VoIP.ms Wiki - User contributions [en]

FAQ

2015-01-21T15:48:50Z

Clark:

[[File:Logofaq.jpg|250px|]]

<span style="font-size: 120%;">__TOC__</span>

Pour la page en Francais veillez cliquez ici: [[Questions_Les_Plus_Fréquentes|Questions Les Plus Fréquentes]]

Haga clic aquí para Español:[[Preguntas frecuentes]]

== Can I port my existing number from another provider to VoIP.ms ==
VoIP.ms does offer LNP service, and your number may be available for porting. Please contact Customer service to find out whether your current number is portable to VoIP.ms network or if you already have an open account, refer to the [[Porting a Number]] guide.

== Why am I receiving calls from an "Asterisk" caller ID? ==
If you are receiving calls with '''“Asterisk”''' as the caller ID name, the first suggestion is to check if the calls appear on your VoIP.ms CDR as incoming calls. If they don't, then you can be sure that these calls are not coming through Voip.ms system since every call that reaches our system will appear on your VoIP.ms CDR.

If this is the case, one of the most probable reasons is a direct SIP call to your Asterisk System or Device, for Spam, SIP campaigns or scanners searching for public IP addresses with SIP port open in order to Brute force.
Unfortunately this is out of Voip.ms control as these calls never reach our system.

However, there are some suggestions you could follow in order to eliminate this inconvenience:

* If you are using some kind of '''Asterisk based system''', please make sure you disable availability to receive “Anonymous calls” and install proper security.

* If you are using an '''ATA, softphone or IP phone''', check for any option to disable these anonymous calls, if there is not, most likely your device doesn't support “Anonymous calls” by default.

* Additionally is recommended to verify your router is '''not set on DMZ mode''' for your ATA device´s internal IP, to make sure your devices and Network are not reachable from public Internet and avoid any unnecessary port forwarding.

* If there is an option on the ATA device to '''restrict incoming calls to only the IP address you are registered with''', switch it on. (This will help to prevent calls from other than voip.ms being accepted)

== Does VoIP.ms offer Calling Card solution to purchase credits for outgoing calls? ==
Voip.ms does not offer a calling card solution. However, you can use your DIDs and forward them to a calling card provider if you desire.
Voip.ms also offers the [[DISA]] feature (resembles the calling card feature). When you call the DID, the system then will ask for a pin number, after entering the caller will be granted with dial tone to make the final call via VoIP while paying the VoIP.ms rates when calling to anywhere in the world.

== Where is VoIP.ms located ? ==

Voip.ms is a Canadian Company. The main office is located in Montreal, Canada and there is also an office in Merida, Mexico for their South America market. Credit Card charges are made in US Dollars (USD) Currency by Swiftvox Inc via Internet secure.

== What are the IP addresses of VoIP.ms´ servers ? ==

You will find the list of all of our servers and their IP addresses [http://wiki.voip.ms/article/Choosing_Server#IPs| Here]

== What server should I use ? ==

Usually in order to get better results, you should choose the server closest to your location. You can still send a ping to any of the servers to check the best response time.

== Do you support TCP for the SIP connection? ==

VoIP.ms supports TCP on almost all of their servers. Currently the only servers that do NOT support TCP are: Atlanta1 and Dallas

== Do you offer Conference calls ? ==

Voip.ms doesn't offer Conference calls as a feature, however if your system or device is capable of establishing a conference call, then you will have no issues running a conference using VoIP.ms

== Does VoIP.ms offer a STUN server ? ==

Voip.ms doesn't provide any kind of STUN server nor is it required to use one with their service. If for any specific reason you wish to do so, you can still find a public STUN server on the web for free if you need one.

== Does VoIP.ms encrypt the communication ? ==

The SIP communication is secure although not encrypted. However, the passwords are MD5 hashed and are not transmitted without encryption when establishing the call.

== Does VoIP.ms provide a Hardware device to use the service ? ==

VoIP.ms does not provide any kind of hardware device, software or system to use the service. The service is a BYOD (Bring your own device).
You should be able to get one from any Communications specialized store, and all SIP-compatible devices are supported. (Almost all VoIP devices support SIP protocol)

== Can I use my existing device with VoIP.ms ? ==

Basically, any device or system which supports SIP or IAX2 protocol will work with Voip.ms system. If you bring your device(ATA, IP phone) from a previous provider, make sure its not locked and you are able to make changes to its configuration.
Feel free to ask Technical support for further details, however they can not help you regarding how to unlock a device from other provider (for legal reasons).

== Can I call an International Toll Free number ? ==

At this moment calls to International Toll Free numbers are not supported. Only to US and Canada toll free numbers.

== Does VoIP.ms offer alternative ports besides 5060 ? ==

Voip.ms offers alternative SIP ports, UDP 5080 and 42872 on all of their servers, You can try those ports in case your ISP does block the SIP PORT 5060 UDP or if you need to use another one.

