be prompted for a User Name and Password, the default credentials are:

'''User Name:''' admin

'''Password:''' admin

After this, you should now be able to see the OBi Web interface.

Now on the left side of the screen please find the next options and follow the next steps:

===Disabling auto-provisioning===

'''**NOTE :''' You may use this guide to configure an OBi110 as well. This is the VoIP.ms recommended configuration versus using the Obihai configuration dashboard (more on this later on this page) and you may also not find all new VoIP.ms servers on the Obihai Dahsboard. In order to make sure there will be no conflicts between this Manual configuration and the Obihai dashboard, please perform the following steps to disable auto-provisioning:

*System Management -> Auto Provisioning -> Auto Firmware Update -> Method : Disabled
*System Management -> Auto Provisioning -> ITSP Provisioning -> Method : Disabled
*System Management -> Auto Provisioning -> OBiTALK Provisioning -> Method : Disabled
*Voice Services -> OBiTALK Service -> Enable : Unchecked

Please note you must remove the check mark from the "default" column, then under "Method" please use the ''''Drop Down Selection'''' and choose '''Disabled'''.

[[File:Obi300_AutoProv.png|450px|thumb|left|Disabling Auto Provisioning]]
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After this, save all changes and you are ready to move on to the actual configuration.
<br />
===Configuring the ITSP Profile===

====General Section====
In this section you will set the name and the DigiMap you will use in the profile you configure. By default you will configure the profile A, unless you use the same device with another provider.

:'''Name''': 100000 (Replace with your 6 digit Main SIP account UserID or Sub Account user name, i.e. 123456 or 123456_sub)<br/>
:'''DigitMap''': Copy the line, including parenthesis, in the Digitmap field in the ITSP Profile and replace the "555" digits in the following lines by the area code of your choice:

::Dial Plan (recommended):
(1xxxxxxxxxx|<1555>[2-9]xxxxxx|<1>[2-9]xxxxxxxxx|911|011xx.|xx.|*xx.|***xxx|4xxx|(Mipd)|[^*]@@.)

Also, some clients have been successful by dialing '''**1 and *97''' for line 1 or '''**2 and *97''' for line 2.

:*If you need to set the dial plan back to Default, you can use this:

(1xxxxxxxxxx|<1>[2-9]xxxxxx|<1>[2-9]xxxxxxxxx|011xx.|***xxx|xx.|(Mipd)|[^*]@@.)

[[File:Step2.png|550px|thumb|left|ITSP profile, General - click to enlarge]]
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====SIP Section====
In this section you can set the server and the port that you wish to register to.

Please note that in order to change the settings, you need to uncheck the Default box on the right hand side.

*ProxyServer: denver.voip.ms (one of VoIP.ms multiple [http://wiki.voip.ms/article/Choosing_Server servers], you can choose the one closest to your location.)
*ProxyServerPort: 5060
*RegistrarServer: denver.voip.ms (one of VoIP.ms multiple servers [http://wiki.voip.ms/article/Choosing_Server servers], you can choose the one closest to your location.)
*RegistrarServerPort: 5060

[[File:Step3.png|550px|thumb|left|ITSP profile, SIP - click to enlarge]]
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Additionally, you may want to change the RegisterExpires value to 300, scroll down, deselect the default box and set the value there from 3600 to 300.

[[File:Step4.png|550px|thumb|left|ITSP profile, SIP (Register Expires)- click to enlarge]]
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===Configuring Voice Services===
<br/>
In this section you can set your Main account/sub_account credentials like User name and Password. The Main account password by default is the same password as the Customer Portal.

*AuthUserName: 100000 (Replace with your 6 digits Main SIP account UserID or Sub Account user name, i.e. 123456 or 123456_sub)
*AuthPassword: ****** (''Your SIP Account Password'')

[[File:Step5.png|550px|thumb|left|Voice Services (SIP Credentials) - click to enlarge]]
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Once you have finished changing all those settings, click on the button ''Submit'' to save the changes and ''reboot your OBi device'', your device should now be registered.

