VoIP.ms Wiki - User contributions [en]

MediaWiki:Sidebar

2026-05-15T21:37:44Z

RP:

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CallerID Filtering

2026-04-22T16:40:06Z

RP:

{| class="wikitable"
|+
|-
! Article en Français !! Artículo en Español
|-
| [https://wiki.voip.ms/article/Identification_de_l%27appelant Français] ||
[https://wiki.voip.ms/article/Filtro_de_Llamadas_(CallerID_Filtering) Español]
|}

This feature allows you to filter incoming calls to your DID numbers to route differently than the current DID routing that come from specific numbers, 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 routing options.
Exact matches will be given priority in the filters that you create, followed by Wildcards and then in priority from top to bottom on the existing CallerID Filtering Rules list.

__TOC__

= Create a Filtering entry =

First you need to go your Customer Portal and click on the "DID Numbers" >> "[https://www.voip.ms/m/callerid_filtering.php CallerID Filtering]" menu option.

[[File:CIDFiltering104222026.png|thumb|none|600px]]

The first thing you need to set is the type of filter you're going to create. You can choose between:
- Anonymous CallerID (This includes 'anonymous','private','restricted','unknown','unavailable','0000000000' & 'id' appearing in the SIP Privacy)
- CallerID not matching the North American NPANXXXXXX format ''(Could block International Calls)''
- All Phone Book
- Phone Book Group
- Specific CallerID Number
- STIR/SHAKEN Attestion Level which will allow you to block all calls being of a specific attestation or lower.

'''Note:''' Attestation A is the highest and most trustworthy calls. While attestation B might be spam and Attestation C and lower (no attestation) are most likely spam.
Do note that even though attestation A is considered trustworthy, there is still a chance that spam calls get signed '''A'''.

'''Note:''' Calls may be flagged as anonymous if they have a SIP Privacy header set to the value id. The purpose of this header is to allow the call to traverse networks where a valid Caller ID is required, while
restricting the Caller ID from showing on the destination device. In this case the caller ID will still show in the CDR even though a separate header has triggered the filter.

[[File:CIDFiltering204222026.png|none|600px]]

Next, select the DID to which you want to apply these filtering rules. You can select "All DIDs" to assign this rule to ALL your DIDs or just "select" some. If you have many DIDs, you can also use the search box to find a specific number.

[[File:Sel_did_filter2.jpg|none|600px]]

To create a whitelist or blacklist group, you will first need to have entered the numbers you want to allow/block to your [[Phone book]], and create a Phone book group. You can then select whether you want the filter to be applied to all the entries on your phone book, or only to a specific group in the phone book.

For the first 4 options you can create a filter and edit it if required. The last option would give you a higher flexibility if you make good use of the '''Wildcards'''.

==Use of Wildcards (Optional)==
X - Matches any digit at the specific location in the number.
* - Matches any number of digits and any digit.

For example, let say that the callerID number is 2145550000:
2145550000, 214*, 214XXX0000, 214XXXXXXX and 214XXX00* are examples that match CallerID 2145550000
214XXX7* and 214XXX are examples that do NOT match the CallerID.

After that, select if you want to apply the filter to all your DID numbers or to a specific number only.

[[File:FilterRouting1.jpg|thumb|none|600px]]

The next options allow you to change the routing of the incoming call. You can route the call to a specific account, an [[Digital Receptionist (IVR)|IVR]], a [[Calling Queues]], [[Time Conditions]], etc. If you click on the '''Show Failover Options''' button, you will be able to choose the routing the call will take if the original destination is busy, unreachable or does not answer. You can also leave a note for the filter you're creating.

=Manage existing Caller ID Filtering Rules =

Once you have created your Filter, it will appear on the bottom of the CallerID Filtering Page. In there, you will be able to search for the filters you have created as well as edit, sort or delete them.

Additionally, you can also bulk delete entries that you would no longer want to filter by either selecting the entries you wish to remove, or simply deleting all of the entries in your callerID Filtering.

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

= Caller ID Names considered as 'Anonymous' =

- Besides the specific "Anonymous" caller ID, the following words are also considered as 'Anonymous' by the Caller ID Filtering Feature:
* 'Anonymous'
* 'Private'
* 'Restricted'
* 'Unknown'
* 'Unavailable'

= Usage Samples =
Here are a few samples of what you can achieve with this feature.

- Your business DID is configured to route to your receptionist's SIP phone. You would like to receive the calls directly on your Cell Phone when the CallerID matches one of your important clients. Simply create a [[Call Forwarding]] entry with your cellphone, then create a CallerID Filtering rule with the CallerID of your client and select your Cell Phone forwarding entry as the routing.

- You have a local number with the 214 area code and do not want Callers from this area code to dial your toll-free number but instead want to playback a message to those callers indicating the local number to call, you can create a Filter with 214* and redirect routing to a pre-recorded message you have uploaded to the system.

= Reseller Configuration =

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

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

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

= CallerID Filtering using the Reseller Interface =

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

Go under the navigation bar on '''[Reseller]''' then click on '''[Manage Rates & Packages]'''
: [[File:CallerIDFiltering_Reseller_1.png|thumb|none|300px]]

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

: [[File:CallerIDFiltering_Reseller_2.png|thumb|none|700px]]

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

: [[File:CallerIDFiltering_Reseller_3.png|thumb|none|700px]]

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

: [[File:CallerIDFiltering_Reseller_4.png|thumb|none|500px]]

1) To add a new CallerID Filtering for your client, or to help your client adding one. Go under the '''[Services]''' at the left navigation bar, then on '''[CallerID Filtering]'''.

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

2) Once on the page, click on '''[Add new CallerID Filtering]''' tab.
You will need to enter some basic information, such as the type of filter, to which DID you would like to apply this filter, a note, and in the '''"Routing"''' tab, where you would like to route this filter.

[[File:CallerIDFiltering_Add_2.png|thumb|none|600px]]

3) Click '''[Save CallerID Filtering]'''

Your CallerID Filtering has been created successfully...

[[category:guides]]

File:CIDFiltering204222026.png

2026-04-22T16:39:32Z

RP:

File:CIDFiltering104222026.png

2026-04-22T16:38:33Z

RP:

CallerID Filtering

2026-04-22T16:35:43Z

RP:

{| class="wikitable"
|+
|-
! Article en Français !! Artículo en Español
|-
| [https://wiki.voip.ms/article/Identification_de_l%27appelant Français] ||
[https://wiki.voip.ms/article/Filtro_de_Llamadas_(CallerID_Filtering) Español]
|}

This feature allows you to filter incoming calls to your DID numbers to route differently than the current DID routing that come from specific numbers, 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 routing options.
Exact matches will be given priority in the filters that you create, followed by Wildcards and then in priority from top to bottom on the existing CallerID Filtering Rules list.

__TOC__

= Create a Filtering entry =

First you need to go your Customer Portal and click on the "DID Numbers" >> "[https://www.voip.ms/m/callerid_filtering.php CallerID Filtering]" menu option.

[[File:CID filtering1.jpg|thumb|none|600px]]

The first thing you need to set is the type of filter you're going to create. You can choose between:
- Anonymous CallerID (This includes 'anonymous','private','restricted','unknown','unavailable','0000000000' & 'id' appearing in the SIP Privacy)
- CallerID not matching the North American NPANXXXXXX format ''(Could block International Calls)''
- All Phone Book
- Phone Book Group
- Specific CallerID Number
- STIR/SHAKEN Attestion Level which will allow you to block all calls being of a specific attestation or lower.

'''Note:''' Attestation A is the highest and most trustworthy calls. While attestation B might be spam and Attestation C and lower (no attestation) are most likely spam.
Do note that even though attestation A is considered trustworthy, there is still a chance that spam calls get signed '''A'''.

'''Note:''' Calls may be flagged as anonymous if they have a SIP Privacy header set to the value id. The purpose of this header is to allow the call to traverse networks where a valid Caller ID is required, while
restricting the Caller ID from showing on the destination device. In this case the caller ID will still show in the CDR even though a separate header has triggered the filter.

[[File:Sel_type_filter1.png|none|600px]]

Next, select the DID to which you want to apply these filtering rules. You can select "All DIDs" to assign this rule to ALL your DIDs or just "select" some. If you have many DIDs, you can also use the search box to find a specific number.

[[File:Sel_did_filter2.jpg|none|600px]]

To create a whitelist or blacklist group, you will first need to have entered the numbers you want to allow/block to your [[Phone book]], and create a Phone book group. You can then select whether you want the filter to be applied to all the entries on your phone book, or only to a specific group in the phone book.

For the first 4 options you can create a filter and edit it if required. The last option would give you a higher flexibility if you make good use of the '''Wildcards'''.

==Use of Wildcards (Optional)==
X - Matches any digit at the specific location in the number.
* - Matches any number of digits and any digit.

For example, let say that the callerID number is 2145550000:
2145550000, 214*, 214XXX0000, 214XXXXXXX and 214XXX00* are examples that match CallerID 2145550000
214XXX7* and 214XXX are examples that do NOT match the CallerID.

After that, select if you want to apply the filter to all your DID numbers or to a specific number only.

[[File:FilterRouting1.jpg|thumb|none|600px]]

The next options allow you to change the routing of the incoming call. You can route the call to a specific account, an [[Digital Receptionist (IVR)|IVR]], a [[Calling Queues]], [[Time Conditions]], etc. If you click on the '''Show Failover Options''' button, you will be able to choose the routing the call will take if the original destination is busy, unreachable or does not answer. You can also leave a note for the filter you're creating.

=Manage existing Caller ID Filtering Rules =

Once you have created your Filter, it will appear on the bottom of the CallerID Filtering Page. In there, you will be able to search for the filters you have created as well as edit, sort or delete them.

Additionally, you can also bulk delete entries that you would no longer want to filter by either selecting the entries you wish to remove, or simply deleting all of the entries in your callerID Filtering.

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

= Caller ID Names considered as 'Anonymous' =

- Besides the specific "Anonymous" caller ID, the following words are also considered as 'Anonymous' by the Caller ID Filtering Feature:
* 'Anonymous'
* 'Private'
* 'Restricted'
* 'Unknown'
* 'Unavailable'

= Usage Samples =
Here are a few samples of what you can achieve with this feature.

- Your business DID is configured to route to your receptionist's SIP phone. You would like to receive the calls directly on your Cell Phone when the CallerID matches one of your important clients. Simply create a [[Call Forwarding]] entry with your cellphone, then create a CallerID Filtering rule with the CallerID of your client and select your Cell Phone forwarding entry as the routing.

- You have a local number with the 214 area code and do not want Callers from this area code to dial your toll-free number but instead want to playback a message to those callers indicating the local number to call, you can create a Filter with 214* and redirect routing to a pre-recorded message you have uploaded to the system.

= Reseller Configuration =

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

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

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

= CallerID Filtering using the Reseller Interface =

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

Go under the navigation bar on '''[Reseller]''' then click on '''[Manage Rates & Packages]'''
: [[File:CallerIDFiltering_Reseller_1.png|thumb|none|300px]]

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

: [[File:CallerIDFiltering_Reseller_2.png|thumb|none|700px]]

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

: [[File:CallerIDFiltering_Reseller_3.png|thumb|none|700px]]

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

: [[File:CallerIDFiltering_Reseller_4.png|thumb|none|500px]]

1) To add a new CallerID Filtering for your client, or to help your client adding one. Go under the '''[Services]''' at the left navigation bar, then on '''[CallerID Filtering]'''.

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

2) Once on the page, click on '''[Add new CallerID Filtering]''' tab.
You will need to enter some basic information, such as the type of filter, to which DID you would like to apply this filter, a note, and in the '''"Routing"''' tab, where you would like to route this filter.

[[File:CallerIDFiltering_Add_2.png|thumb|none|600px]]

3) Click '''[Save CallerID Filtering]'''

Your CallerID Filtering has been created successfully...

[[category:guides]]

Grandstream HandyTone 802 - HT802

2026-03-06T17:32:58Z

RP:

[[File:HT802 Device.jpg|300px|thumb|left]]

The Grandstream HandyTone 802 is a reliable, inexpensive telephone adapter which works with the VoIP.ms service when placed after your broadband internet router.

'''Websites:''' [https://www.grandstream.com/products/gateways-and-atas/analog-telephone-adaptors/product/ht802 Grandstream HT802]

'''Help / Support:''' [http://www.grandstream.com/support Grandstream Support]

<br clear="all" />

__TOC__

----

===Configuring the HandyTone 802===

'''Step 1 - Initial Configurations'''<br>

These instructions are based on HandyTone 802 software version 1.0.3.2 if you are running a different software version some menus and settings may be different.

These instructions are also based on using the HandyTone in its factory default configuration, which obtains a dynamic IP address automatically from your router using DHCP. For information on configuring your HandyTone with a Static IP Address, please refer to the HandyTone user´s manual.

Each step is important in assuring that your device works properly.

''We recommend that you read each step through in its entirety before performing the action indicated in the step.''

'''Step #2 - Plugging the HT802'''<br>

Connect your HandyTone to your router with the supplied Ethernet network cable.

Now connect your phone to the HandyTone. Plugging the cable into the correct FXS Port that you configure.

'''''In most cases, you will only be setting yourself up with FXS PORT 1, so please make sure your telephone is connected to LINE #1.'''''

Finally plug the supplied power cable into the HandyTone.

'''Step #3 - Getting IP address for the GUI'''<br>

Wait 60 seconds after plugging your HT802 in.
Pick up the phone connected to the HT802 and dial *** on it.

Please have a pen and paper ready. You will hear a message - "Enter a menu option", then enter 0 2 on your phone. You will now hear a message giving you the IP address of your HT802 such as - "192.168.001.010" and write this number down.

Open a web browser on your computer such as Chrome or Firefox and enter the IP address you heard in step 4 as the address (I.E. where you would normally enter www.voip.ms).

Please note: Some browsers will require you to remove leading zero's ( 0 's ) in the IP address. For example if you heard "192.168.001.010" you should change this to "''192.168.1.10''".

The Interface has a timeout so please make changes quickly or save/update your settings every couple of minutes.

'''Step #4 - Logging into the device'''<br>

You should now see a page that looks like this:

[[File:HT802 Login.jpg|thumb|none|600px]]

Enter the password for the HT802 in the password field. The default administrator password for the HT802 is '''''admin'''''

After entering the password you should see a screen that looks similar to the one below:

[[File:HT802_FirstPage.PNG|thumb|none|600px]]

'''Step #5 - Configuring device's port FXS'''<br>

Now, click on FXS PORT1 and configure your settings accordingly (as shown below).

* '''Primary SIP Server:''' toronto.voip.ms (Pick one of VoIP.ms multiple [http://wiki.voip.ms/article/Choosing_Server VoIP Servers])
* '''Prefer Primary SIP Server:''' Set to Yes
*'''Outbound Proxy:''' Set the same server you've configured at the '''Primary SIP Server''' field
*'''NAT Traversal:''' Keep-Alive
* '''SIP User ID:''' Replace with your Main SIP account or Subaccount UserID, e.g. 100000 or 100000_sub
*'''Authenticate ID:''' Replace with your Main SIP account or Subaccount UserID, e.g. 100000 or 100000_sub
* '''Authenticate Password:''' ********* (Use your SIP account password - by default this is the same as the Customer Portal)
* '''''Name''''': Outbound CallerID Name (Optional)* '''See the requirements below.'''
*'''DNS Mode:''' Set to "A Record"
* '''SIP Registration:''' Set to Yes
* '''Unregister On Reboot:''' Set to No
* '''Outgoing Call Without Registration:''' Set to Yes
* '''Register Expiration:''' Set to 5
* '''Local RTP Port:''' Set to 10000
* '''''Enable SIP OPTIONS/NOTIFY Keep Alive''''': OPTIONS
* '''Allow Incoming SIP Messages from SIP Proxy Only:''' Set to Yes
* '''Preferred DTMF method:''' In-audio, RFC2833
* '''''Use P-Access-Network-Info Header (if present)''''': Set to No
* '''''Use P-Emergency-info Header (if present)''''': Set to No
* '''Enable Call Features:''' No
* '''Dial Plan:''' {[x*]+}
*'''Preferred Vocoder:''' PCMU, PCMA, G729

'''IMPORTANT''': Outbound CallerID Name
- We suggest entering your outbound Caller ID Name must be in '''capital letters'''. This will appears more clearly/visible on some devices.
- You must '''NOT''' use any special characters, they will not be displayed.
- Some of regular Canadian providers will not show more than '''15 characters max'''. We suggest shrinking or adapt your caller ID.
- Spaces are allowed in a caller id name.

[[File:HT802_FXS_Port.jpg|thumb|none|600px]]

'''Step 6 - Savings the changes'''<br>

Once you have configured the settings above, click the '''Update''' button and then the '''Reboot''' button to save the configurations. Your HT802 will power cycle after you click the reboot button. Please wait at least 30 seconds for the unit to finish power cycling. If you see that the Phone 1 LED (or phone 2 LED, depending on which FXS port you've configured our service for) is a solid blue color, then your unit is configured and ready to make calls.

'''That's it!''' You can now make a phone call.

The area code + the number for calls to the US & Canada

Or

011 + the country code, area code, and number for calls worldwide (you may also use 00 instead of 011).

== Call Encryption - TLS/SRTP ==

To use [[Call_Encryption_-_TLS/SRTP#Configuration_on_SIP_Client | encrypted calls (TLS)]], the following setting on the FXS Port must be changed:

* '''''SIP Transport''''': TLS
* '''''SRTP Mode''''': Enabled and forced

Also make sure '''''Local SIP Port''''' is now at “5061”.

== Preventing Direct IP calls like 100 & 1000 ==

To Prevent Direct IP calls to your device and only allow calls from our service please enable the following 2 options in your FXS Port Configuration Page.

'''Check SIP User ID for incoming INVITE''' - Default is No. Check the incoming SIP User ID in Request URI. If they don’t match, the call will be rejected. If this option is enabled, the device will not be able to make direct IP calls.

'''Allow Incoming SIP Messages from SIP Proxy Only''' - Default is No. Check the incoming SIP messages. If they don’t come from the SIP proxy, they will be rejected. If this option is enabled, the device will not be able to make direct IP calls.

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

User Manual : [http://www.grandstream.com/sites/default/files/Resources/ht80x_user_guide.pdf Download PDF]

Admin Manual : [http://www.grandstream.com/sites/default/files/Resources/ht80x_administration_guide.pdf Download PDF]

[[Category:Analog Telephone Adapters]]

ATA Devices

2026-03-05T02:05:12Z

RP: /* ReadyNet AC1000MS and AC1300MS */

File:Readynet-ac1000mslogo.png

2026-03-05T02:04:57Z

RP:

PBXs

2026-03-04T23:53:22Z

RP: /* FusionPBX */

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

An acronym for '''Private Branch eXchange'''. PBX telephone systems support incoming calls from the outside PSTN, placing calls between 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) - Usually don't support IP at all or support it only with expensive add-on equipment
* '''Converged''' (also known as hybrid) - Support IP and PSTN connections with equal force. Most flexible and cost-effective model
* '''IP-PBX''' - Support only IP connectivity. Any PSTN connectivity must be achieved through external converters (Gateways)

'''Learn More:''' [https://wiki.voip.ms/article/Back_to_Basics_%E2%80%93_What_is_a_PBX%3F Back to Basics - What is a PBX?]
|}

__TOC__

== 3CX Phone System ==

{| cellspacing="0" style="width: 100%; background: white; border: 1px solid #e2e8f0; border-radius: 12px; padding: 25px; margin: 20px 0; box-shadow: 0 2px 10px rgba(0,0,0,0.06);"
|-
| style="width: 300px; vertical-align: top; padding-right: 25px;" |
[[File:3CX Logo.jpg|300px|thumb|center|3CX Phone System]]

| style="vertical-align: top;" |
<div style="font-size: 1.4em; font-weight: 600; color: #1e293b; margin-bottom: 15px;">3CX Phone System</div>

'''Type:''' Software-based IP PBX<br>
'''Platform:''' Open Standards<br>
'''Overview:''' 3CX is a software-based, open standards IP PBX that offers complete Unified Communications, out of the box. 3CX makes installation, management, and maintenance of your PBX so easy that you can effortlessly manage it yourself, whether on an appliance, on your servers or in your cloud account.

<div style="margin-top: 20px;">
[[3CX Phone System|<span style="background: linear-gradient(135deg, #d0382d, #b8261a); color: white; padding: 12px 24px; border-radius: 20px; text-decoration: none; font-weight: 600; display: inline-block; box-shadow: 0 2px 10px rgba(208, 56, 45, 0.3);">General Configuration</span>]]
  
[[3CX_Phone_System_-_v20|<span style="background: linear-gradient(135deg, #d0382d, #b8261a); color: white; padding: 12px 24px; border-radius: 20px; text-decoration: none; font-weight: 600; display: inline-block; box-shadow: 0 2px 10px rgba(208, 56, 45, 0.3);">V20 Configuration</span>]]
</div>
|}

== 3CX StartUP ==

{| cellspacing="0" style="width: 100%; background: white; border: 1px solid #e2e8f0; border-radius: 12px; padding: 25px; margin: 20px 0; box-shadow: 0 2px 10px rgba(0,0,0,0.06);"
|-
| style="width: 300px; vertical-align: top; padding-right: 25px;" |
[[File:3CX_startup_0.png|300px|thumb|center|3CX StartUP]]

| style="vertical-align: top;" |
<div style="font-size: 1.4em; font-weight: 600; color: #1e293b; margin-bottom: 15px;">3CX StartUP</div>

'''Type:''' Hosted Solution<br>
'''Capacity:''' Up to 20-30 users<br>
'''Overview:''' Preconfigured, ready-to-go shared instance. Recommended for most small businesses. StartUP is only available hosted by 3CX.

'''Sign up for your free 3CX business comms system:''' [https://www.3cx.com/signup/ here]

<div style="margin-top: 20px;">
[[3CX StartUP|<span style="background: linear-gradient(135deg, #d0382d, #b8261a); color: white; padding: 12px 24px; border-radius: 20px; text-decoration: none; font-weight: 600; display: inline-block; box-shadow: 0 2px 10px rgba(208, 56, 45, 0.3);">Configuration</span>]]
</div>
|}

== Asterisk ==

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

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

'''Type:''' Open Source PBX<br>
'''Platform:''' Linux/UNIX<br>
'''Created:''' 1999<br>
'''Overview:''' Asterisk is a telephone private branch exchange (PBX), created as open software for Linux and other UNIX-like systems. Like other private branch exchanges, it allows attached telephones to make calls to one another and provides connections to outside lines.

'''Connection Methods to VoIP.ms:'''
* [[Asterisk IAX2|Asterisk (IAX2)]] - Inter-Asterisk protocol
* [[Asterisk SIP|Asterisk (SIP)]] - Standard Session Initiation Protocol
* [[Asterisk PJSIP|Asterisk (PJSIP)]] - Open Source Embedded SIP protocol stack

'''Resources:'''
* [http://www.asteriskdocs.org Free HTML documentation]
* [http://www.asterisk.org Official Asterisk website]

<div style="margin-top: 20px;">
[[Asterisk IAX2|<span style="background: linear-gradient(135deg, #d0382d, #b8261a); color: white; padding: 10px 20px; border-radius: 20px; text-decoration: none; font-weight: 600; display: inline-block; box-shadow: 0 2px 10px rgba(208, 56, 45, 0.3); margin: 5px;">IAX2 Config</span>]]
[[Asterisk SIP|<span style="background: linear-gradient(135deg, #d0382d, #b8261a); color: white; padding: 10px 20px; border-radius: 20px; text-decoration: none; font-weight: 600; display: inline-block; box-shadow: 0 2px 10px rgba(208, 56, 45, 0.3); margin: 5px;">SIP Config</span>]]
[[Asterisk PJSIP|<span style="background: linear-gradient(135deg, #d0382d, #b8261a); color: white; padding: 10px 20px; border-radius: 20px; text-decoration: none; font-weight: 600; display: inline-block; box-shadow: 0 2px 10px rgba(208, 56, 45, 0.3); margin: 5px;">PJSIP Config</span>]]
</div>
|}

== Avaya IP Office ==

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

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

'''Type:''' Enterprise PBX<br>
'''Target:''' Small and Medium Enterprises<br>
'''Overview:''' Avaya is a leading global provider of next-generation business collaboration and communications solutions, providing unified communications, real-time video collaboration, contact center, networking and related services to companies of all sizes around the world.

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

== Cisco IOS ==

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

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

'''Type:''' Router/Switch Operating System<br>
'''Platform:''' Cisco Hardware<br>
'''Overview:''' Cisco IOS (originally Internetwork Operating System) is software used on most Cisco Systems routers and current Cisco network switches. IOS is a package of routing, switching, internetworking and telecommunications functions integrated into a multitasking operating system.

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

== E-MetroTel ==

{| cellspacing="0" style="width: 100%; background: white; border: 1px solid #e2e8f0; border-radius: 12px; padding: 25px; margin: 20px 0; box-shadow: 0 2px 10px rgba(0,0,0,0.06);"
|-
| style="width: 300px; vertical-align: top; padding-right: 25px;" |
[[File:EMT Logo new nortel 7 1 19b (2).png|300px|thumb|center|E-MetroTel - Exceptional Innovation]]

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

'''Type:''' Unified Business Communications<br>
'''Target:''' Global client base<br>
'''Overview:''' E-MetroTel provides unified business communications systems through leading-edge telephony hardware and software. Delivers streamlined migration to complete solutions that tightly integrate digital, IP, analog, and Unified Communications capabilities.

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

== Epygi PBX ==

{| cellspacing="0" style="width: 100%; background: white; border: 1px solid #e2e8f0; border-radius: 12px; padding: 25px; margin: 20px 0; box-shadow: 0 2px 10px rgba(0,0,0,0.06);"
|-
| style="width: 300px; vertical-align: top; padding-right: 25px;" |
[[File:Epygi_logo_300DPI.png|300px|thumb|center|Epygi PBX]]

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

'''Type:''' IP PBX Appliance<br>
'''Experience:''' 18+ years<br>
'''Overview:''' For over 18 years Epygi has been designing, manufacturing, and delivering IP PBX and IP Gateway appliances. They serve customers ranging from small businesses to enterprise companies, each benefiting from Epygi's excellent value and service.

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

== FreePBX / PBX in a Flash ==

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

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

'''Type:''' Open Source Web-based PBX<br>
'''Platform:''' Asterisk-based<br>
'''Overview:''' 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.

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

== FreeSwitch ==

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

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

'''Type:''' Open Source Telephony Platform<br>
'''Created:''' 2006<br>
'''Overview:''' 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. Created to fill the void left by proprietary commercial solutions.

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

== FusionPBX ==

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

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

'''Type:''' Open Source IP PBX<br>
'''Platform:''' FreeSWITCH-based<br>
'''Overview:''' FusionPBX is a full-featured domain based multi-tenant PBX and voice switch for FreeSWITCH. It provides an easy to use web interface for FreeSWITCH configuration and management. FusionPBX provides unlimited extensions, voicemail-to-email, music on hold, call parking, call queues and many other features.

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

== Grandstream UCM 6200 ==

{| cellspacing="0" style="width: 100%; background: white; border: 1px solid #e2e8f0; border-radius: 12px; padding: 25px; margin: 20px 0; box-shadow: 0 2px 10px rgba(0,0,0,0.06);"
|-
| style="width: 300px; vertical-align: top; padding-right: 25px;" |
[[File:Grandstream-Logo-2018.png|300px|thumb|center|Grandstream]]

| style="vertical-align: top;" |
<div style="font-size: 1.4em; font-weight: 600; color: #1e293b; margin-bottom: 15px;">Grandstream UCM 6200</div>

'''Type:''' Appliance-based IP PBX<br>
'''Target:''' SMB and Enterprise<br>
'''Overview:''' Grandstream Networks has been connecting the world since 2002 with SIP Unified Communications solutions. Award-winning solutions serve the small and medium business and enterprises markets, recognized for quality, reliability and innovation.

<div style="margin-top: 20px;">
[[Grandstream UCM6200|<span style="background: linear-gradient(135deg, #d0382d, #b8261a); color: white; padding: 12px 24px; border-radius: 20px; text-decoration: none; font-weight: 600; display: inline-block; box-shadow: 0 2px 10px rgba(208, 56, 45, 0.3);">Configuration</span>]]
</div>
|}

== Incredible PBX 2027 ==

{| cellspacing="0" style="width: 100%; background: white; border: 1px solid #e2e8f0; border-radius: 12px; padding: 25px; margin: 20px 0; box-shadow: 0 2px 10px rgba(0,0,0,0.06);"
|-
| style="width: 300px; vertical-align: top; padding-right: 25px;" |
[[File:IncrediblePBX-VoIPms.jpg|300px|thumb|center|Incredible PBX 2027]]

| style="vertical-align: top;" |
<div style="font-size: 1.4em; font-weight: 600; color: #1e293b; margin-bottom: 15px;">Incredible PBX 2027</div>

'''Type:''' Open Source Unified Communications<br>
'''Platform:''' Multi-platform (CentOS, Rocky, Debian, Ubuntu, Raspbian)<br>
'''Overview:''' GPL aggregation which bundles industry-leading components to provide a production-ready, turnkey unified communications VoIP platform with support for SIP and IAX. Includes Apache, PHP, MariaDB/MySQL, and latest releases of Asterisk and FreePBX.

<div style="margin-top: 20px;">
[[Incredible_PBX_2027|<span style="background: linear-gradient(135deg, #d0382d, #b8261a); color: white; padding: 12px 24px; border-radius: 20px; text-decoration: none; font-weight: 600; display: inline-block; box-shadow: 0 2px 10px rgba(208, 56, 45, 0.3);">Configuration</span>]]
</div>
|}

== PhoneSuite Systems ==

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

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

'''Type:''' Hotel Communication Solutions<br>
'''Industry:''' Hospitality<br>
'''Overview:''' PhoneSuite is a leading provider of hotel voice communication solutions for over 25 years. Leverages expertise in communication technology to provide high-quality, low-power consuming products designed exclusively for the hotel industry.

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

== PortSIP ==

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

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

'''Type:''' All-in-one Communications Platform<br>
'''Features:''' Voice, Video, Messaging, SMS, WhatsApp, Meetings<br>
'''Overview:''' PortSIP is an all-in-one communications platform designed to meet the needs of modern businesses. Offering a comprehensive suite of tools for voice, video, messaging, SMS, WhatsApp, video meetings, contact centers, and team collaboration.

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

== Positron Telecom Systems ==

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

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

'''Type:''' IP PBX Solutions<br>
'''Target:''' Small and Medium Businesses<br>
'''Overview:''' Positron IP PBX solutions offer small and medium sized businesses powerful VoIP phone systems that combine voice and data into one easy-to-use device, also known as Unified Communications. Features traditionally only available to large companies.

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

== ScopServ ScopTEL IPPBX ==

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

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

'''Type:''' IP PBX System<br>
'''Company:''' ScopServ International Inc.<br>
'''Overview:''' ScopServ International Inc. is a global pioneer in IP telephony solutions design (VoIP). The integrated system SCOPTEL IPBX provides one of the most stable environments and richest on the market.

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

== Synway ==

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

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

'''Type:''' Multimedia Gateway<br>
'''Location:''' China<br>
'''Overview:''' Synway, as a world-leading VoIP enabling-technologies provider in China, has been specialized in providing superior Multimedia Gateway, Integrated Multimedia Switch, Telephony Hardware in use for Telecom communications.

<div style="margin-top: 20px;">
[[Synway UC200|<span style="background: linear-gradient(135deg, #d0382d, #b8261a); color: white; padding: 12px 24px; border-radius: 20px; text-decoration: none; font-weight: 600; display: inline-block; box-shadow: 0 2px 10px rgba(208, 56, 45, 0.3);">Configuration</span>]]
</div>
|}

== Ubiquiti Unifi Talk ==

{| cellspacing="0" style="width: 100%; background: white; border: 1px solid #e2e8f0; border-radius: 12px; padding: 25px; margin: 20px 0; box-shadow: 0 2px 10px rgba(0,0,0,0.06);"
|-
| style="width: 300px; vertical-align: top; padding-right: 25px;" |
[[File:UbiquitiLogo.png|300px|x200px|thumb|center|Ubiquiti Unifi Talk]]

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

'''Type:''' Cloud Phone Solution<br>
'''Setup Time:''' Minutes<br>
'''Overview:''' Only minutes to set up a reliable, full-featured phone solution. Part of Ubiquiti's comprehensive networking and communications ecosystem.

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

== VitalPBX ==

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

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

'''Type:''' Free Communications System<br>
'''Platform:''' Linux/Asterisk-based<br>
'''Overview:''' VitalPBX is a free telephone and communications system for companies. Complete platform that can be installed on physical hardware on-site or as a hosted application. Acts as the upper layer interface for the Linux base and Asterisk.

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

== Vodia PBX ==

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

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

'''Type:''' Software-based IP PBX<br>
'''Platform:''' Open Standards<br>
'''Overview:''' Vodia PBX is a software-based, open standards IP PBX that offers complete UC functionality right out of the box. Makes installation, management, and maintenance so easy that you can effortlessly manage it yourself.

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

== Wazo ==

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

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

'''Type:''' Unified Communications Platform<br>
'''Target:''' MSPs and UCaaS providers<br>
'''Overview:''' Wazo is a unified communications technology provider that helps MSPs stand out from the crowd and build unique UCaaS experiences with simple, powerful, flexible APIs and integrations. Global reach with entities in Canada, United States, and France.

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

== Wildix ==

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

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

'''Type:''' Unified Communications<br>
'''Focus:''' Communications-Enabled Business Processes<br>
'''Overview:''' Communications-Enabled Business Processes: Wildix Unified Communications solutions enter the business processes to enhance the workplace efficiency and improve the productivity of your team.

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

== XCALLY ==

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

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

'''Type:''' Omni Channel Contact Center<br>
'''Platform:''' Asterisk-based<br>
'''Overview:''' XCALLY is an innovative Omni Channel solution that integrates Asterisk with Shuttle and Motion technologies. One of the best Contact Center management platforms for multi-channel - voice, chat, email, SMS, fax, and custom channels.

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

== Xorcom ==

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

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

'''Type:''' Business Telephony Solutions<br>
'''Specialization:''' Integrated Solutions<br>
'''Overview:''' Xorcom designs and manufactures integrated business telephony solutions that support both traditional PSTN and VoIP, including IP PBX, Hotel Phone Systems, Virtual PBX Systems, and Multi-tenant PBX.

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

== Yeastar ==

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

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

'''Type:''' VoIP PBX Systems<br>
'''Target:''' SMBs<br>
'''Overview:''' Designed for SMBs, Yeastar S-Series/P-Series VoIP PBX and Yeastar Cloud PBX deliver enterprise-grade communications features along with advanced UC capabilities, bringing solid, reliable and affordable voice solutions.

<div style="margin-top: 20px;">
[[Yeastar|<span style="background: linear-gradient(135deg, #d0382d, #b8261a); color: white; padding: 12px 24px; border-radius: 20px; text-decoration: none; font-weight: 600; display: inline-block; box-shadow: 0 2px 10px rgba(208, 56, 45, 0.3); margin: 5px;">S-Series Configuration</span>]]
[[Yeastar_P-Series|<span style="background: linear-gradient(135deg, #d0382d, #b8261a); color: white; padding: 12px 24px; border-radius: 20px; text-decoration: none; font-weight: 600; display: inline-block; box-shadow: 0 2px 10px rgba(208, 56, 45, 0.3); margin: 5px;">P-Series Configuration</span>]]
</div>
|}

== Legacy Systems ==

{| cellspacing="0" style="width: 100%; background: linear-gradient(135deg, #fef3c7 0%, #fde68a 100%); border: 1px solid #f59e0b; border-radius: 8px; padding: 25px; margin: 25px 0; box-shadow: 0 2px 10px rgba(245, 158, 11, 0.1);"
|-
| style="text-align: left;" |
<div style="font-size: 1.1em; font-weight: 600; color: #92400e; margin-bottom: 10px;">⚠️ The following systems are included for reference but are no longer actively maintained or recommended for new deployments:</div>

* '''[[Elastix|Elastix]]''' - Open source unified communications server. Development has slowed significantly in recent years.
* '''[[NortelBCM|Nortel/Avaya BCM 450 and BCM50 R6]]''' - Legacy Nortel phone system acquired by Avaya. End-of-life with limited support.
* '''[[PBXes.org|PBXes.org]]''' - Hosted PBX service with limited features compared to modern cloud solutions.
* '''[[SIPfoundry|SIPfoundry]]''' - Open source project with limited ongoing development.
* '''[[TalkSwitch|TalkSwitch]]''' - Small business phone system with discontinued development and support.
* '''[[Trixbox|Trixbox]]''' - Project has not been maintained since 2010. You might want to look for alternatives.

For new deployments, we recommend choosing from the modern PBX solutions listed above.
|}

[[Category:PBXes]]

PBXs

2026-03-04T23:51:20Z

RP: /* Ubiquiti Unifi Talk */

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

An acronym for '''Private Branch eXchange'''. PBX telephone systems support incoming calls from the outside PSTN, placing calls between 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) - Usually don't support IP at all or support it only with expensive add-on equipment
* '''Converged''' (also known as hybrid) - Support IP and PSTN connections with equal force. Most flexible and cost-effective model
* '''IP-PBX''' - Support only IP connectivity. Any PSTN connectivity must be achieved through external converters (Gateways)

'''Learn More:''' [https://wiki.voip.ms/article/Back_to_Basics_%E2%80%93_What_is_a_PBX%3F Back to Basics - What is a PBX?]
|}

__TOC__

== 3CX Phone System ==

{| cellspacing="0" style="width: 100%; background: white; border: 1px solid #e2e8f0; border-radius: 12px; padding: 25px; margin: 20px 0; box-shadow: 0 2px 10px rgba(0,0,0,0.06);"
|-
| style="width: 300px; vertical-align: top; padding-right: 25px;" |
[[File:3CX Logo.jpg|300px|thumb|center|3CX Phone System]]

| style="vertical-align: top;" |
<div style="font-size: 1.4em; font-weight: 600; color: #1e293b; margin-bottom: 15px;">3CX Phone System</div>

'''Type:''' Software-based IP PBX<br>
'''Platform:''' Open Standards<br>
'''Overview:''' 3CX is a software-based, open standards IP PBX that offers complete Unified Communications, out of the box. 3CX makes installation, management, and maintenance of your PBX so easy that you can effortlessly manage it yourself, whether on an appliance, on your servers or in your cloud account.

<div style="margin-top: 20px;">
[[3CX Phone System|<span style="background: linear-gradient(135deg, #d0382d, #b8261a); color: white; padding: 12px 24px; border-radius: 20px; text-decoration: none; font-weight: 600; display: inline-block; box-shadow: 0 2px 10px rgba(208, 56, 45, 0.3);">General Configuration</span>]]
  
[[3CX_Phone_System_-_v20|<span style="background: linear-gradient(135deg, #d0382d, #b8261a); color: white; padding: 12px 24px; border-radius: 20px; text-decoration: none; font-weight: 600; display: inline-block; box-shadow: 0 2px 10px rgba(208, 56, 45, 0.3);">V20 Configuration</span>]]
</div>
|}

== 3CX StartUP ==

{| cellspacing="0" style="width: 100%; background: white; border: 1px solid #e2e8f0; border-radius: 12px; padding: 25px; margin: 20px 0; box-shadow: 0 2px 10px rgba(0,0,0,0.06);"
|-
| style="width: 300px; vertical-align: top; padding-right: 25px;" |
[[File:3CX_startup_0.png|300px|thumb|center|3CX StartUP]]

| style="vertical-align: top;" |
<div style="font-size: 1.4em; font-weight: 600; color: #1e293b; margin-bottom: 15px;">3CX StartUP</div>

'''Type:''' Hosted Solution<br>
'''Capacity:''' Up to 20-30 users<br>
'''Overview:''' Preconfigured, ready-to-go shared instance. Recommended for most small businesses. StartUP is only available hosted by 3CX.

'''Sign up for your free 3CX business comms system:''' [https://www.3cx.com/signup/ here]

<div style="margin-top: 20px;">
[[3CX StartUP|<span style="background: linear-gradient(135deg, #d0382d, #b8261a); color: white; padding: 12px 24px; border-radius: 20px; text-decoration: none; font-weight: 600; display: inline-block; box-shadow: 0 2px 10px rgba(208, 56, 45, 0.3);">Configuration</span>]]
</div>
|}

== Asterisk ==

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

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

'''Type:''' Open Source PBX<br>
'''Platform:''' Linux/UNIX<br>
'''Created:''' 1999<br>
'''Overview:''' Asterisk is a telephone private branch exchange (PBX), created as open software for Linux and other UNIX-like systems. Like other private branch exchanges, it allows attached telephones to make calls to one another and provides connections to outside lines.

'''Connection Methods to VoIP.ms:'''
* [[Asterisk IAX2|Asterisk (IAX2)]] - Inter-Asterisk protocol
* [[Asterisk SIP|Asterisk (SIP)]] - Standard Session Initiation Protocol
* [[Asterisk PJSIP|Asterisk (PJSIP)]] - Open Source Embedded SIP protocol stack

'''Resources:'''
* [http://www.asteriskdocs.org Free HTML documentation]
* [http://www.asterisk.org Official Asterisk website]

<div style="margin-top: 20px;">
[[Asterisk IAX2|<span style="background: linear-gradient(135deg, #d0382d, #b8261a); color: white; padding: 10px 20px; border-radius: 20px; text-decoration: none; font-weight: 600; display: inline-block; box-shadow: 0 2px 10px rgba(208, 56, 45, 0.3); margin: 5px;">IAX2 Config</span>]]
[[Asterisk SIP|<span style="background: linear-gradient(135deg, #d0382d, #b8261a); color: white; padding: 10px 20px; border-radius: 20px; text-decoration: none; font-weight: 600; display: inline-block; box-shadow: 0 2px 10px rgba(208, 56, 45, 0.3); margin: 5px;">SIP Config</span>]]
[[Asterisk PJSIP|<span style="background: linear-gradient(135deg, #d0382d, #b8261a); color: white; padding: 10px 20px; border-radius: 20px; text-decoration: none; font-weight: 600; display: inline-block; box-shadow: 0 2px 10px rgba(208, 56, 45, 0.3); margin: 5px;">PJSIP Config</span>]]
</div>
|}

== Avaya IP Office ==

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

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

'''Type:''' Enterprise PBX<br>
'''Target:''' Small and Medium Enterprises<br>
'''Overview:''' Avaya is a leading global provider of next-generation business collaboration and communications solutions, providing unified communications, real-time video collaboration, contact center, networking and related services to companies of all sizes around the world.

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

== Cisco IOS ==

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

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

'''Type:''' Router/Switch Operating System<br>
'''Platform:''' Cisco Hardware<br>
'''Overview:''' Cisco IOS (originally Internetwork Operating System) is software used on most Cisco Systems routers and current Cisco network switches. IOS is a package of routing, switching, internetworking and telecommunications functions integrated into a multitasking operating system.

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

== E-MetroTel ==

{| cellspacing="0" style="width: 100%; background: white; border: 1px solid #e2e8f0; border-radius: 12px; padding: 25px; margin: 20px 0; box-shadow: 0 2px 10px rgba(0,0,0,0.06);"
|-
| style="width: 300px; vertical-align: top; padding-right: 25px;" |
[[File:EMT Logo new nortel 7 1 19b (2).png|300px|thumb|center|E-MetroTel - Exceptional Innovation]]

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

'''Type:''' Unified Business Communications<br>
'''Target:''' Global client base<br>
'''Overview:''' E-MetroTel provides unified business communications systems through leading-edge telephony hardware and software. Delivers streamlined migration to complete solutions that tightly integrate digital, IP, analog, and Unified Communications capabilities.