== Does VoIP.ms offer a referral program ? ==

Voip.ms is not offering any kind of referral program at the moment.

== Can I register 2 different devices with the same account ? ==

It is strongly suggested not to do so, it can cause conflicts while routing the calls and the devices can steal each others' registration.
If you need to register more than one device, please create and use [[Sub Accounts]], you will get new credentials for any additional device.

== How do I port out my number ? ==

If you have a DID number with VoIP.ms and wish to port it out to another provider, you need to always start this process on the new provider's side. They should be able to provide all the information required to start this process.

== How do I change the US48/Canada termination route for my subaccount ? ==

The US48/Canada termination route for the sub-accounts needs to be changed from the Main Menu >> [[Account Settings]] >> Account Routing tab. The route you use for the Main account is what will apply to the whole account, unlike the International route, which is independent per accounts/[[Sub Accounts]].

== Does VoIP.ms offer Distinctive Ring tones ? ==

Distinctive ring tones are not available at the moment, if you are in a situation where you need to know which one of your DIDs is receiving a call, a good option is for you to use the "Caller ID name Prefix" for the DID, to add a specific prefix to DID, different from the others.

== Can I deposit less than $25 USD ? ==

The minimum amount to deposit currently is $25 USD. However, if at anytime you don't feel satisfied with the service provided, you can ask for a refund of the remaining balance on your account.

== What is my Main Account SIP password ? ==

The main account SIP password by default is the same as your customer portal password. If you have not changed the SIP password, it is the same password you use to log into your portal. To change this password please refer to our Account Settings page: [[Account_Settings#Security|Settings Security]]

== Does VoIP.ms offer Call transfers ? ==

VoIP.ms does offer Call transfers, this needs to be requested to the VoIP.ms staff, it is not a setting you can enable from the customer portal.

== Does VoIP.ms offer wholesale rates ? ==

If you are interested on a discount based on Traffic usage or volume, please send an email to sales@voip.ms providing all details (destinations you need to call, average of minutes used per month), in order to receive proper information.

== Does VoIP.ms apply taxes ? ==

All customers with Canada as country will pay for the tax called GST. The customers who are also from Quebec in Canada will pay an additional province tax, PST. This applies for both Paypal and CC payments. The HST will get applied on Canadian provinces that use it.

In order to get further details about how these are applied, feel free to check this link: http://en.wikipedia.org/wiki/Sales_taxes_in_Canada

[[Category: Guides]]

2015-01-08T21:31:08Z

Clark:

[[File:Logo.png]]
One of the benefits of having a VoIP.ms account is that not only do you get both origination and termination services, but VoIP.ms also offers a complete solution for your business or home by providing you with many options and features to make your life easier than ever. Most of the Features listed below are completely free with the service and you will not need to pay to have these awesome benefits. To fully understand each of these features, you can click on the the feature name to access the specific guide for each topic.