== Configuration for usage with call encryption TLS/SRTP ==

===Configuring a Voice line using TLS===

'''NOTE''': This section is optional. It explains in detail how to use encrypted traffic in the device. If you are not certain about how encrypted traffic works, or the benefits of encrypting SIP traffic, please contact technical support for more information.

These devices are compatible with TLS, however, some settings need to be adjusted to have it working. This section will be assuming that your SIP account is already enabled to use TLS, if you have not enabled it yet, please follow these instructions before going further:

For more information on how to enable encrypted traffic for the main account, please click on [[Call_Encryption_-_TLS/SRTP#Activate_This_Option_on_Your_Main_Account | Main account]] or more information on how to enable encrypted traffic for the sub-account [[Call_Encryption_-_TLS/SRTP#Activate_This_Option_on_Your_Sub_Account | sub account]]

In order to use TLS/call encryption with OBi devices, you'll need to modify the following parameters accordingly. Please note that this is only available for some OBi devices, and the screenshots are from the most recent firmware version.

Under Service Providers > ITSP Profile > SIP use the following values:
:'''ProxySeverPort''': 5061
:'''ProxyServerTransport''': TLS
:'''RegistrarServerPort''': 5061
:'''OutboundProxyPort''': 5061
:'''X_OutboundProxyTransport''': TLS

[[File:TLSOBi1.png|550px|thumb|left|ITSP profile, SIP (TLS)- click to enlarge]]
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Under Voice Services > SP Service use the following values:
:'''X_KeepAliveServerPort''': 5061
:'''X_SRTP''': Use SRTP Only

[[File:TLSOBi2.png|550px|thumb|left|Voice Services, SP Service (SRTP)- click to enlarge]]
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Konftel 300Wx IP

2020-02-06T19:39:10Z

Joseph: Created page with "left <br clear="all" /> The wireless conference phone [https://www.konftel.com/en/products/konftel-300wx Konftel 300Wx] allows you to h..."

File:Konftel-300Wx-IP.png

2020-02-06T19:37:12Z

Joseph:

File:Kon300 serverTLS.png

2020-02-06T19:37:05Z

Joseph:

File:Kon300 server.png

2020-02-06T19:36:59Z

Joseph:

File:Kon300 main.png

2020-02-06T19:36:52Z

Joseph:

File:Kon300 extensionTLS.png

2020-02-06T19:36:46Z

Joseph:

File:Kon300 extensionsRegistered.png

2020-02-06T19:36:39Z

Joseph:

File:Kon300 extension.png

2020-02-06T19:36:34Z

Joseph:

File:Kon300 add server.png

2020-02-06T19:36:27Z

Joseph:

SIP Responses

2020-01-24T19:10:35Z

Joseph:

SIP responses are the codes used by ''Session Initiation Protocol'' for communication. They complement the [[SIP Requests]], which are used to initiate action such as a phone conversation. Note that the Reason Phrases of the responses listed below are only the recommended examples, and can be replaced with local equivalents without affecting the protocol.SIP responses are the codes used by Session Initiation Protocol for communication. They complement the SIP Requests, which are used to initiate action such as a phone conversation. Note that the Reason Phrases of the responses listed below are only the recommended examples, and can be replaced with local equivalents without affecting the protocol.