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

== Epygi PBX ==

{| cellspacing="0" style="width: 100%; background: white; border: 1px solid #e2e8f0; border-radius: 12px; padding: 25px; margin: 20px 0; box-shadow: 0 2px 10px rgba(0,0,0,0.06);"
|-
| style="width: 300px; vertical-align: top; padding-right: 25px;" |
[[File:Epygi_logo_300DPI.png|300px|thumb|center|Epygi PBX]]

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

'''Type:''' IP PBX Appliance<br>
'''Experience:''' 18+ years<br>
'''Overview:''' For over 18 years Epygi has been designing, manufacturing, and delivering IP PBX and IP Gateway appliances. They serve customers ranging from small businesses to enterprise companies, each benefiting from Epygi's excellent value and service.

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

== FreePBX / PBX in a Flash ==

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

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

'''Type:''' Open Source Web-based PBX<br>
'''Platform:''' Asterisk-based<br>
'''Overview:''' 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.

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

== FreeSwitch ==

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

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

'''Type:''' Open Source Telephony Platform<br>
'''Created:''' 2006<br>
'''Overview:''' 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. Created to fill the void left by proprietary commercial solutions.

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

== FusionPBX ==

{| cellspacing="0" style="width: 100%; background: white; border: 1px solid #e2e8f0; border-radius: 12px; padding: 25px; margin: 20px 0; box-shadow: 0 2px 10px rgba(0,0,0,0.06);"
|-
| style="width: 300px; vertical-align: top; padding-right: 25px;" |
[[File:FusionPBXlogo.png|300px|thumb|center|3CX Phone System]]

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

'''Type:''' Open Source IP PBX<br>
'''Platform:''' FreeSWITCH-based<br>
'''Overview:''' FusionPBX is a full-featured domain based multi-tenant PBX and voice switch for FreeSWITCH. It provides an easy to use web interface for FreeSWITCH configuration and management. FusionPBX provides unlimited extensions, voicemail-to-email, music on hold, call parking, call queues and many other features.

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

== Grandstream UCM 6200 ==

{| cellspacing="0" style="width: 100%; background: white; border: 1px solid #e2e8f0; border-radius: 12px; padding: 25px; margin: 20px 0; box-shadow: 0 2px 10px rgba(0,0,0,0.06);"
|-
| style="width: 300px; vertical-align: top; padding-right: 25px;" |
[[File:Grandstream-Logo-2018.png|300px|thumb|center|Grandstream]]

| style="vertical-align: top;" |
<div style="font-size: 1.4em; font-weight: 600; color: #1e293b; margin-bottom: 15px;">Grandstream UCM 6200</div>

'''Type:''' Appliance-based IP PBX<br>
'''Target:''' SMB and Enterprise<br>
'''Overview:''' Grandstream Networks has been connecting the world since 2002 with SIP Unified Communications solutions. Award-winning solutions serve the small and medium business and enterprises markets, recognized for quality, reliability and innovation.

<div style="margin-top: 20px;">
[[Grandstream UCM6200|<span style="background: linear-gradient(135deg, #d0382d, #b8261a); color: white; padding: 12px 24px; border-radius: 20px; text-decoration: none; font-weight: 600; display: inline-block; box-shadow: 0 2px 10px rgba(208, 56, 45, 0.3);">Configuration</span>]]
</div>
|}

== Incredible PBX 2027 ==

{| cellspacing="0" style="width: 100%; background: white; border: 1px solid #e2e8f0; border-radius: 12px; padding: 25px; margin: 20px 0; box-shadow: 0 2px 10px rgba(0,0,0,0.06);"
|-
| style="width: 300px; vertical-align: top; padding-right: 25px;" |
[[File:IncrediblePBX-VoIPms.jpg|300px|thumb|center|Incredible PBX 2027]]

| style="vertical-align: top;" |
<div style="font-size: 1.4em; font-weight: 600; color: #1e293b; margin-bottom: 15px;">Incredible PBX 2027</div>

'''Type:''' Open Source Unified Communications<br>
'''Platform:''' Multi-platform (CentOS, Rocky, Debian, Ubuntu, Raspbian)<br>
'''Overview:''' GPL aggregation which bundles industry-leading components to provide a production-ready, turnkey unified communications VoIP platform with support for SIP and IAX. Includes Apache, PHP, MariaDB/MySQL, and latest releases of Asterisk and FreePBX.

<div style="margin-top: 20px;">
[[Incredible_PBX_2027|<span style="background: linear-gradient(135deg, #d0382d, #b8261a); color: white; padding: 12px 24px; border-radius: 20px; text-decoration: none; font-weight: 600; display: inline-block; box-shadow: 0 2px 10px rgba(208, 56, 45, 0.3);">Configuration</span>]]
</div>
|}

== PhoneSuite Systems ==

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

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

'''Type:''' Hotel Communication Solutions<br>
'''Industry:''' Hospitality<br>
'''Overview:''' PhoneSuite is a leading provider of hotel voice communication solutions for over 25 years. Leverages expertise in communication technology to provide high-quality, low-power consuming products designed exclusively for the hotel industry.

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

== PortSIP ==

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

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

'''Type:''' All-in-one Communications Platform<br>
'''Features:''' Voice, Video, Messaging, SMS, WhatsApp, Meetings<br>
'''Overview:''' PortSIP is an all-in-one communications platform designed to meet the needs of modern businesses. Offering a comprehensive suite of tools for voice, video, messaging, SMS, WhatsApp, video meetings, contact centers, and team collaboration.

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

== Positron Telecom Systems ==

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

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

'''Type:''' IP PBX Solutions<br>
'''Target:''' Small and Medium Businesses<br>
'''Overview:''' Positron IP PBX solutions offer small and medium sized businesses powerful VoIP phone systems that combine voice and data into one easy-to-use device, also known as Unified Communications. Features traditionally only available to large companies.

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

== ScopServ ScopTEL IPPBX ==

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

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

'''Type:''' IP PBX System<br>
'''Company:''' ScopServ International Inc.<br>
'''Overview:''' ScopServ International Inc. is a global pioneer in IP telephony solutions design (VoIP). The integrated system SCOPTEL IPBX provides one of the most stable environments and richest on the market.

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

== Synway ==

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

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

'''Type:''' Multimedia Gateway<br>
'''Location:''' China<br>
'''Overview:''' Synway, as a world-leading VoIP enabling-technologies provider in China, has been specialized in providing superior Multimedia Gateway, Integrated Multimedia Switch, Telephony Hardware in use for Telecom communications.

<div style="margin-top: 20px;">
[[Synway UC200|<span style="background: linear-gradient(135deg, #d0382d, #b8261a); color: white; padding: 12px 24px; border-radius: 20px; text-decoration: none; font-weight: 600; display: inline-block; box-shadow: 0 2px 10px rgba(208, 56, 45, 0.3);">Configuration</span>]]
</div>
|}

== Ubiquiti Unifi Talk ==

{| cellspacing="0" style="width: 100%; background: white; border: 1px solid #e2e8f0; border-radius: 12px; padding: 25px; margin: 20px 0; box-shadow: 0 2px 10px rgba(0,0,0,0.06);"
|-
| style="width: 300px; vertical-align: top; padding-right: 25px;" |
[[File:UbiquitiLogo.png|300px|x200px|thumb|center|Ubiquiti Unifi Talk]]

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

'''Type:''' Cloud Phone Solution<br>
'''Setup Time:''' Minutes<br>
'''Overview:''' Only minutes to set up a reliable, full-featured phone solution. Part of Ubiquiti's comprehensive networking and communications ecosystem.

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

== VitalPBX ==

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

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

'''Type:''' Free Communications System<br>
'''Platform:''' Linux/Asterisk-based<br>
'''Overview:''' VitalPBX is a free telephone and communications system for companies. Complete platform that can be installed on physical hardware on-site or as a hosted application. Acts as the upper layer interface for the Linux base and Asterisk.

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

== Vodia PBX ==

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

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

'''Type:''' Software-based IP PBX<br>
'''Platform:''' Open Standards<br>
'''Overview:''' Vodia PBX is a software-based, open standards IP PBX that offers complete UC functionality right out of the box. Makes installation, management, and maintenance so easy that you can effortlessly manage it yourself.

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

== Wazo ==

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

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

'''Type:''' Unified Communications Platform<br>
'''Target:''' MSPs and UCaaS providers<br>
'''Overview:''' Wazo is a unified communications technology provider that helps MSPs stand out from the crowd and build unique UCaaS experiences with simple, powerful, flexible APIs and integrations. Global reach with entities in Canada, United States, and France.

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

== Wildix ==

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

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

'''Type:''' Unified Communications<br>
'''Focus:''' Communications-Enabled Business Processes<br>
'''Overview:''' Communications-Enabled Business Processes: Wildix Unified Communications solutions enter the business processes to enhance the workplace efficiency and improve the productivity of your team.

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

== XCALLY ==

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

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

'''Type:''' Omni Channel Contact Center<br>
'''Platform:''' Asterisk-based<br>
'''Overview:''' XCALLY is an innovative Omni Channel solution that integrates Asterisk with Shuttle and Motion technologies. One of the best Contact Center management platforms for multi-channel - voice, chat, email, SMS, fax, and custom channels.

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

== Xorcom ==

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

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

'''Type:''' Business Telephony Solutions<br>
'''Specialization:''' Integrated Solutions<br>
'''Overview:''' Xorcom designs and manufactures integrated business telephony solutions that support both traditional PSTN and VoIP, including IP PBX, Hotel Phone Systems, Virtual PBX Systems, and Multi-tenant PBX.

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

== Yeastar ==

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

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

'''Type:''' VoIP PBX Systems<br>
'''Target:''' SMBs<br>
'''Overview:''' Designed for SMBs, Yeastar S-Series/P-Series VoIP PBX and Yeastar Cloud PBX deliver enterprise-grade communications features along with advanced UC capabilities, bringing solid, reliable and affordable voice solutions.

<div style="margin-top: 20px;">
[[Yeastar|<span style="background: linear-gradient(135deg, #d0382d, #b8261a); color: white; padding: 12px 24px; border-radius: 20px; text-decoration: none; font-weight: 600; display: inline-block; box-shadow: 0 2px 10px rgba(208, 56, 45, 0.3); margin: 5px;">S-Series Configuration</span>]]
[[Yeastar_P-Series|<span style="background: linear-gradient(135deg, #d0382d, #b8261a); color: white; padding: 12px 24px; border-radius: 20px; text-decoration: none; font-weight: 600; display: inline-block; box-shadow: 0 2px 10px rgba(208, 56, 45, 0.3); margin: 5px;">P-Series Configuration</span>]]
</div>
|}

== Legacy Systems ==

{| cellspacing="0" style="width: 100%; background: linear-gradient(135deg, #fef3c7 0%, #fde68a 100%); border: 1px solid #f59e0b; border-radius: 8px; padding: 25px; margin: 25px 0; box-shadow: 0 2px 10px rgba(245, 158, 11, 0.1);"
|-
| style="text-align: left;" |
<div style="font-size: 1.1em; font-weight: 600; color: #92400e; margin-bottom: 10px;">⚠️ The following systems are included for reference but are no longer actively maintained or recommended for new deployments:</div>

* '''[[Elastix|Elastix]]''' - Open source unified communications server. Development has slowed significantly in recent years.
* '''[[NortelBCM|Nortel/Avaya BCM 450 and BCM50 R6]]''' - Legacy Nortel phone system acquired by Avaya. End-of-life with limited support.
* '''[[PBXes.org|PBXes.org]]''' - Hosted PBX service with limited features compared to modern cloud solutions.
* '''[[SIPfoundry|SIPfoundry]]''' - Open source project with limited ongoing development.
* '''[[TalkSwitch|TalkSwitch]]''' - Small business phone system with discontinued development and support.
* '''[[Trixbox|Trixbox]]''' - Project has not been maintained since 2010. You might want to look for alternatives.

For new deployments, we recommend choosing from the modern PBX solutions listed above.
|}

[[Category:PBXes]]

File:UbiquitiLogo.png

2026-03-04T23:51:05Z

RP:

PBXs

2026-03-04T23:50:23Z

RP: /* FusionPBX */

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

An acronym for '''Private Branch eXchange'''. PBX telephone systems support incoming calls from the outside PSTN, placing calls between 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) - Usually don't support IP at all or support it only with expensive add-on equipment
* '''Converged''' (also known as hybrid) - Support IP and PSTN connections with equal force. Most flexible and cost-effective model
* '''IP-PBX''' - Support only IP connectivity. Any PSTN connectivity must be achieved through external converters (Gateways)

'''Learn More:''' [https://wiki.voip.ms/article/Back_to_Basics_%E2%80%93_What_is_a_PBX%3F Back to Basics - What is a PBX?]
|}

__TOC__

== 3CX Phone System ==

{| cellspacing="0" style="width: 100%; background: white; border: 1px solid #e2e8f0; border-radius: 12px; padding: 25px; margin: 20px 0; box-shadow: 0 2px 10px rgba(0,0,0,0.06);"
|-
| style="width: 300px; vertical-align: top; padding-right: 25px;" |
[[File:3CX Logo.jpg|300px|thumb|center|3CX Phone System]]

| style="vertical-align: top;" |
<div style="font-size: 1.4em; font-weight: 600; color: #1e293b; margin-bottom: 15px;">3CX Phone System</div>

'''Type:''' Software-based IP PBX<br>
'''Platform:''' Open Standards<br>
'''Overview:''' 3CX is a software-based, open standards IP PBX that offers complete Unified Communications, out of the box. 3CX makes installation, management, and maintenance of your PBX so easy that you can effortlessly manage it yourself, whether on an appliance, on your servers or in your cloud account.

<div style="margin-top: 20px;">
[[3CX Phone System|<span style="background: linear-gradient(135deg, #d0382d, #b8261a); color: white; padding: 12px 24px; border-radius: 20px; text-decoration: none; font-weight: 600; display: inline-block; box-shadow: 0 2px 10px rgba(208, 56, 45, 0.3);">General Configuration</span>]]
  
[[3CX_Phone_System_-_v20|<span style="background: linear-gradient(135deg, #d0382d, #b8261a); color: white; padding: 12px 24px; border-radius: 20px; text-decoration: none; font-weight: 600; display: inline-block; box-shadow: 0 2px 10px rgba(208, 56, 45, 0.3);">V20 Configuration</span>]]
</div>
|}

== 3CX StartUP ==

{| cellspacing="0" style="width: 100%; background: white; border: 1px solid #e2e8f0; border-radius: 12px; padding: 25px; margin: 20px 0; box-shadow: 0 2px 10px rgba(0,0,0,0.06);"
|-
| style="width: 300px; vertical-align: top; padding-right: 25px;" |
[[File:3CX_startup_0.png|300px|thumb|center|3CX StartUP]]

| style="vertical-align: top;" |
<div style="font-size: 1.4em; font-weight: 600; color: #1e293b; margin-bottom: 15px;">3CX StartUP</div>

'''Type:''' Hosted Solution<br>
'''Capacity:''' Up to 20-30 users<br>
'''Overview:''' Preconfigured, ready-to-go shared instance. Recommended for most small businesses. StartUP is only available hosted by 3CX.

'''Sign up for your free 3CX business comms system:''' [https://www.3cx.com/signup/ here]

<div style="margin-top: 20px;">
[[3CX StartUP|<span style="background: linear-gradient(135deg, #d0382d, #b8261a); color: white; padding: 12px 24px; border-radius: 20px; text-decoration: none; font-weight: 600; display: inline-block; box-shadow: 0 2px 10px rgba(208, 56, 45, 0.3);">Configuration</span>]]
</div>
|}

== Asterisk ==

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

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

'''Type:''' Open Source PBX<br>
'''Platform:''' Linux/UNIX<br>
'''Created:''' 1999<br>
'''Overview:''' Asterisk is a telephone private branch exchange (PBX), created as open software for Linux and other UNIX-like systems. Like other private branch exchanges, it allows attached telephones to make calls to one another and provides connections to outside lines.

'''Connection Methods to VoIP.ms:'''
* [[Asterisk IAX2|Asterisk (IAX2)]] - Inter-Asterisk protocol
* [[Asterisk SIP|Asterisk (SIP)]] - Standard Session Initiation Protocol
* [[Asterisk PJSIP|Asterisk (PJSIP)]] - Open Source Embedded SIP protocol stack

'''Resources:'''
* [http://www.asteriskdocs.org Free HTML documentation]
* [http://www.asterisk.org Official Asterisk website]

<div style="margin-top: 20px;">
[[Asterisk IAX2|<span style="background: linear-gradient(135deg, #d0382d, #b8261a); color: white; padding: 10px 20px; border-radius: 20px; text-decoration: none; font-weight: 600; display: inline-block; box-shadow: 0 2px 10px rgba(208, 56, 45, 0.3); margin: 5px;">IAX2 Config</span>]]
[[Asterisk SIP|<span style="background: linear-gradient(135deg, #d0382d, #b8261a); color: white; padding: 10px 20px; border-radius: 20px; text-decoration: none; font-weight: 600; display: inline-block; box-shadow: 0 2px 10px rgba(208, 56, 45, 0.3); margin: 5px;">SIP Config</span>]]
[[Asterisk PJSIP|<span style="background: linear-gradient(135deg, #d0382d, #b8261a); color: white; padding: 10px 20px; border-radius: 20px; text-decoration: none; font-weight: 600; display: inline-block; box-shadow: 0 2px 10px rgba(208, 56, 45, 0.3); margin: 5px;">PJSIP Config</span>]]
</div>
|}

== Avaya IP Office ==

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

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

'''Type:''' Enterprise PBX<br>
'''Target:''' Small and Medium Enterprises<br>
'''Overview:''' Avaya is a leading global provider of next-generation business collaboration and communications solutions, providing unified communications, real-time video collaboration, contact center, networking and related services to companies of all sizes around the world.

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

== Cisco IOS ==

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

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

'''Type:''' Router/Switch Operating System<br>
'''Platform:''' Cisco Hardware<br>
'''Overview:''' Cisco IOS (originally Internetwork Operating System) is software used on most Cisco Systems routers and current Cisco network switches. IOS is a package of routing, switching, internetworking and telecommunications functions integrated into a multitasking operating system.

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

== E-MetroTel ==

{| cellspacing="0" style="width: 100%; background: white; border: 1px solid #e2e8f0; border-radius: 12px; padding: 25px; margin: 20px 0; box-shadow: 0 2px 10px rgba(0,0,0,0.06);"
|-
| style="width: 300px; vertical-align: top; padding-right: 25px;" |
[[File:EMT Logo new nortel 7 1 19b (2).png|300px|thumb|center|E-MetroTel - Exceptional Innovation]]

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

'''Type:''' Unified Business Communications<br>
'''Target:''' Global client base<br>
'''Overview:''' E-MetroTel provides unified business communications systems through leading-edge telephony hardware and software. Delivers streamlined migration to complete solutions that tightly integrate digital, IP, analog, and Unified Communications capabilities.

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

== Epygi PBX ==

{| cellspacing="0" style="width: 100%; background: white; border: 1px solid #e2e8f0; border-radius: 12px; padding: 25px; margin: 20px 0; box-shadow: 0 2px 10px rgba(0,0,0,0.06);"
|-
| style="width: 300px; vertical-align: top; padding-right: 25px;" |
[[File:Epygi_logo_300DPI.png|300px|thumb|center|Epygi PBX]]

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

'''Type:''' IP PBX Appliance<br>
'''Experience:''' 18+ years<br>
'''Overview:''' For over 18 years Epygi has been designing, manufacturing, and delivering IP PBX and IP Gateway appliances. They serve customers ranging from small businesses to enterprise companies, each benefiting from Epygi's excellent value and service.

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

== FreePBX / PBX in a Flash ==

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

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

'''Type:''' Open Source Web-based PBX<br>
'''Platform:''' Asterisk-based<br>
'''Overview:''' 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.

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

== FreeSwitch ==

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

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

'''Type:''' Open Source Telephony Platform<br>
'''Created:''' 2006<br>
'''Overview:''' 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. Created to fill the void left by proprietary commercial solutions.

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

== FusionPBX ==

{| cellspacing="0" style="width: 100%; background: white; border: 1px solid #e2e8f0; border-radius: 12px; padding: 25px; margin: 20px 0; box-shadow: 0 2px 10px rgba(0,0,0,0.06);"
|-
| style="width: 300px; vertical-align: top; padding-right: 25px;" |
[[File:FusionPBXlogo.png|300px|thumb|center|3CX Phone System]]

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

'''Type:''' Open Source IP PBX<br>
'''Platform:''' FreeSWITCH-based<br>
'''Overview:''' FusionPBX is a full-featured domain based multi-tenant PBX and voice switch for FreeSWITCH. It provides an easy to use web interface for FreeSWITCH configuration and management. FusionPBX provides unlimited extensions, voicemail-to-email, music on hold, call parking, call queues and many other features.

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

== Grandstream UCM 6200 ==

{| cellspacing="0" style="width: 100%; background: white; border: 1px solid #e2e8f0; border-radius: 12px; padding: 25px; margin: 20px 0; box-shadow: 0 2px 10px rgba(0,0,0,0.06);"
|-
| style="width: 300px; vertical-align: top; padding-right: 25px;" |
[[File:Grandstream-Logo-2018.png|300px|thumb|center|Grandstream]]

| style="vertical-align: top;" |
<div style="font-size: 1.4em; font-weight: 600; color: #1e293b; margin-bottom: 15px;">Grandstream UCM 6200</div>

'''Type:''' Appliance-based IP PBX<br>
'''Target:''' SMB and Enterprise<br>
'''Overview:''' Grandstream Networks has been connecting the world since 2002 with SIP Unified Communications solutions. Award-winning solutions serve the small and medium business and enterprises markets, recognized for quality, reliability and innovation.

<div style="margin-top: 20px;">
[[Grandstream UCM6200|<span style="background: linear-gradient(135deg, #d0382d, #b8261a); color: white; padding: 12px 24px; border-radius: 20px; text-decoration: none; font-weight: 600; display: inline-block; box-shadow: 0 2px 10px rgba(208, 56, 45, 0.3);">Configuration</span>]]
</div>
|}

== Incredible PBX 2027 ==

{| cellspacing="0" style="width: 100%; background: white; border: 1px solid #e2e8f0; border-radius: 12px; padding: 25px; margin: 20px 0; box-shadow: 0 2px 10px rgba(0,0,0,0.06);"
|-
| style="width: 300px; vertical-align: top; padding-right: 25px;" |
[[File:IncrediblePBX-VoIPms.jpg|300px|thumb|center|Incredible PBX 2027]]

| style="vertical-align: top;" |
<div style="font-size: 1.4em; font-weight: 600; color: #1e293b; margin-bottom: 15px;">Incredible PBX 2027</div>

'''Type:''' Open Source Unified Communications<br>
'''Platform:''' Multi-platform (CentOS, Rocky, Debian, Ubuntu, Raspbian)<br>
'''Overview:''' GPL aggregation which bundles industry-leading components to provide a production-ready, turnkey unified communications VoIP platform with support for SIP and IAX. Includes Apache, PHP, MariaDB/MySQL, and latest releases of Asterisk and FreePBX.

<div style="margin-top: 20px;">
[[Incredible_PBX_2027|<span style="background: linear-gradient(135deg, #d0382d, #b8261a); color: white; padding: 12px 24px; border-radius: 20px; text-decoration: none; font-weight: 600; display: inline-block; box-shadow: 0 2px 10px rgba(208, 56, 45, 0.3);">Configuration</span>]]
</div>
|}

== PhoneSuite Systems ==

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

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

'''Type:''' Hotel Communication Solutions<br>
'''Industry:''' Hospitality<br>
'''Overview:''' PhoneSuite is a leading provider of hotel voice communication solutions for over 25 years. Leverages expertise in communication technology to provide high-quality, low-power consuming products designed exclusively for the hotel industry.

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

== PortSIP ==

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

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

'''Type:''' All-in-one Communications Platform<br>
'''Features:''' Voice, Video, Messaging, SMS, WhatsApp, Meetings<br>
'''Overview:''' PortSIP is an all-in-one communications platform designed to meet the needs of modern businesses. Offering a comprehensive suite of tools for voice, video, messaging, SMS, WhatsApp, video meetings, contact centers, and team collaboration.

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

== Positron Telecom Systems ==

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

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

'''Type:''' IP PBX Solutions<br>
'''Target:''' Small and Medium Businesses<br>
'''Overview:''' Positron IP PBX solutions offer small and medium sized businesses powerful VoIP phone systems that combine voice and data into one easy-to-use device, also known as Unified Communications. Features traditionally only available to large companies.

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

== ScopServ ScopTEL IPPBX ==

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

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

'''Type:''' IP PBX System<br>
'''Company:''' ScopServ International Inc.<br>
'''Overview:''' ScopServ International Inc. is a global pioneer in IP telephony solutions design (VoIP). The integrated system SCOPTEL IPBX provides one of the most stable environments and richest on the market.

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

== Synway ==

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

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

'''Type:''' Multimedia Gateway<br>
'''Location:''' China<br>
'''Overview:''' Synway, as a world-leading VoIP enabling-technologies provider in China, has been specialized in providing superior Multimedia Gateway, Integrated Multimedia Switch, Telephony Hardware in use for Telecom communications.

<div style="margin-top: 20px;">
[[Synway UC200|<span style="background: linear-gradient(135deg, #d0382d, #b8261a); color: white; padding: 12px 24px; border-radius: 20px; text-decoration: none; font-weight: 600; display: inline-block; box-shadow: 0 2px 10px rgba(208, 56, 45, 0.3);">Configuration</span>]]
</div>
|}

== Ubiquiti Unifi Talk ==

{| cellspacing="0" style="width: 100%; background: white; border: 1px solid #e2e8f0; border-radius: 12px; padding: 25px; margin: 20px 0; box-shadow: 0 2px 10px rgba(0,0,0,0.06);"
|-
| style="width: 300px; vertical-align: top; padding-right: 25px;" |
[[File:Ubiquiti.jpg|300px|x200px|thumb|center|Ubiquiti Unifi Talk]]

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

'''Type:''' Cloud Phone Solution<br>
'''Setup Time:''' Minutes<br>
'''Overview:''' Only minutes to set up a reliable, full-featured phone solution. Part of Ubiquiti's comprehensive networking and communications ecosystem.

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

== VitalPBX ==

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

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

'''Type:''' Free Communications System<br>
'''Platform:''' Linux/Asterisk-based<br>
'''Overview:''' VitalPBX is a free telephone and communications system for companies. Complete platform that can be installed on physical hardware on-site or as a hosted application. Acts as the upper layer interface for the Linux base and Asterisk.

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

== Vodia PBX ==

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

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

'''Type:''' Software-based IP PBX<br>
'''Platform:''' Open Standards<br>
'''Overview:''' Vodia PBX is a software-based, open standards IP PBX that offers complete UC functionality right out of the box. Makes installation, management, and maintenance so easy that you can effortlessly manage it yourself.

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

== Wazo ==

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

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

'''Type:''' Unified Communications Platform<br>
'''Target:''' MSPs and UCaaS providers<br>
'''Overview:''' Wazo is a unified communications technology provider that helps MSPs stand out from the crowd and build unique UCaaS experiences with simple, powerful, flexible APIs and integrations. Global reach with entities in Canada, United States, and France.

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

== Wildix ==

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

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

'''Type:''' Unified Communications<br>
'''Focus:''' Communications-Enabled Business Processes<br>
'''Overview:''' Communications-Enabled Business Processes: Wildix Unified Communications solutions enter the business processes to enhance the workplace efficiency and improve the productivity of your team.

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

== XCALLY ==

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

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

'''Type:''' Omni Channel Contact Center<br>
'''Platform:''' Asterisk-based<br>
'''Overview:''' XCALLY is an innovative Omni Channel solution that integrates Asterisk with Shuttle and Motion technologies. One of the best Contact Center management platforms for multi-channel - voice, chat, email, SMS, fax, and custom channels.

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

== Xorcom ==

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

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

'''Type:''' Business Telephony Solutions<br>
'''Specialization:''' Integrated Solutions<br>
'''Overview:''' Xorcom designs and manufactures integrated business telephony solutions that support both traditional PSTN and VoIP, including IP PBX, Hotel Phone Systems, Virtual PBX Systems, and Multi-tenant PBX.

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

== Yeastar ==

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

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

'''Type:''' VoIP PBX Systems<br>
'''Target:''' SMBs<br>
'''Overview:''' Designed for SMBs, Yeastar S-Series/P-Series VoIP PBX and Yeastar Cloud PBX deliver enterprise-grade communications features along with advanced UC capabilities, bringing solid, reliable and affordable voice solutions.

<div style="margin-top: 20px;">
[[Yeastar|<span style="background: linear-gradient(135deg, #d0382d, #b8261a); color: white; padding: 12px 24px; border-radius: 20px; text-decoration: none; font-weight: 600; display: inline-block; box-shadow: 0 2px 10px rgba(208, 56, 45, 0.3); margin: 5px;">S-Series Configuration</span>]]
[[Yeastar_P-Series|<span style="background: linear-gradient(135deg, #d0382d, #b8261a); color: white; padding: 12px 24px; border-radius: 20px; text-decoration: none; font-weight: 600; display: inline-block; box-shadow: 0 2px 10px rgba(208, 56, 45, 0.3); margin: 5px;">P-Series Configuration</span>]]
</div>
|}

== Legacy Systems ==

{| cellspacing="0" style="width: 100%; background: linear-gradient(135deg, #fef3c7 0%, #fde68a 100%); border: 1px solid #f59e0b; border-radius: 8px; padding: 25px; margin: 25px 0; box-shadow: 0 2px 10px rgba(245, 158, 11, 0.1);"
|-
| style="text-align: left;" |
<div style="font-size: 1.1em; font-weight: 600; color: #92400e; margin-bottom: 10px;">⚠️ The following systems are included for reference but are no longer actively maintained or recommended for new deployments:</div>

* '''[[Elastix|Elastix]]''' - Open source unified communications server. Development has slowed significantly in recent years.
* '''[[NortelBCM|Nortel/Avaya BCM 450 and BCM50 R6]]''' - Legacy Nortel phone system acquired by Avaya. End-of-life with limited support.
* '''[[PBXes.org|PBXes.org]]''' - Hosted PBX service with limited features compared to modern cloud solutions.
* '''[[SIPfoundry|SIPfoundry]]''' - Open source project with limited ongoing development.
* '''[[TalkSwitch|TalkSwitch]]''' - Small business phone system with discontinued development and support.
* '''[[Trixbox|Trixbox]]''' - Project has not been maintained since 2010. You might want to look for alternatives.

For new deployments, we recommend choosing from the modern PBX solutions listed above.
|}

[[Category:PBXes]]

File:FusionPBXlogo.png

2026-03-04T23:27:25Z

RP:

Call Detail Records

2026-02-23T14:40:41Z

RP: /* Block Calls Directly from your Call Detail Record */

{| class="wikitable"
|+
|-
! Article en Français !! Artículo en Español
|-
| [https://wiki.voip.ms/article/D%C3%A9tails_des_appels Français] || [https://wiki.voip.ms/article/Registro_de_Llamadas_(CDR) Español]
|}

The Call Detail Records (CDR) allows you to obtain detailed information of your incoming and outgoing calls. It also contains different filters to sort the call records in order to view the desired information. You can access all the calls from your account without any type of limitation on the original date.

__TOC__

The CDR can be accessed from our Customer Portal at: Finances >> Call Detail Records. You will be presented with the following screen:

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

== Search Range, Filters & TimeZone ==

The CDR page offers different options to help you sort your records. The first field is the Search Range, followed by the different filters that can help you get the the records of the calls required.

'''The Search Range:''' Under this field you can select a date range up to 92 days at a time to see your Call Detail Records. Please note that you can get records past the 92 days (meaning you can search for calls of any date), but the Search Range itself needs to be 92 or less days.

Under ''Filters and TimeZone'' you will have available the following filters to help you sort your calls:

'''Answered:''' This filter shows or hides calls with ¨Answered¨ status.

'''No Answer:''' This filter shows or hides calls with ¨No Answer¨ status.

'''Busy:''' This filter shows or hides calls with ¨Busy¨ status.

'''Failed:''' This filter shows or hides calls with ¨Failed¨ status.

'''Call Type:''' This filter allows you to choose to show calls with one of the following options:

'''All Calls:''' All Incoming and Outgoing calls will be displayed.
'''Outgoing Calls: All''' Only Outgoing calls of all kind will be displayed.
'''Outgoing Calls: Toll Free''' Only Outgoing calls towards toll free numbers will be displayed.
'''Outgoing Calls: USA''' Only Outgoing calls towards local US numbers will be displayed.
'''Outgoing Calls: CAN''' Only Outgoing calls towards local Canadian numbers will be displayed.
'''Outgoing Calls: USA/Canada''' Only Outgoing calls towards local US and Canadian numbers will be displayed.
'''Outgoing Calls: International''' Only Outgoing calls towards International numbers will be displayed.
'''Incoming Calls: Toll Free''' Only Incoming calls from toll free numbers will be displayed.
'''Incoming Calls: USA''' Only Incoming calls from local US numbers will be displayed.
'''Incoming Calls: CAN''' Only Incoming calls from local Canadian numbers will be displayed.
'''Incoming Calls: CallerID Filtering''' Only incoming calls that have been subject to a CallerID Filtering.
'''Calls to DID:''' Only Inbound calls to DID will be displayed.

'''Call Billing:''' This filter allows you to choose to show calls with one of the following options:

'''All Calls:''' All Free and Billed calls will be displayed.
'''Free Calls:''' Only Free Calls will be displayed.
'''Billed Calls:''' Only Billed calls will be displayed.

'''Account:''' This Filter allows you to choose to show calls with one of the following options:

'''All Accounts:''' Calls from all accounts will be displayed.
'''Main Account:''' Calls from the Main account will be displayed.
'''SubAccounts:''' Calls from Sub Account will be displayed.

'''Time Zone:''' Here you can set the Time Zone to adjust the CDR to your local time.

The entries will be displayed below the filters and in order to assist you to identify your entries. Premium calls will be displayed with a blue asterisk.

'''Gray Arrows:''' The blue arrows on the CDR allow you to sort each field in ascending order.

'''Search:''' With this field, you will be able to search calls per number dialed, number that received the call, description, duration, rate and cost.

== Detail Screen ==

Clicking on the entries will display further details on the call. After clicking on an entry you will be able to see the ''Destination, Description, Caller ID, Account, Disposition, Date, Time, Duration in Seconds, Duration, Rate Type, Rate, Total in US cents, Total in US dollars, and Unique ID''. Please refer to the image below for an example of the details screen:

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

== CallerID Filtering in CDR ==

When you create a rule using the feature "[[CallerID Filtering]]" and route it to any destination, including those that are routed to '''''"SYSTEM"''''', the arrow [[File:CIDFiltering_Arrow.png]] will appear in the CDR in the description.
<br />
[[File:CDR_CIDFiltering.png|750px|border]]

By clicking on the entry where the [[File:CIDFiltering_Arrow.png]] arrow is, you will be able to see the call details, including the destination it is being routed to. ''(Such as to a specific Ring Group as shown in the example below.)''<br />
[[File:CDR_CIDFiltering-Details.png|750px|border]]

== Block Calls Directly from your Call Detail Record ==

As of 02-07-2023, you can now block undesired incoming calls directly from your call detail records. To proceed, simply head into your VoIP.ms portal, Finances, Call Detail Records and look for the incoming call you wish to block. Press the action button on the right of the entry, confirm you wish to proceed and the number will automatically be blocked as well as the button now turning red to state there is an active block for this number.

[[File:CDRquickBLOCK1.png|border|600px]]

[[File:CDRquickBLOCK2.png|border|600px]]

Lastly, you can also consult callers you are currently blocking under DID Numbers menu, CallerID Filtering.

[[File:CDRquickBLOCK3.png|border|600px]]

==Consulting CDR Logs==

Effective August 15, 2024, you can now access detailed logs of both incoming and outgoing calls by simply clicking on the specific call and selecting the "View Call Logs" option.

[[File:CDRLogNew2024-08-28.png|border|700px]] [[File:CDRLogNew2-2024-08-28.png|border|400px]]

With incoming calls, you can access detailed information on the entire journey the caller took to reach you. For instance, if your DID is connected to an IVR, you'll be able to see which options the caller selected, whether they were placed in a queue, and how long they waited before the call was answered or disconnected. This comprehensive data empowers you to optimize your telecommunications setup effectively.

[[File:CDRLogNew3-2024-08-28.png|border|500px]]

===List of Messages found in CDR Logs===
You will find below a list of all the messages that can show in the CDR logs and a quick description.

{| class="wikitable"
|- style="font-weight:bold;"
! CDR Log Message
! Desdcription
|-
| Failover due to 'No Answer' status
| The incoming call to your DID Number was not answered and the call went to the failover ''No Answer''.
|-
| Failover due to 'Busy' status
| The incoming call to your DID Number returned a busy signal and the call went to the failover ''Busy''.
|-
| Failover due to 'Unreachable' status
| The incoming call to your DID Number was not answered because your device was unreachable, and the call went to the failover ''Unreachable''.
|-
| Failover to Voicemail {Voicemail}
| The incoming call to your DID Number went to the voicemail configured on your DID number.
|-
| Hangup: Account suspended
| The account in general was suspended.
|-
| Hangup: Sub-account suspended
| The sub account was suspended and can no longer make or receive calls.
|-
| Hangup: Too many active channels for this DID
| Your DID Number reached its maximum capacity of channels.
|-
| Hangup: Too many active channels for this trunk
| Your main account / sub account reached the maximum amount of channels on outgoing calls.
|-
| Hangup: Channel is busy
| A busy signal was received.
|-
| Hangup: Channel is congested
| A congestion response was received.
|-
| Hangup: Channel is unavailable
| You currently do not have enough channels to pass an outgoing call.
|-
| Hangup: Exiting queue
| A call was in a calling queue and hungup.
|-
| Hangup: Too many active channels for this account
| Your account has reached its maximum channel capacity
|-
| Hangup: Invalid IVR
| The IVR configuration set on the DID is either invalid or deleted.
|-
| Hangup: Invalid recording
| The Recording set on the DID is either invalid or deleted.
|-
| Hangup: Maximum rounds in IVR has been reached
| Caller reached the maximum attempt to make a selection in an IVR and the call was hungup.
|-
| Hangup: Too many invalid choices for this IVR
| Caller made too many invalid choices in the IVR selection menu and the call was hungup.
|-
| Hangup: Too many timeouts for this IVR
| Caller timed out in the IVR too many times in the IVR and the call was hungup.
|-
| Hangup: Too many failed password attempts for DISA
| Caller reaching the DID with a DISA routing entered too many invalid NIPs and the call was hungup.
|-
| Hangup: Too many phone number attempts for DISA
| Caller reaching the DID with a DISA routing attempted to reach too many numbers and the call was hungup.
|-
| Normal hangup
| Normal hangup by the caller or the user.
|-
| Hangup: Attempt to dial a banned number
| A number that is not authorized to be dialed was dialed.
|-
| Hangup: Too many active channels for this extension
| The extension related to a sub account currently has too many concurrent calls.
|-
| Hangup: Country not allowed
| The country the account is trying to call is not authorized in your list of allowed countries.
|-
| Hangup: International calls not allowed
| The main account or sub account does not have international calls enabled.
|-
| Routing to 'Busy' option"
| If the failover option ''If Busy'' has a routing option, it will attempt this routing.
|-
| Routing to 'Number not in service' option
| The DID is routing to System: Number not in Service.
|-
| Routing to 'Number disconnected' option
| The DID is routing to System: Number Disconnected.
|-
| Routing to IVR: $IVR_NAME
| The incoming call received on your DID went to a specific IVR.
|-
| Routing to Queue: $QUEUE_NAME
| The incoming call received on your DID went to a specific Calling Queue.
|-
| Routing to Time condition: $TIMECONDITION_NAME
| The incoming call received on your DID went to a specific Time Condition.
|-
| Routing to account: {$account}
| The incoming call received on your DID went to your main account.
|-
| Routing to voicemail: {$voicemail}
| The incoming call received on your DID went to a specific Voicemail.
|-
| Routing to Ring group: $RINGGROUP_NAME
| The incoming call received on your DID went to a specific Ring Group.
|-
| Routing to Call hunting: $CH_NAME
| The incoming call received on your DID went to a specific Call Hunting.
|-
| Routing to Play recording: $PR_NAME
| The incoming call received on your DID went to a specific Recording.
|-
| Routing to Call forwarding: $CF_NAME
| The incoming call received on your DID went to a specific Call Forwarding.
|-
| Routing to SIP URI: $SU_NAME
| The incoming call received on your DID went to a specific SIP URI.
|-
| Routing to Callback: $CALLBACK_NAME
| The incoming call received on your DID went to a specific Callback.
|-
| Routing to Disa: $DISA_NAME
| The incoming call received on your DID went to a specific DISA.
|-
| Routing to 'Hangup' option
| The DID is routing to System: Hangup.
|-
| Routing to Echo test
| The DID is routing to System: Echo Test.
|-
| Routing to DTMF test
| The DID is routing to System: DTMF Test.
|-
| Caller ID filtering applied for Anonymous
| The incoming call contained an anonymous callerID and was filtered based on your callerID Filtering setting.
|-
| Caller ID filtering applied for Invalid number
| The incoming call contained an invalid callerID and was filtered based on your callerID Filtering setting.
|-
| Caller ID custom filtering applied for: $FILTERING_NUMBER
| The incoming call contained a specific callerID and was filtered based on your callerID Filtering setting.
|-
| IVR reached option 'i' (invalid choice)
| The caller selected an invalid choice and you have the ''i'' option configured in your IVR.
|-
| IVR reached option 't' (no response)
| The caller selected an invalid choice and you have the ''t'' option configured in your IVR.
|-
| IVR key(s) pressed: {$KEY}
| The caller pressed a specific key to reach a routing option within the IVR menu tree.
|-
| Time condition criteria matched
| Caller reached your time condition and the criteria matched.
|-
| Time condition criteria did not match
| Caller reached your time condition and the criteria did not match.
|-
| Time condition matched
| Caller reached your time condition and the criteria matched.
|-
| Time condition did not match
| Caller reached your time condition and the criteria did not match.
|-
| Doing a CNAM lookup
| Your DID has CNAM Lookup enabled and it did a verification.
|-
| Routing to sub-account: $SUBACCOUNT_NAME
| The incoming call received on your DID went to a specific sub account.
|-
| Applying phonebook Speed dial: $PHONEBOOK_NUMBER
| A speed dial entry from your phonebook was dialed.
|-
| Applying phonebook CallerID override: $CALLERID
| A callerID Override from your phonebook was applied.
|-
| Applying phonebook CNAM override: $PHONEBOOK_NAME
| A callerID Name Override from your phonebook was applied.
|-
| Routing to internal extension: {$param} ($EXTENSION)
| A call placed from 1 sub account to another via the internal extension number was placed.
|-
| Received as inbound SIP URI.
| An incoming call coming as a SIP URI was received.
|-
| Routing to sequence: $SECUENCE_NAME
| The incoming call received on your DID went to a specific Sequence.
|-
| Routing to conference: $CONFERENCE_NAME
| The incoming call received on your DID went to a specific Audio Conference.
|-
| Conference Recording: {$param} secs raw duration
| A specific audio conference was recorded and provides the total amount of seconds the recording occured.
|-
| Status is 'Answered'
| The call was answered
|-
| Status is 'No answer'
| The call was not answered.
|-
| Status is 'Busy'
| The call returned a busy signal.
|-
| Status is 'Congestion'
| The call returned a congestion error code.
|-
| Status is 'Channel not available'
| No channels are available to proceed with the call.
|-
| Status is unknown
| No error codes received, but the call ended up not working.
|-
| Call hunting status is 'Answered'
| Call sent to a call hunting was answered.
|-
| Call hunting status is 'No answer'
| Call sent to a call hunting was not answered.
|-
| Call hunting status is 'Busy'
| Call sent to a call hunting returned a busy signal.
|-
| Call hunting status is 'Congestion'
| Call sent to a call hunting returned a congestion error.
|-
| Call hunting status is 'Channel unavailable'
| No channels are available to proceed with the call.
|-
| Call hunting status is unknown
| No error codes received, but the call ended up not working.
|-
| Caller from Canada blocked
| Callers from Canada are blocked to call your toll-free number.
|-
| Caller from Puerto-Rico blocked
| Callers from Puerto Rico are blocked to call your toll-free number.
|-
| Caller from Alaska blocked
| Callers from Alaska are blocked to call your number.
|-
| Outgoing Call started by a Call Forwarding
| This is the outgoing part of a call forwarding entry.
|-
| Call failed: Max. Per Minute Rate Limit for International Calls
| Your account has international calls enabled, but the call was rejected because your current per minute rate limit is lower than the actual cost to call the destination.
|}

[[category:guides]]

Porting FAQ

2026-02-06T23:21:27Z

RP: /* How to port my Google Voice Number to VoIP.ms */

{| class="wikitable"
|+
|-
! Article en Français !! Artículo en Español
|-
| [https://wiki.voip.ms/article/Questions_fr%C3%A9quentes_sur_la_transf%C3%A9rabilit%C3%A9 Français] || [https://wiki.voip.ms/article/Preguntas_frecuentes_LNP Español]
|}

__TOC__

'''Please also check out our other Wiki Guides in Regards to Porting:'''
*[http://wiki.voip.ms/article/Porting_a_Number Porting a Number]
*[http://wiki.voip.ms/article/Porting_a_US_Number Porting a US Number]
*[http://wiki.voip.ms/article/Porting_a_Canadian_Number Porting a Canadian Number]
*[http://wiki.voip.ms/article/Porting_a_Toll_Free_Number Porting a Toll Free Number]
*[http://wiki.voip.ms/article/Port_Rejection Port Rejections]

== How much does it cost to Port a Number to VoIP.ms? ==

As of January 15th, 2020, all portings are free of charge across Canada and US48 for local and toll-free numbers. <br/>
Fax numbers have a $15 USD one-time fee per number.<br>
For International Numbers please send your inquiry to '''ports@voip.ms'''

Also note the following fees apply when porting numbers from US and Canada:

A $8.75 USD fee for a port resubmission after a rejection.<br>
A $50 USD fee for a port cancellation once the order has been submitted to the carrier.<br>
A $250 USD fee for expediting a port with the carrier, if available.