{|border="1"
|+ FEATURES
!Feature !! Summary
|-
! [[Call Detail Records|Call Details Report and Invoice Generation]]
|You will be able to examine detailed information of all the calls coming and going from your Main account and Sub accounts simply from the portal.
Additionally you will find the '''Finances''' Menu, where you will get access to the tool to Generate Invoices according to your needs and also be able to add funds to your account.
|-
! [[Callback]]
|With this feature, you can define a number to be called by our system, in order to receive a dial tone and place outgoing calls through VoIP.ms. This could be useful if you want to place a call and you are not at home or don't have access to your voip device.
|-
! [[Caller ID]]
|Caller ID is a telephone service that transmits the calling party's number to the called party's telephone. When available, the Caller ID number can be complemented with the caller ID name (description e.g. John Smith)
|-
! [[Call Forwarding]]
|A Call Forwarding allows an incoming call to be redirected to a mobile telephone or another telephone number where the desired called party is able to answer. You can set any number, even international numbers.
|-
! [[CallerID Filtering]]
|This feature allows you to filter the incoming calls to your DID numbers given a specific Caller ID number, area code or even anonymous numbers. For example, if you receive annoying incoming calls from a telemarketing company you can create a filter to route all the calls to a recording that plays the message "That number is no longer in service, please hang-up and try again", amongst several other options.
|-
! [[Calling Queues]]
|If you want a solution to manage your incoming calls and have your customer(s) wait on the line while an agent picks up the call, you need to create a Calling Queue entry. This will permit you to have many calls on hold, queued calls in First In-First Out order until agents become available.
|-
! [[Porting a Number|DID Portability]]
|Don't lose that number that you like so much, if you are a VoIP.ms customer and have an existing number with a different provider you can optionally bring it (port it in) to VoIP.ms network. Toll free numbers, local USA/Canada and even International numbers can be ported to VoIP.ms.
The LNP (Local Number Portability) Department handles all porting processes and guides the customer through it. They are also in charge of updating the customer via email with any relevant information about the port every time there is a change on the port request status.
|-
! [[DigitalReceptionist_IVR|Digital Receptionist (IVR)]]
| Interactive Voice Response (IVR), (also know as Auto-attendant) this option can be used to present a recording to the people calling to your DID number and also give them the chance to enter the extension of the person or department they want to reach. For example, you could create an IVR and point one of your DID numbers to it, and when the IVR answers, the caller will hear "Thank you for calling XYZ Inc, for Sales press 1, for Service press 2", etc.
|-
! [[DISA]]
|Direct Inward System Access ( DISA ) allows you to use our system for placing outgoing calls, even if you are not close to any device where you are registering your account or sub account. In this case you just would need to dial to your DID number and to provide a 4 digit PIN number, you can then dial out to any number in the world under our termination rates.
|-
! [[E911]]
|VoIP 911 Service differs from traditional 911 services due to limitations brought on by VoIP technology. It is nearly impossible to detect where a call originates from when placed over the internet. e911 allows users to associate physical addresses with their DIDs, allowing them to have service similar to traditional 911.
'''Important Note: Use of our 911 Service costs a recovery setup fee of $ 1.50 on activation and a regulatory recovery fee of $ 1.50 per DID number activated per month.'''
|-
! Failover for Incoming calls
|Whether your system is unavailable or you are experiencing an Internet or power outage, VoIP.ms offers you the option to redirect the incoming calls to your DID numbers to any location where you can take the call, for example like your cell phone.
Since you are virtually able to select any regular Routing option for the failovers, you can customize your setup in different ways by selecting one of three states, No Answer, Unreachable or Busy.
|-
! Free Calls Between VoIP.ms Customers
|Save even more money with this feature, with VoIP.ms you get free calls from your account to any VoIP.ms DID from US and Canada. So if your friends, family or customers have VoIP.ms DIDs from US/Can, you can be sure that the call will be FREE.
|-
! [[Sub_Accounts#Optional_Settings|Internal Extensions]]
|This feature will allow you to define a quick dial access number to contact the subaccount directly and internally, this way you will be able to reach each a sub account by dialing an extension number E.G "102". This is an amazing feature because calls between extensions are free, so you can have one extension at your office or home in Canada and another one in the US or another country in the world and reach that subaccount dialing only the extension number. With this feature your business can be more efficient and also your friends and family will be in touch easily.
|-
! IP Authentication and IAX support
|Besides the regular method of authentication (Username and password), VoIP.ms also offers the customer the option to authenticate their system via IP address, this way the customer can configure their system to fit their needs.
Even though the native and regular protocol is SIP, VoIP.ms also supports IAX protocol for Customers wanting to communicate this way.
|-
! [[Account_Settings#Notifications|Low Balance Email Notifications]]
|Since it could be time consuming to constantly keep track of the Account balance, VoIP.ms offers you the option to receive an Email notification to your email address, when your account reaches a Low balance limit, below your defined Balance Threshold.
|-
! Music on Hold
|With our services you can easily chose if you want to play "Music on Hold" when the call is on hold state, the caller will hear smooth jazz while they wait on line. Also you can deactivate this option if you don't want to play Music to the caller, this can be done from your '''customer portal >> Main Menu >> Account Settings >> General. '''[http://wiki.voip.ms/article/Account_Settings#General]
|-
! [[Phone book]]
|The Phone Book feature allows you to configure Speed-Dial entries and Caller ID name (CNAM) overrides. For example, let's say you have a Customer, Provider or Relative that you call frequently, you can create a phone book entry in order to make a call using a speed-dial entry of 4 digits long. You can also have a Caller ID name (CNAM) override to identify the incoming calls of an important customer if his number doesn't have a proper Caller ID name (CNAM) linked to it, for free.
|-
! [[Recordings]]
|VoIP.ms allows you to upload an audio file and use it in the different options we have under the DID numbers menu. It can be used with several of the other features of our system such as Digital receptionist, Calling queues, and others.
|-
! [[Ring Groups]]
|The Ring Group feature allows you to have incoming calls to be redirected to different destinations that are included in your Ring Group, where any member of the group is able to answer. When you receive a call to a DID routed to a Ring Group, all members of that group will ring at the same time until one of them answers the call. You can add various types of members to a ring group:
Main Account,[[Sub Accounts]],[[SIP URI]]'s,[[Call Forwarding]].
|-
! [[Reseller Basic Guide| Reseller]]
|The Reseller feature is intended for users who want to initiate a business in the VoIP world by reselling VoIP.ms services, this useful tool can help you create and manage accounts for your own clients, you will be able to assign specific rates and DIDs to them and also provide credentials to register a device or call system. You will be able to handle the billing for your customers depending on your needs and how much you want to earn with your business, setting rates and adding credits for them. Additionally you will find options to customize your reseller portal to have a more personalized site.
|-
! [[SIP_URI|SIP URI]]
|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 or address. The SIP URI resembles an e-mail address.
|-
! [[SMS]]
|
This feature introduced on 2013 will allow you to send and receive messages with your DID Number (US and Canada DID Numbers Only).Currently it is in beta and will remain free (the price will be 1 cent per message starting sometime in 2015).
|-
! [[Sub Accounts]]
|Having a Sub Account allows you to register more than one device to make or receive calls simultaneously, you can also use it as internal extensions for your office or even your house. Many of the features within VoIP.ms make use of the sub accounts. With this guide we are going to learn how to create and use this feature properly.
|-
! [[Time Conditions]]
|Time Conditions is a feature that allows you to route your incoming calls to different destinations depending on the time of the day of the call. For example, you can have a Time condition for your Non-Business hours and all the incoming calls you receive will be routed to a recording or the voicemail directly (you can also route to other options, like IVR, Call Forwarding entry, etc.).
|-
! Vanity Toll Free Numbers
|Are you looking to have that special contact number for your business with specific digits? VoIP.ms offers the option to order a customized Toll Free number with any combination of digits you like.
Either for business or personal use, you will be able to use the tool from the portal and find that number.
|-
! [[Voicemail]]
|VoIP.ms has an advanced Voicemail feature that is free to use, which also provides you the option to forward your messages to your email address as an audio attachment.
In order to use the Voicemail feature with Voip.ms you will have to create a Voicemail entry and then assign your entry to one of your DIDs or Accounts.
|-
|}