=== 1xx = Informational SIP Responses ===

{| class="wikitable" style="border: none;"
|-
! scope="col" | Code
! scope="col" | Name
! scope="col" | Description
|-
| 100 || Trying || Extended search is being performed so a forking proxy must send a 100 Trying response.
|-
| 180 || Ringing || The Destination User Agent has received the INVITE message and is alerting the user of call.
|-
| 181 || Call Is Being Forwarded || Optional, send by Server to indicate a call is being forwarded.
|-
| 182 || Queued || Destination was temporarily unavailable, the server has queued the call until the destination is available.
|-
| 183 || Session Progress || This response may be used to send extra information for a call which is still being set up.
|-
| 199 || Early Dialog Terminated || Send by the User Agent Server to indicate that an early dialogue has been terminated.
|}

=== 2xx = Success Responses ===

{| class="wikitable" style="border: none;"
|-
! scope="col" | Code
! scope="col" | Name
! scope="col" | Description
|-
| 200 || OK || Shows that the request was successful
|-
| 202 || accepted || Indicates that the request has been accepted for processing, mainly used for referrals.
|-
| 204 || No Notification || Indicates that the request was successful but no response will be received.
|}

=== 3xx = Redirection Responses ===

{| class="wikitable" style="border: none;"
|-
! scope="col" | Code
! scope="col" | Name
! scope="col" | Description
|-
| 300 || Multiple Choices || The address resolved to one of several options for the user or client to choose between.
|-
| 301 || Moved Permanently || The original Request URI is no longer valid, the new address is given in the Contact header.
|-
| 302 || Moved Temporarily || The client should try at the address in the Contact field.
|-
| 305 || Use Proxy || The Contact field details a proxy that must be used to access the requested destination.
|-
| 380 || Alternative Service || The call failed, but alternatives are detailed in the message body.
|}

=== 4xx = Request Failures ===

{| class="wikitable" style="border: none;"
|-
! scope="col"| Code
! scope="col"| Name
! scope="col"| Description
|-
| 400 || Bad Request || The request could not be understood due to malformed syntax.
|-
| 401 || Unauthorized || The request requires user authentication. This response is issued by UASs and registrars.
|-
| 402 || Payment Required || (Reserved for future use).
|-
| 403 || Forbidden || The server understood the request but is refusing to fulfill it.
|-
| 404 || Not Found || The server has definitive information that the user does not exist at the (User not found).
|-
| 405 || Method Not Allowed || The method specified in the Request-Line is understood, but not allowed.
|-
| 406 || Not Acceptable || The resource is only capable of generating responses with unacceptable content.
|-
| 407 || Proxy Authentication Required || The request requires user authentication.
|-
| 408 || Request Timeout || Couldn’t find the user in time.
|-
| 409 || Conflict || User already registered (deprecated)
|-
| 410 || Gone || The user existed once but is not available here anymore.
|-
| 411 || Length Required || The server will not accept the request without a valid content length (deprecated).
|-
| 412 || Conditional Request Failed || The given precondition has not been met.
|-
| 413 || Request Entity Too Large || Request body too large.
|-
| 414 || Request URI Too Long || Server refuses to service the request, the Req-URI is longer than the server can interpret.
|-
| 415 || Unsupported Media Type || Request body is in a non-supported format.
|-
| 416 || Unsupported URI Scheme || Request-URI is unknown to the server.
|-
| 417 || Uknown Resource-Priority || There was a resource-priority option tag, but no Resource-Priority header.
|-
| 420 || Bad Extension || Bad SIP Protocol Extension used, not understood by the server.
|-
| 421 || Extension Required || The server needs a specific extension not listed in the Supported header.
|-
| 422 || Session Interval Too Small || The request contains a Session-Expires header field with a duration below the minimum.
|-
| 423 || Interval Too Brief || The expiration time of the resource is too short.
|-
| 424 || Bad Location Information || The request’s location content was malformed or otherwise unsatisfactory.
|-
| 428 || Use Identity Header || The server policy requires an Identity header, and one has not been provided.
|-
| 429 || Provide Referrer Identity || The server did not receive a valid Referred-By token on the request.
|-
| 430 || Flow Failed || A specific flow to a user agent has failed, although other flows may succeed.
|-
| 433 || Anonymity Disallowed || The request has been rejected because it was anonymous.
|-
| 436 || Bad Identity-Info || The request has an Identity-Info header and the URI scheme contained cannot be de-referenced.
|-
| 437 || Unsupported Certificate || The server was unable to validate a certificate for the domain that signed the request.
|-
| 438 || Invalid Identity Header || Server obtained a valid certificate used to sign a request, was unable to verify the signature.
|-
| 439 || First Hop Lacks Outbound Support || The first outbound proxy doesn’t support the “outbound” feature.
|-
| 440 || Max-Breadth Exceeded || If a SIP proxy determined a response context had insufficient Incoming Max-Breadth to carry out a desired parallel fork, and the proxy is unwilling/unable to compensate by forking serially or sending a redirect, that proxy MUST return a 440 response. A client receiving a 440 response can infer that its request did not reach all possible destinations.
|-
| 469 || Bad Info Package || If a SIP UA receives an INFO request associated with an Info Package that the UA has not indicated willingness to receive, the UA MUST send a 469 response, which contains a Recv-Info header field with Info Packages for which UA is willing to receive INFO requests.
|-
| 470 || Consent Needed || The source of the request did not have the permission of the recipient to make such a request.
|-
| 480 || Temporarily Unavailable || Callee currently unavailable.
|-
| 481 || Call/Transaction Does Not Exist || Server received a request that does not match any dialogue or transaction.
|-
| 482 || Loop Detected || Server has detected a loop.
|-
| 483 || Too Many Hops || Max-Forwards header has reached the value ‘0’.
|-
| 484 || Address Incomplete || Request-URI incomplete.
|-
| 485 || Ambiguous || Request-URI is ambiguous.
|-
| 486 || Busy Here || Callee is busy.
|-
| 487 || Request Terminated || Request has terminated by bye or cancel.
|-
| 488 || Not Acceptable Here || Some aspects of the session description of the Request-URI are not acceptable.
|-
| 489 || Bad Event || The server did not understand an event package specified in an Event header field.
|-
| 491 || Request Pending || Server has some pending request from the same dialogue.
|-
| 493 || Undecipherable || UndecipherableRequest contains an encrypted MIME body, which recipient can not decrypt.
|-
| 494 || Security Agreement Required || The server has received a request that requires a negotiated security mechanism.
|}