== How to port my Google Voice Number to VoIP.ms ==

You will first want to make sure that your number is portable to our service by checking here [https://voip.ms/features/local-number-portability Number Portability] then please follow this guide to [https://support.google.com/voice/answer/1316844?hl=en Unlock your Google Voice Number].
You may use your username as the account number and your service address since Google does not list any on their website. If you have further questions please email us at ports@voip.ms

== How do I start the process to port my number(s)? ==
To start the porting request you can log in on your portal -> DID Numbers -> DID Portability and click on Start Procedure to start your porting order.

To port your number over, please verify with your existing service provider that the address listed in your Customer Service Record (CSR)/Account information/Customer portal is a physical address. PO BOX addresses are not valid for portability.

== I am porting many numbers, what is the best way to proceed? ==

In this case, prior to submitting any porting request for many numbers, please email our LNP Department ( ports@voip.ms ) with the batch you wish to port. Our LNP Team will verify your list of numbers and advise what is the best way to proceed with it.

== What type of information will you need in order to port my number(s)? ==
The first thing you will need to provide to us is the Service Address of the number(s) you wish to port. This is the address where you will be using the number(s) and where you would like your E911 services to be listed. Even virtual numbers have a service address listed on the losing carrier’s side.

The service address will help us to fill up the Letter of Authorization (LOA). This form will be sent to our carrier to process your order.

In order to fill out this form, we will need the following information from you:
Company Name (if applicable), Service Address (as it is listed with your current carrier), Service Provider Name, Number to port, Billing Telephone Number (BTN):
This is the main billing number on your account that all of your numbers may be listed under. Please note that it is possible to have several different BTNs depending on how your account is set up.

All of the information needed for filling out the LOA can be supplied by your current phone provider by requesting a Customer Service Record (CSR).

Note: most local number portability does not require a signature as a LOA proof. Although some exceptions apply and will require the owner of the number(s) to hand-sign his current provider invoice. The exceptions are as follow:

* Portability of some ratecenters in Canada.
* Portability of toll-free numbers.
* Porting a number as a virtual Fax DID.

For local numbers in Canada, our LNP department will gladly advise the moment a signed invoice is required.

'''Note''': We accept digital signatures, although they can in some cases be rejected. What is suggested to avoid any rejections is to receive a hand-signed signature.

== Can I port at the same time, one number as a Voice and another as a Fax? ==

Since our Fax and Voice carriers are different, you will need to submit only one port (either for the Voice or Fax) and wait until that port is complete before you submit the second number.
Submitting both ports at the same time will only cause rejections and delays on both orders.

==What is a partial port?==
When transferring numbers to our network, it is necessary to verify the numbers listed on your bill/statement. If you do not wish to port all listed numbers or want to maintain other active services (such as Internet, FAX, etc.), this will be treated as a partial port.

You will need to provide a list detailing the current provider's services and/or numbers, in addition to the number(s) slated for porting. Also, indicate whether you prefer to keep the other services active or have them disconnected during the porting process.

== My current carrier does not provide any invoices, what else can I submit? ==

If your carrier does not provide billing invoices, you can submit a screenshot from the losing carrier’s online portal that shows the customer name, number to port, and account information. The screenshot must also be printed, hand-signed, and sent to us.

== What type of numbers can I port? ==
You can port any local, virtual, and even wireless number as long as we have coverage for that particular area. You can also port your Toll-Free numbers and Fax numbers, although Fax is only available for US and Canada numbers.
Please note that Pager numbers are non-portable.

== Can I have my existing VoIP.ms Voice number ported to your Fax service? ==
If you have already a voice number with us, we can (upon checking portability) transfer it for it to work as a Fax number. The fee is the same as the regular Fax port (15$) since our Voice carriers and Fax carriers are not related, even when both numbers are within our network. If you wish to transfer your Voice number to our Fax service please email us at ports@voip.ms
*Please note that a Fax number will only work as a Fax and not as a Voice number anymore.

== Internal Port (VoIP.ms to VoIP.ms) ==
Do you currently have a phone number (DID) in your account that you want to move to a different account? You can request this transfer through our portal's ticket system. If you need further help, please get in touch with our support team.

== My number is not portable to your network, even though I am close to a city that is portable to you, why is that? ==

The portability of a number depends on our carriers to cover that city (or rate center which is where the number actually belongs) and some towns/cities are not available no matter how close they are to covered cities. In some cases, you can purchase a number from a certain city but it is not available for porting. This again depends on our carriers’ coverage and it is out of our hands.

Some numbers are simply not portable. These include anything from pagers to numbers from a few small rural independent telephone companies to "shared use" toll-free numbers where the customer only has use of the number in one city or one area code. A few loopholes exist for a provider to claim they, not you, "own" a number

No ICA. None of our carriers have an Interconnect Agreement in place with the telephone company that currently holds your number, making it non-portable.

Our managers are always looking for new carriers to cover more rate centers and also our current carriers are always expanding their coverage; we suggest that you come back in the future and verify if your rate center is available for porting. You may do so by logging in on your portal -> DID Numbers -> DID Portability -> Check Availability

== How long does the port process take? ==

On average porting requests take from 5 business days to complete Canadian and US numbers and 2 weeks for Fax numbers; however, this can change depending always on the losing carrier’s cooperation level and also if there is any holiday in between. Please keep in mind any type of change or rejection can delay this process. This is why it is very important to verify the porting information prior to submitting an order.

== What happens behind the scenes while my number is being ported? ==

Once your order has been submitted and received, you will be notified via email. After reviewing your order, we will submit your port request with the information you provided to our carrier. Our carriers then submit the port request to the losing carrier (CLEC) for their approval, sometimes they check the info with their reseller (most carriers resell services from a much larger carrier aka CLECs). At that point, the ball is in the losing CLEC's court. You will be notified of its progress/status via email. If the request is rejected, we will provide you with instructions on how to resolve the issue.
Once we have received a FOC date, you will again be notified via email. Please keep in mind any changes made to a port request can push the FOC date further out or even create a rejection.

== Situations That Can Lead to an Unsuccessful Porting your number to VoIP.ms and How to Overcome Them ==

For more information in this regard, head to [[Port_Rejection | this article that provides all the details]].

== What happens if my port request gets rejected? ==

In the event that we receive a rejection from your carrier, we will provide you with the main reason for the rejection and also instructions on how to resolve the issue. You can check the different and most common rejections by clicking on our Wiki Site [[Port Rejection]].

== I read that I will receive an SMS to approve the port, can you provide further information about this? ==

Wireless carriers have established that, to port away from them, they will send you an SMS that you must reply to with approval within the next 90 minutes of receipt for them to authorize the portability. The SMS is sent only by the losing carrier.

You must stay alert on your phone as soon as you receive our Processing or Resubmitted email, that would mean the port was sent to the carrier and they will soon send the SMS to the number you are porting.

If, for some reason, you are unable to receive the SMS, you will need to check your options with the losing carrier, as they are the ones that will reject your port if they do not receive your approval.

Please note '''the losing carrier will always send the SMS to the number you are porting''', even if you have configured your number as a landline or cannot receive SMS.

Some carriers might say not to do anything about the SMS if you want to move forward. Note that you must always reply with approval, otherwise, they will reject your port, regardless of what they say in the message.
If you have further questions, please email us at ports@voip.ms

== How can I expedite my port request? ==

We will always try to obtain the soonest FOC date available; however, delays may occur due to holidays, rejections, etc. Also, carriers (on our side and also on the losing side) have specific processing times we must honor and both the FCC (for the US) and the CRTC (for Canada) mandate that we cannot take a number unless the current provider gives us permission to. In some special cases, we are able to expedite porting orders; however, some carriers may charge for this and it is also a case-by-case situation. Please email us a ports@voip.ms for details about expediting your number.

== Will I experience any downtime while my numbers are porting? ==

On the day of your port, at some point, your number(s) will be triggered in our system. You should not experience any downtime as long as you have set up everything as you need it and properly routed the phone number(s) in your VoIP.ms portal. You will be able to view your number(s) in your VoIP.ms portal prior to the day of port in order to give you time to set everything up. Please contact Support via Live Chat or Ticket if you need help setting up everything before your number ports. However, it is important to also note that some downtime may occur during the porting day as the number(s) is(are) changing provides and routes.

== When/How will I know my number(s) has/have ported? ==

After your number(s) has/have completed porting, we will make test calls to make sure your number(s) is/are working properly. Once we have verified calls are completed on our network, you will be notified via email that the porting was a success. If there are any problems while test calls are being made, we will work with you until the issue has been resolved.

== My previous carrier is charging me for porting out and/or asking for a 30-day notice, how can I deal with this? Is it legal? ==

Unfortunately, we have no control over the fees the losing carrier charges for porting out their numbers. Prior to submitting an order with us, please verify with them if there is any fee involved.

Regarding the 30-day notice, this is usually for Pay-as-you-go services. Since you were not charged for 30 days from the day you started your phone service with your carrier, you are responsible for payment for the 30 days following the cancellation of your account (which, in most cases, occurs due to number porting). This has always been stated in the terms of service of almost every carrier.
Some carrier contracts mention that the customer will “Pay for 30 days after the transfer is requested", and not after notice is given (by the customer). The one who requests the transfer is the new carrier.
Unfortunately, the CRTC (or FCC) does not regulate phone company policies.

== What is the difference between Geographical and National numbers? ==

Non-geographic numbers or Nationals are not tied to specific locations. Nationals can be directed to any existing landline, mobile or international number.
Because they are not restricted to a limited area, Nationals allow businesses to be flexible; they are important for businesses who aim to have a presence across the nation.

== What is a RespOrg? ==

The port for Toll-Free numbers is a bit different from porting local numbers. RespOrg is the communication carrier responsible for managing your Toll-free number(s) and they use the SMS800 system to do so.

== What are all of these acronyms: LOA, FOC, BTN, LSP, CSR, LSR, PON? ==

For a better understanding of the porting process, we provide the following list of the most common terms used in the VoIP LNP Business:

LOA - Letter of Authorization. This is the form we will need to fill out when porting your number(s). This form gives us permission to request, on your behalf, to port your number(s) to a different phone provider.

FOC - Firm Order Confirmation. This is the date on which a phone provider will complete the port of a number(s).

BTN - Billing Telephone Number. This is the main telephone number listed on your account. Depending on what type of account you have, you could have just one BTN or you could have several. When porting numbers to VoIP.ms you want to be sure to ask your current phone provider what the BTN(s) is/are for the number(s) you wish to port.

LSP - Losing Service Provider (aka Losing Carrier). This is the phone provider that your number(s) is/are leaving from.

CSR - Customer Service Record. Every carrier has a record that has your number(s), address, and Billing Telephone Number (BTN)(s). Provides correct porting details for the numbers associated with an account. Please note that the information listed on the CSR may not be the same information listed on the billing portion of your account.

LSR - Local Service Request. This is the actual order request that the losing carrier will receive from the gaining provider. All of the information on this request must match the losing carrier’s (CSR) in order for them to issue a FOC date.

PON – Porting Order Number. This is the confirmation number that is created once the order has been sent to the losing carrier.

CLEC - Competitive Local Exchange Carrier or Upstream carrier. Most carriers are resellers from larger upstream carriers.

TN = Telephone Number.

TF = Toll-Free Number.

EU = End User or Customer. This is you :)

NPA NXX = Number Plan Area, commonly called Area Code. NXX refers to the three digits of a phone number immediately following the area code. In the number (555) 777-0000, the NPA is “555” and the NXX is “777”.

== I am being asked for my PIN and IMEI why is that? ==

When porting a wireless / cell number, some extra information is needed. For US numbers you must provide the PIN/Password on the current account.
The PIN stands for Personal Identification Number (PIN) which it's usually a 4-digit number combination. PIN numbers are often used for wireless numbers to avoid the number being ported by accident to another carrier. You may ask the current carrier for the proper PIN to use.

For Canadian Wireless numbers, we will need only your account number; unless the account number and the number porting are the same (meaning it’s a prepaid account); then we will need the IMEI.
<br>
'''Also, for Canadian Wireless numbers, as soon as the losing carrier receives our order, they will send either an SMS or email to the owner regarding approval for the current port. Please note that once you receive this notice, you have 90 minutes to approve the port, otherwise the losing carrier will reject it.'''

== What is the IMEI? ==

The IMEI is the International Mobile Equipment Identity is a 15-digit number, usually unique, to identify mobile phones, as well as some satellite phones. It is usually found printed inside the battery compartment of the phone.
To retrieve the IMEI, enter *#06# and the IMEI will be displayed on your phone.

== When porting, I am being asked for information that I have never provided to my current carrier, why is that? ==

When porting your number you will be asked by our system to provide certain information which is important for our carriers to properly port your number(s). Some carriers may not use nor require the SSN, PIN, Name, Service address, etc when first subscribed with them; however, our carriers do require this information to process the order accordingly. This is the same when Porting Out, some carriers may ask you for information we don't request when opening an account with us.

== 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 advise what information is needed to start this process.

Once your new carrier confirms your number was ported out successfully and it is properly working on their end, please delete it from your account with us; you may do so by log in on your account -> DID Billing-> Delete number. You will also need to please click Yes on the pop-up window when deleting the number. ("Check here if you ported this number to another network").
In some cases, we are also advised that your number was ported out, in which case we will remove it from your account and email you regarding so.

Please open a ticket to our LNP Department or send an email to ports@voip.ms to find out what is the port out information for your number or if you get a rejection.

== Can I have updates regarding my Port Out? ==

Per industry standards, the winning carrier is the only one that is provided with details on a port, this is why they are the only ones that can provide more info about this Port Out.
When it comes to Port Outs, the losing party (VoIP.ms) is never provided with updates or details, therefore we are unable to see what is happening with that order.
Upstream carriers (our carriers) send rejections and details to the winning carrier only, kindly refer to them for more info about this.
If your new carrier does not provide any updates, feel free to contact authorities and report your new carrier to them since as per authorities' rules only they can and are obligated to provide any and all details to their customers.

== Can a Port In/Out be rejected for Negative Balance?==
Per industry standards, a negative balance is not a reason to reject a port. That means that in no circumstance a port can be declined for the negative balance.

Please consider this when porting numbers either In or Out.

== Can I block a Port Out from a customer of mine? ==

As per FCC (for US and Toll Frees) and CRTC/CCTS (for Canada) regulations, the only reason for a Port Out to be rejected is if the End-User information provided by the winning party does not match with what our carriers have on file or (in case of local numbers) if the number is not in service/disconnected.
As long as the winning provider submits the correct information and the number is properly working, it must be released.

== Can I port out a number I have purchased from your portal? ==

We allow port-out requests for all our numbers. Please note; however, that in the case of numbers purchased from our stock, a port away fee may apply, case by case.

For more information about porting out your number, please email LNP Department at ports@voip.ms

== When Porting Out my number the new carrier requests the PIN, what is my PIN? ==

Please note that numbers that are ported into us or purchased from our portal, are not assigned any passwords/PINs. This could be a system rejection that the gaining carrier provides before sending the LSR to our underlying carrier.
If this number is being ported to a wireless provider, they may need to leave the PIN field blank or make up a PIN. If the port still gets rejected, kindly ask the winning carrier for both the PON and the exact reason and email us at ports@voip.ms for us to verify with our carriers.

== My current carrier says the number has already been released or the order has not been received on their side yet. ==

The porting process is dealt with between CLECs and (sometimes) resellers. However, some resellers are not aware of the port, not even when the number has left their side. We have no reason why we should not send an order as long as all the proper information is provided. We will submit the order, most likely, on the same day; however, since our carrier works as well with some other upstream carriers, the order might take some time before reaching the losing party.
It is not odd that, if your order was rejected, your current carrier will not see it since most carriers do not save records of rejected orders.
Some carriers also say that the number has already been released/ported when in fact it has not. Our system is always monitoring when a number has been released without prior notice, if your number was in fact released, we will be notified about it and we will email you immediately after adding the number to your account, to avoid missing calls.

== I need my number to be ported on weekend or on a specific date, can I request this? ==

LNP Departments are closed on weekends; therefore no numbers can be ported during Saturdays or Sundays, or holidays. Please note that if you need a specific date, this is only available for some US local, US TF numbers, and Canada local numbers. You will need to leave a note in the Step 5 Notes section when filling your porting request and we will ask our carrier if such a date can be met. Please remember that this is not always possible; however, we will do our best to meet it.

== My port is now processing or has a FOC date, can I now cancel my number with the losing carrier to avoid any extra charges? ==

It is important to note that you must wait until the number is fully ported to our system (we will email you about this) before you can cancel the account with the old carrier. If the order is canceled before the number is fully ported, you might get a rejection or even lose the number. Please be cautious.

== Why are purchased numbers cheaper than ported numbers? ==

This is because porting a number involves different carriers that must be paid for the transfer. Also, it is easier to purchase a number and configure it than to transfer an existing number from another carrier.

== My number has been disconnected by my current carrier and I am not receiving any calls, has it been ported? ==

Please note that the number will always belong to the current provider until we email you regarding its completion. The number must be active at all times during the porting process. Disconnected numbers are not portable. If the losing carrier disconnected the number claiming that it has ported, you must verify if you have received a completion notice from us. If not, you must contact the current carrier and ask them to reactivate the number as soon as possible. Remember that the number is still on their end since it has not been ported.
Some carriers may disconnect the number just to stop the port and then activate it again on their side.

== My porting process has been completed and I have enabled SMS on my number, why can't I receive messages yet? ==

The SMS feature can take up to 48 hours (24 hours on average) after the porting process has been completed to become fully active. Please allow this much time before fully testing the SMS functionality on your recently ported in number.

== Now that the number has been ported to VoIP.ms, who cancels the old account? ==

Once we confirm the number has been successfully ported to us, check to see if your account is in fact closed and you are not continuing to be charged by the old carrier. If the account wasn't closed, please contact the losing carrier and ask them to cancel your old account if it's no longer useful to you anymore. This is to avoid any extra charges, as some carriers still show the ported number on their end, even when the number was ported out.
Please note that the cancellation can only be done by the customer since he/she is the one paying for the old carrier's services. We can't contact them nor cancel any services/lines with them.
Some carriers might say otherwise just to keep on billing you. Please demand the cancellation of the old account if it's no longer useful to you.
Note that neither your account with us nor the number will be affected by this cancellation, since they are now with us.

== I currently have a CNAM on my number, when porting will I keep it? ==

If you have a CNAM with the current carrier when porting your number, it is most likely that the current CNAM is deleted when the transfer is completed, showing some random word such as the number itself, the customer's first or last name, or simply “unknown”. In some special cases, the CNAM is kept as it was.

To avoid this, please verify if your number can have CNAM updated before starting the porting process. If so, after the number is ported, please test your number to verify your CNAM values and if they are not what you wish, contact our Support Staff ( Support@voip.ms ) to have the CNAM updated.

== At what point and how can I configure my porting number? ==

When the FOC is received, your number will be added to our system. From that point, you can configure it and prepare it for the day the port completes. Your number will be reachable from anyone on our network at this time. Remember that if you need further assistance with your configuration, you can always open a ticket to our Support Staff who will gladly help you.

== Pro Bonus Tip ==

A pro tip we can share to avoid disruption when porting numbers is to kindly ask your current carrier to do a temporary forward to VoIP.ms temporary numbers you may purchase on our portal beforehand starting the porting process, this way, you will now miss any calls at all!

[[Category: Guides]]

Desvío de Llamadas

2026-01-28T20:27:08Z

RP: /* Cargos */

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

__TOC__

Un Desvío de llamadas permite redirigir una llamada entrante a un teléfono móvil u otro número de teléfono donde la persona que desea llamar pueda responder. Puede configurar cualquier número, incluso números internacionales.

== Video tutorial ==

[[Image:CallForwardThumbnail.png|200px|link=https://www.youtube.com/watch?v=k1NsFxBQ8gQ]]

== Configurar un desvío de llamadas ==
=== Crear entrada de desvío de llamadas ===
<br clear="all" />
*El primer paso sería crear una entrada de Desvío de llamadas, puede hacerlo desde el portal del cliente>> DID Numbers>> Call Forwarding.

IMAGEN
<div><ul>
<li style="display: inline-block;"> [[File:Callfwd.JPG|thumb|none|500px|Call Forwarding menu]] </li>
<li style="display: inline-block;"> [[File:Call fwd page.JPG|thumb|none|530px|Call Forwarding page]] </li>
</ul></div>

*Después de eso, solo necesita dar click en el botón de " Add Forwarding" y verá la siguiente pantalla:
<br clear="all" />
[[File:Callfwdentry.jpg|thumb|none|650px|Call Forwarding entry]]
<br clear="all" />

'''Phone Number''': Ingrese aquí el número de teléfono al que desea que se envíen las llamadas entrantes de su DID. Para números de USA o Canadá, puede configurar el número con 10 dígitos o incluso usar el prefijo 1, por ejemplo, 403XXXXXXX o 1403XXXXXXX le dará el mismo resultado. Para números internacionales, asegúrese de ingresar el número con el prefijo 00 o 011 (por ejemplo, 0052999XXXXXXX para un número en México). También puede usar el 033 o 044 para anular la ruta [Value vs Premium | Value o Premium]. También asegúrese de haber habilitado las llamadas internacionales en su [Account Settings | account].
'''CallerID Override''': Esta configuración es opcional y le permite enviar el [[Caller ID]] de su elección al nûmero al cual redirigirá la llamada, ésto sirve para reconocer de dónde proviene la llamada, tenga en cuenta que la confiabilidad del paso del número [caller ID] dependerá de la ruta que está utilizando, recuerde que para las rutas internacionales el [Caller ID] no está 100% garantizado.

Nota: Si el desvío de llamadas con un CallerID override se asigna a un grupo de timbre, entonces ese número de CallerID override será el que pasará a todas las llamadas enviadas que formen parte de ese grupo de timbre.
Por ejemplo, digamos que tiene su teléfono celular configurado como desvío de llamadas, puede poner su número DID como [[caller ID]] override y de esta manera cuando vea una llamada entrante en su teléfono celular de el número DID, sabrá que esto es una llamada de voip.ms. Si deja esta configuración en blanco, recibirá el número [caller ID] de la persona que llama a su número DID.

'''Description''': Esto también es opcional, esto puede ayudarlo a identificar una entrada de Desvío de llamadas en particular.

'''DTMF Digits''': esta configuración es opcional y le permitirá ingresar los dígitos que se marcarán una vez que la llamada se conecte a su número de desvío de llamadas.
Por ejemplo, puede desviar las llamadas de un número a un [Digital Recptionist (IVR) | ivr] y enviar automáticamente un número de extensión de esta manera.

'''Pause''': la cantidad de segundos que el sistema esperaría antes de enviar los dígitos DTMF que configuró con la opción anterior. Ésta también es una configuración opcional.

=== Enrutar las llamadas entrantes desde DID a su desvío de llamadas ===

Una vez que haya creado una entrada de desvío de llamadas, puede asignarla a muchos números DID como desee sin necesidad de volver a crearla.
Debe ir al portal del cliente > DID Numbers > [Manage DID].

Debe seleccionar el número DID que desea aplicarle un desvio de llamadas y luego hacer clic en el botón "Editar DID" (el icono con el lápiz), también puede seleccionar más de un DID y hacer clic en los botones de "Edit Selection"

En este punto, debe estar en la página Edit DID Settings, la única configuración que debe cambiar es "Routing" . Debe seleccionar la opción "Call Forwarding" y luego seleccionar el número al cual desea desviar sus llamadas el cual encontrará en el menú desplegable a la derecha.

[[File:Cll_F01.png|none|700px|center]]
<br clear="all" />
Para finalizar, aplique los cambios y debe tener el Desvío de llamadas funcionando para el DID.
Nota: Cuando se aplica el desvío de llamadas a un DID, el tiempo de timbre de su DID aún afectará la llamada y la persona que llama irá al buzón de voz o a su conmutación por error (fail over options) si usted no responde la llamada. Ésto aplica siempre y cuando el buzón de voz o contestador automático del número al cual desvirá la llamada no termine primero que el de su número DID.

== Cargos ==
Tenga en cuenta que cuando desvía una llamada, se aplican los cargos entrantes normales de acuerdo con su plan DID y con la tarifa de terminación normal también se aplica al
número de destino durante la duración de la llamada. Por ejemplo, digamos que tiene un DID de Dallas con el plan Por minuto (tarifa entrante a $ 0.01) y desvía la llamada a un número de teléfono celular de EE. UU., la llamada de desvío se cobrará de la siguiente manera:

<nowiki>
(Tarifa entrante del número DID) + (tarifa de terminación al destino) = (costo total de la llamada por minuto)
$ 0.01 + $ 0.01 = $ 0.02 por minuto.
</nowiki>

=Creación masiva de entradas de desvío de llamadas=
En la parte superior derecha de la página, puede seleccionar « Importar desvíos » para cargar un archivo con todas las entradas que desea crear.

A continuación, se detalla información adicional sobre los archivos aceptados y los campos requeridos:
* Formato del archivo: CSV (valores separados por comas) con los campos en este orden exacto:
* Número, Anulación de Caller ID, Descripción, Dígitos DTMF, Pausa, Encabezado de desvío, Cliente revendedor
* Número (obligatorio): Número de teléfono de destino al que se reenviarán las llamadas. Debe incluir el código de país (ej.: 12141231234 para EE. UU.). No puede estar vacío.
* Anulación de Caller ID (opcional): Número de identificación de llamada que se mostrará al destino reenviado (ej.: 5551234567). Dejar vacío para usar el Caller ID original.
* Descripción (opcional): Nombre descriptivo para este desvío (ej.: « Mi oficina », « Teléfono de casa »). Dejar vacío si no es necesario.
* Dígitos DTMF (opcional): Dígitos numéricos enviados como tonos DTMF después de que la llamada sea contestada (ej.: 123 para extensión o navegación IVR). Dejar vacío si no es necesario.
* Pausa (opcional): Tiempo en segundos a esperar antes de enviar los dígitos DTMF (ej.: 0.5, 1.0, 2.5). Se aceptan valores decimales. Valor predeterminado: 0 si está vacío.
* Encabezado de desvío (opcional): Incluir el encabezado SIP Diversion en la llamada. Valores: yes para habilitar; no para deshabilitar. Predeterminado: no si está vacío.
* Cliente revendedor (opcional): ID de cliente para cuentas de revendedor (ej.: 12345). Use 0 si no está asignado. Predeterminado: 0 si está vacío.

'''Nota''': Solo una entrada de desvío por línea. Las líneas vacías serán ignoradas.

'''Nota''': Si se selecciona « Sobrescribir entradas de desvío de llamadas existentes », cualquier desvío existente con un número coincidente será reemplazado por los nuevos valores.

=Desvío de llamadas mediante la interfaz para revendedores=

La función está disponible para su cliente a través de la interfaz para revendedores. Debe habilitar esta función en su paquete para que pueda usarla.

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

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

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

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

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

A continuación, desplace hacia abajo y busque la función '''Call Forwarding'''', y actívela.

: [[File:CallForwarding_Reseller_4.png|thumb|none|500px]]

1) Para añadir una entrada de Desvío de llamadas a su cliente, o para ayudar a su cliente a añadir una. Vaya a ''[Services]'' en la barra de navegación de la izquierda, y luego a '''[Call Forwarding]'''
: [[File:CallForwarding_Add.png|thumb|none|400px]]

2) Haga clic en la pestaña Add new Call Forwarding y añada el '''Phone Number''' al cual se va a desviar la llamada. También puede introducir una '''Description''' para facilitar la búsqueda.
: [[File:CallForwarding_Add2.png|thumb|none|600px]]

3) Rellene el formulario y haga clic en el botón '''[Save Call Forwarding]'''.

Cuando crea el desvío de llamadas, podrá asignarlo a cualquier sección de enrutamiento. Como por ejemplo un IVR, enrutamiento DID, Opciones de conmutación por error, Grupo de timbre, Caceria de llamadas, etc...

[[category:guides]]

Renvoi d'appel

2026-01-28T20:25:57Z

RP:

{| class="wikitable"
|+
|-
! Article in English !! Artículo en Español
|-
| [https://wiki.voip.ms/article/Call_Forwarding English] || [https://wiki.voip.ms/article/Desv%C3%ADo_de_Llamadas_(Call_Forwarding) Español]
|}

__TOC__

Un '''Renvoi d'appel''' va permettre à un appel entrant d'être redirigé vers un téléphone mobile ou un autre numéro de téléphone. Vous pouvez choisir n'importe quel numéro (États-Unis, Canada ou International).

== Vidéo Tutoriel ==

[[Image:CallForwardThumbnail.png|200px|link=https://www.youtube.com/watch?v=k1NsFxBQ8gQ]]

== Configuration d'un renvoi ==

=== Créer un renvoi ===

La première étape serait de créer une entrée, vous pouvez le faire sur votre portail de client

Numéros DID >> [https://www.voip.ms/m/callforwarding.php Renvoi d'appel].

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

Après ceci vous devez cliquez sur "Ajouter Renvoi" et vous verrez les options suivantes:

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

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

'''Numéro de téléphone''': Entrez ici le numéro de téléphone souhaité pour rediriger les appels entrants. Pour les numéros des États-Unis ou du Canada, vous pouvez définir le numéro à 10 chiffres ou même utiliser le préfixe 1, par exemple 403XXXXXXX ou 1403XXXXXXX vous donnera le même résultat.

Pour les numéros '''internationaux''', assurez-vous d''''entrer le numéro avec le préfixe 001 ou 011''' (par exemple, pour un certain numéro 00152999XXXXXXX au Mexique). Vous pouvez également utiliser le 033 ou 044 pour remplacer la route [[Value vs Premium (Français)|Value ou Premium]]. Assurez-vous également que vous ayez activé les appels internationaux de votre [[Paramètres du compte#Restrictions_du_Compte|Compte]].

'''Outrepassement de l'identification de l'appelant''': Ce paramètre est facultatif et il vous permet d'envoyer l'[[ID de l'appelant]] de votre choix pour la destination afin de reconnaître où provient l'appel. Veuillez noter que l'[[ID de l'appelant]] dépendra de la route (value ou premium) que vous utilisez. Pour les numéros internationaux l'[[ID de l'appelant]] n'est pas garanti.

Par exemple, disons que vous avez votre téléphone portable pour le renvoi d'appel, vous pourrez mettre votre numéro DID comme ID de l'appelant. De cette façon, lorsque vous recevrez un appel entrant sur votre téléphone portable à partir du numéro DID, vous saurez que c'est une appel de VoIP.ms. Alternativement, si vous laissez ce paramètre vide, vous recevrez l'ID de l'appelant du numéro de la personne qui appelle votre numéro DID.

'''Description''': C'est également facultatif. Ceci vas vous aider à identifier l'entrée (renvoi d'appel) que vous allez créer.

'''Chiffres DTMF''': Ce paramètre est facultatif et vous permet d'entrer les chiffres qui seraient passés une fois que l'appel est connecté à votre numéro de renvoi d'appel. Par exemple, vous pourriez transférer un numéro à une [[Réceptionniste virtuelle IVR | Réceptionniste virtuelle (IVR)]] et passez un numéro de poste de cette façon.

'''Pause''': Le nombre de secondes que le système attendra avant d'envoyer les chiffres DTMF que vous avez définis avec l'option ci-dessus. C'est aussi un paramètre optionnel.

=== Configurez votre Numéro DID avec votre renvoi d'appel ===

Une fois que vous aurez créé une entrée de renvoi d'appel, vous pourrez le configurer avec le numéro DID que vous souhaitez sans avoir à le recréer. Vous devez vous rendre à votre portail client > Numéros DID > [[Gérer les numéros DID | Gestion des DID]]

Vous devez sélectionner le numéro DID que vous voulez renvoyer, puis cliquez sur le bouton '''Modifier DID''' (l'icône avec le crayon).

À cette étape, vous devriez être dans les paramètres du numéro DID, le seul paramètre que vous devez changer est la section '''Routage'''. Vous devez sélectionner l'option '''Renvoi d'appel''' puis sélectionnez le numéro dans le menu déroulant sur la droite.

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

== Frais ==

Veuillez noter que lorsque vous transférez un appel, les frais normaux entrants s'appliquent en fonction de votre forfait et le tarif des appels sortants est également appliqué pour le numéro de destination pour la durée de l'appel. Par exemple disons que vous avez un DID de Montréal avec le tarif par minute (débit entrant à 0,01 $) et vous transférez l'appel vers un numéro de téléphone cellulaire États-Unis, le renvoi d'appel sera facturé comme suit:

(Tarif d'appel entrant du numéro DID) + (Tarif de l'appel sortant) = (Coût total de votre appel)
$0.01 + $0.01 = $0.02 par minute.

=Création en masse des entrées de transfert d’appels=
En haut à droite de la page, vous pouvez sélectionner « Importer les transferts » pour téléverser un fichier contenant toutes les entrées que vous souhaitez créer.
Voici plus d’informations sur les fichiers acceptés et les champs requis :

*Format du fichier : CSV (valeurs séparées par des virgules) avec les champs dans cet ordre exact :
*Numéro, Remplacement du Caller ID, Description, Chiffres DTMF, Pause, En-tête de diversion, Client revendeur
*Numéro (obligatoire) : Numéro de téléphone de destination vers lequel les appels seront transférés. Doit inclure l’indicatif du pays (ex. : 12141231234 pour les États-Unis). Ne peut pas être vide.
*Remplacement du Caller ID (optionnel) : Numéro de l’identification de l’appelant à afficher vers la destination transférée (ex. : 5551234567). Laisser vide pour utiliser le Caller ID d’origine.
*Description (optionnelle) : Nom convivial pour ce transfert (ex. : « Mon bureau », « Téléphone domicile »). Laisser vide si non nécessaire.
*Chiffres DTMF (optionnel) : Chiffres numériques envoyés comme tonalités DTMF après la prise d’appel (ex. : 123 pour une extension ou une navigation IVR). Laisser vide si non nécessaire.
*Pause (optionnelle) : Temps en secondes à attendre avant l’envoi des chiffres DTMF (ex. : 0,5 ; 1,0 ; 2,5). Les valeurs décimales sont acceptées. Par défaut : 0 si vide.
*En-tête de diversion (optionnel) : Inclure l’en-tête SIP Diversion dans l’appel. Valeurs : yes pour activer ; no pour désactiver. Par défaut : no si vide.
*Client revendeur (optionnel) : ID client pour les comptes revendeurs (ex. : 12345). Utiliser 0 si non attribué. Par défaut : 0 si vide.

'''Note''': Une seule entrée de transfert par ligne. Les lignes vides seront ignorées.

'''Note''': Si l’option « Écraser les entrées de transfert d’appels existantes » est cochée, toute entrée existante avec un numéro correspondant sera remplacée par les nouvelles valeurs.

=Créer un renvoi en utilisant l'interface revendeur=

Cette fonction est disponible pour votre client via l'interface revendeur. Vous devez activer cette fonction dans votre forfait afin de leur donner la possibilité d'en tirer parti.

Allez sous la barre de navigation sur '''[Revendeur]''' puis cliquez sur '''[Gestion des tarifs et des forfaits]'''
: [[File:CallForwarding_Reseller_1_FR.png|thumb|none|300px]]

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

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

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

: [[File:CallForwarding_Reseller_3_FR.png|thumb|none|700px]]

Ensuite, faites défiler vers le bas et recherchez la fonction "Renvoi d'appel", puis activez-la.

: [[File:CallForwarding_Reseller_4_FR.png|thumb|none|500px]]

1) Pour ajouter une nouveau Renvoi d'appel à votre client, ou pour aider votre client à en ajouter une. Via la barre de navigation de gauche, allez sur l'option '''[Services]''', puis '''[Call Forwarding]'''.
: [[File:CallForwarding_Add.png|thumb|none|400px]]

2) Cliquez sur l'onglet "Add new Call Forwarding" et ajoutez le '''numéro de téléphone''' vers lequel le renvoi doit être effectué. Vous pouvez également entrer une '''description''' pour faciliter la recherche.
: [[File:CallForwarding_Add2.png|thumb|none|600px]]

3) Remplissez le formulaire et cliquez sur le bouton '''[Save Call Forwarding]'''.

Lorsque votre renvoi d'appel est créé, vous pourrez le sélectionner dans n'importe quelle section d'acheminement. Comme dans un IVR, routage DID, Basculement, Groupe de sonnerie, Recherche d'appel, etc...

[[category:guides en français]]

Renvoi d'appel

2026-01-28T20:24:40Z

RP: /* Configurez votre Numéro DID avec votre renvoi d'appel */

{| class="wikitable"
|+
|-
! Article in English !! Artículo en Español
|-
| [https://wiki.voip.ms/article/Call_Forwarding English] || [https://wiki.voip.ms/article/Desv%C3%ADo_de_Llamadas_(Call_Forwarding) Español]
|}

__TOC__

Un '''Renvoi d'appel''' va permettre à un appel entrant d'être redirigé vers un téléphone mobile ou un autre numéro de téléphone. Vous pouvez choisir n'importe quel numéro (États-Unis, Canada ou International).

== Vidéo Tutoriel ==

[[Image:CallForwardThumbnail.png|200px|link=https://www.youtube.com/watch?v=k1NsFxBQ8gQ]]

== Configuration d'un renvoi ==

=== Créer un renvoi ===

La première étape serait de créer une entrée, vous pouvez le faire sur votre portail de client

Numéros DID >> [https://www.voip.ms/m/callforwarding.php Renvoi d'appel].

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

Après ceci vous devez cliquez sur "Ajouter Renvoi" et vous verrez les options suivantes:

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

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

'''Numéro de téléphone''': Entrez ici le numéro de téléphone souhaité pour rediriger les appels entrants. Pour les numéros des États-Unis ou du Canada, vous pouvez définir le numéro à 10 chiffres ou même utiliser le préfixe 1, par exemple 403XXXXXXX ou 1403XXXXXXX vous donnera le même résultat.

Pour les numéros '''internationaux''', assurez-vous d''''entrer le numéro avec le préfixe 001 ou 011''' (par exemple, pour un certain numéro 00152999XXXXXXX au Mexique). Vous pouvez également utiliser le 033 ou 044 pour remplacer la route [[Value vs Premium (Français)|Value ou Premium]]. Assurez-vous également que vous ayez activé les appels internationaux de votre [[Paramètres du compte#Restrictions_du_Compte|Compte]].

'''Outrepassement de l'identification de l'appelant''': Ce paramètre est facultatif et il vous permet d'envoyer l'[[ID de l'appelant]] de votre choix pour la destination afin de reconnaître où provient l'appel. Veuillez noter que l'[[ID de l'appelant]] dépendra de la route (value ou premium) que vous utilisez. Pour les numéros internationaux l'[[ID de l'appelant]] n'est pas garanti.

Par exemple, disons que vous avez votre téléphone portable pour le renvoi d'appel, vous pourrez mettre votre numéro DID comme ID de l'appelant. De cette façon, lorsque vous recevrez un appel entrant sur votre téléphone portable à partir du numéro DID, vous saurez que c'est une appel de VoIP.ms. Alternativement, si vous laissez ce paramètre vide, vous recevrez l'ID de l'appelant du numéro de la personne qui appelle votre numéro DID.

'''Description''': C'est également facultatif. Ceci vas vous aider à identifier l'entrée (renvoi d'appel) que vous allez créer.

'''Chiffres DTMF''': Ce paramètre est facultatif et vous permet d'entrer les chiffres qui seraient passés une fois que l'appel est connecté à votre numéro de renvoi d'appel. Par exemple, vous pourriez transférer un numéro à une [[Réceptionniste virtuelle IVR | Réceptionniste virtuelle (IVR)]] et passez un numéro de poste de cette façon.

'''Pause''': Le nombre de secondes que le système attendra avant d'envoyer les chiffres DTMF que vous avez définis avec l'option ci-dessus. C'est aussi un paramètre optionnel.

=== Configurez votre Numéro DID avec votre renvoi d'appel ===

Une fois que vous aurez créé une entrée de renvoi d'appel, vous pourrez le configurer avec le numéro DID que vous souhaitez sans avoir à le recréer. Vous devez vous rendre à votre portail client > Numéros DID > [[Gérer les numéros DID | Gestion des DID]]

Vous devez sélectionner le numéro DID que vous voulez renvoyer, puis cliquez sur le bouton '''Modifier DID''' (l'icône avec le crayon).

À cette étape, vous devriez être dans les paramètres du numéro DID, le seul paramètre que vous devez changer est la section '''Routage'''. Vous devez sélectionner l'option '''Renvoi d'appel''' puis sélectionnez le numéro dans le menu déroulant sur la droite.