'''If you have more inquires regarding the Features, feel free to contact the support team via the Live Chat or the Ticket system.

For the '''Live Chat''' you will find one link on the VoIP.ms web site, just scroll down and you will find the '''Live Support Access''' on the left, also if you have already an account you will find this at the top on the right of your customer portal.

To send request via the '''Ticket System''' you can open a new one from your customer portal >> Support >> New Ticket or send us an email to '''support@voip.ms'''

2015-01-06T15:13:05Z

Clark:

FAQ

2014-12-31T14:38:24Z

Clark: /* Do you support TCP for the SIP connection? */

__TOC__

Pour la page en Francais veillez cliquez ici: [[Questions_Les_Plus_Fréquentes|Questions Les Plus Fréquentes]]

Haga clic aquí para Español:[[Preguntas frecuentes]]

== Can I port my existing number from another provider to VoIP.ms ==
VoIP.ms does offer LNP service, and your number may be available for porting. Please contact Customer service to find out whether your current number is portable to VoIP.ms network or if you already have an open account, refer to the [[Porting a Number]] guide.

== Why am I receiving calls from an "Asterisk" caller ID? ==
If you are receiving calls with '''“Asterisk”''' as the caller ID name, the first suggestion is to check if the calls appear on your VoIP.ms CDR as incoming calls. If they don't, then you can be sure that these calls are not coming through Voip.ms system since every call that reaches our system will appear on your VoIP.ms CDR.

If this is the case, one of the most probable reasons is a direct SIP call to your Asterisk System or Device, for Spam, SIP campaigns or scanners searching for public IP addresses with SIP port open in order to Brute force.
Unfortunately this is out of Voip.ms control as these calls never reach our system.

However, there are some suggestions you could follow in order to eliminate this inconvenience:

* If you are using some kind of '''Asterisk based system''', please make sure you disable availability to receive “Anonymous calls” and install proper security.

* If you are using an '''ATA, softphone or IP phone''', check for any option to disable these anonymous calls, if there is not, most likely your device doesn't support “Anonymous calls” by default.

* Additionally is recommended to verify your router is '''not set on DMZ mode''' for your ATA device´s internal IP, to make sure your devices and Network are not reachable from public Internet and avoid any unnecessary port forwarding.

* If there is an option on the ATA device to '''restrict incoming calls to only the IP address you are registered with''', switch it on. (This will help to prevent calls from other than voip.ms being accepted)

== Does VoIP.ms offer Calling Card solution to purchase credits for outgoing calls? ==
Voip.ms does not offer a calling card solution. However, you can use your DIDs and forward them to a calling card provider if you desire.
Voip.ms also offers the [[DISA]] feature (resembles the calling card feature). When you call the DID, the system then will ask for a pin number, after entering the caller will be granted with dial tone to make the final call via VoIP while paying the VoIP.ms rates when calling to anywhere in the world.

== Where is VoIP.ms located ? ==

Voip.ms is a Canadian Company. The main office is located in Montreal, Canada and there is also an office in Merida, Mexico for their South America market. Credit Card charges are made in US Dollars (USD) Currency by Swiftvox Inc via Internet secure.