=== 5xx = Server Errors ===

{| class="wikitable" style="border: none;"
|-
! scope="col" | Code
! scope="col" | Name
! scope="col" | Description
|-
| 500 || Server Internal Error || The server could not fulfill the request due to some unexpected condition.
|-
| 501 || Not Implemented || The SIP request method is not implemented here.
|-
| 502 || Bad Gateway || The server, received an invalid response from a downstream server while trying to fulfill a request.
|-
| 503 || Service Unavailable || The server is in maintenance or is temporarily overloaded and cannot process the request.
|-
| 504 || Server Time-out || The server tried to access another server while trying to process a request, no timely response.
|-
| 505 || Version Not Supported || The SIP protocol version in the request is not supported by the server.
|-
| 513 || Message Too Large || The request message length is longer than the server can process.
|-
| 555 || Push Notification Service Not Supported || The server does not support the push notification serviced specified in the pn-provider SIP URI parameter.
|-
| 580 || Precondition Failure || The server is unable or unwilling to meet some constraints specified in the offer.
|}

=== 6xx = Global Failures===

{| class="wikitable" style="border: none;"
|-
! scope="col" | Code
! scope="col" | Name
! scope="col" | Description
|-
| 600 || Busy Everywhere || All possible destinations are busy.
|-
| 603 || Decline || Destination cannot/don't wish to participate in the call, no alternative destinations.
|-
| 604 || Does Not Exist Anywhere || The server has authoritative information that the requested user does not exist anywhere.
|-
| 606 || Not Acceptable || The user’s agent was contacted successfully but some aspects of the session description were not acceptable.
|-
| 607 || Unwanted || The called party did not want his call from the calling party. Future attempts from the calling party are likely to be similarly rejected.
|}