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

=Création en masse des entrées de transfert d’appels=
En haut à droite de la page, vous pouvez sélectionner « Importer les transferts » pour téléverser un fichier contenant toutes les entrées que vous souhaitez créer.
Voici plus d’informations sur les fichiers acceptés et les champs requis :

*Format du fichier : CSV (valeurs séparées par des virgules) avec les champs dans cet ordre exact :
*Numéro, Remplacement du Caller ID, Description, Chiffres DTMF, Pause, En-tête de diversion, Client revendeur
*Numéro (obligatoire) : Numéro de téléphone de destination vers lequel les appels seront transférés. Doit inclure l’indicatif du pays (ex. : 12141231234 pour les États-Unis). Ne peut pas être vide.
*Remplacement du Caller ID (optionnel) : Numéro de l’identification de l’appelant à afficher vers la destination transférée (ex. : 5551234567). Laisser vide pour utiliser le Caller ID d’origine.
*Description (optionnelle) : Nom convivial pour ce transfert (ex. : « Mon bureau », « Téléphone domicile »). Laisser vide si non nécessaire.
*Chiffres DTMF (optionnel) : Chiffres numériques envoyés comme tonalités DTMF après la prise d’appel (ex. : 123 pour une extension ou une navigation IVR). Laisser vide si non nécessaire.
*Pause (optionnelle) : Temps en secondes à attendre avant l’envoi des chiffres DTMF (ex. : 0,5 ; 1,0 ; 2,5). Les valeurs décimales sont acceptées. Par défaut : 0 si vide.
*En-tête de diversion (optionnel) : Inclure l’en-tête SIP Diversion dans l’appel. Valeurs : yes pour activer ; no pour désactiver. Par défaut : no si vide.
*Client revendeur (optionnel) : ID client pour les comptes revendeurs (ex. : 12345). Utiliser 0 si non attribué. Par défaut : 0 si vide.

'''Note''': Une seule entrée de transfert par ligne. Les lignes vides seront ignorées.

'''Note''': Si l’option « Écraser les entrées de transfert d’appels existantes » est cochée, toute entrée existante avec un numéro correspondant sera remplacée par les nouvelles valeurs.

== Frais ==

Veuillez noter que lorsque vous transférez un appel, les frais normaux entrants s'appliquent en fonction de votre forfait et le tarif des appels sortants est également appliqué pour le numéro de destination pour la durée de l'appel. Par exemple disons que vous avez un DID de Montréal avec le tarif par minute (débit entrant à 0,01 $) et vous transférez l'appel vers un numéro de téléphone cellulaire États-Unis, le renvoi d'appel sera facturé comme suit:

(Tarif d'appel entrant du numéro DID) + (Tarif de l'appel sortant) = (Coût total de votre appel)
$0.01 + $0.01 = $0.02 par minute.

=Créer un renvoi en utilisant l'interface revendeur=

Cette fonction est disponible pour votre client via l'interface revendeur. Vous devez activer cette fonction dans votre forfait afin de leur donner la possibilité d'en tirer parti.

Allez sous la barre de navigation sur '''[Revendeur]''' puis cliquez sur '''[Gestion des tarifs et des forfaits]'''
: [[File:CallForwarding_Reseller_1_FR.png|thumb|none|300px]]

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

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

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

: [[File:CallForwarding_Reseller_3_FR.png|thumb|none|700px]]

Ensuite, faites défiler vers le bas et recherchez la fonction "Renvoi d'appel", puis activez-la.

: [[File:CallForwarding_Reseller_4_FR.png|thumb|none|500px]]

1) Pour ajouter une nouveau Renvoi d'appel à votre client, ou pour aider votre client à en ajouter une. Via la barre de navigation de gauche, allez sur l'option '''[Services]''', puis '''[Call Forwarding]'''.
: [[File:CallForwarding_Add.png|thumb|none|400px]]

2) Cliquez sur l'onglet "Add new Call Forwarding" et ajoutez le '''numéro de téléphone''' vers lequel le renvoi doit être effectué. Vous pouvez également entrer une '''description''' pour faciliter la recherche.
: [[File:CallForwarding_Add2.png|thumb|none|600px]]

3) Remplissez le formulaire et cliquez sur le bouton '''[Save Call Forwarding]'''.

Lorsque votre renvoi d'appel est créé, vous pourrez le sélectionner dans n'importe quelle section d'acheminement. Comme dans un IVR, routage DID, Basculement, Groupe de sonnerie, Recherche d'appel, etc...

[[category:guides en français]]

Call Forwarding

2026-01-28T20:23:13Z

RP: /* Bulk Creation of Call Forwarding Entries */

{| class="wikitable"
|+
|-
! Article en Français !! Artículo en Español
|-
| [https://wiki.voip.ms/article/Renvoi_d%27appel Français] || [https://wiki.voip.ms/article/Desv%C3%ADo_de_Llamadas_(Call_Forwarding) Español]
|}

__TOC__

A '''Call Forwarding''' allows an incoming call to be redirected to a mobile telephone or other telephone number where the desired called party is able to answer. You can set any number, even international numbers.

== Tutorial Video ==

[[Image:CallForwardThumbnail.png|200px|link=https://www.youtube.com/watch?v=k1NsFxBQ8gQ]]

== Setup a Call Forward ==

=== Create the Call Forwarding entry ===
<br clear="all" />

*The first step would be to create a Call Forwarding entry, you can do this from your Customer Portal>> DID Numbers>> Call Forwarding.

<div><ul>
<li style="display: inline-block;"> [[File:CFWD120250803v2.png|thumb|none|500px|Call Forwarding menu]] </li>
<li style="display: inline-block;"> [[File:CFWD220250803.png|thumb|none|530px|Call Forwarding page]] </li>
</ul></div>

*After that you only need to click on the "Add Forwarding" button and you will see the follow screen:
<br clear="all" />
[[File:CFWD320250803.png|thumb|none|650px|Call Forwarding entry]]
<br clear="all" />

'''Phone Number''': Enter here the phone number that you want the redirected incoming calls to your DID to be sent to. For USA or Canada numbers you can set the number with 10 digits or even using the prefix 1, for example 403XXXXXXX or 1403XXXXXXX will give you the same result. For international numbers please make sure to enter the number with prefix 00 or 011 (e.g. 0052999XXXXXXX for a number in Mexico), you can also use the 033 or 044 to override the [[Value vs Premium|Value or Premium]] route. Also make sure that you have enabled the International Calls in your [[Account Settings|account]].

'''CallerID Override''': This setting is optional and it lets you send the [[Caller ID]] of your choice to the destination in order to recognize where the call comes from, please note that the [[caller ID]] number reliability will depend on the route that you're using and for International Routes the [[Caller ID]] is not 100% guaranteed.

Note: If the Call Forwarding with a CallerID Override is assigned to a Ring Group then that CallerID Override Number will
be on all calls sent through that Ring Group.

For example lets say that you have your Cellphone set as call forward, you can put your DID number as [[caller ID]] override and this way when you see an incoming call in your cellphone from the DID number, you will known that this is a call from voip.ms. If you leave this setting blank, you will receive the [[caller ID]] number of the person calling your DID number.

'''Description''': This is also optional, this can help you identify a particular Call Forwarding entry.

'''DTMF Digits''': This setting is optional, and would allow you to enter the digits that would be dialed once the call is connected to your call forward number. For example, you can forward a number to an [[Digital Receptionist (IVR)|ivr]] and automatically send an extension number this way.

'''Pause''': The amount of seconds the system would wait before it sends the DTMF digits you set with the option above. This is also an optional setting.

=== Route incoming calls from DID to your Call Forwarding ===

Once you have created a call forwarding entry, you can assign it to as many DID numbers as you want without needing to create it again.
You need to go to your Customer Portal >> DID Numbers >> [[Manage DID]].

You need to select the DID number you want to forward and then click on '''Edit DID''' button (the icon with the pencil), also you can select more than one DID and click on the '''Edit Selection''' buttons.

At this point you should be in the Edit DID Settings page, the only setting you should change is the '''Routing'''. You need to select the '''Call Forwarding''' option and then select the number from the drop down menu on the right.

[[File:Cll_F01.png|none|700px|center]]
<br clear="all" />

To finish, apply the changes and you should have the Call Forwarding working for the DID.

Note: When Call Forwarding a DID your DID`s Ring Time still affects the call and the caller will go to Voicemail or your Failover for
No Answer, providing your Destination does not time out first (Cell Voicemail or Answering Machine).

== Charges ==

Please note that when you forward a call, normal inbound charges apply according to your DID plan and the normal termination rate is also applied for the destination number for the duration of the call. For example let's say that you have a DID from Dallas with the Per minute plan (incoming rate at $0.01) and you forward the call to an US cellphone number, the forward call will be charged as follow:

<nowiki>
(Incoming rate of the DID number) + (termination rate to the destination) = (total cost of call per minute)
$0.01 + $0.01 = $0.02 per minute.
</nowiki>

= Reseller Configuration =

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

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

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

=Bulk Creation of Call Forwarding Entries=
On the top right of the page, you can select '''Import Forwarding''' to upload a file with all the entries you wish to create.
Here's some more information on the files accepted and required fields:

* File Format: CSV (Comma Separated Value) with fields in this exact order:
* Number, CallerID Override, Description, DTMF Digits, Pause, Diversion Header, Reseller Client
* Number (Required): The destination phone number where calls will be forwarded. Must include country code (e.g., 12141231234 for USA). Cannot be empty.
* CallerID Override (Optional): The caller ID number to display to the forwarded destination (e.g., 5551234567). Leave empty to use the original caller ID.
* Description (Optional): A friendly name for this forwarding (e.g., "My Office", "Home Phone"). Leave empty if not needed.
* DTMF Digits (Optional): Numeric digits to be sent as DTMF tones after the call is answered (e.g., 123 for extension or IVR navigation). Leave empty if not needed.
* Pause (Optional): Time in seconds to wait before sending DTMF digits (e.g., 0.5, 1.0, 2.5). Accepts decimal values. Defaults to 0 if empty.
* Diversion Header (Optional): Include SIP Diversion Header in the call. Values: yes to enable; no to disable. Defaults to no if empty.
* Reseller Client (Optional): Client ID for reseller accounts (e.g., 12345). Use 0 for not assigned. Defaults to 0 if empty.

'''Note''': Only one forwarding per line. Empty lines will be ignored.

'''Note''': If "Overwrite Existing Call Forwarding Entries" is checked, any existing forwarding with a matching Number will be replaced with the new values.

=Configuration Using the Reseller Interface=

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

Go under the navigation bar on '''[Reseller]''' then click on '''[Manage Rates & Packages]'''
: [[File:CallForwarding_Reseller_1.png|thumb|none|300px]]

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

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

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

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

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

: [[File:CallForwarding_Reseller_4.png|thumb|none|500px]]

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

2) Click on the tab Add new Call Forwarding and the '''Phone Number''' to forward to. You can also enter a '''Description''' for searchability.
: [[File:CallForwarding_Add2.png|thumb|none|600px]]

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

When your call forwarding is created, you will be able to select it any routing section. Such as in an IVR, DID Routing, Failover, Ring Group, Call Hunting, etc...

[[category:guides]]

Call Forwarding

2026-01-28T20:22:58Z

RP: /* Reseller Configuration */

{| class="wikitable"
|+
|-
! Article en Français !! Artículo en Español
|-
| [https://wiki.voip.ms/article/Renvoi_d%27appel Français] || [https://wiki.voip.ms/article/Desv%C3%ADo_de_Llamadas_(Call_Forwarding) Español]
|}

__TOC__

A '''Call Forwarding''' allows an incoming call to be redirected to a mobile telephone or other telephone number where the desired called party is able to answer. You can set any number, even international numbers.

== Tutorial Video ==

[[Image:CallForwardThumbnail.png|200px|link=https://www.youtube.com/watch?v=k1NsFxBQ8gQ]]

== Setup a Call Forward ==

=== Create the Call Forwarding entry ===
<br clear="all" />

*The first step would be to create a Call Forwarding entry, you can do this from your Customer Portal>> DID Numbers>> Call Forwarding.

<div><ul>
<li style="display: inline-block;"> [[File:CFWD120250803v2.png|thumb|none|500px|Call Forwarding menu]] </li>
<li style="display: inline-block;"> [[File:CFWD220250803.png|thumb|none|530px|Call Forwarding page]] </li>
</ul></div>

*After that you only need to click on the "Add Forwarding" button and you will see the follow screen:
<br clear="all" />
[[File:CFWD320250803.png|thumb|none|650px|Call Forwarding entry]]
<br clear="all" />

'''Phone Number''': Enter here the phone number that you want the redirected incoming calls to your DID to be sent to. For USA or Canada numbers you can set the number with 10 digits or even using the prefix 1, for example 403XXXXXXX or 1403XXXXXXX will give you the same result. For international numbers please make sure to enter the number with prefix 00 or 011 (e.g. 0052999XXXXXXX for a number in Mexico), you can also use the 033 or 044 to override the [[Value vs Premium|Value or Premium]] route. Also make sure that you have enabled the International Calls in your [[Account Settings|account]].

'''CallerID Override''': This setting is optional and it lets you send the [[Caller ID]] of your choice to the destination in order to recognize where the call comes from, please note that the [[caller ID]] number reliability will depend on the route that you're using and for International Routes the [[Caller ID]] is not 100% guaranteed.

Note: If the Call Forwarding with a CallerID Override is assigned to a Ring Group then that CallerID Override Number will
be on all calls sent through that Ring Group.

For example lets say that you have your Cellphone set as call forward, you can put your DID number as [[caller ID]] override and this way when you see an incoming call in your cellphone from the DID number, you will known that this is a call from voip.ms. If you leave this setting blank, you will receive the [[caller ID]] number of the person calling your DID number.

'''Description''': This is also optional, this can help you identify a particular Call Forwarding entry.

'''DTMF Digits''': This setting is optional, and would allow you to enter the digits that would be dialed once the call is connected to your call forward number. For example, you can forward a number to an [[Digital Receptionist (IVR)|ivr]] and automatically send an extension number this way.

'''Pause''': The amount of seconds the system would wait before it sends the DTMF digits you set with the option above. This is also an optional setting.

=== Route incoming calls from DID to your Call Forwarding ===

Once you have created a call forwarding entry, you can assign it to as many DID numbers as you want without needing to create it again.
You need to go to your Customer Portal >> DID Numbers >> [[Manage DID]].

You need to select the DID number you want to forward and then click on '''Edit DID''' button (the icon with the pencil), also you can select more than one DID and click on the '''Edit Selection''' buttons.

At this point you should be in the Edit DID Settings page, the only setting you should change is the '''Routing'''. You need to select the '''Call Forwarding''' option and then select the number from the drop down menu on the right.

[[File:Cll_F01.png|none|700px|center]]
<br clear="all" />

To finish, apply the changes and you should have the Call Forwarding working for the DID.

Note: When Call Forwarding a DID your DID`s Ring Time still affects the call and the caller will go to Voicemail or your Failover for
No Answer, providing your Destination does not time out first (Cell Voicemail or Answering Machine).

== Charges ==

Please note that when you forward a call, normal inbound charges apply according to your DID plan and the normal termination rate is also applied for the destination number for the duration of the call. For example let's say that you have a DID from Dallas with the Per minute plan (incoming rate at $0.01) and you forward the call to an US cellphone number, the forward call will be charged as follow:

<nowiki>
(Incoming rate of the DID number) + (termination rate to the destination) = (total cost of call per minute)
$0.01 + $0.01 = $0.02 per minute.
</nowiki>

= Reseller Configuration =

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

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

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

=Bulk Creation of Call Forwarding Entries=
On the top right of the page, you can select '''Import Forwarding''' to upload a file with all the entries you wish to create.
Here's some more information on the files accepted and required fields:

* File Format: CSV (Comma Separated Value) with fields in this exact order:
* Number, CallerID Override, Description, DTMF Digits, Pause, Diversion Header, Reseller Client
* Number (Required): The destination phone number where calls will be forwarded. Must include country code (e.g., 12141231234 for USA). Cannot be empty.
* CallerID Override (Optional): The caller ID number to display to the forwarded destination (e.g., 5551234567). Leave empty to use the original caller ID.
* Description (Optional): A friendly name for this forwarding (e.g., "My Office", "Home Phone"). Leave empty if not needed.
* DTMF Digits (Optional): Numeric digits to be sent as DTMF tones after the call is answered (e.g., 123 for extension or IVR navigation). Leave empty if not needed.
* Pause (Optional): Time in seconds to wait before sending DTMF digits (e.g., 0.5, 1.0, 2.5). Accepts decimal values. Defaults to 0 if empty.
* Diversion Header (Optional): Include SIP Diversion Header in the call. Values: yes to enable; no to disable. Defaults to no if empty.
* Reseller Client (Optional): Client ID for reseller accounts (e.g., 12345). Use 0 for not assigned. Defaults to 0 if empty.

'''Note''': Only one forwarding per line. Empty lines will be ignored.
'''Note''': If "Overwrite Existing Call Forwarding Entries" is checked, any existing forwarding with a matching Number will be replaced with the new values.

=Configuration Using the Reseller Interface=

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

Go under the navigation bar on '''[Reseller]''' then click on '''[Manage Rates & Packages]'''
: [[File:CallForwarding_Reseller_1.png|thumb|none|300px]]

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

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

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

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

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

: [[File:CallForwarding_Reseller_4.png|thumb|none|500px]]

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

2) Click on the tab Add new Call Forwarding and the '''Phone Number''' to forward to. You can also enter a '''Description''' for searchability.
: [[File:CallForwarding_Add2.png|thumb|none|600px]]

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

When your call forwarding is created, you will be able to select it any routing section. Such as in an IVR, DID Routing, Failover, Ring Group, Call Hunting, etc...

[[category:guides]]

Sub Cuentas (Sub Accounts)

2026-01-28T20:17:44Z

RP: /* Configuración de Revendedor (Reseller Configuration) */

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

Tener una subcuenta le permite registrar más de un dispositivo para hacer o recibir llamadas simultáneamente, también puede usarla como una extensión interna para su oficina o incluso su casa. Muchas de las funciones de voip.ms utilizan subcuentas.
Con esta guía aprenderemos a crear y utilizar esta función correctamente.

__TOC__

== Cómo crear una Subcuenta ==

Puede crear tantas subcuentas como sea necesario y no hay ningún cargo adicional por hacerlo. Primero debe ingresar a su portal de clientes y seleccionar "Create Sub Account"(Crear Sub cuentas) en el menú de subcuentas (under Sub account menu).

[[File:Create subacc.jpg|800px]]

=== Opciones Básicas ===

Estas son las opciones básicas que necesita configurar para poder crear una Subcuenta.

'''Protocol''': Aquí puede seleccionar cuál protocolo utilizara con esta subcuenta, puede elegir entre SIP o IAX2. Esta opción dependerá del dispositivo que utilizará con esta subcuenta. SIP es el protocolo recomendado y más utilizado.

'''Authentication type''': Esta opción no está disponible si utiliza la cuenta principal. Sin embargo para las subcuenta(s) usted puede elegir entre '''User/Password Authentication''' o '''IP Authentication'''. Esta opción depende del dispositivo que está utilizando.

-- '''User/Password Authentication''': Esta es la opción recomendada y la soportada por la mayoría de los dispositivos.

-- '''IP Authentication''': Recomendada sólo para usuarios avanzados. Mayormente es utilizada para servidores Asterisk o [[PBXs|PBX]]. Cuando elige esta opción usted puede indicar cuál es la dirección IP de su dispositivo/servidor.

'''Note:''' Cuando usa la autenticación por IP, el estado del registro (Registration Status) no será desplegado en la página principal de su portal de usuario VoIP.ms.

'''Username''': Este es el nombre de usuario para la subcuenta. Es posible utilizar caracteres alfanuméricos en este campo (asegúrese que su dispositivo también soporta los caracteres alfanuméricos en el campo de usuario). El nombre de usuario es indicado con el siguiente formato: '''{Main Account}_{username}'''.

Por ejemplo, digamos que su cuenta es ''100000'' y usted asigna el nombre de usuario como ''casa'', el nombre de usuario que deberá utilizar en su dispositivo será ''100000_casa''.

'''Password/IP Address''': Aquí puede ingresar la contraseña que usará con esta subcuenta. La contraseña debe tener un mínimo de 6 caracteres, aunque es recomendable usar una contraseña con más caracteres y que ademas incluya caracteres alfanuméricos. Es muy recomendable cambiar la contraseña cada par de meses, como una recomendación de seguridad.

Si está utilizando IP Authentication aquí puede ingresar la dirección IP de su servidor/dispositivo. El formato es "X.X.X.X", un ejemplo sería: 201.202.203.204.

[[File:IP AUTH.jpg|800px]]

'''Trunk''': Esta opción solamente estará disponible si el protocolo elegido para la subcuenta es IAX2. Puede dejar seleccionada la opción estándar ("Send mini voice packets"), o seleccionar "Send trunk packets" para agrupar la información de voz saliente en paquetes de troncal, cada uno con su propia marca de tiempo.

[[File:Trunk.jpg]]

'''Device type''': Es importante que se asegure de seleccionar el tipo de dispositivo correcto que se utilizará con esta subcuenta para poder recibir llamadas entrantes de forma adecuada en el dispositivo.

'''CallerID Number''': Aquí puede ingresar el número de [[Numero Identificador (Caller ID)|callerID]] que desea enviar, esto funcionará si utiliza un dispositivo [[Devices|ATA]], teléfono IP o [[Softphones|Softphone]], y su dispositivo no es capaz de enviar el [[Numero Identificador (Caller ID)|callerID]]. Asegúrese de ingresar un número válido para garantizar que la llamada se conecte. De igual forma utilice únicamente números en este campo y el [[Numero Identificador (Caller ID)|callerID]] debe ser de 10 dígitos. ''Opcional''

'''Allow International Calls''': Con esta opción podrá habilitar o des-habilitar las llamadas internacionales para esta subcuenta.

'''International Route''': Con esta opción puede seleccionar la ruta que se utilizará para realizar llamadas internacionales (llamadas que NO sean a EEUU o Canadá) con esta subcuenta. Llamadas a Estados Unidos o Canadá usarán la ruta USA48/Canada que tenga seleccionada en sus [[Configuraciones de la Cuenta (Account Settings)|account settings]].

'''Music on Hold''': Esta opción le permite seleccionar si desea o no que la persona que le llame a su número o extensión escuche música mientras la llamada se encuentra en espera.
<br>
Las siguientes opciones para MOH (Music on Hold) se encuentran disponibles: <br>

*'''No Music, Intermittent Bleep:''' Silencio, pero con un sutil e intermitente sonido para que la persona en espera sepa que aun esta en la linea.
*'''Away in the Tropics:''' Música caribeña, estas canciones ofrecen sonidos de ukelele, tambores y guitarras.
*'''Coffee and Sunrise:''' Música animada sin ser festiva, y positiva sin ser muy sonriente.
*'''Coffee Shop Acoustic:''' Música relajada, el sonido de la guitarra acústica es ideal para crear una buena atmósfera.
*'''Easy Listening:''' Música suave y casual.
*'''Guitar Alchemy:''' Bellas armonías y secuencias progresivas de cuerdas que crean una cálida experiencia musical
*'''Happy Endings:''' Música animada con un estilo comercial. Guitarras, tambores, ukelele, armónica y campanas.
*'''Light and Casual:''' Música tranquila con sensación positiva
*'''Orchestral Moods:''' Emotivos y dramáticos cuentos contados por violines, pianos y orquestas completas
*'''Piano Mix:''' Melodías suaves de piano.
*'''Rock Me Easy:''' Música agradable para crear un ambiente de relajación.
*'''Spa Sounds:''' Música instrumental suave, lenta y tranquila.

Usando los siguientes codigos, usted puede probar las diferentes opciones de MOH: <br>

'''*** 89''' Away in the Tropics <br>
'''*** 90''' Coffee and Sunrise <br>
'''*** 91''' Coffee Shop Acoustic <br>
'''*** 92''' Easy Listening <br>
'''*** 93''' Guitar Alchemy <br>
'''*** 94''' Happy Endings <br>
'''*** 95''' Light and Casual <br>
'''*** 96''' Orchestral Moods <br>
'''*** 97''' Piano Mix <br>
'''*** 98''' Rock Me Easy <br>
'''*** 99''' Spa Sounds<br>
'''***100''' No Music, Intermittent Bleep<br>

'''Account name or description''': Con esta opción podrá identificar fácilmente cada subcuenta.

=== Opciones Avanzadas ===

Con estas opciones puede determinar cuál codec va a permitir utilizar, el modo de DTMF y las opciones de NAT para cada subcuenta. Estas opciones son recomendadas para usuarios avanzados.

[[File:Subacc adv opt.jpg|800px]]

'''Allowed Codecs:''' Esta opción le permite seleccionar los codecs que se utilizarán con esta subcuenta, puede seleccionar si desea habilitar todos los codecs o sólo algunos.
'''Nota:''' Es recomendable que seleccione '''Allow All''' y sólo lo cambie si tiene alguna razón específica para hacerlo.

'''DTMF Mode:''' Esto le permite seleccionar el modo de DTMF (tonos) que se utilizará con esta subcuenta. Si selecciona ''AUTO'' el modo ''RFC2833 (AVT)'' va ser utilizado por defecto y se cambiará automáticamente a ''INBAND'' si una de las dos partes de la llamada no soporta ''RFC2833''.

'''Nota''': Es recomendable, mas no es un requisito, que seleccione el mismo modo de DTMF en su dispositivo.

'''NAT (Network Address Translation)''': Ponga esta opción en '''YES''' si está utilizando NAT. Si no está seguro de qué significa esta opción, es recomendable que la deje en '''YES'''.

=== Opciones Adicionales ===

Aunque estas son consideradas opciones adicionales, son utilizadas para asignar un número de extensión interna, buzón de voz y el tiempo de timbrado para la subcuenta.

[[File:Subcuentaoptional2.png|800px]]

'''Internal Extension Number''': Aquí puede asignar el número de extensión interno que se usará para realizar llamadas entre subcuentas. El número de extensión que ingrese en este campo tendrá el prefijo ''10'', puede ingresar entre 1 a 10 dígitos. Por ejemplo si usted ingresa ''55'' el número de extensión resultante será ''1055''.

'''Nota''': Asegúrese que las subcuentas se encuentran registradas al mismo servidor de VoIP.ms, para que pueda realizar las llamadas internas.
Las llamadas entre extensiones son gratuitas para quien envía la llamada y también para quien la recibe.

'''Habilitar identificador de llamadas interno/nombre de identificador de llamadas interno:''' Le permite establecer un nombre de identificador de llamadas específico que se verá solo en llamadas internas entre usted y los usuarios de su subcuenta. Además, su número de extensión se mostrará en sus llamadas internas en lugar de su número de identificación de llamadas de 10 dígitos.

==== Subcuenta como SIP URI externa ====
----
Para utilizar una subcuenta como una [[Dirección URI (SIP URI)|SIP URI]] externa, solamente es necesario que primero la habilite como una extensión interna. Por ejemplo, digamos que su extensión interna es 2 (102 con el prefijo 10), usted puede alcanzar la subcuenta vía SIP desde otra red con la siguiente dirección URI: '''1000002@server.voip.ms''' (Reemplace server.voip.ms con el servidor al que la subcuenta esta conectado y el numero 2 con el numero de su extensión interna).
----

'''Internal Extension VoiceMail''': Aquí puede elegir cual [[Buzón de voz (Voicemail)|buzón de voz]] estará asociado con esta subcuenta.

'''Nota:''' Si asocia un [[Buzón de voz (Voicemail)|buzón de voz]] con su subcuenta, cuando haya nuevos mensajes en su buzón, se enviará una notificación a su dispositivo.
Esto llevará a diferentes resultados dependiendo del tipo de dispositivo que utilice. Usualmente, usted recibirá un tono distintivo
o una luz indicadora parpadeando.

'''Internal Extension Ringing Time''': Este es el tiempo que su teléfono se mantendrá timbrando cuando reciba una llamada directamente a la extensión. Cada 5 segundos equivalen a un timbre.

==== Interceptación de llamadas (Call Pick-up) ====

La función **Interceptación_de_llamadas** permite que un dispositivo registrado bajo una subcuenta pueda contestar una llamada entrante dirigida a otro dispositivo registrado bajo una subcuenta diferente.

'''Por ejemplo:'''
Si tanto Alex como Bob están recibiendo llamadas en sus respectivos dispositivos y Bob necesita ausentarse, Alex puede contestar la llamada que iba dirigida a Bob directamente desde su propio dispositivo, asegurando que la llamada no se pierda.

Esta funcionalidad es especialmente útil en entornos de trabajo compartidos, como centros de atención telefónica o recepciones, donde los miembros del equipo pueden apoyarse mutuamente atendiendo llamadas en nombre de otros.

Para más información sobre cómo usar la función de Interceptación de llamadas, consulta este [[Interceptación_de_llamadas | artículo]].

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

=== Configuración de Revendedor (Reseller Configuration) ===

Si usted utiliza la interfaz de revendedor ([[Guía Básica de Reseller|Reseller Interface]]), puede asociar cada subcuenta con uno de sus clientes.

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

'''Reseller Client''': Aquí puede seleccionar el cliente con el cual desea asociar la subcuenta. Primero necesitará crear la cuenta de su cliente en la sección de [[Guía Básica de Reseller|Reseller]] en su portal de usuario.

'''Reseller client package''': Aquí puede elegir el paquete que desea asignarle a su cliente. Primero necesita crear el paquete en la sección de [[Guía Básica de Reseller|Reseller]].

'''next billing date''': Aquí puede asignar la próxima fecha de cobro. Usualmente el sistema asigna este valor en forma automática, pero si usted cambio el paquete asociado a esta subcuenta en forma manual, probablemente desee ajustar la próxima fecha de cobro.

'''Charge setup fees now''': Una vez que seleccione esta opción, los cargos mensuales por el paquete serán cargados a la cuenta de su cliente. Usted puede utilizar esta opción cuando haya realizado algún cambio en el paquete del cliente.

==Creación masiva de subcuentas==
También puede crear subcuentas de forma masiva desde su portal de clientes. Para hacerlo, acceda a su portal de clientes y vaya a Subcuentas > Administrar subcuentas. Una vez allí, en la parte superior derecha de la página encontrará la opción Importar subcuentas. Haga clic en ella, descargue la plantilla, complétela y cárguela. Esto creará automáticamente las subcuentas según la plantilla cargada.

== Reportes de Subcuentas (Sub Account Reports) ==

Usted puede ver la cantidad de minutos, llamadas y el total que ha gastado por cada subcuenta. Esta información es útil incluso si usted no hace uso de la interfaz de reseller. Puede acceder a esta opción desde el sub menú '''Sub Accounts Reports''' que se encuentra en el menú '''Sub Accounts''' en su portal de usuario.

[[File:Subacc report.jpg|800px]]

'''Report Range''': Es posible desplegar un reporte con un rango de hasta 92 días (3 meses).

'''Minutes''': Este valor muestra el total de minutos que han sido utilizados por esta subcuenta durante el periodo seleccionado. Los valores están expresados usando tiempo decimal.

'''Calls''': Este valor muestra el número total de llamadas realizadas por la subcuenta durante el periodo seleccionado.

'''Amount Spent''': Este es el total que se ha gastado con este subcuenta durante el período seleccionado. Expresado en dólares americanos.

== Uso de las subcuentas ==

Una vez que haya creado una subcuenta, es posible utilizarla con la mayoría de las herramientas que están disponibles dentro de su portal de usuario. Por ejemplo puede usarla con su [[Recepcionista Digital|IVR]], como uno de los agentes para recibir las llamadas de su [[Llamadas en Cola (Calling Queues)|calling queue]], utilizarlas directamente con sus número DID, como una [[Dirección URI (SIP URI)|SIP URI]] externa para poder recibir llamadas desde otras redes, etc.

[[category:Guías]]

Sous Comptes

2026-01-28T20:17:15Z

RP: /* Sécurité */

{| class="wikitable"
|+
|-
! Article in English !! Artículo en Español
|-
| [https://wiki.voip.ms/article/Sub_Accounts English] ||
[https://wiki.voip.ms/article/Sub_Cuentas_(Sub_Accounts) Español]
|}

Avoir un sous-compte vous offre la possibilité d’enregistrer plus d’un appareil pour recevoir et faire des appels, ou même de recevoir des appels sur plus d’un appareil simultanément. Vous pouvez également utiliser les extensions internes pour votre bureau ou pour votre domicile. Beaucoup d’autres options font l’usage des sous-comptes. Avec ce guide, vous serez en mesure de créer et d’utiliser cette option correctement.

__TOC__

== Comment créer un sous-compte ==

Vous pouvez créer autant de sous-comptes que vous le désirez et il n’y a pas de frais additionnel. Vous devrez premièrement aller dans votre portail client, Sous-Comptes > [https://www.voip.ms/m/subaccount.php Création de Sous-Comptes].

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

===Options de base===

Ce sont les options de base que vous devrez configurer pour créer un sous-compte correctement.
*'''Protocole''': Vous pouvez sélectionner SIP ou IAX2 comme protocole. Cela dépendra de l’appareil utilisé. Le protocole recommandé est SIP.

*'''Type d'authentification''': Ceci est une option exclusive aux sous-comptes. Vous pouvez choisir entre nom d’utilisateur / mot de passe ou par IP statique. Cela dépendra une fois de plus du type d’appareil utilisé.

:'''Authentification nom d'utilisateur / mot de passe ''': Ceci est l’option recommandée étant donné que la majorité des appareils supporte seulement cette option.

:'''Authentification IP statique''': SIP uniquement, recommandé pour les utilisateurs expérimentés. Majoritairement utilisé pour les serveurs PBX ou Astérisk. Quand vous sélectionnez cette option, vous pouvez mettre l’adresse IP de votre appareil / serveur.

'''Note:''' Quand vous faites usage de l’authentification par IP statique, le status d’enregistrement n’apparaîtra pas dans votre portail.

*'''Nom d'utilisateur''': Cela sera le nom d’utilisateur du sous-compte. Vous pouvez utiliser lettres et chiffres pour le nom d’utilisateur (assurez-vous que votre appareil supporte l’un des deux types de caractères). Le format du nom d’utilisateur sera comme suit: '''{compte principal}_{nom d’utilisateur}'''.

Par exemple: disons que votre compte est le ''100000'' et le nom d’utilisateur a été mis en tant que ''maison''. Vous mettrez donc dans votre appareil ''100000_maison''.

*'''Mot de passe / adresse IP''': Ici vous mettrez le mot de passe pour l’usage du sous-compte. Il y a des réglages pour les mots de passe (voir ci-bas):

:#Doit comporter au moins 8 caractères
:#Doit contenir au moins une lettre en majuscule
:#Doit contenir au moins une lettre en minuscule
:#Doit contenir au moins un chiffre
:#Permet l'affichage alphanumérique et les caractères suivants: ! # $% & / () =? * [] _:. , {} + -

Si vous faites l’usage de l’authentification par '''IP statique''', vous pourrez donc mettre l’adresse IP de votre appareil / serveur. Format : ''201.202.203.204''.

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

*'''Trunk''': Disponible seulement si vous utilisez IAX2. Cette fonction sert à activer le mode "Trunk IAX2". Dans ce mode, les "Frames Média " seront regroupées dans des "Paquets trunk", chacun avec ses propres horodateurs, au lieu de minis paquets de voix individuelles.

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

*'''Type d'appareil''': Assurez-vous de sélectionner le bon type d’appareil que vous utiliserez avec ce sous-compte afin de recevoir correctement les appels entrants.

*'''Numéro d'identification de l'appelant''': Si vous utilisez un adaptateur de téléphone analogue (ATA), un téléphone IP ou un téléphone logiciel, c'est ici que vous devez en régler les paramètres. Il est important de s'assurer de transmettre une identification de l'appelant valide pour assurer une terminaison appropriée. À cet endroit, vous aurez la possibilité de faire afficher l'un de vos numéros DID entrants comme numéro d'identification d'appel sortant. Vous pourrez aussi utiliser un numéro personnalisé ne figurant pas dans vos listes de numéros DID, cependant '''vous devez respecter les normes en vigueur par la réglementation à cet effet.''' '''Utilisez toujours un numéro qui vous appartient ou auquel vous avez une autorisation d'utiliser comme numéro d'affichage d'appel sortant. ''' Si vous avez un PBX qui permet de gérer le numéro d'identification d'appel sortant, vous aurez une option à cet effet. Afin de pouvoir définir un nom en affichage de sortie, généralement ceci devra être réglé à partir de votre appareil ATA/PBX/Softphone etc. ''Remarque: l'utilisation d'un numéro sans frais comme identifiant d'appelant sortant n'est pas recommandée, en particulier lorsque vous appelez des numéros sans-frais.''

*'''Routage au Canada''': Cette opération permet de définir la route utilisée par le système pour les appels effectués au Canada.

*'''Routage international''': Cela définit la route que le système utilisera lorsque vous effectuez un appel vers une destination internationale (à l'extérieur de la zone USA48 et du Canada).

*'''Autoriser les appels internationaux''': Lorsque cette fonction est désactivée, les appels effectués vers les destinations situées hors des É.-U. / Canada seront automatiquement rejetés.

*'''Autoriser *225 pour solde''': Lorsque cette fonction est activée, les appels effectués vers *225 fourniront le solde actuel du compte VoIP.ms.

*'''Musique en attente''': La plupart des téléphones IP et des téléphones logiciels envoient un signal au serveur VoIP auquel ils sont connectés lorsque vous appuyez sur le bouton "HOLD". Si vous souhaitez faire jouer de la musique pour la personne que vous mettez en attente, vous pouvez la sélectionner ici.
<br />

Vous avez les choix suivants:<br />

{| class="wikitable" border: none;"
! colspan="3" | Musique en attente
|-
! scope="col" | Code de teste
! scope="col" | Nom
! scope="col" | Description
|-
| '''***89''' || Away in the Tropics || d'Hawaï aux Caraïbes, ces tonalités offrent une variété de sons d'ukulélés, de percussions métalliques et de guitares.
|-
| '''***90''' || Coffee and Sunrise || Des tonalités agréables sans être trop intenses, qui sont positives sans être trop exaltées.
|-
| '''***91''' || Coffee Shop Acoustic || Des pistes de guitares acoustiques apaisantes qui procurent une atmosphère de détente et un environnement libre de stress.
|-
| '''***92''' || Easy Listening || Une variété de mélodies douces et décontractées.
|-
| '''***93''' || Guitar Alchemy || Des harmoniques astucieuses accompagnées de séquences d'accords progressifs pour créer une expérience musicale à la fois joyeuse et réconfortante.
|-
| '''***94''' || Happy Endings || Une combinaison de guitares, de batteries, de ukulélés, d'harmonicas et de cloches.
|-
| '''***95''' || Light and Casual || Des chansons apaisantes au ton léger et agréable.
|-
| '''***96''' || Orchestral Moods || Des histoires racontées sur fond de violons, de pianos et d'orchestres.
|-
| '''***97''' || Piano Mix || une variété de merveilleuses harmonies de piano
|-
| '''***98''' || Rock me Easy || une musique réconfortante pour créer une atmosphère de détente.
|-
| '''***99''' || Spa Sounds || Une variété d'instruments joués dans une atmosphère apaisante et sereine.
|-
| '''***100''' || Aucune musique, bip intermittent || Silencieux, mais émet un léger bip intermittent pour indiquer aux appelants qu'ils sont toujours en attente.
|}

*'''Nom du compte ou description''': Cette option est pour vous aider à identifier chaque sous-compte.

=== Options avancées ===

Ces options offrent la possibilité de changer les codecs utilisés, le mode DTMF et NAT pour chaque sous-compte. Ces options sont recommandées seulement pour usage avancé.

[[File:Advsubfr.png|thumb|none|800px]]

*'''Codecs autorisés:''' Les codecs sont utilisés pour convertir un signal vocal analogique version codé numériquement. Puisque la voix et le son sont analogues, ils doivent être convertis (ou encodés) en format numérique adapté à la transmission par internet. Une fois que le signal atteint sa destination, il doit être décodé à nouveau de sorte que l’interlocuteur puisse entendre ce que vous dites.

'''G.722 Codec'''

Le codec '''G.722 ou voix haute définition ('''HD) est un codec audio à large bande qui fonctionne à un taux d'échantillonnage élevé. La fréquence d'échantillonnage plus élevée permet au codec G.722 de fournir des signaux audio d'une clarté supérieure à celle du codec G.711.<br>
Le codec G.722 et la voix haute définition '''sont disponibles''' '''et entièrement supportés lors d’appels SIP où le codec G.722 est configuré aux deux extrémités de l'appel (appelant et appelé)''' et fonctionne également pour tous les appels internes entre clients VoIP.ms, enregistrements système ou messages.<br/>

Le codec G.722 n'est actuellement pas disponible pour la communication entre serveurs/POP différents

En raison de la nature de sa technologie et des limitations actuelles imposées par les fournisseurs de téléphonie standard existants (c.-à-d. ne prenant pas en charge le codec G.722), '''la voix haute définition n'est actuellement pas disponible''' pour les appels sur des lignes téléphoniques régulières ou mobiles.

*'''Mode DTMF:''' Lorsque vous appuyez sur un bouton sur votre téléphone, ce dernier envoie une combinaison de tonalités audibles qui peuvent être séparées à l'autre extrémité pour déterminer la touche que vous avez appuyée.

*'''NAT (Traduction d'adresse réseau):''' Sélectionnez "Oui" si vous êtes derrière un NAT, et "Non" si vous ne l'êtes pas. Si vous ne savez pas ce que signifie ce paramètre, nous vous recommandons fortement de laisser le champ à "Oui".

*'''Expiration maximale:''' définit le délai maximum (en secondes) jusqu'à l'expiration de l'enregistrement d'un périphérique ou d'un système téléphonique. La valeur par défaut est 3600 mais la plage peut aller de 60 à 3600 secondes. Si votre appareil ou votre système téléphonique demande un délai d'expiration d'enregistrement inférieur sur le serveur, celui-ci sera respecté. L'expiration maximale s'applique également aux abonnements, aux indicateurs de message en attente (MWI) et à la présence (BLF).

*'''Le délai d'expiration du RTP''': définit le délai (en secondes) sans activité RTP (silence) avant la fin de l'appel lorsque l'appel n'est PAS en attente. Si le champ est laissé vide, la valeur par défaut de 60 secondes sera utilisée. Le délai d'attente RTP peut aller de 1 à 3600 secondes, mais il doit toujours être égal ou inférieur au délai d'attente du RTP.

*'''Le délai d'attente du RTP''': définit le délai (en secondes) sans activité RTP (silence) avant de mettre fin à un appel en attente. Si le champ est laissé vide, la valeur par défaut de 600 secondes sera utilisée. Le délai d'attente du RTP peut aller de 1 à 3600 secondes, mais il doit toujours être égal ou supérieur au délai d'expiration du RTP.

Remarque: '''les paramètres Expiration maximale, Délai d'expiration du RTP et Délai d'attente du RTP''' sont disponibles indépendamment pour le compte principal et les sous-comptes.

===Configuration Revendeur===

Si vous faites usage de notre fonction de revendeur, vous pouvez associer chaque sous-compte à l’un de vos clients.

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

*'''Revendeur client''': Vous pouvez sélectionner le client que vous voulez associer avec ce sous-compte. Vous devez en premier lieu créer votre client sous la section revendeur dans votre portail client.

*'''Paquet de client revendeur''': Vous pouvez sélectionner le paquet que vous voulez associer à votre client. Vous devez en premier lieu créer le paquet sous la section Revendeur.

*''' Prochaine date de facturation''': Vous pouvez spécifier la prochaine date de facturation pour votre client. Normalement, le système la configure automatiquement, par contre vous pouvez modifier l’option manuellement.

*''' Facturer frais d'activation maintenant''': Une fois vous avez sélectionné cette option, le frais mensuel du paquet sera chargé. Par exemple, vous pouvez utiliser cette option lorsque vous avez fait un changement au niveau du paquet de votre client.

=== Paramètres facultatifs ===

Malgré que ces options soient facultatives, vous pouvez les utiliser pour assigner une extension interne à votre sous-compte, une boîte vocale ou un temps de sonnerie.

[[File:souscomptefacultatif2.png|thumb|none|700px]]

*'''Numéro de poste interne''': Il s'agit du numéro de poste utilisé pour communiquer entre les sous-comptes appartenant à ce compte. Si vous inscrivez le 55, votre poste sera 1055.

Note: Assurez-vous que tous les sous-comptes soient enregistrés sous le même serveur afin de faire utilisation de cette fonction. De plus, les appels entre extensions sont gratuites.