== What are the IP addresses of VoIP.ms´ servers ? ==

You will find the list of all of our servers and their IP addresses in the Choosing Server page:
http://wiki.voip.ms/article/Choosing_Server

== What server should I use ? ==

Usually in order to get better results, you should choose the server closest to your location. You can still send a ping to any of the servers to check the best response time.

== Do you support TCP for the SIP connection? ==

VoIP.ms supports TCP on almost all of their servers. Currently the only servers that do NOT support TCP are: Atlanta1 and Dallas

== Do you offer Conference calls ? ==

Voip.ms doesn't offer Conference calls as a feature, however if your system or device is capable of establishing a conference call, then you will have no issues running a conference using VoIP.ms

== Does VoIP.ms offer a STUN server ? ==

Voip.ms doesn't provide any kind of STUN server nor is it required to use one with their service. If for any specific reason you wish to do so, you can still find a public STUN server on the web for free if you need one.

== Does VoIP.ms encrypt the communication ? ==

The SIP communication is secure although not encrypted. However, the passwords are MD5 hashed and are not transmitted without encryption when establishing the call.

== Does VoIP.ms provide a Hardware device to use the service ? ==

VoIP.ms does not provide any kind of hardware device, software or system to use the service. The service is a BYOD (Bring your own device).
You should be able to get one from any Communications specialized store, and all SIP-compatible devices are supported. (Almost all VoIP devices support SIP protocol)

== Can I use my existing device with VoIP.ms ? ==

Basically, any device or system which supports SIP or IAX2 protocol will work with Voip.ms system. If you bring your device(ATA, IP phone) from a previous provider, make sure its not locked and you are able to make changes to its configuration.
Feel free to ask Technical support for further details, however they can not help you regarding how to unlock a device from other provider (for legal reasons).

== Can I call an International Toll Free number ? ==

At this moment calls to International Toll Free numbers are not supported. Only to US and Canada toll free numbers.

== Does VoIP.ms offer alternative ports besides 5060 ? ==

Voip.ms offers alternative SIP ports, UDP 5080 and 42872 on all of their servers, You can try those ports in case your ISP does block the SIP PORT 5060 UDP or if you need to use another one.

== Does VoIP.ms offer a referral program ? ==

Voip.ms is not offering any kind of referral program at the moment.

== Can I register 2 different devices with the same account ? ==

It is strongly suggested not to do so, it can cause conflicts while routing the calls and the devices can steal each others' registration.
If you need to register more than one device, please create and use [[Sub Accounts]], you will get new credentials for any additional device.

== How do I port out my number ? ==

If you have a DID number with VoIP.ms and wish to port it out to another provider, you need to always start this process on the new provider's side. They should be able to provide all the information required to start this process.

== How do I change the US48/Canada termination route for my subaccount ? ==

The US48/Canada termination route for the sub-accounts needs to be changed from the Main Menu >> [[Account Settings]] >> Account Routing tab. The route you use for the Main account is what will apply to the whole account, unlike the International route, which is independent per accounts/[[Sub Accounts]].

== Does VoIP.ms offer Distinctive Ring tones ? ==

Distinctive ring tones are not available at the moment, if you are in a situation where you need to know which one of your DIDs is receiving a call, a good option is for you to use the "Caller ID name Prefix" for the DID, to add a specific prefix to DID, different from the others.

== Can I deposit less than $25 USD ? ==

The minimum amount to deposit currently is $25 USD. However, if at anytime you don't feel satisfied with the service provided, you can ask for a refund of the remaining balance on your account.

== What is my Main Account SIP password ? ==

The main account SIP password by default is the same as your customer portal password. If you have not changed the SIP password, it is the same password you use to log into your portal. To change this password please refer to our Account Settings page: [[Account_Settings#Security|Settings Security]]

== Does VoIP.ms offer Call transfers ? ==

VoIP.ms does offer Call transfers, this needs to be requested to the VoIP.ms staff, it is not a setting you can enable from the customer portal.

== Does VoIP.ms offer wholesale rates ? ==

If you are interested on a discount based on Traffic usage or volume, please send an email to sales@voip.ms providing all details (destinations you need to call, average of minutes used per month), in order to receive proper information.

== Does VoIP.ms apply taxes ? ==

All customers with Canada as country will pay for the tax called GST. The customers who are also from Quebec in Canada will pay an additional province tax, PST. This applies for both Paypal and CC payments. The HST will get applied on Canadian provinces that use it.

In order to get further details about how these are applied, feel free to check this link: http://en.wikipedia.org/wiki/Sales_taxes_in_Canada

[[Category: Guides]]

Choosing Server

2014-12-30T16:20:51Z

Devices

2014-12-26T21:03:12Z

Clark:

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.
* '''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.