*''' Activer l'identification de l'appelant interne / Nom de l'identifiant de l'appelant interne:''' Vous permet de définir un nom d'ID de l'appelant spécifique qui sera vu uniquement lors des appels internes entre vous et les utilisateurs de votre sous-compte. De plus, votre numéro de poste apparaîtra sur vos appels internes au lieu de votre numéro d'identification de l'appelant à 10 chiffres.

*''' Messagerie Vocale de poste interne''': Ici, vous pouvez sélectionner le temps de sonnerie utilisé lorsque vous recevez des appels à ce poste interne.

'''Note:''' Si vous associez votre messagerie vocale à ce sous-compte, vous recevrez une alerte de messagerie quand quelqu’un vous laissera un message.
Cela amènera un résultat différent dépendamment de l’appareil utilisé.
Par exemple, en utilisant un appareil ATA Linksys, vous recevrez une tonalité périodique. Les téléphones IP ont une tonalité saccadée.

*''' Temps de sonnerie de poste externe''': Ici, vous pouvez sélectionner le temps de sonnerie utilisé lorsque vous recevez des appels à ce poste interne.

==== Interception d'appels (Call Pick-up) ====

La fonction **d'interception d'appels** permet à un appareil enregistré sous un sous-compte de répondre à un appel destiné à un autre appareil enregistré sous un sous-compte différent.

'''Par exemple :'''
Si Alex et Bob reçoivent des appels sur leurs appareils respectifs et que Bob doit s’absenter, Alex peut répondre à l’appel entrant destiné à Bob directement depuis son propre appareil, évitant ainsi que l’appel ne soit manqué.

Cette fonctionnalité est particulièrement utile dans des environnements de travail partagés, comme les centres d'appels ou les zones de réception, où les membres d'une équipe doivent pouvoir s'entraider en répondant aux appels les uns pour les autres.

Pour plus d’informations sur l’utilisation de la fonction Call Pickup, consultez cet [[Interception_d%27appels | article]].

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

== Sécurité ==

Nous fournissons des paramètres supplémentaires afin d'améliorer davantage la sécurité de vos systèmes avec notre service.

[[File:IP_Restrict_Fr.jpg|thumb|none|700px]]

'''Restriction d'Adresse IP''': L'activation de cette option vous fournira une couche de sécurité supplémentaire dans laquelle les appels sortants ne seront autorisés qu'à partir de l'adresse IP ou de la plage d'adresse IP spécifiée.

'''Restriction PdP''': L'activation de cette option vous fournira une couche de sécurité supplémentaire dans laquelle les appels sortants ne seront autorisés que depuis les points de présence sélectionnés.

==== L’utilisation du sous-compte comme SIP URI externe ====

Pour faire l'utilisation du sous-compte comme [[SIP URI FR | SIP URI]] externe, vous devez seulement activer l’option d’extension interne. Par exemple, disons que l’extension interne est le 2 (102 avec le 10 prédéterminé), vous pouvez rejoindre directement via SIP avec une adresse ressemblant à: 1000002@serveur.voip.ms (remplacez serveur.voip.ms par le serveur où votre sous-compte est enregistré et le 2 par l’extension interne de votre sous-compte).

==Création en masse de sous-comptes==
Vous pouvez également créer des sous-comptes en masse à partir de votre portail client. Pour ce faire, rendez-vous dans votre portail client, puis Sous-comptes > Gestion des sous-comptes. Une fois sur cette page, une option Importer des sous-comptes est disponible en haut à droite. Cliquez dessus, téléchargez le modèle, remplissez-le, puis importez-le. Les sous-comptes seront automatiquement créés selon le modèle importé.

== Enregistrement des sous-comptes ==

Vous pouvez vérifier l’enregistrement de vos sous-comptes sur la page principale de votre portail client. Vos sous-comptes apparaîtrons sous '''État d'enregistrement du sous-compte''' (dans la page d'accueil) et vous pourrez également rechercher un sous-compte en question à l’aide de la boîte de recherche.

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

== Rapport de sous-comptes ==

Vous pouvez voir le nombre de minutes, nombre d’appels et même le montant que le sous-compte a fait comme dépense. Cette information est utile même si vous n’êtes pas un revendeur. Vous pouvez accéder à ce rapport sous votre portail client, Sous-Comptes > Rapport de sous-comptes.

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

''' Plage de rapports''': Vous pouvez sélectionner jusqu’à une marge de 92 jours (3 mois).

'''Minutes''': Le nombre de minutes utilisé par ce sous-compte dans la marge sélectionnée et ce, en décimales.

'''Appels''': Le nombre d’appels que ce sous-compte a fait dans la marge sélectionnée.

''' Montant dépensé''': Le montant du total dépensé pour le sous-compte en question pour la marge sélectionnée.

== L’utilisation d’un sous-compte ==

Une fois que vous aurez créé un sous-compte, vous pourrez l’utiliser avec la majorité de nos fonctionnalités. Par exemple, la [[Réceptionniste virtuelle IVR | Réceptionniste virtuelle (IVR)]], [[File d'attente]], l’utiliser comme [[SIP URI FR | SIP URI]] et plusieurs autres.

[[category:guides en français]]

Sub Accounts

2026-01-28T20:15:31Z

RP: /* Reseller Configuration */

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

Having a Sub Account allows you to register more than one device to make or receive calls simultaneously, you can also use it as an internal extension for your office or even your house. Many of the features within voip.ms make use of sub accounts. With this guide we are going to learn how to create and use this feature properly.

__TOC__

== Blog Article ==

[https://wiki.voip.ms/article/How_to_use_IP_authentication_with_a_Sub_Account How to Use IP Authentication with a Sub Account]

== How to Create a Sub Account ==

You can create as many Sub Accounts as required and there's no extra charge for doing this. First you need to enter your Customer Portal and select "Create Sub Account" under the Sub Account Menu.

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

===Basic Options ===

These are the basic options you need to configure in order to properly create a Sub Account.

'''Protocol''': Here you can select either SIP or IAX2 protocol. This will depend on the device you want to use with this Sub Account. SIP is the recommended protocol.

'''Authentication type''': This is an option that you don't have with your main account. You can choose between '''User/Password Authentication''' or '''IP Authentication'''. Again this setting will depend on the device you're using.

-- '''User/Password Authentication''': This is the recommended setting and most devices only support this option.

-- '''IP Authentication''': Recommended for advanced users. Mostly this is used for PBX or Asterisk-based servers. When you select this option you can set the IP of your device/server.

'''Note:''' When you use IP Authentication the registration status will not be displayed in your Customer Portal: Home Page.

'''Username''': This is the username of the sub-account. You can use alphanumeric characters for the username (make sure your device supports the use of alphanumeric characters in the user field). The format of the username will be as follow: '''{Main Account}_{username}'''.

For example, let say that your account is ''100000'' and you set the username as ''home'' the username that you're going to use in your device will be ''100000_home''.

'''Password/IP Address''': Here you can set the password to use for this sub account. The minimum is 6 characters, although it's strongly suggested that you use more characters and also use alphanumeric characters to create a strong password.

If you're using IP Authentication here you can set the IP address of your device/server. Format e.g. 201.202.203.204.

[[File:IP AUTH.jpg|thumb|none|700px]]

'''Trunk''': This option only becomes available if IAX2 is the chosen protocol. You can leave the default to send mini voice packets, or select "Send trunk packets" to group outgoing media frames into trunk packets, each with their own timestamps.

[[File:Trunk.jpg|thumb|none|700px]]

'''Device type''': Make sure to select the correct type of device you're going to use with this subaccount to properly receive incoming calls.

'''CallerID Number''': If you are using an analog telephone adapter (ATA), IP phone, or softphone, this is where you need to adjust the settings. It is important to ensure that you transmit a valid caller ID to ensure proper termination. Here you will have the option of having one of your incoming DID numbers displayed as an outgoing call identification number. You can also use a personalized number that does not appear in your lists of DID numbers. However ''you must respect the standards in force by the regulations to this effect. Always use a number that belongs to you or to that you have permission to use as the outgoing call display number.'' If you have a PBX that manages the outgoing call identification number, you will have an option for this. In order to be able to define a name in output display, generally this will have to be set from your ATA / PBX / Softphone device, etc. ''Nota: Using a Toll free outbound caller id is not recommended, especially when calling Toll-Free numbers.''

'''Canada Routing''': This defines the route the system will use when you place a call to Canada with your Sub Account.

'''International Route''': You can select the route that is going to be used for International Calls from this sub-account.

'''Allow International Calls''': With this setting, you can enable or disable the International Calls from the sub-account.

'''Allow *225 for Balance''': When Enabled, calls placed to *225 will provide the Current Balance of the VoIP.ms account. When Disabled calls placed to *225 will be rejected.

'''Music on Hold''': This setting allows you to select whether or not the person calling your DID number or extension will hear Music while the call is on hold.<br/>
The following options for music are available:<br/>

{| class="wikitable" border: none;"
! colspan="3" | Music on Hold
|-
! scope="col" | Test Code
! scope="col" | Name
! scope="col" | Description
|-
| '''***89''' || Away in the Tropics || From Hawaii to the Caribbean, these tracks deliver sounds of ukulele, steel drums, and steel guitars.
|-
| '''***90''' || Coffee and Sunrise || Uplifting without being perky, and positive without being too smiley.
|-
| '''***91''' || Coffee Shop Acoustic || Soothing, acoustic guitar tracks makes for a relaxed feel and sets a great atmosphere.
|-
| '''***92''' || Easy Listening || Smooth, casual tunes.
|-
| '''***93''' || Guitar Alchemy || Clever harmonics and progressive chord sequences to create a joyful and warming musical experience.
|-
| '''***94''' || Happy Endings || Uplifting, commercial style. Guitar, drums, ukulele, harmonica and bells.
|-
| '''***95''' || Light and Casual || Soothing and peaceful songs with a light, positive feeling.
|-
| '''***96''' || Orchestral Moods || Emotional and dramatic tales spun by violins, pianos and full orchestras.
|-
| '''***97''' || Piano Mix || Smooth Piano
|-
| '''***98''' || Rock me Easy || Feel-good music to create a relaxing atmosphere.
|-
| '''***99''' || Spa Sounds || Soft, slow and serene instrumentals.
|-
| '''***100''' || Aucune musique, bip intermittent || Silent, but with a subtle intermittent bleep sound to let your callers know that they are still on the line while they hold.
|}

'''Account name or description''': This setting can help you to easily identify each sub-account.

=== Advanced Options ===

These options allow you to set the allowed codecs, DTMF modes and NAT for each subaccount. These settings are recommended for advanced users only.

[[File:Advsettingssub.png|800px]]

*'''Allowed Codecs:''' This setting allows you to select the codecs that you're going to use, you can choose to allow one or more codecs at the time.

'''G.722 Codec'''

The '''codec G.722 or VoIP HD audio''' is a wideband audio codec that operates at a high data sampling rate. The higher sampling rate allows the G.722 codec to provide higher clarity of audio signals than G.711.<br>
Codec G.722 and HD voice is '''available''' and fully supported when performing SIP calls '''where both ends of the calls have codec G.722 configured''' and will also work for internal calls between customers within VoIP.ms '''as long as they are in the same POP server''', system recordings or messages.<br>

Codec G.722 is currently not supported for communication between different POP/servers.

Due to the nature of its technology and the current limitations with the existing regular phone providers (not supporting G.722), '''HD voice is currently not available''' when calling external regular phone lines or mobiles.

*'''DTMF Mode:''' This allows you to select the DTMF mode that is going to be used with this sub account. If you set this to ''AUTO'' the ''RFC2833 (AVT)'' is going to be used and automatically switch to ''INBAND'' if the other end doesn't support ''RFC2833''.
'''Note:''' Its recommended that you select the same DTMF mode in your device.

*'''NAT (Network Address Translation):''' This setting should be set to ''Yes'' if you're behind a NAT, if not set to ''No''. If you're unsure what this setting means, is highly recommended that you leave it to ''Yes''.

*'''Encrypted SIP Traffic''': This setting allows you to encrypt the communication between your device and our server, by using the SIP-TLS ''(Transport Layer Security)'' and SRTP ''(Secure Real-Time Transport Protocol)'' protocol.

Before using this feature, please consult the article about the [[Call Encryption - TLS/SRTP]]

'''Note:''' If enabled, all SIP traffic calls will be encrypted for this sub account.
''' ''' Please note that if encrypted calls are enabled then you need to configure your device to make and receive encrypted calls.

*'''Max Expiry:''' Sets the '''maximum''' amount of time (in seconds) until a device or phone system registration expires. The default value is 3600 but the range can go from 60 to 3600 seconds. If your device or phone system requests a lower registration expiry time to the server, this will be respected. MaxExpiry also applies to subscriptions, Message-Waiting Indicators (MWI) as well as Presence (BLF).

*'''RTP Time Out:''' Sets the amount of time (in seconds) of '''no RTP activity (silence)''' before the call is terminated when the call is '''NOT''' on Hold. If the field is left empty the default value of 60 seconds will be used. RTP time out can range from 1 to 3600 seconds but it must always be equal or lower than RTP Hold Time Out.

*'''RTP Hold Time Out:''' Sets the amount of time (in seconds) of '''no RTP activity (silence)''' before terminating a call that is '''On-Hold'''. If the field is left empty the default value of 600 seconds will be used. RTP Hold Time Out can range from 1 to 3600 seconds but it must always be equal or greater than RTP Time Out.

Note: '''Max Expiry, RTP Time Out, and RTP Hold Time Out''' settings are available for the main account and subaccounts independently.

=== Optional Settings ===

Although these are optional settings, you can use these settings to assign an internal extension number, voicemail and Ring Time for your sub account.

[[File:subaccountoptional2.png|800px]]

'''Internal Extension Number''': You set an internal extension number in order to call between Sub Accounts under the same Main Account. The extension number you enter in this field will have a leading 10, you can enter from 1 to 10 digits. For example if you set ''55'' the extension number will be ''1055''.
Note: Make sure that the sub accounts are registered to the same server in order to make an internal call. The call to an internal extension is FREE.

'''Enable Internal CallerID / Internal CallerID Name:''' Allows you to set a specific callerID Name that will be seen only on internal calls between you and your sub account users. Additionally, your extension number will show on your internal calls instead of your 10digit callerID number.

==== Sub Account as an External SIP URI ====

To use a sub account as an external [[SIP URI]], you only need to enable it as an Internal Extension first. For example, let's say your Internal Extension is set to 2 (102 with the leading 10), you can be reached directly via SIP from another network with a URI that is going to look like this: ''1000002@server.voip.ms'' (Replace server.voip.ms by the server you are registered to, and the 2 by your internal extension).

'''Internal Extension VoiceMail''': Here you can select which [[voicemail]] is going to be associated with this sub account. Please set this to be able to access a particular Voice Mailbox using *97 from the device registered with this sub account's credentials, even if you are not using Extensions.

'''Note:''' If you associate a [[voicemail]] with this subaccount, you are going to receive a '''Message Waiting Indicator''' when you have new messages in your mailbox.
This will lead to different results depending on your type of adapter, soft phone or IP phone.
For example, when using a Linksys ATA adapter, it's usually a stutter tone with periodic ring, IP phones usually use a stutter dial tone
when you pick up the line and are equipped with a blinking light, soft phones usually show a Voicemail Icon in the dial display.

'''Internal Extension Ringing Time''': This is the amount of time the phone will stay ringing when you call this internal extension directly. 5 sec = 1 ring.

==== call pick-up ====

Call Pick-up is a feature that allows a device registered under one sub-account to answer an incoming call intended for a device registered under a different sub-account.

'''For example:'''
If both Alex and Bob are receiving calls on their respective devices and Bob needs to step away, Alex can answer Bob’s incoming call directly from his own device, ensuring no calls are missed.

This functionality is particularly useful in shared work environments, such as call centers or reception areas, where team members may need to support one another by handling calls on each other’s behalf.

For more information on how to use Call Pickup, consult this [[Call_Pickup | article]].

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

=== Security ===

We provide some additional settings to further enhance the security of your systems with our service.

[[File:IP_Restrict_En.jpg|thumb|none|700px]]

'''IP Restriction''': Enabling this option will provide you with an extra security layer where outgoing calls will be authorized only from the IP address or IP range specified.

'''POP Restriction''': Enabling this option will provide you with an extra security layer where outgoing calls will be authorized only from the Points of Presence (POPs) selected.

=== Reseller Configuration ===

Also if you're using the [[Reseller Basic Guide|Reseller Interface]], you can associate each sub account with one of your customers.

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

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

'''Reseller client package''': Here you can select the package that you want to assign to your customer. You need first to create a package in the [[Reseller Basic Guide|Reseller section]].

'''next billing date''': Here you can set the next billing date. Usually the system sets this automatically, but if you manually change the package associated to this subaccount, you can change the next billing date manually.

'''Charge setup fees now''': Once you check this option, the monthly fee for the package is going to be charged. You can use this option when you have applied a change in the package of the customer.

== Sub Account Bulk Creation ==

You can also create in bulk sub accounts from your customer portal. To do so, head to your customer portal, Sub Account, Manage Sub Accounts.
Once there, on the top right of the page there is the option '''Import Sub Account'''. Click on it, download the template, fill the template and upload it. This will automatically create the sub accounts for you based on the template uploaded.

== Sub Account Registration ==

You can check your sub accounts registration at the Main Page of the Customer Portal. Your sub accounts will appear listed under '''Sub Account Registration Status''' and you will be able to search or filter them using the '''Sub-Account Filter''' searchbox.

[[File:SubaccountRegistration.png|800px]]

== Sub Account Reports ==

You can see the amount of minutes, number of calls and even the amount spent for each sub account. This information is useful even if you're not using the [[Reseller Basic Guide|Reseller Interface]]. You can access this from your Customer Portal>>Sub Accounts>>Sub Accounts Reports menu.

[[File:Subacc report.jpg|800px]]

'''Report Range''': You can display a report with a range of 92 days only (3 months).

'''Minutes''': The number of minutes used by this sub account in the given range. Expressed using decimal time.

'''Calls''': The number of calls made for this sub account in the given range.

'''Amount Spent''': This is the amount spent for this sub account in the given range.

== Use of the subaccount ==

Once you have created one subaccount, you can use it with most of the features available within your Customer Portal, for example with the [[Digital Receptionist (IVR)]], as an agent to receive calls from [[Calling Queues]], Routed directly to your DID numbers, as an external [[SIP URI]] to receive incoming calls from another networks, etc.

[[category:guides]]

Reseller Basic Guide

2025-12-16T22:07:23Z

RP: /* Create a new package */

{| class="wikitable"
|+
|-
! Article en Français !! Artículo en Español
|-
| [[Revendeur_Guide_Elementaire | Français]] ||
[[Guía_Básica_de_Reseller | Español]]
|}

The Reseller feature is intended for users who want to initiate a business in the VoIP world. This useful tool can help you create and manage accounts for your own clients, you will be able to assign specific rates for them and also provide credentials to register a DID number. 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 we will provide you with information about how to customize your Reseller portal to have a more personalized site.

The Reports section will allow you to check the Call detail records from the customers and review the Financial Reports related to their accounts. To activate your Reseller section, you need first to login to the Customer Portal then go to Reseller >> Reseller Main >> Activate

This guide will focus on the basic information to run the Reseller section properly. Here you can learn how to:

'''1. Manage Rates and Packages'''

'''2. Manage your clients'''

'''3. Use the Reseller Configuration'''

'''4. Generate Call and Profit Reports'''

== Manage Rates and Packages. ==

This section will focus on how to create packages for the reseller feature, how to edit the rates depending on the destination and how to set a benefit for the calls your clients make. It is important to note that all the calls your customers make are charged to you as per our regulars fees, however on the reseller portal, they will see the charges according to the rates you set here.

=== Create a new package ===

From the Customer Portal go to Reseller >> Manage Rates and Packages >> Create a new Package.

[[File:CreatePackage2.png|thumb|none|600px]]

'''Package name''': Set a package name.

'''Outbound Markup''': Here you set the benefit you want to have on the calls that your clients make, you can set a fixed amount or a percentage, for this example if you set a 20%, your package will be set as follows ([cost rate] + $ fixed amount) + 20 %.

Example : for 1 cent per minute, the sell rate will be (0.01 + 0.0000) + 20 % = 0.012

'''Pulse''': This is the charge increments that you will use for this package. For example, a pulse of 60 will make calls charged by the minute. You can read more about this in our [[Calls Cost]] article. '''Do note that this solely affects outgoing calls.'''

'''Canada Route & International Route''': These are the routes that the package will use on outgoing calls. For more information between the differences between Value and Premium, please read our article [[Value vs Premium]]

'''Voip.ms to Voip.ms Calls''': This option will make it so calls between VoIP.ms DID numbers are free. Applies to calls to USA and Canada VoIP.ms Numbers.

Under the '''Fees''' Tab you will be able to enter a value that you would charge as monthly base rate and a setup rate when connecting the package to a client.

The '''Free Minutes''' Tab will let you set a total of free calls, inbound & outbound for the package. If a vlue is entered in this field, the client will have free minutes (inbound & outbound calculated) up to the number of minutes entered here. Free minutes will be calculated from the invoice date for the package and will be calculated in total rounded minutes( for example, 3 minutes 11 seconds will be calculated as 4 minutes). The destinations added to the ''Free Zones'' at Reseller > Manage Rates & Packages > Edit Rates > Manage free zone will be able to spend these Free Minutes.

The '''Reseller System Configuration''' tab offers several advanced options for the package:

* '''User can edit Sub Account''': If enabled, the end user will be able to modify details of the sub account such as Password, Device Type, Authentication Type, CallerID Override, Music On Hold, Allowed Codecs, DTMF Mode and NAT.

* '''Type of Configuration''': This option determines the configuration of the services that will be shown in the Reseller Customer Portal. If the Default Configuration is set, the Services displayed will be the ones configured at Reseller > Client Branding and Access > Services tab. If the 'Package Configuration' is set, you will be able to set them in the Services section below.

* '''View Setting''': You can set if you want to display the information in the Client Interface on ''Simple View'' or ''Standard View'' here. Simple View is a feature that allows you to show your voipinterface.voip.ms client inferface in a simpler way to your clients. This changes the balance owned to amount due, and hides some features like :
**Call Types (busy, answered, failed & noanswer) filters in the CDR
**This month and today`s amount spent on the Balance page
**Services specific information (Register type, Allowed codecs, etc) on the services page
**And some other information that would not be useful for "vonage style" clients

* '''Services''': In this section you can select which services will be available and which services will not be available for your customers, if you are using the "Package Configuration" option in the Type of Configuration section. You are able for instance to set your customers assigned to this specific package the allowance of buying DIDs, provide them the ability to edit or cancel DIDs (same applies for Virtual Fax DIDs), setup an IVR, ring group and many more! (These Options are only available in the New Version of the reseller system).

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

=== Edit Rates ===

If you want to set different rates depending on the destination of the call, you can do this on the Edit Rates section from the package, all the custom rates you enter there will override the global markup. You can set a custom rate for all the calls to USA or Canada, by country or Calling code.

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

You can also set your '''Free Zones''' here, which will be the destinations that will be included in the Free Minutes offered in the package.

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

'''CAP''': ''Add call codes under a desired rate'' : This option allows you to specify a rate per minute, ''e.g. $0.10'',
This will add all the destinations/area codes (local or international) that have a per minute rate equal or below $0.10 as free destinations.
Note that you can subsequently delete some area codes in the left panel.

== Manage your clients ==

Now that you have a package, you need to add a client, this section will focus on how to create a client and associate it with a [http://wiki.voip.ms/article/Sub_Accounts sub account], as well as adding payments and charges for each client, assigning a DID number and also the proper way to delete a client account that is not longer needed.

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

=== Adding a client ===

From the customer portal go to Reseller menu >> Manage Clients acct >> Create a new client account.

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

Fill the form with the client information and choose between soft and hard balance, when finished click on create account. A customer in HARD balance will be suspended if he is on zero or negative balance, while a customer in SOFT balance will be able to keep making use of the service even if their balance is zero or negative.

On the Manage section, you can add a new package or a DID number. To add numbers to your clients you need to order the number from your Customer Portal first. Don't forget to set a Per Minute rate and/or Monthly fee for your number, as incoming calls are charged independently from the outgoing calls.

Once the client account has been created, you will need to create a [http://wiki.voip.ms/article/Sub_Accounts sub account] for it as well.

Go to Sub accounts menu >> Create Sub account. During the Sub Account creation process, you will reach the Reseller configuration section, here you can choose:

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

*Reseller client

*Reseller client Package

*Next Billing date

*And if you decide to charge the set up fees now, make sure to check the appropriate box.

If you already have a [http://wiki.voip.ms/article/Sub_Accounts sub account] created, you can modify it at Sub Accounts >> Edit Sub Account >> Manage and add a package there.

=== Delete a client ===

You can delete a client on the Edit section from the Client's account, however note that before doing this you need to be sure to unconnect any package or DID number related to the client, you can do this from the Manage section on Client´s account.

Also note that you can delete more than 1 customer at a time by checkboxing the account on the left side and then pressing the button '''Delete selected visible elements'''.

[[Image:MassDeleteReseller.png|600px|border]]

=== Adding Payments manually ===

You can add payments manually from the customer portal >> Reseller menu >> Manage Clients acct >> Manage Client, then select the Payment button under the Balance panel.

=== Add Charges manually ===

You can add charges manually from the customer portal >> Reseller menu >> Manage Clients acct >> Manage Client, then select the Charge button under the Balance panel.

=== Add Phone & Fax numbers to your clients (DIDs) ===

You can add phone and fax numbers to your clients from the '''Customer portal''' >> '''Reseller menu''' >> '''Manage Clients acct''' >> '''Manage Client''' >> '''Add a Phone number''' / '''Add a Fax number'''.

Please note that you must order at least 1 DID number before you can assign it to a client.

[[File:Adding_a_Fax_and_Phone_number.png|border|700px]]

When you click on '''add a phone number''' [[File:link_DID.png]] you will be invited to set up the DID numbers you wish to associate to your customer. Don't forget to set a Per Minute rate and/or Monthly fee for your number, as '''incoming calls are charged independently''' from the outgoing calls.
You will be able to enter manually the DID and/or choose one from the drop-down menu. Note that you can add more than one DID to your customer at a time.

[[File:Reseller_AddPhone2_Manual-Type.png|border|700px]]

You will also be able to click on the button '''[Advanced]''' to use the multi-selection tools. You will see a complete list of available DID(s) to connect. Once it is done, click '''[Save DIDs]'''

[[File:Reseller_AddPhone2_Advanced_DID.png|border|700px]]

Please have in mind the following instructions:
:* Click on a DID and drag the mouse up or down for multiple selections.
:* Press 'Ctrl' and keep it pressed while you click on the DIDs for multiple selections.
:* Press 'Shift' + 'Up' or 'Down' key to select the next/previous DID in the list.
:* If you click on a DID without pressing 'Ctrl', all previous selection will be lost.
:* The 'Select all' button will select all DIDs automatically.
:* The 'Save DIDs' button will override the selected DIDs list in the form.
:* The 'Cancel' button will discard all selection and close the selection screen.

When you have selected the DIDs you wish to link to your customer, you may proceed to customize them.

[[File:Reseller_AddPhone2_Advanced_DID_Selected.png|border|700px]]

You may add a setup and monthly fee ''(If you specified a charge, this will charge your client when the DID is added)''. You may also indicate a per-minute rate for the inbound calls. The per minute rate is independent then your package you have linked to your customer. However, if you indicate a bundle of free minutes in your package, the per minute rate will only be applied after they reach the limits of free minutes allowed. ''(Note that this section is optional.)''

[[File:Reseller_AddPhone2_fee.png|border|700px]]

If you wish to forward every incoming caller called your selected DID in the previous step, you will need to check the box and fill both sections. The phone number where you want to forward every call and a description. Note that your customer and you will still be able to change the routing if you want to route the DID to another destination, such as sub-account, IVR, etc. ''(Note that this section is optional.)''

[[File:Reseller_AddPhone2_fwd.png|border|700px]]

If you wish to create a new voicemail associated with the DID, you will need to check the box and fill the required information. The voicemail will be used when, per instance, the default routing (sub-account) is unreachable. ''(Note that this section is optional.)''

[[File:Reseller_AddPhone2_vm.png|border|700px]]

=== Setting a Default Account ===

If you have more than 1 sub account set for your customer, you can either decide which sub account will display by default on their portal, or allow them to decide. This will help remove confusion whenever your customers log into their account and being prompted to verify which account to select, thus making it more streamlined for your customers.

==== Selecting the default account for your customer ====

By heading into Reseller, Manage Client's Account and then clicking on '''Manage Client''' (Yellow gear), you will have the option to set which sub account will be used by default.

[[File:changeaccount1.png|border|700px]]

==== Having your Customer Set a Default Account ====

Your customers can also set this or change it by logging into their own portal, pressing '''Change Account''', selecting the account they wish to set as default, checkboxing the option to set it as default and then pressing '''Change Account'''

[[File:changeaccount2.png|border|700px]]

== Use the Reseller Configuration ==

It is always important to show a good brand image to your customer, for that reason we offer the option to customize our reseller interface to have the best presentation possible. There is also a useful feature on the site that can allow us to integrate your client's payments with a paypal account, so they can be added automatically as funds to the client.

=== Branded Interface ===

At the top of the screen you will see two available options, these are:

'''Version of the reseller system:''' This is the version of the Reseller System that will be displayed to your clients. The New Version of the Reseller System offers several customization options as well access to most of the VoIP.ms services, while the Old Version offers limited customization and access to basic services such as direct SIP routing and [[Call Forwarding]]

'''Configure Reseller Version:''' Determines which version of the Reseller System you will be configuring on this section. Enabling the New Version will offer you the option to modify the color of the panels, buttons and background of your Client Portal, as well as previewing it before making any changes.

=== Interface and Access ===

'''Clients Interface Access:''' This section will list the URL that you can give your clients to access their Account.

'''Auto Connect Form for Clients:''' This is an HTML form that you can use to automatically send your clients to their interface without needing authorization.

'''DID Access and Fax Access:''' You can set here the pricing for any DIDs that your clients purchase using the options at Services > Order DIDs or Virtual Faxes > Order Faxes on their portal and, the pricing for E911 in case you have enabled these services for them.

[[File:DID_and_Fax_Access.png]]

=== Branding Management ===

In this section you will be able to change the Title that will appear in your Reseller Customer Portal page, as well as any of the colors of the interface, including the buttons* and the panels*, as well as selecting a Header, logo* and login background* for the interface.

<nowiki>*Options only available in the New Version of the reseller system.</nowiki>

=== Display Elements ===

Here you can choose whether or not you'd like to display the SIP and IAX password(s) in the Reseller Interface, as well as allowing your customers the option to reset their passwords in the Login Reseller Interface.

Lastly, you will be presented with an option to integrate an automated signup form into your website. If so, you can link your signup menu option to the URL presented there. You can also select which fields will be shown in order to sign up with a new account with you.

=== Services ===

View Setting: You can set if you want to display the information in the Client Interface on 'Simple View' here. Simple view is a feature that allows you to show your voipinterface.voip.ms client inferface in a simpler way to your clients. This changes the balance owned to amount due, and hides some features like :

*Call Types (busy, answered, failed & noanswer) filters in the CDR
*This month and today`s amount spent on the Balance page
*Services specific information (Register type, Allowed codecs, etc) on the services page
*And some other information that would not be useful for "vonage style" clients

Services*: In this section you can select which services will be available and which services will not be available for your customers, if you are using the "Default Configuration" option in the Type of Configuration section of the Package settings.

<nowiki>*Options only available in the New Version of the reseller system</nowiki>

'''Fax Service (beta)'''

You can enable the Virtual Fax Service for your clients in order to allow them use all the fax functionalities and give them the ability to order Fax DIDs from their portal.

Order Faxes: If enabled, your clients will be able to purchase Fax numbers from their portal.

''It is important that you have set the proper rates and fees in the 'DID and Fax Access' option in the Reseller configuration so they are applied when a purchase is done by the customer.''

Manage Faxes: If enabled, your clients will have available in their portal the options 'Manage Fax Numbers', 'Send Fax', 'My Faxes', 'My Folders' and, 'Email to Fax'.

'''Enhanced 911 (beta)'''

Enabling this service will allow your clients to enable and disable e911 from their portal for the numbers associated with their accounts.

'''E911 Activation'''
Keeping this enabled will notify your customers the moment they have activated their e911 service. You can disable it if you wish to not notify them following the activation.

''It is important that you have set the proper fees in the 'DID and Fax Access' option in the Reseller configuration so they are applied when this service is enabled by the customer.''

=== Text and Messages ===

In this tab you will be able to personalize the Welcome Text that will appear in the Reseller Customer Portal. You are also able to customize the Emails that will be sent when a Customer's Balance is running low, as well as when they request and complete a Password Reset.

== Payment ==
If you want your clients to deposit directly into your PayPal account, you can activate the PayPal integration. All payments will be sent to your PayPal account and automatically credited to your client interface.

Instructions :
#Enter your PayPal email address in the "PayPal email account" field
#Enter the amounts that you want to offer, without decimal and separated by a comma

[[File:PaymentIntegration.png|thumb|none|600px]]
Paypal configuration process, for a Paypal [[#Using_a_Regular_type_account|Regular]] or [[#Using a Business type account|Business]] type account.

=== Using a Regular type account ===
----
'''1)''' Log on your Paypal account, click on the '''Setting icon''' ''(top-right)''

'''2)''' Click on "Seller Tools" tab.

'''3)''' Where '''"Instant Payment Notification"''', click '''[Update]'''.
:[[File:Reseller_PP_Regular_1-3.png|thumb|none|550px]]

'''4)''' Click the '''"[Choose IPN Settings]"''' button.
:[[File:Reseller_PP_Regular_4.PNG|thumb|none|550px]]

'''5)''' Enter the '''"Notification URL"''' and select '''"Receive IPN Message (Enabled)"''' and click the '''[SAVE]''' button.
'''Notification URL:''' <nowiki>http://www.voipinterface.net/paynotify.php</nowiki>
:[[File:Reseller_PP_Regular_5.PNG|thumb|none|550px]]

'''6)''' Done!

'''We strongly recommend making a live test with an amount of $1.'''

=== Using a Business type account ===
----
'''1)''' Log on your Paypal account, click on the '''Setting icon''' ''(top-right)'', then click on '''Account Settings'''.
:[[File:Reseller_PP_Business_1.png|thumb|none|600px]]

'''2)''' Click on '''"Website payments"''', where the left navigation bar.
:[[File:Reseller_PP_Business_2.png|thumb|none|600px]]

'''3)''' Where '''"Instant Payment Notification"''', click '''[Update]'''.
:[[File:Reseller_PP_Business_3.png|thumb|none|600px]]

'''4)''' Click the '''"[Choose IPN Settings]"''' button.
:[[File:Reseller_PP_Business_4.PNG|thumb|none|600px]]

'''5)''' Enter the '''"Notification URL"''', select '''"Receive IPN Message (Enabled)"''' then click the '''[Save]''' button.
'''Notification URL:''' <nowiki>http://www.voipinterface.net/paynotify.php</nowiki>
:[[File:Reseller_PP_Business_5.PNG|thumb|none|600px]]

'''6)''' Done!
:[[File:Reseller_PP_Business_6.PNG|thumb|none|600px]]

'''We strongly recommend making a live test with an amount of $1.'''

=== Charging taxes with PayPal ===
----
VoIP.ms does not apply any taxes to the Reseller’s end customer payments. The customer is responsible and has the freedom to apply taxes to his clients as required.<br>
Charging taxes is also possible through PayPal by configuring this directly within the Business Account.

See screenshots detailing this process below:

<div><ul>
<li style="display: inline-block;"> [[File:PP-profile.png|thumb|none|500px|Click to enlarge]] </li>
<li style="display: inline-block;"> [[File:PP-selling.png|thumb|none|350px|Click to enlarge]] </li>
<li style="display: inline-block;"> [[File:PP-sellin-tools-taxes.png|thumb|none|350px|Click to enlarge]] </li>
</ul></div>

== Host Name ==

In order to personalize the interface with a hostname

#Configure a host with an A RECORD in your DNS that points to 208.100.60.92
#Enter in the field bellow the fully qualified host + domain name (e.g: interface.mydomain.com)
#Click "submit"

There is a one time fee of $10.00 for adding your hostname to our system.

== Reports ==

Here you can see the Call Detail Records for the calls your customers have made and received, with the rates we have established on the packages created, showing a net total of the benefit earned as well.

Reports are divided in the following categories:

*'''Global CDR''': Display cdr report for all your reseller clients calls. You can see all the calls (incoming and outgoing) from all your clients.

*'''Per User CDR''': Display cdr report for a specific client. This filter help you to search calls only for one specific client.

*'''Per User Faxes''': Display Fax report for all your clients or for a specific client.

*'''Financial report''': Month to month revenues ,deposits, costs related to the reseller clients. You can see the financial information by month.

*'''Sub accounts report''': List of all sub accounts connected to the reseller clients ,with related information. Here you will find a list of your clients, [http://wiki.voip.ms/article/Sub_Accounts sub account], package and DID number associated with each one.

*'''DID(s) Report''': List of all DID's connected to the reseller clients, with related information.

=== CDR export for your customers ===

From your whitelabelled portal, your customers will be able to see their CDR, see a graphical usage report and even export it as CSV.

First, by having your customers head into their portal and then under '''My CDR''', they will gain access to all their past calls.

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

From there, they can additionally select one of the following options to export the data:

- Copy: Have all the data copied so that they can paste it where they want (Excel, Word, notepad, etc).

- Excel: Download it in an excel format.

- CSV: Download it in a CSV format.

[[File:CDRreport2.png|thumb|none|600px]]

=== Graphical usage report ===

If they prefer a graphic representation of their own traffic, this can be done by heading into '''Graphical Usage Reports'''. From there, different graphs are available.

Note that they can also be downloaded.

'''Top 10 Destinations by calls:'''
The top 10 destinations (inbound being included) by terms of amounts of calls.
[[File:Graphicalreport.png|thumb|none|600px]]

'''Top 10 Destinations by minutes:'''
The top 10 destinations (inbound being included) by terms of minutes.
[[File:Graphicalreport2.png|thumb|none|600px]]

'''Daily Spending:'''
Shows how much it was spent in total on a daily basis.
[[File:Graphicalreport3.png|thumb|none|600px]]

'''Calls and Minutes:'''
This will show how many calls were done and lengthiest call on a daily basis.
[[File:Graphicalreport4.png|thumb|none|600px]]

'''Daily Call Status:'''
Will show how many calls were answered, not answered, returned busy signal or that failed on a daily basis.
[[File:Graphicalreport5.png|thumb|none|600px]]

== DNO List ==

A Do Not Originate (DNO) List is a collection of phone numbers that are not intended to be used for outbound calls. These numbers are reserved exclusively for inbound communication, such as customer service lines, government agency contacts, or financial institution support numbers.

The primary goal of a DNO List is to prevent fraud and spoofing. Scammers frequently impersonate trusted organizations by spoofing caller IDs using well-known numbers. By registering these numbers in a DNO List, any outbound call attempt using them as a Caller ID is automatically flagged and blocked to help prevent misuse.

You can manually add numbers to your DNO list if:
You've received a report indicating misuse.
The number is confirmed to never initiate outbound calls.
When adding a number, you can include a note for reference. Once the number is on the list, the system will automatically block any outbound calls that attempt to use it as the Caller ID.

To access the DNO list, simply head to Reseller Menu, DNO

[[File:DNO1.png|thumb|none|600px]]

===Adding a Number Into Your DNO List===

====Manually Adding a Number====

To add a number into your DNO list, simply:

''1.'' Add the number you do not wish to have any origination in your account in the field '''CallerID to Block''

''2.'' Add a description to recall the reason or the company related to the number.

====Adding a list via CSV====
You can import your DNO List using a CSV (Comma-Separated Values) file. Simply click on "Import DNO CSV List" and upload a file formatted with two columns: Number,Note.

Examples of format:
5145556666,Bank Institution
5146667777,Report from agency
When uploading your DNO List via CSV, please keep the following in mind:
Duplicate entries will be ignored to prevent conflicts
If you'd like to update existing records, enable the option "Overwrite Existing DNO Entries". In this case, the system will only update the note field for any repeated numbers.

[[category:guides]]

Reseller Detailed Guide

2025-12-16T17:53:38Z

RP:

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= Introduction =

VoIP.ms Reseller Portal is intended to help our customers resell our services to end-user client.

With the reseller portal, you can create a white label interface and create a client account. Your client will have access to a full range of telephony features, allowing them to manage their phone system autonomously.

As a reseller, you will have the ability to choose the level of feature management your clients will have access to. Depending on if you already have your billing system or not, the Reseller Portal can be configured to handle client's payment through '''"Paypal"''' or with our '''"API"''', used to only to manage the client's phone system or both.

This guide describes how to activate and customize the reseller portal, explains how to build packages, how to create and manage client's account and finally also explains the reports.

Through the guide, we will use the term '''"Client"''' to refer to the '''resellers' end users'''.

To activate your Reseller Interface and to start to customize it, you will need to go to your VoIP.ms Customer Portal, and go on the menu '''"Reseller"''' where the navigation bar, and click on the '''"Reseller Main"''' link.<br />
At the bottom of the page, you will find the button '''[Activate]'''.

= Reseller Configuration =

: The reseller configuration page, allows you to manage the major part of your reseller portal and what the end-user see, such as the appearance of your portal, the color of the buttons, your logo, the element/service that you would like to display/withdraw to the end-user view.

: It will also give you the possibility to integrate your PayPal account and manage the amount that they can deposit in their account.
: This page also offers the option to choose a custom hostname ''(URL)'' to give direct access to your reseller portal by using your own domain name.

: To access the Reseller Configuration page, you will need to go on the navigation bar and go over '''"Reseller"''' then, clicking on '''"Reseller configuration"'''.

== Branded Interface ==

=== Version of the reseller system ===

: Once the Reseller Portal is activated, the next step is to select the version of the reseller system.
: The portal is in constant evolution and brought us to develop a new interface with a nicer look.
: To get the most recent version of the portal and all the capabilities, select '''"New Version"'''.

:: [[File:Reseller_Guide_Detailed_Activate_NewVersion.png|650px|border]]

== Interface & Access ==

: The Interface & Access tab in Reseller configuration is divided into three parts. '''"Client Interface Access"''', '''"Auto Connect Form for Clients"''' and '''"DID and FAX Access"'''.

=== Client Interface Access ===

: The purpose of this section is to show the direct link to your reseller interface in order to give your clients access to their account, CDR and manage their services.
: This can be useful to insert a link/button in your business customer portal or indirect communication.

: You can use your own domain name ''(URL)'' refer to the "Hostname" section.

: The purpose of this section is to create a direct link, and HTML code to your reseller interface in order to give your clients access to their account, CDR, and manage the services.

: This can be useful to insert a link/button in your business customer portal.
: Your client will simply have to fill in the login fields via your website, and they will be automatically redirected to his portal.

<div style="margin-left:50px;
padding:5px;
font-family: Consolas, Monaco, Lucida Console, Liberation Mono, DejaVu Sans Mono, Bitstream Vera Sans Mono, Courier New, monospace;
color: silver;
border-left: 6px solid rgb(208, 56, 45, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: black;
width:58%;">

: <form action="''<nowiki>https://www.voipinterface.net/site/login.php</nowiki>''" method="POST" target="_NEW">
:: <input type="HIDDEN" name="UID" value="''{Your_VoIPms_Account_Number}''">
:: <input type="HIDDEN" name="action" value="DirectLogin">
:: <input type="HIDDEN" name="col_email" value="''{Your_client_email}''">
:: <input type="HIDDEN" name="col_password" value="''{Your_client_password}''">
:: <input type="SUBMIT" name="SUBMIT" value="Connect">
: </form>

</div>

<br/>

<div style="margin-left:50px;
padding:5px;
border-left: 6px solid rgb(208, 56, 45, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(51, 51, 51, 0.08);
width:59%;">

'''NOTE''' 💬
: ➜ Replace '''{Your_VoIPms_Account_Number}''' by your VoIP.ms 6-digits account number.
: ➜ You can replace the URL in '''''action=''''' by your domain name ''(URL)'' if you have created it. Refer to the '''[[#Hostname|Hostname]]''' section.
</div>

<div style="margin-left:50px;
clear:left;
padding:5px;
border-left: 6px solid rgb(66, 192, 251, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(66, 192, 251, 0.05);
width:59%;">

🔗 '''Related topic with this section''' | Go to the [[#Hostname|Hostname]] tab section.
</div>

=== DID and FAX Access ===

: With this section, you can set a default rate that will be applied to your client's DIDs, ''(Voice and Fax)'' such as the one-time setup fee, when they buy or you add a new DID ''(Voice and Fax)'' into their account, the recurring monthly fee and the per-minute incoming rate.

<div style="margin-left:50px;
padding:5px;
border-left: 6px solid rgb(208, 56, 45, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(51, 51, 51, 0.08);
width:59%;">

'''NOTE''' 💬: ➜ When connecting a DID manually to a client's account, you will be able to customize these fees.

</div>

=== Voice DIDs and Faxes [Beta] ===

: '''Setup''': For every regular phone number '''(DID voice or Fax)''' number, the Setup fee is the '''One-time charge''' that will be billed to your client when you connect a new DID into your client's account.
: However, if you allow the purchase of DIDs through the client's interface ''(in the package)'', this will be the '''one-time fee''' by default that your client will be charged at the end of the order process.
: The amount will be deducted from his virtual account balance.

: '''Monthly cost''': When your customer purchases a new '''(DID voice or Fax)''' number through their client's interface, your client will be billed this monthly recurring rate each month.
: It can be edited when you manually connect a DID to your client's account.

: '''Per minute''': When a DID is connected to your client's account, all incoming calls will be billed at this rate per minute.
: The rate per minute can be edited when you manually connect a DID number to your client’s account.

=== E911 Fees [Beta] ===

: '''Setup''': When a client activates a '''"e911 record"''' for a DID they have in their account, this will be the '''one-time setup fee''' they will be billed on their account balance.
: This fee is '''not recurring''' and is applied to each new record they submitted to the e911 database.

: '''Monthly cost''': When an e911 record is created and successfully activated for a specific DID, the monthly cost entered will be the one that your client '''will be charged each month''' at the same time as his DID's monthly fee ''(the DID's billing date)''.

=== SMS/MMS Fees [Beta] ===

: '''Incoming SMS''': This is the amount your client will be charged each time a standard incoming text message (SMS) is received.

: '''Outgoing SMS''': This will be the fee your client will be charged each time a standard text message is sent.

: '''Incoming MMS''': This is the amount that will be charged to your customer each time they receive an incoming multi-media text message (MMS).

: '''Outgoing MMS''': This will be the fee your client will be charged each time a multi-media text message (MMS) is sent.

=== Call Recording Fees [Beta] ===

: '''Per-Minute''': When the call recording feature is enabled for your client's sub-account and/or voice DID, this will be the '''per minute charge''' that will be billed to their account. ''(To give him access to this feature, it must be activated in his "Package").''

== Branding Management ==

: In this section, you will be able to customize the look and some components that are appearing on your '''Reseller client's interface'''.

: It will give you the possibility to customize multiple things, such as the title of your reseller client's interface and the colors for the client's interface to reflect your brand.
: Use the button '''[preview]''' to see the changes made before saving your changes.

=== Title and Colors ===

: The title will appear as the '''title of the window''' and at the '''top of the client's interface''' when a user is logged in.
: The color of the '''texts''', '''layouts''' such as the background, navigation bar, links and text can be changed with the HTML color code or with the color picker.
: A preview button is also available to see the color changed before applying your change.

=== Edit buttons ===

: Some buttons will be displayed in the client's interface to manage different services.
: You have different types of buttons, such as the '''primary''', the '''success''', the '''exit''' and the '''delete''' buttons.
: Depending on the action your client would like to do, the proper button will be reflected.
: The '''background color''' of the button can be customized as well as the '''text color'''.
: They can be customized by entering the '''HTML color code''' or by using the '''color picker'''.

=== Edit Panels ===

: Some display panels are used on different cards in the reseller client's interface such as cards to display the charges and the payments, the total of call etc.

=== Images ===

: With this section, you can customize multiple part of the client’s interface by adding your logo at different places of the portal.
: You can customize the '''background image''' and the '''logo''' used at the '''login page''', the '''header image''' when your client is logged in the client's interface and the '''logo''' over the content page ''(on top of the service column)''.
: Use the '''[Preview]''' button to have a preview of your images before applying your changes.

: Logo/Images can be uploaded, or you can use a URL.

== Display Elements ==

: This tab allows your client to manage his SIP/IAX password and the information that can be displayed or hidden in the client interface.

=== Display SIP and IAX password ===

: This tool allows '''showing''' or '''hide''' the end-user password in their client interface under '''[Services]''' and '''[My Services]'''.

=== Display and allow reset password ===
:[[File:Reseller_Guide_Detailed_Display_ForgotPassword.png|left|border|150px]]

: With this feature, you can '''show''' or '''hide''' the link '''"Forgot your password?"''' on the login page of your client interface.
: If displayed: The end-user will receive an email to reset his password.

: You can customize the body of the email under the "Message" tab of the Reseller Configuration page.
: See the section Message below to know more.

<div style="margin-left:50px;
clear:left;
padding:5px;
border-left: 6px solid rgb(66, 192, 251, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(66, 192, 251, 0.05);
width:59%;">

🔗 '''Related topic with this section''' | Go to the [[#Messages|Messages]] tab section.
</div>

=== Signup Form Fields ===

: Use this option if you want to give your clients the opportunity to open their client account autonomously.
: To open an account, Clients will have to provide each information selected by check box.

: When a client creates his account, he will receive an email that his account is awaiting activation.
: The new account will have the status '''"WAIT"''' at the page '''"Manage Client's Accounts"''' under the reseller portal view, which means that you must activate the client account through your portal, whereby he will not be able to access the interface.
: At this point, you will be able to create a sub-account for this client to allow them to connect a device such as a PBX, an IP Phone, a softphone or an ATA.

: When the account is activated, the default account balance management is set to '''"SOFT"'''.
: The client's account will not be suspended if their balance reaches $0.
: The client will be able to use the service, even if the balance in their accounts is $0.
: If happening, their accounts may reach a negative balance. This will represent the amount they owe you.

: Once the client's account is activated, you can set the balance management to '''"HARD"'''.
: The client's account will be suspending when his account balance reaches $0.

<div style="margin-left:50px;
clear:left;
padding:5px;
border-left: 6px solid rgb(66, 192, 251, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(66, 192, 251, 0.05);
width:59%;">

🔗 '''Related topics with this section''' | Go to the [[#Manage_Client's_Account|Manage Client's Account]], [[#Balance_Management|Balance Management]] section.
</div>

: When the customer account is activated, they will be able to start using their account in a '''standalone mode'''. This means they will have access to the default service view that you have set up in the '''[Services]''' tab in the main reseller configuration. But no package associated.

<div style="margin-left:50px;
clear:left;
padding:5px;
border-left: 6px solid rgb(66, 192, 251, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(66, 192, 251, 0.05);
width:59%;">

🔗 '''Related topic with this section''' | Go to the [[#Service|Service]] section.
</div>

== Services ==
: In this section, we explain how to manage which service or feature the end-user has access to manage in the client interface.

=== View settings ===

: With this option, you will have the possibility to choose between 3 types of view, '''standard view''', '''Simple View''' and '''Simple View (Hide Service Page)'''.

: The simple view: will allows your client to see his client interface portal in a simpler way. This will hide some features like
:: - The Call Types filters in the CDR ''(busy, answered, failed and no answer)''.
:: - The panel "This month" and "today's" amount spent on the Balance page.
:: - Some services specific information ''(Register type, Allowed codecs, etc.)'' on the services page
:: - And some other information that would not be useful for the user.

=== Services (Show) ===

: In this section, you can select which services will be displayed in the client's interface and which services will not be available for your client.

<div style="margin-left:50px;
padding:5px;
border-left: 6px solid rgb(208, 56, 45, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(51, 51, 51, 0.08);
width:59%;">

'''NOTE''' 💬: ➜ This is the by default view.
</div>

: If you have not chosen to customize the display view and the services authorized in your package plan, we suggest disabling the services in this page and use the in-package service settings instead. In-package service settings will give you more control over how you want to manage the offered service per client and per user within a client’s account.

<div style="margin-left:50px;
clear:left;
padding:5px;
border-left: 6px solid rgb(66, 192, 251, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(66, 192, 251, 0.05);
width:59%;">

🔗 '''Related topic with this section''' | Go to the [[#Manage_Rates_and_Packages|Manage Rates & Packages]] section.
</div>

== Messages ==
[[File:Reseller_Guide_Detailed_Dashboard_WelcomeText.png|right|border|400px]]
: This tab will allow you to customize various texts and messages of the client's interface, such as the '''Welcome Text''', the '''Low Balance Email''', '''Reset Password Recovery''' Email and '''Reset Password Completed Email'''.

=== Welcome Text ===

: This is the message appearing in the Dashboard section of the client's interface.
: The first thing the end-user see when he is accessing the client's interface is the dashboard, including your custom '''Welcome message'''.
: You can fully customize it by choosing the font, color, size and add images with the text editor.

=== Low Balance Email ===

: If you have configured your client to receive a notification when their balance reaches a certain threshold, you can customize this email they will receive.
: You can customize the body of the email by adding some essential information, such as the current balance and their threshold.
: These are the possible [variable] you can add to the email. Each [variable] will be replaced by the proper information when sent.
:: '''[firstname]''': Will be replaced by the client's first name stated in the client's account.
:: '''[lastname]''': Will be replaced by the client's last name stated in the client's account.
:: '''[balance]''': Will be replaced by your client's current balance account
:: '''[threshold]''': Will be replaced by your client’s balance threshold set in his account.

=== Reset Password URL Recovery Email ===

: You can customize the body of the email your clients will receive when they use the link "Forgot my password" from the login screen to reset their client's account password.
<div style="margin-left:50px;
padding:5px;
border-left: 6px solid rgb(208, 56, 45, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(51, 51, 51, 0.08);
width:59%;">

'''NOTE''' 💬: ➜ The email title and body field are mandatory.
</div>

: These are the possible '''[variable]''' you can add to the email. Each '''[variable] will be replaced''' by the proper information when sent.

:: '''[firstname]''': Will be replaced by your client's first name stated in the client's account.
:: '''[lastname]''': Will be replaced by your client's last name stated in the client's account.

<div style="margin-left:50px;
clear:left;
padding:5px;
border-left: 6px solid rgb(66, 192, 251, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(66, 192, 251, 0.05);
width:59%;">

🔗 '''Related topic with this section''' | Go to the [[#Display_Elements|Display Elements]] tab section.
</div>

=== Reset Password Completed Email ===

: When your client has reset his password successfully, they will receive a second email confirming that the password has been reset successfully. You can also customize this one.
<div style="margin-left:50px;
padding:5px;
border-left: 6px solid rgb(208, 56, 45, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(51, 51, 51, 0.08);
width:59%;">

'''NOTE''' 💬: ➜ The email title and body field are mandatory.
</div>

: These are the possible '''[variable]''' you can add to the email. Each '''[variable] will be replaced''' by the proper information when sent.

:: '''[firstname]''': Will be replaced by the client’s first name stated in the client's account.
:: '''[lastname]''': Will be replaced by the client’s last name stated in the client's account.

<div style="margin-left:50px;
clear:left;
padding:5px;
border-left: 6px solid rgb(66, 192, 251, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(66, 192, 251, 0.05);
width:59%;">

🔗 '''Related topic with this section''' | Go to the [[#Display_Elements|Display Elements]] tab section.
</div>

== Payment ==
: The reseller portal and client's interface offer a payment processing tool.
: You can link your business PayPal account to receive your client deposit directly into your business PayPal account.
: All payments made in the client's interface are automatically credited to your client's virtual balance.
: To enable this feature, you will need to input your PayPal email account, input the amounts your customers can deposit.
: Each amount will be displayed via a drop-down menu. The different amounts that you have established must be separated by a comma.

<div style="margin-left:50px;
padding:5px;
border-left: 6px solid rgb(208, 56, 45, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(51, 51, 51, 0.08);
width:59%;">

'''NOTE''' 💬
: Activating "payment" is '''not mandatory'''. <br/>
: ''You still have the ability to credit/charge the client's account manually in your '''"Manage Client's Accounts"''' page, <br/>
: or simply leverage our API method to credit/charge manually your client's account.''
</div>

== Hostname ==
: By default, you have the generic URL including your VoIP.ms ID in it to allow your client to access their client interface. The domain name is voipinterface.net.
: You can customize this URL with a custom hostname associated with your current domain name. You can request the change for a '''one-time fee of $10'''.
: Once done, your client will have the ability to use your custom hostname to access your white-label client interface.

: In your domain management tool, you will need to configure a hostname with an '''A RECORD''' in your '''DNS''' that points to '''208.100.60.92'''.
: The hostname usually consists of '''host'''.domain.tld.

: For example, if your domain is ''voipxyzservice.com'' you should enter a hostname that looks like '''''interface'''.voipxyzservice.com''.

= Manage Rates and Packages =

: This section of the reseller portal will allow you to manage custom packages and associate them to your client with his sub-account. To create a new package click on the button [[File:Reseller_Guide_Detailed_CreateNewPackage.png|100px|Create a new package]].
: You can customize your outbound markup, your default rate per minute for a specific destination, create a bundle of free minutes or free zone and customize the services and features you want to give your client access to.

: In this section, we will review these tabs and their content.
:: [[File:Reseller_Guide_Detailed_CreatePackage_tabs.png|border|350px]]

== Create/Edit new package ==
=== Details Tab ===
: '''Package name''': The package name will be the name of your package.
: The package name will appear in the transaction history of your client.

: '''Outbound markup''': The outbound markup is a benefit you want to have on the outbound calls. You can set a fix amount or a percentage, or both.
: The Outbound Markup is calculated on top of your per-minute cost rate.

:: Example: Your VoIPms Cost is: 0.009
:: Fix Rate: 0.0025$
:: Percentage Rate: 20%
:: Your client will be charged: 0.0138$/minutes

: '''Pulse''':
: The "Pulse" is the billing increment.
: This is the charge increment you will use for this specific package.
: For instance, a pulse of 60 will make calls charged by the minute or having a pulse of 6 will make calls being charged every 10 seconds. '''Do note that this solely affects outgoing calls. '''
: You can read more about this in our Calls Cost article.

: '''Canada and International Route''':
: For termination (outbound call), VoIP.ms offers two different routes (Value and Premium) for Canada and only a Premium route option for the United States.
: You can choose between Value and Premium Route for client's International calls.
<div style="margin-left:50px;
padding:5px;
border-left: 6px solid rgb(208, 56, 45, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(51, 51, 51, 0.08);
width:59%;">

'''NOTE''' 💬: ➜ Features like DTMF, DISA, Callback and Caller ID can only be guaranteed while using Premium Route.

</div>
: '''VoIP.ms to VoIP.ms Calls''':
: By checking this box, you will allow free calls between VoIP.ms customer.
: The free internal extension to the extension calling is free by default.

=== Fees Tab ===
: '''Monthly fee''':
: Here, you can input a value you want to charge as the monthly base rate.
<div style="margin-left:50px;
padding:5px;
border-left: 6px solid rgb(208, 56, 45, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(51, 51, 51, 0.08);
width:59%;">

'''NOTE''' 💬
: ➜ If you connect a package with a monthly fee on more than one client's connected sub-account, it will multiply the monthly charges by each connected package.

</div>
: '''Setup fee''':
: This is to charge a predefined one-time fee when connecting the package to a client.

=== Free minutes Tab ===
: '''Free total''':
: This option is to offer free minutes (inbound & outbound calculated).
: Free minutes will be calculated from the billing date for the package and will be calculated in total rounded minutes (for example, 3 minutes 11 seconds will be calculated as 4 minutes).

=== Reseller system configuration ===
:'''Sub account''':
[[File:Reseller_Guide_Detailed_UserEditSubAccount.png|right|border|300px|Client can edit his Sub account (If enabled)]]
:: '''User can edit sub account'''
::: If you allow your client to edit their sub accounts, they will have access to manage the sub account setting.
::: '''If enabled''', the end-user will be able to modify details of the sub account such as Password, Device Type, Authentication Type, CallerID Override, Music on Hold, Allowed Codecs, DTMF Mode and NAT.
::: The description of this is also described in the client's interface guide.

----

: '''Services configuration on the reseller'''
:: '''Type of configuration'''
::: This option determines the configuration of the services that will be shown in the client's interface.

: '''Default configuration'''
:: When the Default Configuration is set, the services displayed will be the ones configured at '''[Reseller]''' > '''[Client Branding and Access]''' > '''[Services tab]'''.

: '''Package configuration'''
:: If the '''[Package Configuration]''' is selected, you will be able to set which services a client or a user within the same client's account will be able to manage.

: '''View Setting'''
:: The View Setting options offer the ability to display or not information in the CDR and on the Balance page or to remove the service page view in a bulk way for a specific user.

:: '''Your current view is set to:'''
::: '''Standard View'''
:::: The client's user will see every available information and service.

::: '''Simple View''':
:::: This hides some features like:
:::: Call Types ''(busy, answered, failed & no answer)'' filters in the CDR
:::: This month and today's amount spent on the Balance page
:::: Services-specific information (Register type, Allowed codecs, etc.) on the services page

::: '''Simple View (Hide Services Page)''':
:::: This view will change the same displayed information as "Simple View", but will also hide the service page and withdraw all the service/feature management options.

: '''Services''' (Show)
:: In this section, you can select which services will be available and which services will not be available for your client or a specific user within your client's account.
:: Simply enable or disable the service to allow which feature the client can (or not) manage by themselves.

:: Moreover, you can withdraw all money-related components from the client's interface.
:: Additional information at the Tips & Trick section.

: DIDs/incoming calls charges:
<div style="margin-left:50px;
clear:left;
padding:5px;
border-left: 6px solid rgb(66, 192, 251, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(66, 192, 251, 0.05);
width:59%;">

🔗 '''Related topic with this section''' | Go to the [[#Add_Phone/Fax_DIDs_to_your_clients|Add Phone/Fax DIDs to your clients]] guide section.
</div>

: Edit your packages
<div style="margin-left:50px;
clear:left;
padding:5px;
border-left: 6px solid rgb(66, 192, 251, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(66, 192, 251, 0.05);
width:59%;">

🔗 '''Related topic with this section''' | Go to the [[#Manage_Rates_and_Packages|Manage Rates and Packages]] guide section.
</div>

=== Sub Account Relation With Package ===
: The package and the sub account are closely related. You need to link the package to the sub-account and not to the client's account.
: That being said, in a Hosted-PBX setup, you can create packages to reflect a '''"per user"''' pricing, allow or restrain your client from managing specific features.

: If you connect your own PBX, you can create a package that will reflect the overall billing/pricing of the client and chose which service/feature they can manage by themselves.

== Edit Rate – (Per Area rates)==

: If you want to set different rates depending on the destination of the call, you can do this with the '''edit rates [[File:VoIPms_edit_rates_yellow.png|edit rates yellow icon]]''' icon from the package, all the custom rates you enter there will override the global markup. You can set a custom rate for all the calls to USA or Canada, by country or Calling code.

=== Packages specific settings ===
: Rates changed per area with this function will affect only the related packages and the related user connected to this package.

=== Free Zone ===
: This function allows offering free zone calling in bulk per country, per calling code/prefix or per a maximum calling per minute cost rate.

= Manage client's account =

: This part of the reseller portal is to create and manage the client's account.
: When arriving on this page, there is a quick view of all active client's accounts.
: We can find the User ID, the Email / Client username, Phone number, Status, and the Last login date.

: It is also where you can Edit Client '''(1)''', Manage Client '''(2)''' and Login to clients' interface '''(3)'''
: See picture below:

::[[File:Reseller_Guide_Detailed_Manage_Clients_Account.png|650px|border]]

== How to add (create) a client ==

: To create a client, simply go to over the '''[Reseller]''' tab, then '''[Manage Client's Accounts]''' and '''[Create a new client account]'''.
: Once you click on '''[Create a new client account]''' you will arrive on this page.
: See below the picture, the description of each field.

:: '''Firstname*''' = First name of your client.
:: '''Last name*''' = Last name of your client.
:: '''Company''' = Name of the company of your client. ''(if applicable)''.
:: '''Address''' = Address of service of your client.
:: '''City''' = City of the service address of your client.
:: '''State''' = City of the service address of your client.
:: '''Country''' = Country of the service address of your client.
:: '''Zip''' = Zip Code of the service address of your client.

:: '''Email''' = The email address of your client. It will also be used as the username for your client to have access to the client's interface.
:: '''Password''' = Your client password to access the client's interface.

:: '''Phone number''' = The contact phone number of your client.
:: '''Next billing date''' = The next date the charges associated to the packages will be applied to the client's account.
:: '''Setup fees''' = When you created the Packages, if you did input a Setup fee, you can charge them at the creation of the client's account by checking this box.

: '''Balance Management''':
:: You can use this option to automatically suspend your client’s services if their account balance reaches 0$.

: '''Sub account''':
:: You can create or connect an already existing sub account to the client's account.

=== Sub account connections ===

: Packages are connected to the sub account and the sub account is connected to the client's account.
: You need at least one sub account connected to the client's account to enable a package to the client's account.

== Edit Client ==
: This function is to edit client's information, delete client's accounts or set a Low balance email threshold.
: Once a client's account is created, you will see his information layout like this:

== Client information ==
: Here you can edit previously input client's information.
: Refer to the previous section for more information.

=== Balance Management ===
: You can use this option to automatically suspend your client's service if their balance reaches 0$.

=== Delete Client ===

: This function is to delete a client's account completely. This action cannot be reversed.
=== Low Balance Email ===
: You can set the portal to send an email to the client when his balance reaches a threshold of your choice between 10$ and 500$.
: The email can be sent to a custom email address which it can be different from the client’s interface username.

== Manage client ==

: Once you click on the [[File:VoIPms_manage_client_button.png|Manage client]] icon''', you will have access to Assign Package, apply payment or charges manually, see the history of transactions of the client, link or unconnect phone or fax numbers and unconnect sub account.
: Finally, you get an overall view of what is connected to a client's account and the routing of the DIDs for this given client.

=== Assign a package ===

: [[File:Reseller_Guide_Detailed_Manage_Clients_Assign_Package.png|border|left]] This function allows assigning a package to a sub account.
: Once you click on '''[assign package]''', you will be requiring selecting an existing sub account or create a new one.
: When you click on '''[Next]''', you will land on a sub account configuration page.
: If it is a new sub account, simply fill all the needed information.
: On this page, to complete the package assignment, scroll down to '''"Reseller Configuration"''', use the drop-down menu and select the appropriate package.
: Go to the bottom of the and click "'''Update Account"'''
:: [[File:Reseller_Guide_Detailed_Manage_Client_AssignPackage.png]]

=== Add Payments/Charges - Transaction history ===

: [[File:Reseller_Guide_Detailed_Manage_Clients_Pmt-Charge-History.png|border|left]] This option simply gives the ability to write payment or charges manually to the client's account.
: It is also where we can see the history of transactions.
: Whether is a payment or a charge, it is possible to write the amount and the description of the transaction.
: Each manual transaction will be dated for the same day in the transaction history.
: The transaction history will show each payment/deposit and each charge.
: You also have the option to delete a transaction.

=== Add Phone/Fax DIDs to your clients ===
: This function is to connect a DID to an already linked sub account.
: It is also here the reseller can set incoming per minute rates and DIDs monthly charges, a call forwarding and set-up a voicemail.
: In the column '''[# Phone / Fax numbers]''' click on the link [add phone numbers] to connect a DID to a client's account.
: The following section will explain the DID connections page and rates configuration.

: '''Connect a DID for charging'''
: This submenu will offer, already purchased/ported-in, available DIDs you have in your VoIP.ms account.
: Use the drop-down menu to select the DID or click on '''[Advance]''' to connect multiple DIDs to the same sub-account.

: '''Setting up the charges for those DID(s)'''
: This submenu is useful to set a monthly charge for the DID(s), a set-up fee and a custom per minute rate.
: If in the previous submenu, you selected multiple DIDs, the monthly charge will be multiplicated by the same number of DIDs selected. Same result for the setup rate.
: The option '''[Set per minute rate]''' is to set a per-minute cost on the incoming call received on a given DID.

: '''Forwarding these DID(s) to a local number'''
This submenu allows creating a call forwarding to a local phone number as a cell phone instead of routing the incoming calls to an IP phone or an IVR.
<div style="margin-left:50px;
padding:5px;
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box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(51, 51, 51, 0.08);
width:59%;">

'''NOTE''' 💬
: ➜ The dual connections billing rules apply here. You will be billed per minute for the '''incoming''' call '''AND''' be billed for the '''outgoing call''' to establish the connection to the local phone number.
</div>

: '''Add a voicemail'''
: This submenu allows creating and connecting a voicemail to the selected DID(s).
: To create a voicemail from this menu, simply check the box "Add a voicemail for these number(s)" and complete all the field.
: At the field "Email" mention the email address, the client wants to receive his voicemail messages in a wave format.

: Finally, click on '''[Connect DID(s) to client]''' to confirm the connection of those DIDs to the given client.

=== How to connect a sub-account to a client ===
: There are two methods to connect an existing or a new sub-account to a client.

: '''Assign package'''
::* Under the '''[Sub Accounts]''' tab, go to '''[Manage Sub-Accounts]'''.
::* Locate the existing sub account you wish to connect and click on the '''edit [[File:VoIPms_edit_yellow.png|Yellow Edit icon]] icon'''
::* Scroll down and locate the '''[Reseller configuration]''' section.
::* Choose your '''client''', '''package''' and '''billing date'''. You can apply the '''one-time''' setup fee by checking the box. ''(Only if you have specified a one-time setup fee in your package.)''
::: [[File:Reseller_Guide_Detailed_Manage_Client_AssignPackage.png]]

: '''Create a sub account'''
::* Under the '''[Sub Accounts]''' tab, go to '''[Create Sub-Account]'''.
::* Fill in all the necessary fields for your client.
::* Locate the '''[Reseller configuration]''' section.
::* Choose your '''client''', '''package''' and '''billing date'''. You can apply the '''one-time''' setup fee by checking the box. ''(Only if you have specified a one-time setup fee in your package.)''
::: [[File:Reseller_Guide_Detailed_Manage_Client_AssignPackage.png]]

----

=== How to unconnect sub-account or DID ===
: Before deleting a client’s account or to simply withdraw a user and/or a DID within a client’s account, you will be required to '''unconnect''' the sub account and/or the DID.
: Simply go to the '''“Manage Client’s accounts”''' page ''(Under Reseller)'' then click on the '''"Manage client" [[File:VoIPms_manage_client_button.png|Manage client]] icon''' beside the chosen client.
: Once in the client’s account, it is possible to individually '''unconnect''' [[File:VoIPms_manage_client_unlink.png|Unlink]] a sub account (user/device), a DID or a fax number.

: '''Unconnect a Sub account:'''
:: [[File:Reseller_Guide_Detailed_Manage_Client_Unconnect-SubAccount.png|650px]]

: '''Unconnect a Voice DID:'''
:: [[File:Reseller_Guide_Detailed_Manage_Client_Unconnect-DID.png|650px]]

: '''Unconnect a FAX number:'''
:: [[File:Reseller_Guide_Detailed_Manage_Client_Unconnect-FAX.png|650px]]

----

=== Activate client ''(autonomous signup)'' ===

: When your client signed up directly with your signup link. Their account will be automatically created with a '''"WAIT"''' status.
: You will need to activate the account through your portal in the section '''"Manage Client's Accounts"'''.

: Click on the '''edit [[File:VoIPms_edit_yellow.png|Yellow Edit icon]] icon''' beside the profile of your client.
: When in the profile you will find the [[File:VoIPms_manageclient_activate-client.png|90px|Activate client]] '''[Activate client]''' button and the '''Delete client''' button.

:: [[File:Reseller_Guide_Detailed_Manage_Client_WAIT_edit.png|650px]]
:: [[File:Reseller_Guide_Detailed_Manage_Client_WAIT_Activate.png|650px]]

== Login into client's interface ==
: In the "Manage client's Accounts", you will find a blue arrow button [[File:Reseller_Guide_Detailed_Manage_Clients_LoginAsClient.png]],
: This function is to access the client's interface to see it as the client sees it.
: It can be useful to create a feature component on behalf of the client.
<div style="margin-left:50px;
padding:5px;
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box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(51, 51, 51, 0.08);
width:58%;">

'''NOTE''' 💬
: ➜ To link any Hosted-PBX feature to a client's account and make it appear to the client's interface, you need to build the component from the client interface.

</div>

= Reports =
: In the '''[Reports]''' Tab you can generate the '''Call Detail Records''' for the calls your clients have made and received.
: This will show you the total net earnings gained by establishing the rates on the created packages and gather the information, either per user or for all customers connected to the reseller portal.

: Reports are divided in the following categories:

== Global CDR ==
: For a selected period, this report gives a Call Detail Record overall view of '''all users combined'''.
: The cost, the selling price and net profit for each call.
: You can also see all the calls ''(incoming and outgoing)'' from all your clients.

== Per User CDR ==
: For a selected period, this report gives a Call Detail Record overall view of '''a specific user'''.
: Your cost, your selling price and the net profit for each call.

== Per User FAX ==
: For a selected period, this report gives an overall view '''of a specific user''' faxes usage.
: It also displays the cost, the selling price and the net profit for each fax.

== Financial Report ==
: This is a month-to-month view of revenues, deposits and costs related to all the reseller’s clients.

== Sub Accounts Report ==
: List of all sub-accounts connected to the reseller clients, with related information.
: Here you will find a list of the sub account's associated clients, sub-account, sub account’s associated packages and DID numbers associated with each.

== DIDs Report ==
: List of all DID's connected to the reseller's clients, with related information such as related package, associated sub accounts and the rate center of each DID.

= Useful API methods with the reseller portal =

<div style="margin-left:50px;
padding:5px;
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box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(51, 51, 51, 0.08);
width:58%;">

⌛ Coming soon. Please refer to the API documentation. https://voip.ms/m/apidocs.php

</div>

= Reseller Tips & Tricks =

'''Tip #1''' - Use your custom ''hostname'' as a VoIP PoP .
----
: If you have your own '''DNS server''' managing your domain name or if your registrar offers you the possibility to manage the '''"DNS zone"''' of your domain name, you can create a new host that will point to the IP address of the desired PoP server.

::* The new record needs to be a '''type A''' ''(A Records)'' with the value of your choice. ''(E.g. "voip1" that will be '''voip1'''.domain.tld)''.
::* The '''TTL''' value will be 300 seconds and the '''pointer/IP''' will be the IP address of the wished PoP Server.
::* If the class is needed to specify, indicate '''IN''' ''(for internet)''.

'''Tips #2''' - Using your own payment gateway.
----
: By using our '''SOAP and REST/JSON API''', you have the ability to use your own payment gateway instead of the integrated Paypal API option.

:You will simply need to add into your customer’s database the '''USER ID''' associated to your client database.
: When your client uses your existing customer portal to process his payment, you can execute a query using our API when the payment is successfully added.
: You will simply need to use the method '''"addPayment"''' of our API, specify the '''UserID''', the Amount and a description.
: The payment will automatically be added into the proper Client account under the VoIP.ms reseller client portal.
: If wished, you can indicate in the welcome message, a link to redirect your customer if they would like to add funds to your billing portal.

: It also possible to use the method '''"addCharge"''' if examples your customer requests a refund or if you would like to apply any charge to clients.

:::: '''https://voip.ms/m/apidocs.php'''

'''Tips #3''' - Link feature to client's account.
----
: To link any Hosted-PBX feature to a client’s account and make it appear to the client’s interface, you need to build it from the client interface.

: For any service that your customer needs to manage and having access to it, the new entry needs to be created directly from its client interface.
: For example, if you create a new voicemail from your VoIP.ms portal, your customer will not be able to see it. You must create this voicemail.
: You can connect '''"as your customer"''' by going to the option '''"management of customer accounts."'''

'''Tips #4''' - Turn a reseller client into a SMS/MMS contact center.
----
: If you prefer to use your own PBX and use us as your SIP trunk provider, but you need to give your client access to SMS/MMS feature. You can create a package for this needs and allow only '''SMS/MMS Message''' service under "Package configuration".

: You will need to connect the sub-account to this client and the package created and attach the DID to the client ''(Manage client's account)''. Your client will be able to use the white label portal to send and receive SMS/MMS without having the ability to manage other service/feature.

'''Tips #5''' - Phonebook for custom caller ID Name.
----
: For each incoming call that the caller ID number is received, the system will check to see if there is a matching entry in the phonebook, in this case, if your customer wishes to associate a specific information or name to a specific caller ID number, an entry can be created in the phonebook with that information. Each time this caller calls one of your DIDs using this "Caller ID number", the name specified in the phonebook will be displayed.

'''Tips #6''' - Turn a reseller client into Fax service manager.
----
: If you prefer to use your own PBX and use us as your SIP trunk provider, but you need to give your client access to send/receive and manage faxes. You can create a package for this needs and allow only '''Manage Faxes''' service under "Package configuration".

: You still will need to connect a sub-account to this client ''(it could be a sub-account that they will never be connected)'' this is simply to have the ability to link a package with the restriction to your client. Then you will need to attach the Fax DID to the client ''(Manage client's account)''. Your client will be able to use the white label portal to send/receive and manage faxes without having the ability to manage other service/feature.

= Videos - Diving Into the Reseller Portal =

:* [https://www.youtube.com/watch?v=QsCOsStTpUo&list=PLhH3IcHzMUd_uUWNNuYI3syotXSt7aN8D&index=2| Diving Into the Reseller Portal - Part 1]
:* [https://www.youtube.com/watch?v=pkqcgTw9UOE&list=PLhH3IcHzMUd_uUWNNuYI3syotXSt7aN8D| Diving Into the Reseller Portal - Part 2]
:* [https://www.youtube.com/watch?v=83ArwHjMyQQ&list=PLhH3IcHzMUd_uUWNNuYI3syotXSt7aN8D&index=3| Diving Into the Reseller Portal - Part 3]

</div>

Costo del servicio

2025-12-02T19:22:32Z

RP:

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

<div style="margin-left:20px;
clear:left;
padding:5px;
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box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(208, 144, 45, 0.08);
width:70vw;">

:'''💬 NOTE 💰 '''
: En este momento, todas las '''tarifas y precios''' mostrados en nuestro sitio web y en el portal de clientes, así como cualquier depósito realizado a través de nuestro procesador de pagos mediante tarjeta de crédito o a través de su institución financiera para transferencias bancarias, se procesan '''en dólares estadounidenses (USD)'''. Al usar billeteras digitales como PayPal o billeteras móviles como Apple Pay y Google Pay, las transacciones también se procesan '''en dólares estadounidenses (USD)''', pero es posible que ya se realice una conversión de moneda y se muestre el monto convertido a su moneda local en el momento del pago. Tenga en cuenta que su institución financiera también puede imponer tarifas y tasas de cambio adicionales para las transacciones con tarjeta de crédito o transferencias bancarias.
</div>

[http://voip.ms/ VoIP.ms] es básicamente un proveedor a nivel mundial de llamadas salientes y entrantes. En el sistema de [http://voip.ms/ VoIP.ms], las llamadas entrantes y salientes se cobran de manera independiente. El servicio puede ser usado para llamadas salientes únicamente, ya que no es necesario adquirir un número DID si no desea recibir llamadas.
[http://voip.ms/ VoIP.ms] le permite utilizar uno de sus propios [[Devices | dispositivos ]] ) (hardware como Linksys ATA o software como [[X-Lite|X-lite]]).
[http://voip.ms/ VoIP.ms] es un proveedor de servicios de prepa y debe depositar un mínimo de $ 15 en su cuenta para poder utilizar el servicio. Todos los cargos se deducirán de ese saldo. Su cuenta no podrá ni recibir ni realizar llamadas una vez que su saldo llegue a cero.

Registrarse es gratis. Puede navegar, verificar funciones, configurar su dispositivo y marcar números de prueba de eco y prueba DTMF incluso sin agregar fondos.

El cliente puede cambiar la ruta (Value / Premium) a premium en la sección [[Configuraciones de la Cuenta (Account Settings)]] del portal del cliente. De forma predeterminada, las cuentas nuevas están configuradas para utilizar la ruta de value. La ruta Value / Premium se aplica por separado para Canadá y llamadas internacionales. La ruta Premium es la única ruta disponible para U.S.

La ruta ('''Value o Premium''') puede ser editada por el cliente en la sección de '''"Configuración de cuenta"''' desde el portal. Por defecto, las cuentas nuevas están configuradas en la ruta Value. Las rutas Value y Premium se configuran de manera separada para Canadá, las llamadas internacionales y Toll Free. Para Estados Unidos (US48) solo se maneja con '''ruta Premium'''.

== Llamadas Salientes ==

=== Tarifas a USA ===

*Ruta Premium: $ 0.0100 (1 ¢) por minuto
*Números de lada sin costo (Toll free) Value Route: Gratis
*Números de lada sin costo (Toll free) Ruta Premium: $ 0.0106 (1.06 ¢) por minuto
*e411: $ 0.99 por llamada. Debe ser activado por el cliente en las configuraciónes de la cuenta

=== Tarifas a Canadá ===

*Ruta Value: la mayor parte de Canadá a $ 0.0052 (1/2 ¢)
*Ruta Premium: $ 0.009 (0.9 ¢) por minuto
*Las tarifas para todos los destinos de Canadá se encuentran en la sección de tarifas [http://www.voip.ms/rates.php rates]

=== Tarifas Internacionales ===

Lista disponible en la sección de [http://www.voip.ms/rates.php rates]

Frecuencia de actualización de tarifa: A veces, se bajaran o subiran las tarifas en algunos destinos si es necesario

=== Incrementos de cobro ===

*USA y Canadá: incremento de 6 segundos
*Internacional: incremento de 6 segundos
*México: 60 segundos inicial / incremento de 60 segundos
*Consulte [[Costo por llamada]] para obtener más información.

== Llamadas entrantes ==

Disponibilidad de números DID de USA y Canadá
*48 estados continentales de US
*10 provincias canadienses
*3 territorios canadienses

=== Precio DID al por minuto de USA / Canadá ===

*Precio mensual:
USA: $1.10
Canadá: $1.10

*Por minuto entrante:
$ 0.009 a $ 0.0125

*Tarifa única de activación: $ 0.40
*Incremento de facturación: 6 segundos
*Canales: 25
*Uso previsto: Cualquiera

=== Precio DID mensual fijo de U.S./Canadá ===

*Precio mensual:
USA: $ 4.95
Canadá: $ 4.95

*Por minuto entrante: gratis, hasta 3500 minutos al mes
*Tarifa única de activación: $ 0.85
*Incremento de facturación: no aplicable
*Canales: 2
*Uso previsto: residencial, hasta 3500 minutos ENTRANTES al mes

== Funciones adicionales ==

*E911 / 911 $ 1.50 al mes
*Búsqueda de [Numero Identificador (Caller ID) | Nombre de identificador de llamas]] de CNAM: $ 0.8 centavos por consulta.
-Cuentas de buzón de voz [[Buzón de voz (Voicemail)]] ilimitadas

La búsqueda de CNAM se cobrará si la llamada coincide con un número de (callerID) de América del Norte (NPANXXXXXX), ya sea que se responda o no la llamada.
No se le cobra por Búsqueda CNAM si el número está en su directorio telefónico del portal del cliente.

== Precios de números DID con lada sin costo (Toll Free numbers) ==

=== Del proveedor de USA===
Acepta llamadas de USA de forma predeterminada y se puede solicitar desbloquearlo para aceptar llamadas entrantes de Canadá también.

*Costo mensual: $ 0.99
*Por minuto entrante:
*USA / $ 0.019 (1.9 ¢)
*Canadá (solicitandolo únicamente, desbloqueado a solicitud) / $ 0.08 (8 ¢)
*Alaska (solicitandolo únicamente, desbloqueado a solicitud) / $ 0.17
*Puerto Rico (solicitandolo únicamente, desbloqueado a solicitud) / $ 0.095
*Incremento de facturación: 6 segundos
*Tarifa única de activación: no aplicable
*Canales: 25
*Un número personalizado de lada sin costo conlleva un cargo de activación de $15 y tiene un tiempo de entrega aproximado de 1 a 5 días hábiles.

=== Del proveedor de Canadá ===
Acepta llamadas entrantes tanto de USA como de Canadá, todas a la misma tarifa de entrada.

*Cuota mensual: $ 1.25
*Por minuto entrante:
*Canadá: $ 0.027 (2.7 ¢)
*Estados Unidos: $ 0.027 (2.7 ¢)
*Alaska: $ 0,17
*Puerto Rico: $ 0.095
*Incremento de facturación: 6 segundos
*Canales: 25
*Un número personalizado de lada sin costo conlleva un cargo de activación de $ 30.

== Precios DID internacionales ==
El precio varía según el país. Desde su portal del cliente, consulte DID Numbers >Order DID> International.
'''Por favor, póngase en contacto con nuestro departamento de ventas enviando un correo electrónico a [mailto:ventas@voip.ms ventas@voip.ms] para obtener más información'''

====Facturación de DID Internacionales====

*Tarifa mensual fija
*Llamadas entrantes ilimitadas
*Canales: 2 '''Nota: Algunos DID internacionales pueden aumentarse adquiriendo un [[PRI_virtual | PRI Virtual]]. Por favor, póngase en contacto con Ventas para obtener más información.
*Cuota de configuración pagada al ordenar el DID. Los pedidos se tratan como pedidos pendientes y se requiere documentación adicional para completarlos. El tiempo de entrega depende de la rapidez y exactitud con la que se proporcionen la información requerida.
*Cuota mensual facturada el 1 de cada mes

[[category:Guías]]

Coût des Services Rendus

2025-12-02T19:21:46Z

RP: /* Plan Taux Fixe É.-U. / Canada */

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

<div style="margin-left:20px;
clear:left;
padding:5px;
border-left: 6px solid rgb(208, 149, 45, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(208, 144, 45, 0.08);
width:70vw;">

:'''💬 NOTE 💰 '''
: Actuellement, '''tous les tarifs et prix''' affichés sur notre site et dans le portail client, ainsi que tous les dépôts effectués via notre processeur de paiement par carte de crédit ou par virement bancaire auprès de votre institution financière, sont traités en '''dollars américains (USD)'''. Lorsque vous utilisez des portefeuilles numériques comme PayPal ou des portefeuilles mobiles tels qu'Apple Pay et Google Pay, les transactions sont également traitées en '''dollars américains (USD)''', mais une conversion de devise peut déjà être effectuée, affichant le montant converti en votre monnaie locale au moment du paiement. Veuillez noter que votre institution financière peut également imposer des frais et des taux de change supplémentaires pour les transactions par carte de crédit ou par virement bancaire.
</div>

'''VoIP.MS''' est essentiellement un service de fournisseur dans tout le monde pour Termination Calls ('''appels sortants''') et Origination ('''appels entrants'''). Dans le système VoIP.ms, les appels entrants et sortants sont facturés séparément. Le service peut être utilisé uniquement pour les appels sortants. Il n'est pas nécessaire pour vous d'obtenir un DID (numéro de téléphone) si vous n'avez pas besoin de recevoir des appels entrants.