==3COM 3108 Wireless Phone==

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

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

'''Company:''' 3COM

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

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

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

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

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

'''Company:''' Aastra

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

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

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

==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 (analogue, ISDN, VoIP) and connections for 10 internal participants (analogue, 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 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 softswitch 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]]
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==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 (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 analogue hardware while migrating to voice over IP.

[[Cisco SPA112|See Configuration Details]]

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==Cisco WRP400==

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

'''Product:''' WRP400

'''Company:''' Cisco

'''Overview:''' The Cisco WRP400 wi-fi router connects to VoIP service through a wired broadband Internet connection and provides two virtual telephone (FxS) lines to which standard touch-tone 'phones may be connected, four 100baseT downlink ports for a wired local area network and full-featured wireless B/G networking. This device is effectively a complete local area network and a two-line analogue telephone adapter in one box.

If wired Internet is not available, the WRP400 is able to fall back to a mobile Internet connection as uplink if the appropriate hardware is connected to its USB port.

The telephony capabilities of the WRP400 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]]

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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 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 provide voice communication over an IP network. It provides traditional features, such as call forwarding, redialing, speed dialing, transferring calls, conference calling, and accessing voice mail. Calls can be made or received with a handset, headset or speaker.
Your Cisco IP Phone provides a web interface for the phone user that allows you to configure some features of your phone by using a web browser.
This article will guide you through the steps for basic configuration to make it work with VoIP.ms

[[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-feature telephone that provides voice communication over an IP network. This phone functions like 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 SPA2100 Phone Adapter==

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

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

'''Company:''' Cisco

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

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

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

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

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

'''Company:''' Cisco

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

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

[[Cisco_SPA2102_Phone_Adapter_with_Router|See Configuration Details]]
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==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 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 office and IP PBX applications. These do not provide a virtual line for connecting analogue 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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==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 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 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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==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 HandyTone 486==

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

'''Product:''' Grandstream HandyTone 486

'''Company:''' Grandstream

'''Overview:''' Grandstream's award-wining HandyTone-486 is an all-in-one VoIP integrated access device that features superb audio quality, rich functionalities, high level of integration, compactness and ultraaffordability. 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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==Mediatrix 4100 Series==

[[File:Mediatrix4102s.jpg|300px|thumb|left|Mediatrix 4102]]

'''Product:''' : Mediatrix 4102S and 4100 Series running DGW 2.0 firmware

'''Company:''' Media5 Corporation

'''Overview:''' The Mediatrix® 4100 Series device is a security ready VoIP gateway, connecting up to two analog phones and/or faxes, as well as a PC or a home router to a broadband modem. The Mediatrix 4102 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 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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==OBi 100/110==

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

'''Product:''' OBi 100/110

'''Company:''' 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 a traditional phone service. If you do not have a traditional phone service at home, then the OBi100 is probably the right product to get.

[[OBi 100/110|See Configuration Details]]
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==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 speaker phone on cordless handset

[[Panasonic KX-TGP 550|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 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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==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 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 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 built-in, full-duplex speakerphone and three-party conference bridging.

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

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

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

'''Company:''' Snom

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

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

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

[[Snom_m3_VoIP_Phone|See Configuration Details]]
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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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==Yealink SIP-T28P (VSRF)==

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

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

'''Company:''' Yealink

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

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

[[Yealink_SIP-T28P|See Configuration Details]]
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[[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]]
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PBXs

2014-12-26T21:01:54Z

Clark:

An acronym for Private Branch eXchange. PBX telephone systems support incoming calls from the outside PSTN, placing calls between users' phones (also known as extensions) and other phones or the outside PSTN, conferencing other users together, recording voicemails and a variety of other advanced telecommunication functions. PBX systems are broadly broken into several categories: traditional (also known as legacy); converged (also known as hybrid) or pure IP, aka IP-PBX.
Traditional PBX systems usually either don't support IP at all or they support it only with expensive add-on equipment.
Converged PBX systems support IP and PSTN connections with equal force. It is the most flexible and cost-effective model.
IP-PBX systems, as the name implies, support only IP connectivity. Any PSTN connectivity must be achieved through external converters, known as Gateways.

==3CX Phone System==

[[File:3CX Logo.jpg|300px|thumb|left|3CX Phone System]]
<br><br>
3CX Phone System for Windows is a software-based IP PBX that replaces a proprietary hardware PBX / PABX. 3CX’s IP PBX has been developed specifically for Microsoft Windows and is based on the SIP standard – making it easier to manage and allowing you to use any SIP phone (software or hardware).