VoIP.ms vous permet d'utiliser un de vos propres [[Devices]] (dispositifs comme un ATA Linksys (Adaptateur téléphonique) ou un logiciel comme [[Softphones#Zoiper_Classic | Zoiper]]).

'''VoIP.ms''' est un fournisseur de services prépayés. Vous devez effectuer un dépôt minimum de $15 dans votre compte pour pouvoir utiliser les services. Tous les frais seront déduits du solde de votre compte. Votre compte ne sera pas en mesure de recevoir ou passer des appels lorsque votre solde atteint zéro.

L'inscription est libre de toute charge. Vous pouvez naviguer autour, vérifier les caractéristiques, configurez votre appareil et le tester avec notre "'''Test de qualité sonore'''", même sans l'ajout de fonds.

La '''Route''' (Value ou Premium) peut être sélectionnée au choix par le client dans les [[Paramètres du compte]] sur le portail client. Par défaut, les nouveaux comptes sont configurés pour utiliser la route "Value". Les routes Value/Premium s'appliquent séparément pour US48/CA et l'international.

__TOC__

== Appels sortants ==

=== Tarifs É.-U. ===
:* Premium Route: 0,01 $ (1¢) par minute
:* Numéro sans frais (800): gratuit
:* Numéros sans frais route Premium (800): 0,0106 $ (1,06 ¢) par minute
:* E411: 0,99 $ par appel. Il doit être activé par le client dans les paramètres du compte.

=== Tarif pour le Canada ===

:* '''Route Value''': La plupart du Canada $0.0052 par minute.
:* '''Premium route''': $0.009 par minute.
:* Toutes les destinations du Canada sont sur la liste des tarifs. [http://www.voip.ms/rates.php Tarifs]

=== Tarif International ===
:* Liste disponible sur [http://www.voip.ms/rates.php Tarifs].
:* Fréquence actualisation des tarifs: Rare. Parfois, nous allons augmenter ou baisser le taux sur certaines destinations, si nécessaire.

=== Facturation par Tranches ===

:* '''É.-U. et Canada''': 6 secondes
:* '''International''': 6 secondes
:* '''Mexique''': 60 secondes initiales / 60 secondes.

== Appels Entrants ==
: '''Disponibilité des numéros É.-U./Canada '''
::* 48 États des États-Unis
::* 10 Provinces du Canada
::* 3 territoires du Canada

=== Plan par minute pour les numéros É.-U./ Canada ===
:* '''Taux mensuel''':
:: '''É.-U.''': <s>($0.99 à $1.49)</s> $1.10
:: '''Canada''': <s>($0.99 à $1.99)</s> $1.10<br/>
::* '''Taux par appel reçu par minute''':
::: <s>($0.01 à $0.0149)</s> $0.009 à $0.0125
::* '''Frais d'activation''': $0.40 <s>($0.50)</s>
::* '''Facturation par tranches''': 6 secondes
::* '''Canaux (channels)''': Haute capacité de canaux ''(25 canaux sur ce plan, ça veux dire que vous pouvez recevoir jusqu’à 25 appels en même temps)''

=== Plan Taux Fixe É.-U. / Canada ===
: '''Taux mensuel''':
::* '''É.-U.''': $4.95
::* '''Canada''': $4.95
:* '''Taux par appel reçu par minute''': gratuit jusqu’à 3500 minutes.
:* '''Frais d'activation''': <s>($1.00)</s> $0.85
:* '''Canaux (channels)''': 2 (Votre numéro DID pourra recevoir jusqu’à 2 appels en même temps)
:* '''Utilisation''': Résidentiel

===Service Supplémentaires===
:* [[Service E911]]/911 $1.50 par mois
:* CNAM [[ID de l'appelant]] name Lookup - $0.008 par recherche.
:* [[Messagerie vocale]] il y a aucune limite de boites vocales.

==Prix des Numéros Sans Frais (Toll Free Numbers) ==
===Pour un opérateur des É.-U.===
: Il accepte les appels des États-Unis par défaut et il peut être débloqué sur demande pour accepter les appels entrants provenant du Canada aussi.
:* Taux Mensuel: $0.99
:* Taux par appel reçu par minute:
:* É.-U. / <s>$0.024</s> $0.019
:* '''Canada''' ''(Déblocage sur demande'') / $0.08
:* '''Alaska''' ''(Déblocage sur demande)'' / $0.17
:* '''Puerto Rico''' ''(Déblocage sur demande)'' / $0.095
:* '''Facturation par tranches''': 6 secondes
:* '''Frais d'activation''': Aucun
:* '''Canaux (channels)''': 25 (Augmentation par requête)

===Depuis un opérateur canadien===
:Il accepte les appels entrants des États-Unis et du Canada, tous aux mêmes prix.
::* '''Taux Mensuel''': $1.49
::* Taux par appel reçu par minute:
::* É.-U. / $0.032
::* Facturation par tranches: 6 secondes
::* Canaux (channels): 25 (Augmentation par requête)

:* Taux Mensuel: <s>($1.49)</s> $1.25
:* Taux par appel reçu par minute:
:: Canada: <s>$0.032</s> $0.027
:: É.-U.: <s>$0.032</s> $0.027
:: Alaska: $0.17
:: Puerto Rico: $0.095
:* Facturation par tranches: 6 secondes
:* Canaux (channels): 25

==Prix des numéros internationaux==

:* Prix varie selon le pays, veuillez consulter votre compte > Numéros DID > [https://www.voip.ms/m/dids.php | Commander numéros DID] > [https://www.voip.ms/m/intldids.php | Numéros Internationaux].
:: '''Veuillez contacter notre service commercial en envoyant un e-mail à [mailto:ventes@voip.ms ventes@voip.ms] pour plus d'informations'''

====Facturation des DID internationaux====
:'''Forfait mensuel'''
:* Appels entrants illimités
:* '''Canaux''': 2 '''''Remarque : Certains DID internationaux peuvent être augmentés en acquérant un [[PRI_Virtuel | PRI Virtuel]]. Veuillez contacter le service de vente pour plus d'informations.''
:* Frais de configuration payés lors de la commande du DID. Les commandes sont traitées comme des commandes en attente et des documents supplémentaires sont nécessaires pour les compléter. Le délai de livraison dépend de la rapidité et de l'exactitude des informations requises.
:* Frais mensuels facturés le 1er de chaque mois

Coût des Services Rendus

2025-12-02T19:20:48Z

RP:

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

<div style="margin-left:20px;
clear:left;
padding:5px;
border-left: 6px solid rgb(208, 149, 45, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(208, 144, 45, 0.08);
width:70vw;">

:'''💬 NOTE 💰 '''
: Actuellement, '''tous les tarifs et prix''' affichés sur notre site et dans le portail client, ainsi que tous les dépôts effectués via notre processeur de paiement par carte de crédit ou par virement bancaire auprès de votre institution financière, sont traités en '''dollars américains (USD)'''. Lorsque vous utilisez des portefeuilles numériques comme PayPal ou des portefeuilles mobiles tels qu'Apple Pay et Google Pay, les transactions sont également traitées en '''dollars américains (USD)''', mais une conversion de devise peut déjà être effectuée, affichant le montant converti en votre monnaie locale au moment du paiement. Veuillez noter que votre institution financière peut également imposer des frais et des taux de change supplémentaires pour les transactions par carte de crédit ou par virement bancaire.
</div>

'''VoIP.MS''' est essentiellement un service de fournisseur dans tout le monde pour Termination Calls ('''appels sortants''') et Origination ('''appels entrants'''). Dans le système VoIP.ms, les appels entrants et sortants sont facturés séparément. Le service peut être utilisé uniquement pour les appels sortants. Il n'est pas nécessaire pour vous d'obtenir un DID (numéro de téléphone) si vous n'avez pas besoin de recevoir des appels entrants.

VoIP.ms vous permet d'utiliser un de vos propres [[Devices]] (dispositifs comme un ATA Linksys (Adaptateur téléphonique) ou un logiciel comme [[Softphones#Zoiper_Classic | Zoiper]]).

'''VoIP.ms''' est un fournisseur de services prépayés. Vous devez effectuer un dépôt minimum de $15 dans votre compte pour pouvoir utiliser les services. Tous les frais seront déduits du solde de votre compte. Votre compte ne sera pas en mesure de recevoir ou passer des appels lorsque votre solde atteint zéro.

L'inscription est libre de toute charge. Vous pouvez naviguer autour, vérifier les caractéristiques, configurez votre appareil et le tester avec notre "'''Test de qualité sonore'''", même sans l'ajout de fonds.

La '''Route''' (Value ou Premium) peut être sélectionnée au choix par le client dans les [[Paramètres du compte]] sur le portail client. Par défaut, les nouveaux comptes sont configurés pour utiliser la route "Value". Les routes Value/Premium s'appliquent séparément pour US48/CA et l'international.

__TOC__

== Appels sortants ==

=== Tarifs É.-U. ===
:* Premium Route: 0,01 $ (1¢) par minute
:* Numéro sans frais (800): gratuit
:* Numéros sans frais route Premium (800): 0,0106 $ (1,06 ¢) par minute
:* E411: 0,99 $ par appel. Il doit être activé par le client dans les paramètres du compte.

=== Tarif pour le Canada ===

:* '''Route Value''': La plupart du Canada $0.0052 par minute.
:* '''Premium route''': $0.009 par minute.
:* Toutes les destinations du Canada sont sur la liste des tarifs. [http://www.voip.ms/rates.php Tarifs]

=== Tarif International ===
:* Liste disponible sur [http://www.voip.ms/rates.php Tarifs].
:* Fréquence actualisation des tarifs: Rare. Parfois, nous allons augmenter ou baisser le taux sur certaines destinations, si nécessaire.

=== Facturation par Tranches ===

:* '''É.-U. et Canada''': 6 secondes
:* '''International''': 6 secondes
:* '''Mexique''': 60 secondes initiales / 60 secondes.

== Appels Entrants ==
: '''Disponibilité des numéros É.-U./Canada '''
::* 48 États des États-Unis
::* 10 Provinces du Canada
::* 3 territoires du Canada

=== Plan par minute pour les numéros É.-U./ Canada ===
:* '''Taux mensuel''':
:: '''É.-U.''': <s>($0.99 à $1.49)</s> $1.10
:: '''Canada''': <s>($0.99 à $1.99)</s> $1.10<br/>
::* '''Taux par appel reçu par minute''':
::: <s>($0.01 à $0.0149)</s> $0.009 à $0.0125
::* '''Frais d'activation''': $0.40 <s>($0.50)</s>
::* '''Facturation par tranches''': 6 secondes
::* '''Canaux (channels)''': Haute capacité de canaux ''(25 canaux sur ce plan, ça veux dire que vous pouvez recevoir jusqu’à 25 appels en même temps)''

=== Plan Taux Fixe É.-U. / Canada ===
: '''Taux mensuel''':
::* '''É.-U.''': <s>($4.95</s>
::* '''Canada''': <s>$4.95 </s>
:* '''Taux par appel reçu par minute''': gratuit jusqu’à 3500 minutes.
:* '''Frais d'activation''': <s>($1.00)</s> $0.85
:* '''Canaux (channels)''': 2 (Votre numéro DID pourra recevoir jusqu’à 2 appels en même temps)
:* '''Utilisation''': Résidentiel

===Service Supplémentaires===
:* [[Service E911]]/911 $1.50 par mois
:* CNAM [[ID de l'appelant]] name Lookup - $0.008 par recherche.
:* [[Messagerie vocale]] il y a aucune limite de boites vocales.

==Prix des Numéros Sans Frais (Toll Free Numbers) ==
===Pour un opérateur des É.-U.===
: Il accepte les appels des États-Unis par défaut et il peut être débloqué sur demande pour accepter les appels entrants provenant du Canada aussi.
:* Taux Mensuel: $0.99
:* Taux par appel reçu par minute:
:* É.-U. / <s>$0.024</s> $0.019
:* '''Canada''' ''(Déblocage sur demande'') / $0.08
:* '''Alaska''' ''(Déblocage sur demande)'' / $0.17
:* '''Puerto Rico''' ''(Déblocage sur demande)'' / $0.095
:* '''Facturation par tranches''': 6 secondes
:* '''Frais d'activation''': Aucun
:* '''Canaux (channels)''': 25 (Augmentation par requête)

===Depuis un opérateur canadien===
:Il accepte les appels entrants des États-Unis et du Canada, tous aux mêmes prix.
::* '''Taux Mensuel''': $1.49
::* Taux par appel reçu par minute:
::* É.-U. / $0.032
::* Facturation par tranches: 6 secondes
::* Canaux (channels): 25 (Augmentation par requête)

:* Taux Mensuel: <s>($1.49)</s> $1.25
:* Taux par appel reçu par minute:
:: Canada: <s>$0.032</s> $0.027
:: É.-U.: <s>$0.032</s> $0.027
:: Alaska: $0.17
:: Puerto Rico: $0.095
:* Facturation par tranches: 6 secondes
:* Canaux (channels): 25

==Prix des numéros internationaux==

:* Prix varie selon le pays, veuillez consulter votre compte > Numéros DID > [https://www.voip.ms/m/dids.php | Commander numéros DID] > [https://www.voip.ms/m/intldids.php | Numéros Internationaux].
:: '''Veuillez contacter notre service commercial en envoyant un e-mail à [mailto:ventes@voip.ms ventes@voip.ms] pour plus d'informations'''

====Facturation des DID internationaux====
:'''Forfait mensuel'''
:* Appels entrants illimités
:* '''Canaux''': 2 '''''Remarque : Certains DID internationaux peuvent être augmentés en acquérant un [[PRI_Virtuel | PRI Virtuel]]. Veuillez contacter le service de vente pour plus d'informations.''
:* Frais de configuration payés lors de la commande du DID. Les commandes sont traitées comme des commandes en attente et des documents supplémentaires sont nécessaires pour les compléter. Le délai de livraison dépend de la rapidité et de l'exactitude des informations requises.
:* Frais mensuels facturés le 1er de chaque mois

Service Cost

2025-12-02T19:19:17Z

RP: /* VoIP.ms Pricing Overview */

{| cellspacing="0" style="width: 100%; border: 0; padding: 0; background: transparent;"
|-
| style="vertical-align: top;" |

{| cellspacing="0" style="width: 100%; background: white; border: 1px solid #e2e8f0; border-radius: 8px; padding: 35px; margin-bottom: 25px;"
|-
| style="text-align: center;" |

<div style="font-size: 3.2em; font-weight: 200; color: #1e293b; margin-bottom: 15px; letter-spacing: -2px;">Understanding VoIP.ms Pricing</div>

<div style="font-size: 1rem; color: #64748b; margin-bottom: 25px;">
'''[https://wiki.voip.ms/article/Co%C3%BBt_des_Services_Rendus 🇨🇦 Français]''' • '''[https://wiki.voip.ms/article/Costo_del_servicio 🇲🇽 Español]'''
</div>

__TOC__

<div style="background: #fef3c7; border: 1px solid #f59e0b; border-radius: 8px; padding: 20px; margin: 30px 0; text-align: left;">
<div style="font-size: 1.1em; font-weight: 600; color: #92400e; margin-bottom: 10px;">💰 PRICING TRANSPARENCY</div>
<div style="color: #92400e;">
'''All rates and prices''' displayed on our website and in the customer portal, as well as any deposits made through our payment processor via credit card or your financial institution for bank wire transfers, are processed in '''US dollars (USD)'''. When using digital wallets like PayPal or mobile wallets such as Apple Pay and Google Pay, transactions are also processed in '''US dollars (USD)''' but they may already do a currency conversion and display the amount converted to your local at the time of payment. Please note that your financial institution may also impose additional currency exchange fees and rates for credit card transactions or bank wire transfers.<br/><br/>
1) '''VoIP.ms Complete Pricing:''' https://voip.ms/pricing<br/>
2) '''VoIP.ms Savings Calculator:''' https://voip.ms/pricing#roi_tool_container
</div>
</div>

<div style="text-align: left; margin-top: 50px;">

== VoIP.ms vs. Traditional Phone Company: Key Differences ==

VoIP.ms is a Voice over Internet Protocol (VoIP) service that provides voice termination ('''outgoing calls''') and origination ('''incoming calls''') globally. It operates on a fundamentally different model than traditional bundled phone plans.

<div style="margin: 30px 0;">

=== Service Model Comparison ===

</div>

<div style="margin-bottom: 25px;">

'''VoIP.ms Approach:'''

</div>

* Pay-per-use billing - charges based on actual usage rather than fixed monthly fees

* No contractual commitments - service can be discontinued without early termination fees

* Itemized billing showing per-minute rates for different call types and destinations

* On-demand feature activation and phone number (DID) management

* International calling at published per-minute rates

* Various features included in the base service

<div style="margin: 25px 0;">

'''Traditional Bundle Plans:'''

</div>

* Fixed monthly fees typically including set minute allowances (published or not)

* Often require 1-2 year service agreements

* Package pricing that may include unused services

* Standardized feature sets with potential upgrade fees

* May have restrictions on international calling or require separate international plans

<div style="margin: 30px 0;">

'''Technical Requirements:''' VoIP.ms requires compatible hardware (such as analog telephone adapters) since it doesn't provide physical phone equipment. It does, however, provide a softphone. More information: [[VoIP.ms_Softphone|VoIP.ms Softphone]]

</div>

<div style="margin: 30px 0;">

'''Account Structure:''' The service operates on a prepaid model requiring a minimum $15 account deposit. All usage charges deduct from this balance, and service becomes unavailable when the balance reaches zero. Account registration is free, allowing users to explore features and test functionality before funding the account.

</div>

</div>

<div style="text-align: left; margin-top: 50px;">

== VoIP.ms Pricing Overview ==

</div>

<center>
{| class="wikitable" style="border-collapse: collapse; margin: 20px auto; width: 80%;"
! style="background: #d0382d; color: white; padding: 12px; width: 30%;" |
! style="background: #d0382d; color: white; padding: 12px; width: 35%;" | '''United States (Contiguous 48 states)'''
! style="background: #d0382d; color: white; padding: 12px; width: 35%;" | '''Canada'''
|-
| style="padding: 12px; font-weight: 500;" | '''Local Phone Number'''
| style="padding: 12px; text-align: center;" | $1.10/month
| style="padding: 12px; text-align: center;" | $1.10/month
|-
| style="padding: 12px; font-weight: 500;" | '''Make calls'''
| style="padding: 12px; text-align: center;" | $0.01/min
| style="padding: 12px; text-align: center;" | $0.0052/min (one fifth of a cent)
|-
| style="padding: 12px; font-weight: 500;" | '''Receive calls'''
| style="padding: 12px; text-align: center;" | $0.009/min (one ninth of a cent)
| style="padding: 12px; text-align: center;" | $0.009/min (one ninth of a cent)
|-
| style="padding: 12px; font-weight: 500;" | '''Emergency Services'''
| style="padding: 12px; text-align: center;" | $1.50/month
| style="padding: 12px; text-align: center;" | $1.50/month
|-
| style="padding: 12px; text-align: center; background: #f8f9fa; font-size: 0.9em;" colspan="3" | '''For the most updated pricing and complete rate information, please visit:''' https://voip.ms/pricing
|}
</center>

<center>
<div style="background: #fee; border: 2px solid #d0382d; border-radius: 8px; padding: 25px; margin: 30px auto; text-align: center; width: 80%;">
<div style="font-size: 1.3em; font-weight: 600; color: #d0382d; margin-bottom: 15px;">🏠 Home users?</div>
<div style="color: #d0382d; margin-top: 15px;">
Get our '''Unlimited* inbound calling plan''' for just '''$4.95/month'''
</div>
</div>
</center>

<div style="text-align: left;">

'''Additional Services:'''

* '''International Calling:''' Rates vary by destination - view complete pricing at https://voip.ms/pricing

* '''USA Toll-Free Numbers:''' $0.99/month + $0.019/min inbound (custom numbers +$15 setup)

* '''Canadian Toll-Free Numbers:''' $1.25/month + $0.027/min inbound (custom numbers +$30 setup)

* '''CNAM Caller ID Lookup:''' $0.008 per query

* '''International DID Numbers:''' Pricing varies by country - view complete pricing at https://voip.ms/pricing

* '''Messaging/Texting (US/CAN):''' $0.0075/SMS and $0.02/SMS

* '''Call Recording:''' $0.0025/min

* '''Transcripts (Voicemail, Call, etc.):''' $0.05/min

* '''Microsoft Teams:''' Starting at $4.50/voice license monthly and $2.00/messaging license monthly

* '''Advanced Features:''' Voicemail, Call Encryption, Call Forwarding, Digital Receptionist (IVR) - '''All Included'''

</div>

<div style="text-align: left; margin: 50px 0;">

== Frequently Asked Questions ==

</div>

<center>
{| class="wikitable" style="border-collapse: collapse; margin: 20px auto; width: 95%;"
! style="background: #4a5568; color: white; padding: 12px; width: 20%;" | '''Category'''
! style="background: #4a5568; color: white; padding: 12px; width: 35%;" | '''Question'''
! style="background: #4a5568; color: white; padding: 12px; width: 45%;" | '''Answer'''
|-
| rowspan="3" style="padding: 12px; font-weight: 500; vertical-align: top; text-align: left;" | Getting Started
| style="padding: 12px; font-weight: 500; vertical-align: top; text-align: left;" | How much does it cost to get started with VoIP.ms?
| style="padding: 12px; vertical-align: top; text-align: left;" | Account registration is completely free. However, you need to deposit a minimum of $15 to activate calling services. This prepaid balance doesn't expire and is used to pay for your actual usage.
|-
| style="padding: 12px; background: #fafbfc; font-weight: 500; vertical-align: top; text-align: left;" | Are there any contracts or monthly commitments?
| style="padding: 12px; background: #fafbfc; vertical-align: top; text-align: left;" | No contracts required. VoIP.ms operates on a pay-as-you-go model, and you can cancel anytime without penalties or early termination fees.
|-
| style="padding: 12px; font-weight: 500; vertical-align: top; text-align: left;" | What payment methods do you accept?
| style="padding: 12px; vertical-align: top; text-align: left;" | We accept PayPal, Visa, MasterCard, American Express, Google Pay, Apple Pay, and wire transfers. All transactions are processed in $USD.
|-
| rowspan="3" style="padding: 12px; background: #fafbfc; font-weight: 500; vertical-align: top; text-align: left;" | Call Rates
| style="padding: 12px; background: #fafbfc; font-weight: 500; vertical-align: top; text-align: left;" | How are calls billed?
| style="padding: 12px; background: #fafbfc; vertical-align: top; text-align: left;" | Calls are billed in 6-second increments for all destinations except some international destinations that are billed at 60-second increments. For example, a 27-second call to Montreal, Quebec is charged as 30 seconds.
|-
| style="padding: 12px; font-weight: 500; vertical-align: top; text-align: left;" | Are there separate charges for local vs. long-distance calls?
| style="padding: 12px; vertical-align: top; text-align: left;" | No, there are no "local" or "long-distance" distinctions with VoIP. All calls to United States for instance are $0.01/minute regardless of where you're calling from or to.
|-
| style="padding: 12px; background: #fafbfc; font-weight: 500; vertical-align: top; text-align: left;" | What's the difference between Value and Premium routes?
| style="padding: 12px; background: #fafbfc; vertical-align: top; text-align: left;" | For USA calls, only Premium routes are available at $0.01/minute. For Canada and international destinations, Value routes offer lower rates while Premium routes provide consistently higher quality using Tier-1 carriers.
|-
| rowspan="2" style="padding: 12px; font-weight: 500; vertical-align: top; text-align: left;" | Additional Services
| style="padding: 12px; font-weight: 500; vertical-align: top; text-align: left;" | What about fax services?
| style="padding: 12px; vertical-align: top; text-align: left;" | Virtual Fax requires a dedicated fax DID at $1.99/month + $0.029/minute for usage. Regular voice DIDs cannot be used for fax.
|-
| style="padding: 12px; background: #fafbfc; font-weight: 500; vertical-align: top; text-align: left;" | How much do advanced features cost?
| style="padding: 12px; background: #fafbfc; vertical-align: top; text-align: left;" | Most advanced features are included free: voicemail, call forwarding, digital receptionist, call queues, and over 70 telephony features.
|-
| rowspan="4" style="padding: 12px; font-weight: 500; vertical-align: top; text-align: left;" | Billing & Account Management
| style="padding: 12px; font-weight: 500; vertical-align: top; text-align: left;" | When am I charged for services?
| style="padding: 12px; vertical-align: top; text-align: left;" | All charges are deducted from your prepaid account balance in real-time. Monthly fees for DIDs are charged on your anniversary date.
|-
| style="padding: 12px; background: #fafbfc; font-weight: 500; vertical-align: top; text-align: left;" | What happens if my balance reaches zero?
| style="padding: 12px; background: #fafbfc; vertical-align: top; text-align: left;" | Your account cannot make or receive calls once the balance reaches zero. You'll need to add funds to restore service.
|-
| style="padding: 12px; font-weight: 500; vertical-align: top; text-align: left;" | Can I get a refund if I don't like the service?
| style="padding: 12px; vertical-align: top; text-align: left;" | Yes, you can request a refund for your remaining balance within 90 days of your deposit if you decide to cancel your account.
|-
| style="padding: 12px; background: #fafbfc; font-weight: 500; vertical-align: top; text-align: left;" | How do I track my usage and costs?
| style="padding: 12px; background: #fafbfc; vertical-align: top; text-align: left;" | Your customer portal provides detailed call logs, transaction history, and usage reports. You can also generate invoices for any date range.
|-
| rowspan="2" style="padding: 12px; font-weight: 500; vertical-align: top; text-align: left;" | Business & Integration
| style="padding: 12px; font-weight: 500; vertical-align: top; text-align: left;" | Are there volume discounts for businesses?
| style="padding: 12px; vertical-align: top; text-align: left;" | For high-volume usage or reseller inquiries, contact [mailto:sales@voip.ms sales@voip.ms] for custom pricing discussions.
|-
| style="padding: 12px; background: #fafbfc; font-weight: 500; vertical-align: top; text-align: left;" | How much does it cost to port my existing number?
| style="padding: 12px; background: #fafbfc; vertical-align: top; text-align: left;" | Number porting is free for USA and Canada numbers. The process typically takes 5 business days for local and toll-free numbers.
|-
| rowspan="2" style="padding: 12px; font-weight: 500; vertical-align: top; text-align: left;" | International Services
| style="padding: 12px; font-weight: 500; vertical-align: top; text-align: left;" | How much do international calls cost?
| style="padding: 12px; vertical-align: top; text-align: left;" | International rates vary widely by destination, typically ranging from $0.005-$0.50+ per minute. Check the complete rate table at https://voip.ms/pricing
|-
| style="padding: 12px; background: #fafbfc; font-weight: 500; vertical-align: top; text-align: left;" | Can I get international phone numbers?
| style="padding: 12px; background: #fafbfc; vertical-align: top; text-align: left;" | Yes, international DIDs are available in 100+ countries with varying pricing. See https://voip.ms/pricing for country pricing and availability.
|}
</center>

|}

[[category:guides]]

Servicio E911

2025-11-25T23:25:42Z

RP: /* Llamada de emergencia (e911) sin verificacion */

{| class="wikitable"
|+
|-
! Article in English !! Article en Français
|-
| [https://wiki.voip.ms/article/E911 English] ||
[https://wiki.voip.ms/article/Service_E911 Français]
|}

__TOC__

== Articulo de Blog ==
[https://wiki.voip.ms/article/Enhanced_911_and_VoIP Enhanced 911 and VoIP]

== E911 ==

El sistema básico del 911 funciona identificando la ubicación de la persona que llama por el número de teléfono fijo. La llamada se enruta automáticamente al PSAP (punto de respuesta de seguridad pública más cercano) y el despachador en ese punto se comunica con el personal de servicios de emergencia más cercano para atender la llamada.

Sin embargo, con los teléfonos inalámbricos y voip, el sistema 911 original no podía identificar la ubicación de una persona que llamaba y, si la persona que llamaba no podía identificar o describir una ubicación, el personal de emergencia se veía obstaculizado para brindar asistencia.

E911 es la solución para esto. Cuando se realiza una llamada de emergencia (e911) a través de la red VoIP.MS En los EE. UU., la dirección física que ingresará en el momento del registro e911 para un DID específico se transmitirá a su PSAP local, proporcionando al despachador en el PSAP la ubicación exacta donde se requiere ayuda. Tenga en cuenta que "Puerto Rico" ahora se puede aprovisionar para e911, sin embargo, no funciona de la misma manera que el resto de los EE. UU. La información de la dirección no se transmitirá ni se mostrará en el PSAP, por lo que es necesario que la persona que llama proporcione la dirección verbalmente al llamar.

El servicio 911 canadiense se maneja de manera un poco diferente. Todas las llamadas canadienses al centro de llamadas de National 911 son respondidas por un despachador que accederá a la base de datos del proveedor del servicio 911 para obtener la ubicación y también preguntará verbalmente a la persona que llama por su ubicación antes de transferir la llamada al PSAP de la persona que llama dando verbalmente el PSAP la ubicación y luego conectar a la persona que llama. Esta es la forma en que se tratan todas las llamadas VOIP al 911 en Canadá

Para activar este servicio, puede encontrar esta opción en su Portal>>DID Numbers >> E911.

'''Nota Importante:'''

<span style="color:red">'''Tenga en cuenta que si alguno de sus números DID con voip.ms no está suscrito al servicio E911, intentar hacer una llamada'''
'''al 911 resultará en "ocupado" (Busy signal) debido a que su CallerID (el número que manda, que debe ser un número válido)'''
'''no estará en la base de datos de voip.ms; el resultado será una llamada fallida.'''
'''Igualmente asegúrese de leer los "Términos y condiciones del servicio"(Terms of Service) al final de la página. En caso de tener
'''alguna duda o pregunta con respecto a este tema en inglés, por favor contacte a un miembro de nuestro staff en español
'''y con mucho gusto le atenderemos.'''</span>

''' <span style="color:red"> Actualización a octubre de 2021 </span> '''

Debido a las nuevas regulaciones para los servicios E911, los softphones en dispositivos móviles comenzarán a usar el marcador móvil nativo para realizar llamadas a los servicios de emergencia. '''Es posible que no todos los dispositivos de softphone proporcionen esta funcionalidad y se recomienda encarecidamente entablar una estrecha comunicación con el proveedor de softphone si tiene más preguntas.'''

[[File:E9112.JPG|thumb|none|600px]]

El uso del servicio e911 tiene un costo de recuperación de $1.50 USD por activación y una cuota menusal de $1.50 USD por cada número que tenga activo este servicio. VoIP.MS no obtiene ganancias por este servicio, simplemente cobra lo que se necesita para poder proveerlo.

Este servicio esta disponible solamente para números de Canada y US, incluyendo números Toll Free. Solamente tiene que dar click en el botón Apply para comenzar el proceso y habiltar este servicio
para su número.

[[File:E911_new.png|thumb|none|600px]]

Después usted puede leer los Términos y Condiciones para el uso de este servicio. Por favor lealo con detenimiento y después haga click en "I agree to the Terms and Conditions" si quiere habilitar el servicio.

Haga click aquí para leer los Términos y Condiciones de este servicio. [[E911#VoIP.ms_911.2Fe911_Emergency_Service:_Terms_of_Service|E911 TOS]]

Una nueva ventana aparecerá pidiendo información que será usada con este servicio. Por favor introduzca la información requerida y porfavor verifique la dirección sea correcta, ya que esta será usada para atender la emergencia cuando sea solicitada. Después haga click en el botón de validación para confirmar que todo este correcto y al final usted recibirá un correo electrónico de confirmación.

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

Una vez que su servicio e911 esté activo, debe configurar su número DID como su número de identificación de llamadas salientes para su cuenta principal o subcuentas que usará para marcar.

Para configurar su [[Caller ID]] a su main account, puede hacerlo en su Portal >> Main Menu >> [[Configuraciones de la Cuenta (Account Settings)|Account Settings]] >> General tab >> "[[Caller ID]] number" option.

Si va a usar una subcuenta para hacer la llamada, puede asignar el [[Caller ID]] en su Portal >> [[Sub Cuentas (Sub Accounts)|Sub accounts]] >> Manage [[Sub Cuentas (Sub Accounts)|Sub accounts]] >> Edit >> [[Caller ID]] number.

Para comprobar que su identificador de llamadas funciona correctamente, puede marcar 1-555-555-0911 desde la red VoIP.ms. El sistema reproducirá su identificador de llamadas, luego hará una breve pausa y reproducirá el resultado de la prueba. Para asegurarse de que su dispositivo / interruptor esté configurado correctamente para e911, debe asegurarse de que su número de identificación de llamadas coincida exactamente con el número DID que está activado para el 911. Esa es la única forma de identificarlo correctamente. De lo contrario, su llamada al 911 no se realizará. El identificador de llamadas con el formato correcto consta de un número de 10 dígitos, idéntico a su DID. Por ejemplo, si su DID es 555.555.1234, su número de identificación de llamadas debe ser 5555551234.

Si en algún momento necesita cambiar la dirección física con la que dió de alta el servicio, puede seleccionar la opción "Modify" y dar click en el botón "Apply". Con esto usted podrá cambiar la información relacionada a este servicio y después de que ésta haya sido aprobada, recibirá una confirmación por correo electrónico de que su información ha sido actualizada.

[[File:E911Enable.png|thumb|none|600px]]

== Llamada de emergencia (e911) sin verificacion==

VoIP.ms ahora conectará las llamadas al 911 desde cualquier número dentro de Estados Unidos o Canadá, incluso si el servicio E911 no está configurado, para garantizar su seguridad en situaciones de emergencia.

Se aplicará un cargo de $75 dólares por cada tentativa de llamada al 911 si el servicio E911 no está activo en su número. Le recomendamos que active el servicio E911 para evitar este cargo y garantizar que su ubicación se comparta con precisión con los servicios de emergencia.

== VoIP.ms 911/e911 Servicio de Emergencia: Términos del servicio ==

La Compañía quiere asegurarse de que los Clientes sean conscientes de las importantes diferencias en el funcionamiento de los Servicios de Emergencia al utilizar servicios VoIP en comparación con el servicio telefónico tradicional. A continuación, se describe lo que los Clientes deben tener en cuenta.  

'''Enrutamiento de llamadas de emergencia'''
Para los residentes de los Estados Unidos, cuando un Cliente realiza una llamada de emergencia, la Compañía intentará enrutarlas automáticamente a través de un proveedor de servicios de terceros al Centro de Respuesta de Seguridad Pública ("PSAP") correspondiente a la dirección registrada del Cliente en su cuenta. Sin embargo, no se garantiza la entrega de la ubicación física del Cliente a su PSAP local. Es posible que la ubicación del Cliente no se proporcione al despachador del PSAP. En tales ocasiones, será responsabilidad exclusiva del Cliente proporcionar al despachador su nombre, ubicación (o la ubicación de la emergencia) e información de contacto para recibir asistencia de los servicios de emergencia. Para los residentes de Canadá, la llamada de emergencia del Cliente se enviará directamente a un centro de llamadas de emergencia para confirmar su identidad y ubicación, y luego se transferirá de inmediato al PSAP local.

'''Limitaciones debido a las redes VoIP'''
Debido a las diversas dependencias de las redes VoIP, la Compañía no puede garantizar que una llamada de emergencia del Cliente se realice. Muchas condiciones, como la pérdida de energía, acceso a Internet o conectividad, y otras condiciones, pueden hacer que los servicios de emergencia sean inoperables. La Compañía no tiene control sobre este tipo de situaciones y, por lo tanto, no puede ser considerada responsable de tal inoperabilidad. La Compañía tomará medidas comercialmente razonables para prevenir interrupciones del servicio dentro de su red.  

'''CallerID de salida'''
Para que la información de la dirección de emergencia sea transmitida al despachador del PSAP local del Cliente, el valor del CallerID de salida del Cliente debe estar configurado en el DID específico para el cual se adquieren los servicios de emergencia. Al aceptar estos Términos, se considera que el Cliente ha configurado el número de CallerID de salida al DID que ha habilitado para los servicios de emergencia al realizar una llamada de emergencia de salida. La Compañía ha agregado una extensión en su red donde todos los Clientes pueden llamar para probar el valor de su CallerID. En cualquier momento, un Cliente puede probar su valor de CallerID de salida marcando '1-555-555-0911' a través de la red de la Compañía.  

'''Limitaciones en los servicios de emergencia'''
LOS CLIENTES ENTIENDEN LAS LIMITACIONES DE LOS SERVICIOS DE EMERGENCIA DE LA COMPAÑÍA Y ASUMEN TODA LA RESPONSABILIDAD Y LIBERAN A LA COMPAÑÍA DE CUALQUIER RESPONSABILIDAD POR EL USO DE LOS SERVICIOS DE EMERGENCIA, Y ADEMÁS ACEPTAN MANTENER INDEMNES A LA COMPAÑÍA, SUS OFICIALES, DIRECTORES, EMPLEADOS Y AGENTES POR CUALQUIER DAÑO, YA SEA DIRECTO O INDIRECTO, QUE PUEDA RESULTAR DE: (1) LOS SERVICIOS DE EMERGENCIA PROPORCIONADOS POR LA COMPAÑÍA (INCLUYENDO, PERO NO LIMITADO A SITUACIONES DE INDISPONIBILIDAD DE LOS SERVICIOS DE EMERGENCIA COMO SE DESCRIBE EN ESTOS TÉRMINOS Y LA INFORMACIÓN INCOMPLETA O INCORRECTA PROPORCIONADA POR EL CLIENTE); (2) EL INCUMPLIMIENTO DEL CLIENTE DE OBTENER ACCESO A LOS SERVICIOS DE EMERGENCIA CONVENCIONALES COMO PARTE DE UNA SUSCRIPCIÓN DE LÍNEA TELEFÓNICA DE OTRA COMPAÑÍA BAJO UN ACUERDO SEPARADO; (3) EL INCUMPLIMIENTO O RETRASO DEL CLIENTE EN UTILIZAR LOS SERVICIOS DE EMERGENCIA CONVENCIONALES. LOS CLIENTES QUE REVENDE LOS SERVICIOS ADEMÁS ACEPTAN SER RESPONSABLES DE NOTIFICAR, Y SE COMPROMETEN A NOTIFICAR, A SUS CLIENTES, CONTRATISTAS, AGENTES, EMPLEADOS, ASOCIADOS, ACCIONISTAS, SOCIOS Y CUALQUIER OTRO USUARIO POTENCIAL DE LOS SERVICIOS DE LA COMPAÑÍA SOBRE LA NATURALEZA Y LIMITACIONES DE LOS SERVICIOS DE EMERGENCIA. SI UN CLIENTE NO SE SIENTE CÓMODO CON LAS LIMITACIONES DE LAS LLAMADAS DE EMERGENCIA, DEBE CONSIDERAR UN MEDIO ALTERNATIVO PARA ACCEDER A LOS SERVICIOS DE EMERGENCIA TRADICIONALES, YA QUE EL REGISTRO EN LOS SERVICIOS DE EMERGENCIA ES OBLIGATORIO EN LA MAYORÍA DE LOS PAÍSES. ADEMÁS, EL CLIENTE ACEPTA QUE LA COMPAÑÍA NO TIENE RESPONSABILIDAD EN RELACIÓN CON LA CALIDAD DEL CONSEJO Y LOS SERVICIOS PROPORCIONADOS POR UN PSAP.

'''Precios'''
Se cobrará al Cliente una tarifa de configuración y una tarifa de recuperación regulatoria por mes por cada DID enviado a la base de datos de servicios de emergencia; consulte los precios. Esta tarifa no es reembolsable.

'''Regulación: Sección aplicable a los clientes que residen en los Estados Unidos o Canadá'''
Debido a recientes disposiciones y regulaciones de la FCC/CRTC, todos los Clientes que utilicen los servicios de la Compañía como su principal proveedor telefónico residencial o comercial deben activar los Servicios de Emergencia 911 en al menos uno de sus DIDs. De acuerdo con las regulaciones, los Clientes también deben asegurarse de que sus sistemas de comunicación permitan a todos los usuarios marcar siempre el 911, sin la necesidad de marcar un prefijo, y que, si se marca el 911 desde el sistema de comunicación del Cliente, el personal y las autoridades pertinentes sean informados de inmediato de la emergencia. La Compañía se esfuerza por proporcionar al Cliente los servicios necesarios para cumplir con las reglas y regulaciones, de acuerdo con estos Términos. No obstante, el Cliente acepta cumplir con dichas regulaciones y entiende que la Compañía no asumirá ninguna responsabilidad en caso de incumplimiento de dichas regulaciones por parte del Cliente.  

'''Regulación: Sección aplicable a los clientes que residen fuera de los Estados Unidos o Canadá'''
Los reguladores de telecomunicaciones fuera de Canadá y los Estados Unidos generalmente disponen que todos los Clientes que utilicen servicios VoIP como su principal proveedor telefónico residencial o comercial deben activar los Servicios de Emergencia en al menos uno de sus DIDs. Esta obligación se aplica a cualquier Cliente que resida en los siguientes países: Austria, Australia, Bélgica, Bulgaria, Croacia, República Checa, Chipre, Dinamarca, Estonia, Finlandia, Francia, Alemania, Grecia, Hong Kong, Hungría, Irlanda, Israel, Italia, Japón, Letonia, Lituania, Países Bajos, Nueva Zelanda, Noruega, Polonia, Portugal, Rumania, Singapur, Eslovaquia, Eslovenia, Corea del Sur, España, Suecia, Turquía y Reino Unido. Si un Cliente utiliza los Servicios para finalizar llamadas con un CallerID que no está registrado en su cuenta, el Cliente reconoce y acepta que la Compañía no podrá finalizar las llamadas de Servicios de Emergencia debido a la falta de la información necesaria y declina toda responsabilidad con respecto a este asunto. En este caso, el Cliente debe comunicarse con el operador con el que está registrado su número telefónico para activar los Servicios de Emergencia con ellos, ya que será su responsabilidad finalizar sus llamadas de Servicios de Emergencia.  

'''Uso de los Servicios de Emergencia fuera del país de residencia del Cliente'''
Los reguladores generalmente exigen que se registre una dirección local cuando se activan los Servicios de Emergencia para un número telefónico. Si un Cliente utiliza los Servicios para la terminación de llamadas locales en un país en el que no tiene una dirección de residencia, el Cliente puede no ser capaz de registrarse y activar los Servicios de Emergencia en dicho país y puede experimentar dificultades para comunicarse con los Servicios de Emergencia en su país de residencia. La Compañía declina toda responsabilidad con respecto a esta limitación de servicio, de acuerdo con esta sección de los Términos.  

==Precios de los servicios de emergencia==

{| class="wikitable" style="vertical-align:middle;"
|- style="font-weight:bold; text-align:center;"
! País
! Tarifa única
! Tarifa recurrente
|-
| Alemania
| 1,50 $
| 2,50 $
|-
| Australia
| 1,50 $
| 2,50 $
|-
| Austria
| 1,50 $
| 2,50 $
|-
| Bélgica
| 1,50 $
| 2,50 $
|-
| Bulgaria
| 1,50 $
| 2,50 $
|-
| Canadá
| 1,50 $
| 1,50 $
|-
| Chipre
| 1,50 $
| 2,50 $
|-
| Croacia
| 1,50 $
| 2,50 $
|-
| Dinamarca
| 1,50 $
| 2,50 $
|-
| Eslovaquia
| 1,50 $
| 2,50 $
|-
| Eslovenia
| 1,50 $
| 2,50 $
|-
| España
| 1,50 $
| 2,50 $
|-
| Estados Unidos
| 1,50 $
| 1,50 $
|-
| Estonia
| 1,50 $
| 2,50 $
|-
| Finlandia
| 1,50 $
| 2,50 $
|-
| Francia
| 1,50 $
| 2,50 $
|-
| Grecia
| 1,50 $
| 2,50 $
|-
| Hungría
| 1,50 $
| 2,50 $
|-
| Irlanda
| 1,50 $
| 2,50 $
|-
| Italia
| 1,50 $
| 2,50 $
|-
| Letonia
| 1,50 $
| 2,50 $
|-
| Lituania
| 1,50 $
| 2,50 $
|-
| Noruega
| 1,50 $
| 2,50 $
|-
| Nueva Zelanda
| 1,50 $
| 2,50 $
|-
| Países Bajos
| 1,50 $
| 2,50 $
|-
| Polonia
| 1,50 $
| 2,50 $
|-
| Portugal
| 1,50 $
| 2,50 $
|-
| Reino Unido
| 1,50 $
| 2,50 $
|-
| República Checa
| 4,50 $
| 2,50 $
|-
| Rumania
| 1,50 $
| 2,50 $
|-
| Suecia
| 1,50 $
| 2,50 $
|}

=Usando el 911 en la interfaz de revendedor =

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

Tenga en cuenta que el DID debe estar vinculado a su cliente. (Reseller> Manage client's accounts> Haga clic en Manage client donde está su cliente).