[[3CX Phone System|3CX Phone System Configuration]]
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<br><br><br><br><br>

==Asterisk (SIP)==
[[File:TC-22622-MainIcon.gif‎‎|300px|x200px|thumb|left|Asterisk]]
<br><br>
Asterisk turns an ordinary computer into a communications server. Asterisk powers IP PBX systems, VoIP gateways, conference servers and is used by small businesses, large businesses, call centers, carriers and governments worldwide.
[[Asterisk (SIP)|Asterisk (SIP) Configuration]]
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<br><br><br><br><br>

==Asterisk (IAX2)==
[[File:TC-22622-MainIcon.gif‎‎|300px|x200px|thumb|left|Asterisk]]
<br><br>
Asterisk turns an ordinary computer into a communications server. Asterisk powers IP PBX systems, VoIP gateways, conference servers and is used by small businesses, large businesses, call centers, carriers and governments worldwide.
[[Asterisk (IAX2)|Asterisk (IAX2) Configuration]]
<div style="width:100%;overflow:hidden;clear:both"></div>
<br><br><br><br><br>

==Cisco IOS==
[[File:Cisco-new-logo-should-be.gif‎‎‎|300px|x200px|thumb|left|Cisco IOS]]
<br><br>
Cisco IOS (originally Internetwork Operating System) is software used on most Cisco Systems routers and current Cisco network switches. (Earlier switches ran CatOS.) IOS is a package of routing, switching, internetworking and telecommunications functions integrated into a multitasking operating system.
[[Cisco IOS|Cisco IOS Configuration]]
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<br><br><br><br><br>

==Elastix==

[[File:Elastix_Logo.jpg‎|300px|x200px|thumb|left|Elastix]]
<br><br>
Elastix is an open source unified communications server software that brings together: IP PBX, Email, IM and Faxing. The Elastix functionality is based on open source projects including Asterisk, HylaFAX, Openfire and Postfix. Those packages offer the PBX, fax, instant messaging and email functions, respectively, Elastix runs on CentOS operating system.
[[Elastix|Elastix Configuration]]
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<br><br><br><br><br>

==FreePBX / PBX in a Flash (SIP)==

[[File:FreePBX_Logo.jpg‎|300px|thumb|left|FreePBX]]
<br><br>
As an open source, web-based PBX solution, FreePBX is easy to customize and adapt to your changing needs. FreePBX is currently being used to manage the business communications of all sizes and types of businesses from small one person SOHO businesses, to multi-location corporations and call centers.
[[FreePBX / PBX in a Flash (SIP)|FreePBX / PBX in a Flash (SIP) Configuration]]
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<br><br><br><br><br>

==FreePBX / PBX in a Flash (IAX2)==

[[File:PIAF.gif‎‎|300px|thumb|left|PBX in a Flash]]
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If you've longed for the good ol' days of Asterisk@Home, welcome back to the new steroid-enhanced version. PBX in a Flash™ is the Lean, Mean Asterisk Machine designed to meet the needs of hobbyists as well as business users and VARs. You'll have a high-performance turnkey Asterisk PBX that's easy to upgrade with dozens of add on scripts to provide virtually any feature you can imagine.
[[FreePBX / PBX in a Flash (IAX2)|FreePBX / PBX in a Flash (IAX2) Configuration]]
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==FreeSwitch==

[[File:Fslogo.png‎‎|300px|thumb|left|FreeSwitch]]
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FreeSWITCH is a scalable open source cross-platform telephony platform designed to route and interconnect popular communication protocols using audio, video, text or any other form of media. It was created in 2006 to fill the void left by proprietary commercial solutions. FreeSWITCH also provides a stable telephony platform on which many telephony applications can be developed using a wide range of free tools.
[[FreeSwitch|FreeSwitch Configuration]]
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==Nortel/Avaya BCM 450 and BCM50 R6==

[[File:Avaya.jpg|300px|thumb|left|Avaya BCM]]
<br><br>
Avaya BCM450 (formerly Nortel BCM 450) and the BCM50 with VOIP or SIP Trunk keycode license.

The BCM system is a popular legacy Nortel phone system that uses classic Nortel "Meridian" M and T series digital sets and their Unistim IP phones.
[[NortelBCM|BCM Configuration]]
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[[category:PBXes]]

==PBXes.org==

[[File:Pbxeshead.png‎|300px|thumb|left|PBXes]]
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PBXes is a hosted PBX running like a voicemail service or an email server in a data center. Like a classical PBX it serves as the bridge between your organisation (the extensions) and the public telephone network (the trunks). Thus it optimises both your internal and external communications. As trunks any SIP providers worldwide may be used.

[[PBXes.org|PBXes Configuration]]
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[[category:PBXes]]

==SIPfoundry==

[[File:Sipfoundry-logo.png‎]]
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SIPfoundry is the home of the sipXecs open source unified communications and collaboration project. If you are looking to replace your aging (IP) PBX with an all software modern communications solution that scales to mid-size and large enterprise, look no further. We are community dedicated to the development of SIP and XMPP based communications solutions. Founded in 2004 as an independent 501(3)c not-for-profit corporation based in Massachusetts, SIPfoundry is governed by an independent board of directors. SIPfoundry is open an invites all interested parties to cooperate and collaborate. While the sipXecs project is the largest active project at SIPfoundry, we are open to make available our infrastructure to other interested projects. A commercial enterprise edition of the sipXecs solution is available from eZuce with full support, software maintenance, training, indemnification, and other services.

To learn how to configure sipXecs to work with voip.ms, follow this 10 minute guide here:

[http://myitdepartment.net/blog/191 Configuration]

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'''Known Issues'''

RTP Time Out when a person leaves a Voicemail resulting in only a 60 second voicemail being possible.

Fix: Please add the following line to your record_waste_resources.xml or vars.xml file this will make RTP keep alive packets to continue to be sent while recordings are made.

<action application="set" data="record_waste_resources=true"/>

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==Talkswitch==

[[File:TalkSwitch.png|300px|thumb|left|Talkswitch]]
<br><br>
Unlike other small business telephone systems, TalkSwitch is easy to install, saving you time and money; you can even do it yourself. And with its easily configured settings, moving employees or changing the way your phone system handles calls is a snap.

[[TalkSwitch|Talkswitch Configuration]]
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==Trixbox==

[[File:Trixbox_logo.jpg|300px|thumb|left|Trixbox]]
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Trixbox Community Edition began in 2004 as the massively popular Open Source IP-PBX project named Asterisk@Home. Since then, it has grown into the world's most popular distribution of Asterisk with over 65,000 downloads per month. trixbox is known for its flexibility to satisfy the needs of custom deployments and will continue to be FREE (as in beer and freedom).
'''Trixbox has not been maintained since June 2010. Customers should look for alternatives.'''
[[Trixbox|Trixbox Configuration]]
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2014-12-26T19:04:26Z

Clark:

{| cellspacing="0" style="width: 100%; border: 0; padding: 0 0 0 0; background: transparent;"
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<div style="text-align:center; font-family:Arial; word-spacing: 60px"><ins>'''[[Bienvenue|Français]]'''</ins> <ins>'''[[Bienvenido|Español]]'''</ins> </div>

[[Image:Facebook-40.png|right|link=https://www.facebook.com/VoIP.ms]] [[Image:Google-plus-40.png‎|right|link=https://www.google.com/+voipms]] [[Image:Twitter-40.png|right|link=https://twitter.com/voipms]]

__TOC__

<table style="font-family:Arial; width:60%; border: 1px black solid; border-collapse:collapse" align="center">
<tr style="background-color:orange; color: black">
<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|Can I Port My Number...?]] </td>
<td>[[Cisco_SPA112|Cisco SPA112]]</td>
</tr>
<tr>
<td>[[Call Forwarding|Call Forwarding]] </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_different_devices_with_the_same_account_.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>[[FAQ#What_are_the_IP_addresses_of_VoIP.ms.C2.B4_servers_.3F|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 VoIP.ms Community Wiki!

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

Please note that this is a work in progress and continuously evolving.

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 support@voip.ms requesting for edit permission for the username you use in this wiki.
<br><br><br>
[[Image:Download.jpg|left|50x75px|link=https://livechat.boldchat.com/aid/2947277729005480016/bc.chat?cwdid=60236691424546376&url=https%3A//www.voip.ms/]] '''Live Chat Support''' [[Image:Download (1).png|50x75px|]] '''1.877.7.VOIP.MS or 1.214.615.8599''' [[Image:Download.png|50x75px|]] '''support@voip.ms'''

==PBX==
An acronym for Private Branch eXchange. PBX telephone systems support incoming calls from the outside PSTN, placing calls between users' phones (also known as extensions) and other phones or the outside PSTN, conferencing other users together, recording voicemails and a variety of other advanced telecommunication functions. PBX systems are broadly broken into several categories: traditional (also known as legacy); converged (also known as hybrid) or pure IP, aka IP-PBX.

* '''Traditional PBX''' systems usually either don't support IP at all or they support it only with expensive add-on equipment.
* '''Converged PBX''' systems support IP and PSTN connections with equal force. It is the most flexible and cost-effective model.
* '''IP-PBX''' systems, as the name implies, support only IP connectivity. Any PSTN connectivity must be achieved through external converters, known as Gateways.

'''PBX Examples'''

If you have a PBX and would like to see some examples on how to configure it, you can check our [[PBXs|PBX's]] article
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==Devices==
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.
* '''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.

'''Devices Examples'''<br>
If you have a specific device and would like to see some examples on how to configure it, you can check our [[Devices|Devices]] article
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==Softphones==
A softphone is a software program for making telephone calls over the Internet using a general purpose computer, 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.

'''Softphone Examples'''

If you have a Softphone and would like to see examples on how to configure it, you can check our [[Softphones|Softphones]] article