Vaya debajo de la barra de navegación en '''[Reseller]''' y luego haga clic en '''[Manage Rates and Packages]'''
: [[File:e911_Reseller_1.png|thumb|none|300px]]

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

: [[File:e911_Reseller_2.png|thumb|none|700px]]

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

: [[File:e911_Reseller_3.png|thumb|none|700px]]

Luego, desplácese hacia abajo y busque la función '''"e911"''' y habilítela.

: [[File:e911_Reseller_4.png|thumb|none|500px]]

1) Para agregar una entrada e911 para su cliente, o para ayudar a su cliente a agregar una. Vaya debajo de '''[Services]''' en la barra de navegación izquierda, luego en '''[e911]'''
: [[File:e911_Add.png|thumb|none|400px]]

2) En la fila donde se encuentra el DID deseado, haga clic en el botón''' [ENABLED]'''.
: [[File:e911_Add2.png|thumb|none|600px]]

3) Su cliente '''debe leer''' y '''aceptar los Términos y Condiciones'''.

4) Complete el formulario y haga clic en el botón '''[Validate]'''.

[[category:Guías]]

Service E911

2025-11-25T23:25:15Z

RP: /* Appel d'urgence (e911) non-vérifié */

{| class="wikitable"
|+
|-
! Article in English !! Artículo en Español
|-
| [https://wiki.voip.ms/article/E911 English] || [https://wiki.voip.ms/article/Servicio_E911 Español]
|}

== Renseignements E911 ==
Le service 911 de base, fonctionne en localisant le lieu d'appel de l'appelant, par le numéro de téléphone de la ligne de terre. L'appel est automatiquement acheminé vers le point de sécurité publique le plus prêt, d'où le répartiteur de ce point, contacte le personnel du service d'urgence le plus prêt, pour ainsi faire face à l'appel.

Cependant, avec la venue des téléphones sans fil et téléphones IP, le système 911 d'origine est devenu incapable de localiser les appelants, et dans le cas où l'appelant n'est pas en mesure d'identifier ou de décrire le lieu où il se trouve, ceci représente une entrave pour le personnel d'urgence quant à l'assistance rendue. Les adresses de Puerto Rico peuvent maintenant être provisionnées. Par contre elles fonctionnent différemment du reste des États-Unis, mais similairement au Canada.

E911 est la solution à ce problème. Lorsqu'un appel d'urgence (E911) est placé à travers le réseau VoIP.ms, l'adresse physique que vous avez fournie au moment d'enregistrer le service pour un DID spécifique, sera transmise à votre PSAP local, fournissant ainsi au répartiteur l'emplacement exact où l'aide est requise.

== Activation du service ==

Vous pouvez activer ce service par votre portail client, Numéros DID > [https://www.voip.ms/m/me911.php E911].

<span style="color:red">
'''Note Important:'''<br/>
Notez que si votre DID avec VoIP.ms n'est pas enregistré au service, vous ne serez pas en mesure de composer 911.
Cela tombera dans une tonalité occupée, étant donné que le [[ID de l'appelant | numéro d'identification de l'appelant]] ne sera pas dans la base de données.
Cela n'atteindra pas la destination désirée. Assurez-vous aussi de lire les Termes d'utilisation à la fin de la page.
</span>

''' <span style="color:red"> Mise à jour octobre 2021 </span> '''

En raison de la nouvelle réglementation pour les services E911, les softphones sur les appareils mobiles commenceront à utiliser le numéroteur mobile natif pour passer des appels aux services d'urgence. '''Il est possible que tous les appareils de softphone ne fournissent pas cette fonctionnalité et il est fortement suggéré d'entrer en communication étroite avec le fournisseur de softphone pour d'autres questions.'''

[[File:911Menu.png|thumb|none|600px]]

L'utilisation du service engendre un frais d'activation de $1,50 ainsi qu'un frais régulier mensuel de $1,50 par numéro DID activé. VoIP.ms ne fait pas un sous sur l'utilisation de ce service. Ces coûts sont simplement ce qui doit être payé pour fournir ce service.

Veuillez noter que vous pouvez activer ce service uniquement pour les numéros canadiens ou US (y compris les numéros sans frais (Toll-Free) des États-Unis ou du Canada). Vous n'avez qu'à cliquer sur le bouton "Activer" afin de débuter le processus d'activation de ce service pour un numéro.

[[File:911nouveau.png|thumb|none|600px]]

Ensuite, vous pouvez vérifier les termes et conditions relatifs à ce service. Assurez-vous de les lire attentivement et cliquez sur "Je suis d'accord avec les conditions" si vous souhaitez activer le service.

Cliquez ici pour lire les Termes & conditions relatives à ce service. [[#Conditions d'utilisation|E911 TOS]]

Une nouvelle fenêtre demandera quelques informations nécessaires qui seront utilisées pour le service. Il suffit de remplir les champs requis et s'assurer d'entrer la bonne adresse. Enfin, cliquez sur "Valider" et si tout est conforme, vous recevrez alors un courriel de confirmation.

[[File:911Remplir.png|thumb|none|600px]]

== Appel d'urgence (e911) non-vérifié ==

VoIP.ms acheminera désormais les appels au 911 à partir de n’importe quel numéro aux États-Unis ou au Canada, même si le service E911 n’est pas configuré, afin d’assurer votre sécurité en cas d’urgence.

Des frais de 75 $ s’appliqueront pour chaque tentative d'appel au 911, si le service E911 n’est pas activé sur votre numéro. Nous vous recommandons d’activer le service E911 pour éviter ces frais et pour que votre emplacement soit transmis avec précision aux services d’urgence.

== Utilisation du service ==

Une fois votre service activé, vous devrez définir votre numéro DID comme votre numéro d'identification de l'appelant (Caller ID) pour appels sortants, pour le compte principal ou sous-compte que vous utiliserez pour appeler.

Vous pouvez définir le numéro d'identification de l'appelant (Caller ID) de votre compte principal, dans votre portail client, Menu principal > Paramètres du compte > Généralités > Numéro d'identification de l'appelant.

Si vous prévoyez utiliser un compte secondaire pour composer, vous pouvez définir le numéro d'identification de l'appelant, dans votre portail client, Sous-comptes > [https://www.voip.ms/m/managesubaccount.php Gestion des sous-comptes] > Modifier > '''Numéro d'identification de l'appelant'''.

Pour tester que votre [[ID de l'appelant | numéro d'identification de l'appelant]] fonctionne correctement, vous pouvez composer 1-555-555-0911 à partir du réseau VoIP.ms. Le système vous répétera votre [[ID de l'appelant | numéro d'identification de l'appelant]], puis fera une courte pause, et jouera le résultat du test. Pour vous assurer que votre appareil / commutateur est correctement configuré pour le E911, vous devez vous assurer que votre [[ID de l'appelant | numéro d'identification de l'appelant]] correspond exactement au numéro DID qui est activé pour le service. C'est la seule façon de vous identifier correctement. Sinon, votre appel au E911 ne passera pas.

Le bon format du [[ID de l'appelant | numéro d'identification de l'appelant]] est composé de 10 chiffres, identiques à votre DID. Par exemple, si votre DID est 555.555.1234, votre [[ID de l'appelant | numéro d'identification de l'appelant]] devrait être 5555551234.

Si jamais vous deviez changer votre adresse, il vous suffirait de sélectionner l'option "Modifier", ensuite, vous seriez en mesure de modifier vos informations E911 et une fois l'information approuvée, vous recevriez une confirmation par courriel et le service serait mis à jour.

[[File:911Active.png|thumb|none|600px]]

== Conditions d'utilisation ==

La Société souhaite s'assurer que les clients sont conscients des différences importantes dans la manière dont les services d'urgence fonctionnent lorsqu'ils utilisent des services VoIP par rapport au service téléphonique traditionnel. Veuillez trouver ci-dessous ce que les clients doivent garder à l'esprit. Acheminement des appels d'urgence.  

'''Acheminement des appels d'urgence'''
Pour les résidents des États-Unis, lorsqu'un client effectue un appel d'urgence, la Société tentera de router automatiquement son appel via un fournisseur de services tiers vers le Point de Réponse de Sécurité Publique («PSAP») correspondant à l'adresse enregistrée du client sur son compte. Toutefois, la transmission de la localisation physique du client à son PSAP local n'est pas garantie. Il est possible que la localisation du client ne soit pas fournie au dispatcher du PSAP. Dans de tels cas, il sera de la seule responsabilité du client de fournir au dispatcher son nom, sa localisation (ou celle de l'urgence) et ses informations de contact pour recevoir l'assistance du service d'urgence. Pour les résidents canadiens, un appel d'urgence d'un client sera directement envoyé à un centre d'appels d'urgence confirmant son identité et sa localisation, puis immédiatement transféré au PSAP local.

'''Limitations dues aux réseaux VoIP'''
En raison des diverses dépendances des réseaux VoIP, la Société ne peut pas et ne garantit pas que l'appel d'urgence d'un client sera acheminé. De nombreuses conditions telles que la perte de courant, l'accès à Internet ou la connectivité, et/ou plusieurs autres conditions peuvent rendre les services d'urgence inopérants. La Société n'a pas de contrôle sur ces types de situations et ne peut donc pas être tenue responsable de cette inopérabilité. La Société prendra des mesures commercialement raisonnables pour prévenir les interruptions de service au sein de son réseau.  

'''Identification de l'appelant sortant (CallerID)'''
Pour que les informations d'adresse des services d'urgence soient transmises au dispatcher PSAP local du client, la valeur de l'ID de l'appelant sortant du client doit être définie sur le DID spécifique pour lequel il achète le service d'urgence. Par conséquent, en acceptant ces Termes, un client est considéré comme ayant défini le numéro de l’identifiant de l’appelant sortant sur le DID pour lequel il a activé les services d'urgence lorsqu'il effectue un appel d'urgence sortant. La Société a ajouté une extension à son réseau où tous les clients peuvent appeler pour tester leur valeur de CallerID. À tout moment, un client peut tester sa valeur de l’identifiant de l’appelant sortant en composant le ‘1-555-555-0911’ via le réseau de la Société.  

'''Limitations des services d'urgence'''
LES CLIENTS COMPRENNENT LES LIMITATIONS DES SERVICES D'URGENCE DE LA SOCIÉTÉ ET ASSUMENT TOUTE RESPONSABILITÉ ET RESPONSABILITÉ ET DÉGAGENT LA SOCIÉTÉ DANS CETTE MESURE POUR L'UTILISATION DES SERVICES D'URGENCE ET CONVIENNENT EN OUTRE DE DÉGAGER DE RESPONSABILITÉ LA SOCIÉTÉ, SES DIRIGEANTS, DIRECTEURS, EMPLOYÉS ET AGENTS POUR TOUT DOMMAGE, DIRECT OU INDIRECT, QUI POURRAIT RÉSULTER : (1) DES SERVICES D'URGENCE FOURNIS PAR LA SOCIÉTÉ (Y COMPRIS MAIS SANS S'Y LIMITER, LES SITUATIONS D'INDISPONIBILITÉ DES SERVICES D'URGENCE COMME DÉCRITES DANS CES TERMES ET LES INFORMATIONS DE LOCALISATION INCOMPLÈTES OU INCORRECTES FOURNIES PAR LE CLIENT) ; (2) L'ÉCHEC DU CLIENT À OBTENIR L'ACCÈS AUX SERVICES D'URGENCE CONVENTIONNELS DANS LE CADRE D'UN ABONNEMENT À UNE LIGNE TÉLÉPHONIQUE D'UNE AUTRE SOCIÉTÉ SOUS UN ACCORD SÉPARÉ ; (3) L'ÉCHEC OU LE RETARD DU CLIENT À UTILISER LES SERVICES D'URGENCE CONVENTIONNELS. LES CLIENTS QUI REVENT LES SERVICES CONVIENNENT QU'ILS SONT RESPONSABLES D'INFORMER, ET ACCEPTENT D'INFORMER, LEURS CLIENTS, CONTRACTANTS, AGENTS, EMPLOYÉS, ASSOCIÉS, ACTIONNAIRES, PARTENAIRES ET TOUT AUTRE UTILISATEUR POTENTIEL DES SERVICES DE LA SOCIÉTÉ SUR LA NATURE ET LES LIMITATIONS DES SERVICES D'URGENCE. SI UN CLIENT N'EST PAS À L'AISE AVEC LES LIMITATIONS DES APPELS D'URGENCE, IL DOIT ENVISAGER UN MOYEN ALTERNATIF POUR ACCÉDER AUX SERVICES D'URGENCE TRADITIONNELS, CAR L'INSCRIPTION AUX SERVICES D'URGENCE EST OBLIGATOIRE DANS LA PLUPART DES PAYS. DE PLUS, LE CLIENT ACCEPTE QUE LA SOCIÉTÉ N'AIT AUCUNE RESPONSABILITÉ EN RELATION AVEC LA QUALITÉ DES CONSEILS ET DES SERVICES FOURNIS PAR UN PSAP.

'''Tarification'''
Le Client sera facturé des frais de mise en place de récupération et des frais réglementaires de récupération par mois pour chaque DID soumis à la base de données des services d'urgence – voir tarification. Ces frais ne sont pas remboursables.

'''Réglementation : Section applicable aux clients résidant aux États-Unis ou au Canada'''
En raison des récentes décisions et réglementations de la FCC/CRTC, tous les Clients qui utilisent les services de la Société comme leur principal opérateur téléphonique résidentiel ou commercial doivent activer les services d'urgence 911 sur au moins l'un de leurs DIDs. Conformément aux réglementations, les Clients doivent également s'assurer que leurs systèmes de communication permettent à tous les utilisateurs de composer le 911 sans avoir besoin de composer un préfixe et que, si le 911 est composé à partir du système de communication du Client, le personnel compétent et les autorités soient rapidement informés de l'urgence. La Société s'efforce de fournir au Client les services nécessaires pour se conformer aux règles et réglementations conformément à ces Termes. Nonobstant ce qui précède, par les présents Termes, le Client accepte de se conformer à ces réglementations et comprend que la Société n'assumera aucune responsabilité en cas de violation de ces réglementations par le Client. 

'''Réglementation : Section applicable aux clients résidant en dehors des États-Unis ou du Canada'''
Les régulateurs de télécommunication hors du Canada et des États-Unis stipulent généralement que tous les Clients utilisant des services VoIP comme leur principal opérateur téléphonique résidentiel ou commercial doivent activer les Services d'urgence sur au moins l'un de leurs DIDs. Cette obligation s'applique à tout Client résidant dans les pays suivants : Autriche, Australie, Belgique, Bulgarie, Croatie, République tchèque, Chypre, Danemark, Estonie, Finlande, France, Allemagne, Grèce, Hong Kong, Hongrie, Irlande, Israël, Italie, Japon, Lettonie, Lituanie, Pays-Bas, Nouvelle-Zélande, Norvège, Pologne, Portugal, Roumanie, Singapour, Slovaquie, Slovénie, Corée du Sud, Espagne, Suède, Turquie et Royaume-Uni. Si un Client utilise les Services pour terminer des appels avec un identifiant de l’appelant qui n'est pas enregistré dans son compte, le Client reconnaît et accepte que la Société ne pourra pas terminer les appels des Services d'urgence en raison du manque des informations nécessaires et décline toute responsabilité à cet égard. Dans ce cas, le Client doit communiquer avec l'opérateur chez qui son numéro de téléphone est enregistré pour activer les Services d'urgence avec eux car il sera de leur responsabilité de terminer vos appels de Services d'urgence.  

'''Utilisation des services d'urgence en dehors du pays de résidence du client'''
Les régulateurs exigent généralement qu'une adresse locale soit enregistrée lors de l'activation des Services d'urgence pour un numéro de téléphone. Si un Client utilise les Services pour terminer des appels locaux dans un pays où il n'a pas d'adresse de résidence, il se peut qu'il ne puisse pas s'inscrire et activer les Services d'urgence dans ledit pays et qu'il rencontre des difficultés à communiquer avec les Services d'urgence dans son pays de résidence. La Société décline toute responsabilité concernant cette limitation de service conformément à cette section des Termes.  

==Tarification des services d'urgence==

{| class="wikitable" style="vertical-align:middle;"
|- style="font-weight:bold; text-align:center;"
! Pays
! Frais uniques
! Frais récurrents
|-
| Allemagne
| 1,50 $
| 2,50 $
|-
| Australie
| 1,50 $
| 2,50 $
|-
| Autriche
| 1,50 $
| 2,50 $
|-
| Belgique
| 1,50 $
| 2,50 $
|-
| Bulgarie
| 1,50 $
| 2,50 $
|-
| Canada
| 1,50 $
| 1,50 $
|-
| Chypre
| 1,50 $
| 2,50 $
|-
| Croatie
| 1,50 $
| 2,50 $
|-
| Danemark
| 1,50 $
| 2,50 $
|-
| Espagne
| 1,50 $
| 2,50 $
|-
| Estonie
| 1,50 $
| 2,50 $
|-
| États-Unis
| 1,50 $
| 1,50 $
|-
| Finlande
| 1,50 $
| 2,50 $
|-
| France
| 1,50 $
| 2,50 $
|-
| Grèce
| 1,50 $
| 2,50 $
|-
| Hongrie
| 1,50 $
| 2,50 $
|-
| Irlande
| 1,50 $
| 2,50 $
|-
| Italie
| 1,50 $
| 2,50 $
|-
| Lettonie
| 1,50 $
| 2,50 $
|-
| Lituanie
| 1,50 $
| 2,50 $
|-
| Norvège
| 1,50 $
| 2,50 $
|-
| Nouvelle-Zélande
| 1,50 $
| 2,50 $
|-
| Pays-Bas
| 1,50 $
| 2,50 $
|-
| Pologne
| 1,50 $
| 2,50 $
|-
| Portugal
| 1,50 $
| 2,50 $
|-
| République tchèque
| 4,50 $
| 2,50 $
|-
| Roumanie
| 1,50 $
| 2,50 $
|-
| Royaume-Uni
| 1,50 $
| 2,50 $
|-
| Slovaquie
| 1,50 $
| 2,50 $
|-
| Slovénie
| 1,50 $
| 2,50 $
|-
| Suède
| 1,50 $
| 2,50 $
|}

=e911 via l'interface revendeur=

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

Notez que le DID doit être lié à votre client. (Revendeur > Gestion des comptes clients > Cliquez sur '''Gérer''' où se trouve votre client.)

Dirigez-vous sous la barre de navigation sur [Revendeur] puis cliquez sur [Gestion des tarifs et des forfaits]
: [[File:e911_Reseller_1_FR.png|thumb|none|300px]]

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

: [[File:e911_Reseller_2_FR.png|thumb|none|700px]]

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

: [[File:e911_Reseller_3_FR.png|thumb|none|700px]]

Puis défiler vers le bas et trouvez la fonction '''"e911"''', et activez-la.

: [[File:e911_Reseller_4_FR.png|thumb|none|500px]]

1) Pour ajouter une IVR à votre client, ou pour aider votre client à en ajouter une. Allez sous le '''[Services]''' de la barre de navigation de gauche, puis sur '''[e911]'''
: [[File:e911_Add.png|thumb|none|400px]]

2) Dans la ligne où le DID souhaité, cliquez sur le bouton ACTIVÉ [ENABLED].
: [[File:e911_Add2.png|thumb|none|600px]]

3) Votre client doit '''lire''' et '''accepter les conditions générales'''.

4)Remplissez le formulaire et cliquez sur le bouton Valider [Validate].

[[category:Guides en français]]

Emergency Services

2025-11-25T23:24:30Z

RP: /* Unverified Emergency Calls (E911) */

{| class="wikitable"
|+
|-
! Article en Français !! Artículo en Español
|-
| [https://wiki.voip.ms/article/Service_E911 Français] ||
[https://wiki.voip.ms/article/Servicios_de_Emergencia Español]
|}

== Article blog ==
[https://wiki.voip.ms/article/Enhanced_911_and_VoIP Enhanced 911 and VoIP]

== Emergency Services ==

For customers residing in <b>United States</b>, when you make an emergency call, VoIP.ms will attempt to automatically route your call through a third-party service provider to the Public Safety Answering Point ("PSAP") corresponding to your address of record on your account.

For customers residing in <b>Canada</b>, your emergency call will be directly sent to an emergency call center confirming your identity and location, and then immediately transferred to the local PSAP.

For customers residing <b>outside United States or Canada</b>, telecommunication regulators generally provide that all Customers who are using VoIP services as their primary residential or business telephone carrier must activate Emergency Services on at least one of their DIDs.

For emergency services address information to be passed to your local PSAP dispatcher, you must set your outbound CallerID value to the specific DID you are purchasing emergency service for.

Due to the various dependencies of VoIP networks, VoIP.ms cannot and does not guarantee your emergency call will go through. Many conditions such as loss of power, Internet access or connectivity and/or several other conditions may cause emergency services to be inoperable.

Emergency Services have a one-time fee <b>starting at $ 1.50 and a monthly fee starting at $ 1.50 per month per DID number activated per month</b>. For complete pricing, visit: https://www.voip.ms/residential/pricing

VoIP.ms has added an extension to its network where all VoIP.ms users may call to test their CallerID. At any time, you may test your outbound CallerID by dialing ’1-555-555-0911’ through VoIP.ms’ network and the system will playback your CallerID, then make a short pause, and play the test result. <b>As a reminder, if it is not correctly configured, your call will not go through</b>.

For all the details, refer to our Terms of Service: https://www.voip.ms/terms-of-service



You can activate this service at your Customer Portal >> DID Numbers >> Emergency Services.

'''Important Note:'''
<span style="color:red">'''Note that if your DID from Voip.ms is not subscribed to the Emergency Services, attempting to make a call to 911 will result in a
'''busy signal, since your callerID (which should be a valid DID or number) won't be in Voip.ms Database. This will result by not'''
'''reaching the desired destination. Also make sure to read the Terms and Condition on this page.'''</span>

''' <span style="color:red"> Update as of October 2021 </span> '''

Due to new regulations for Emergency Services, softphones on mobile devices will start using the native mobile dialer to place calls to the emergency services. '''It is possible that not all softphone devices will provide this functionality and it is highly suggested to come into close communication with the softphone provider for further questions.'''

[[File:Emergency Services option.png|thumb|none|600px]]

Use of Emergency Services costs a recovery setup fee of $ 1.50 on activation and a regulatory recovery fee of $ 1.50 per DID number activated per month. VoIP.MS does not make a cent on this charge, it is simply what must be paid to provide this service.

Please note you can enable this service only for Canadian or US numbers (including USA or Canadian toll free numbers).
You just have to click on the checkbox button in order to start the process to enable the service for this number.

[[File:Enable emergency services.png|thumb|none|600px]]

After this you can verify the Terms & conditions regarding of this service. Just please read them carefully and type "I agree" if you wish to enable the emergency services.

Click here to read Terms & conditions regarding this service. [[E911#VoIP.ms_911.2Fe911_Emergency_Service:_Terms_of_Service|Emergency services TOS]]

A new window will request some required information that will be used for the emergency services. Just fill the required fields and be sure to use the correct address. After this, click on the Validate button and if everything is fine, you will receive an email confirmation.

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

Once your emergency service is active, you have to set your DID number as your outgoing [[Caller ID]] number for your main account or [[Sub Accounts]] you are going to use to dial out.

You can set the [[Caller ID]] number for your main account at your Customer Portal >> Main Menu >> [[Account Settings]] >> General tab >> "[[Caller ID]] number" option.

If you are going to use a [[Sub Accounts|sub account]] to dial out, you can set the [[Caller ID]] for it inside your Customer Portal >> [[Sub Accounts|sub account]] >> Manage [[Sub Accounts|sub account]] >> Edit >> [[Caller ID]] number.

To test that your [[Caller ID]] is working properly, you can Dial 1-555-555-0911 from VoIP.ms Network. The system will playback your [[Caller ID]], then make a short pause, and play the test result. To make sure your device/switch is correctly configured for emergency services, you must ensure that your [[Caller ID]] number matches exactly the DID number that is activated for emergency services. That is the only way to identify you correctly. Otherwise your 911 call will not go through. Correctly formatted [[Caller ID]] consist of a 10 digits number, identical to your DID. For example, if your DID is 555.555.1234, your [[Caller ID]] number should be 5555551234.

If at any time you need to change your address, you just need to select the "Modify" icon (The one with the pencil and paper). After this you will be able to change your emergency services information and after the information has been approved, you will be confirmed by email and the emergency services will be updated.

[[File:E911Enable.png|thumb|none|600px]]

== Unverified Emergency Calls (E911) ==

We’ve introduced an important update to help keep you safe. From now on, 911 calls will connect from any number within the U.S. or Canada, even if E911 service isn’t configured.

This change ensures that you can reach emergency services in critical situations, no matter your setup.

Please note that a $75 fee applies per 911 call attempts if E911 is not active on your number. We strongly recommend enabling E911 to avoid this fee and ensure your location is automatically shared with emergency responders.

== VoIP.ms 911/e911 Emergency Service: Terms of Service ==

EMERGENCY COMMUNICATIONS

The Company wants to make sure that Customers are aware of important differences in the way Emergency Services operate when using VoIP services when compared with traditional telephone service. Please find below what Customers need to keep in mind.  

'''Routing of Emergency Calls'''
For United States residents, when a Customer makes an emergency call, the Company will attempt to automatically route its call through a third-party service provider to the Public Safety Answering Point (“PSAP”) corresponding to the Customer’s address of record on its account. However, the delivery of the Customer’s physical location to its local PSAP is not guaranteed. It is possible that the Customer’s location will not be provided to the PSAP dispatcher On such occasions, it will be the Customer’s sole responsibility to give the dispatcher its name, location (or location of the emergency) and contact information to receive emergency service assistance. For Canadian residents, a Customer’s emergency call will be directly sent to an emergency call center confirming its identity and location, and then immediately transferred to the local PSAP.

'''Limitations Due to VoIP Networks'''
Due to the various dependencies of VoIP networks, the Company cannot and does not guarantee a Customer’s emergency call will go through. Many conditions such as loss of power, Internet access or connectivity and/or several other conditions may cause emergency services to be inoperable. The Company does not have control over those types of situations and therefore cannot be held liable of such inoperability. The Company will take commercially reasonable measures to prevent service outages within its network.  

'''Outbound CallerID'''
For emergency services address information to be passed to a Cusomter’s local PSAP dispatcher, the Customer’s outbound CallerID value must be set to the specific DID it is purchasing emergency service for. Therefore, by agreeing to these Terms, a Customer is deemed to have set the outbound CallerID number to the DID it has enabled emergency services for when making an outbound emergency call.  The Company has added an extension to its network where all Customers may call to test their CallerID value. At any time, a Customer may test its outbound CallerID value by dialing ‘1-555-555-0911’ through the Company’s network.  

'''Limitations on Emergency Services'''
CUSTOMERS UNDERSTAND THE LIMITATIONS OF THE COMPANY’S EMERGENCY SERVICES AND ASSUMES ALL LIABILITY AND RESPONSIBILITY, AND RELEASES THE COMPANY TO SUCH EXTENT, FOR THE USE OF EMERGENCY SERVICES,AND FURTHER AGREES TO HOLD THE COMPANY, ITS OFFICERS, DIRECTORS, EMPLOYEES AND AGENTS HARMLESS FOR ANY DAMAGE, WHETHER DIRECT OR INDIRECT THAT MAY RESULT FROM : (1) THE EMERGENCY SERVICES PROVIDED BY THE COMPANY (INCLUDING BUT NOT LIMITED TO SITUATIONS OF UNAVAILABILITY OF EMERGENCY SERVICES AS DESCRIBED IN THESE TERMS AND INCOMPLETE OR INCORRECT LOCATION INFORMATION PROVIDED BY THE CUSTOMER); (2) CUSTOMER’S FAILURE TO OBTAIN ACCESS TO CONVENTIONAL EMERGENCY SERVICE AS PART OF A TELEPHONE LINE SUBSCRIPTION FROM ANOTHER COMPANY UNDER SEPARATE AGREEMENT; (3) CUSTOMER’S FAILURE OR DELAY IN UTILIZING CONVENTIONAL EMERGENCY SERVICE. CUSTOMERS WHO RESELL THE SERVICES FURTHER AGREE THAT THEY ARE RESPONSIBLE FOR NOTIFYING, AND AGREE TO NOTIFY, THEIR CUSTOMERS, CONTRACTORS, AGENTS, EMPLOYEES, ASSOCIATES, SHAREHOLDERS, PARTNERS, AND ANY OTHER POTENTIAL USER OF THE COMPANY’S SERVICES OF THE NATURE AND LIMITATIONS OF THE EMERGENCY SERVICES.  IF A CUSTOMER IS NOT COMFORTABLE WITH THE LIMITATIONS OF EMERGENCY CALLS, THE CUSTOMER MUST CONSIDER AN ALTERNATE MEANS FOR ACCESSING TRADITIONAL EMERGENCY SERVICES, AS REGISTRATION TO EMERGENCY SERVICES IS MANDATORY IN MOST COUNTRIES. FURTHERMORE, THE CUSTOMER AGREES THAT THE COMPANY HAS NO LIABILITY IN RELATION TO THE QUALITY OF THE ADVICE AND SERVICES PROVIDED BY A PSAP.

==Pricing for Emergency Services==

{| class="wikitable"
|- style="font-weight:bold; vertical-align:bottom;"
! Country
! One-time fee
! Recurring Fee
|-
| style="vertical-align:bottom;" | Australia
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | Austria
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | Belgium
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | Bulgaria
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | Canada
| $ 1.50
| $ 1.50
|-
| style="vertical-align:bottom;" | Croatia
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | Cyprus
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | Czech Republic
| $ 4.50
| $ 2.50
|-
| style="vertical-align:bottom;" | Denmark
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | Estonia
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | Finland
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | France
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | Germany
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | Greece
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | Hungary
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | Ireland
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | Italy
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | Latvia
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | Lithuania
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | Netherlands
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | New Zealand
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | Norway
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | Poland
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | Portugal
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | Romania
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | Slovakia
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | Slovenia
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | Spain
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | Sweden
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | United Kingdom
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | United States
| $ 1.50
| $ 1.50
|}

=Using 911 using the Reseller Interface=

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

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

Go under the navigation bar on '''[Reseller]''' then click on '''[Manage Rates & Packages]'''
: [[File:e911_Reseller_1.png|thumb|none|300px]]

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

: [[File:e911_Reseller_2.png|thumb|none|700px]]

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

: [[File:e911_Reseller_3.png|thumb|none|700px]]

Then scroll down and find the feature "'''e911'''", and enable it.

: [[File:e911_Reseller_4.png|thumb|none|500px]]

1) To add a e911 entry for your client, or to help your client adding one. Go under the '''[Services]''' at the left navigation bar, then on '''[e911]'''
: [[File:e911_Add.png|thumb|none|400px]]

2) In the row where the wished DID, click on the [ENABLED] button.
: [[File:e911_Add2.png|thumb|none|600px]]

3) Your client '''must''' read and '''Agreed the Terms and Conditions'''.

4) Complete the form and click the '''[Validate]''' button.

[[category:guides]]

Servicio E911

2025-11-14T20:08:34Z

RP: /* E911 */

{| class="wikitable"
|+
|-
! Article in English !! Article en Français
|-
| [https://wiki.voip.ms/article/E911 English] ||
[https://wiki.voip.ms/article/Service_E911 Français]
|}

__TOC__

== Articulo de Blog ==
[https://wiki.voip.ms/article/Enhanced_911_and_VoIP Enhanced 911 and VoIP]

== E911 ==

El sistema básico del 911 funciona identificando la ubicación de la persona que llama por el número de teléfono fijo. La llamada se enruta automáticamente al PSAP (punto de respuesta de seguridad pública más cercano) y el despachador en ese punto se comunica con el personal de servicios de emergencia más cercano para atender la llamada.

Sin embargo, con los teléfonos inalámbricos y voip, el sistema 911 original no podía identificar la ubicación de una persona que llamaba y, si la persona que llamaba no podía identificar o describir una ubicación, el personal de emergencia se veía obstaculizado para brindar asistencia.

E911 es la solución para esto. Cuando se realiza una llamada de emergencia (e911) a través de la red VoIP.MS En los EE. UU., la dirección física que ingresará en el momento del registro e911 para un DID específico se transmitirá a su PSAP local, proporcionando al despachador en el PSAP la ubicación exacta donde se requiere ayuda. Tenga en cuenta que "Puerto Rico" ahora se puede aprovisionar para e911, sin embargo, no funciona de la misma manera que el resto de los EE. UU. La información de la dirección no se transmitirá ni se mostrará en el PSAP, por lo que es necesario que la persona que llama proporcione la dirección verbalmente al llamar.

El servicio 911 canadiense se maneja de manera un poco diferente. Todas las llamadas canadienses al centro de llamadas de National 911 son respondidas por un despachador que accederá a la base de datos del proveedor del servicio 911 para obtener la ubicación y también preguntará verbalmente a la persona que llama por su ubicación antes de transferir la llamada al PSAP de la persona que llama dando verbalmente el PSAP la ubicación y luego conectar a la persona que llama. Esta es la forma en que se tratan todas las llamadas VOIP al 911 en Canadá

Para activar este servicio, puede encontrar esta opción en su Portal>>DID Numbers >> E911.

'''Nota Importante:'''

<span style="color:red">'''Tenga en cuenta que si alguno de sus números DID con voip.ms no está suscrito al servicio E911, intentar hacer una llamada'''
'''al 911 resultará en "ocupado" (Busy signal) debido a que su CallerID (el número que manda, que debe ser un número válido)'''
'''no estará en la base de datos de voip.ms; el resultado será una llamada fallida.'''
'''Igualmente asegúrese de leer los "Términos y condiciones del servicio"(Terms of Service) al final de la página. En caso de tener
'''alguna duda o pregunta con respecto a este tema en inglés, por favor contacte a un miembro de nuestro staff en español
'''y con mucho gusto le atenderemos.'''</span>

''' <span style="color:red"> Actualización a octubre de 2021 </span> '''

Debido a las nuevas regulaciones para los servicios E911, los softphones en dispositivos móviles comenzarán a usar el marcador móvil nativo para realizar llamadas a los servicios de emergencia. '''Es posible que no todos los dispositivos de softphone proporcionen esta funcionalidad y se recomienda encarecidamente entablar una estrecha comunicación con el proveedor de softphone si tiene más preguntas.'''

[[File:E9112.JPG|thumb|none|600px]]

El uso del servicio e911 tiene un costo de recuperación de $1.50 USD por activación y una cuota menusal de $1.50 USD por cada número que tenga activo este servicio. VoIP.MS no obtiene ganancias por este servicio, simplemente cobra lo que se necesita para poder proveerlo.

Este servicio esta disponible solamente para números de Canada y US, incluyendo números Toll Free. Solamente tiene que dar click en el botón Apply para comenzar el proceso y habiltar este servicio
para su número.

[[File:E911_new.png|thumb|none|600px]]

Después usted puede leer los Términos y Condiciones para el uso de este servicio. Por favor lealo con detenimiento y después haga click en "I agree to the Terms and Conditions" si quiere habilitar el servicio.

Haga click aquí para leer los Términos y Condiciones de este servicio. [[E911#VoIP.ms_911.2Fe911_Emergency_Service:_Terms_of_Service|E911 TOS]]

Una nueva ventana aparecerá pidiendo información que será usada con este servicio. Por favor introduzca la información requerida y porfavor verifique la dirección sea correcta, ya que esta será usada para atender la emergencia cuando sea solicitada. Después haga click en el botón de validación para confirmar que todo este correcto y al final usted recibirá un correo electrónico de confirmación.

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

Una vez que su servicio e911 esté activo, debe configurar su número DID como su número de identificación de llamadas salientes para su cuenta principal o subcuentas que usará para marcar.

Para configurar su [[Caller ID]] a su main account, puede hacerlo en su Portal >> Main Menu >> [[Configuraciones de la Cuenta (Account Settings)|Account Settings]] >> General tab >> "[[Caller ID]] number" option.

Si va a usar una subcuenta para hacer la llamada, puede asignar el [[Caller ID]] en su Portal >> [[Sub Cuentas (Sub Accounts)|Sub accounts]] >> Manage [[Sub Cuentas (Sub Accounts)|Sub accounts]] >> Edit >> [[Caller ID]] number.

Para comprobar que su identificador de llamadas funciona correctamente, puede marcar 1-555-555-0911 desde la red VoIP.ms. El sistema reproducirá su identificador de llamadas, luego hará una breve pausa y reproducirá el resultado de la prueba. Para asegurarse de que su dispositivo / interruptor esté configurado correctamente para e911, debe asegurarse de que su número de identificación de llamadas coincida exactamente con el número DID que está activado para el 911. Esa es la única forma de identificarlo correctamente. De lo contrario, su llamada al 911 no se realizará. El identificador de llamadas con el formato correcto consta de un número de 10 dígitos, idéntico a su DID. Por ejemplo, si su DID es 555.555.1234, su número de identificación de llamadas debe ser 5555551234.

Si en algún momento necesita cambiar la dirección física con la que dió de alta el servicio, puede seleccionar la opción "Modify" y dar click en el botón "Apply". Con esto usted podrá cambiar la información relacionada a este servicio y después de que ésta haya sido aprobada, recibirá una confirmación por correo electrónico de que su información ha sido actualizada.

[[File:E911Enable.png|thumb|none|600px]]

== Llamada de emergencia (e911) sin verificacion==

VoIP.ms ahora conectará las llamadas al 911 desde cualquier número dentro de Estados Unidos o Canadá, incluso si el servicio E911 no está configurado, para garantizar su seguridad en situaciones de emergencia.

Se aplicará un cargo de $75 dólares por cada llamada al 911 completada si el servicio E911 no está activo en su número. Le recomendamos que active el servicio E911 para evitar este cargo y garantizar que su ubicación se comparta con precisión con los servicios de emergencia.

== VoIP.ms 911/e911 Servicio de Emergencia: Términos del servicio ==

La Compañía quiere asegurarse de que los Clientes sean conscientes de las importantes diferencias en el funcionamiento de los Servicios de Emergencia al utilizar servicios VoIP en comparación con el servicio telefónico tradicional. A continuación, se describe lo que los Clientes deben tener en cuenta.  

'''Enrutamiento de llamadas de emergencia'''
Para los residentes de los Estados Unidos, cuando un Cliente realiza una llamada de emergencia, la Compañía intentará enrutarlas automáticamente a través de un proveedor de servicios de terceros al Centro de Respuesta de Seguridad Pública ("PSAP") correspondiente a la dirección registrada del Cliente en su cuenta. Sin embargo, no se garantiza la entrega de la ubicación física del Cliente a su PSAP local. Es posible que la ubicación del Cliente no se proporcione al despachador del PSAP. En tales ocasiones, será responsabilidad exclusiva del Cliente proporcionar al despachador su nombre, ubicación (o la ubicación de la emergencia) e información de contacto para recibir asistencia de los servicios de emergencia. Para los residentes de Canadá, la llamada de emergencia del Cliente se enviará directamente a un centro de llamadas de emergencia para confirmar su identidad y ubicación, y luego se transferirá de inmediato al PSAP local.

'''Limitaciones debido a las redes VoIP'''
Debido a las diversas dependencias de las redes VoIP, la Compañía no puede garantizar que una llamada de emergencia del Cliente se realice. Muchas condiciones, como la pérdida de energía, acceso a Internet o conectividad, y otras condiciones, pueden hacer que los servicios de emergencia sean inoperables. La Compañía no tiene control sobre este tipo de situaciones y, por lo tanto, no puede ser considerada responsable de tal inoperabilidad. La Compañía tomará medidas comercialmente razonables para prevenir interrupciones del servicio dentro de su red.  

'''CallerID de salida'''
Para que la información de la dirección de emergencia sea transmitida al despachador del PSAP local del Cliente, el valor del CallerID de salida del Cliente debe estar configurado en el DID específico para el cual se adquieren los servicios de emergencia. Al aceptar estos Términos, se considera que el Cliente ha configurado el número de CallerID de salida al DID que ha habilitado para los servicios de emergencia al realizar una llamada de emergencia de salida. La Compañía ha agregado una extensión en su red donde todos los Clientes pueden llamar para probar el valor de su CallerID. En cualquier momento, un Cliente puede probar su valor de CallerID de salida marcando '1-555-555-0911' a través de la red de la Compañía.  

'''Limitaciones en los servicios de emergencia'''
LOS CLIENTES ENTIENDEN LAS LIMITACIONES DE LOS SERVICIOS DE EMERGENCIA DE LA COMPAÑÍA Y ASUMEN TODA LA RESPONSABILIDAD Y LIBERAN A LA COMPAÑÍA DE CUALQUIER RESPONSABILIDAD POR EL USO DE LOS SERVICIOS DE EMERGENCIA, Y ADEMÁS ACEPTAN MANTENER INDEMNES A LA COMPAÑÍA, SUS OFICIALES, DIRECTORES, EMPLEADOS Y AGENTES POR CUALQUIER DAÑO, YA SEA DIRECTO O INDIRECTO, QUE PUEDA RESULTAR DE: (1) LOS SERVICIOS DE EMERGENCIA PROPORCIONADOS POR LA COMPAÑÍA (INCLUYENDO, PERO NO LIMITADO A SITUACIONES DE INDISPONIBILIDAD DE LOS SERVICIOS DE EMERGENCIA COMO SE DESCRIBE EN ESTOS TÉRMINOS Y LA INFORMACIÓN INCOMPLETA O INCORRECTA PROPORCIONADA POR EL CLIENTE); (2) EL INCUMPLIMIENTO DEL CLIENTE DE OBTENER ACCESO A LOS SERVICIOS DE EMERGENCIA CONVENCIONALES COMO PARTE DE UNA SUSCRIPCIÓN DE LÍNEA TELEFÓNICA DE OTRA COMPAÑÍA BAJO UN ACUERDO SEPARADO; (3) EL INCUMPLIMIENTO O RETRASO DEL CLIENTE EN UTILIZAR LOS SERVICIOS DE EMERGENCIA CONVENCIONALES. LOS CLIENTES QUE REVENDE LOS SERVICIOS ADEMÁS ACEPTAN SER RESPONSABLES DE NOTIFICAR, Y SE COMPROMETEN A NOTIFICAR, A SUS CLIENTES, CONTRATISTAS, AGENTES, EMPLEADOS, ASOCIADOS, ACCIONISTAS, SOCIOS Y CUALQUIER OTRO USUARIO POTENCIAL DE LOS SERVICIOS DE LA COMPAÑÍA SOBRE LA NATURALEZA Y LIMITACIONES DE LOS SERVICIOS DE EMERGENCIA. SI UN CLIENTE NO SE SIENTE CÓMODO CON LAS LIMITACIONES DE LAS LLAMADAS DE EMERGENCIA, DEBE CONSIDERAR UN MEDIO ALTERNATIVO PARA ACCEDER A LOS SERVICIOS DE EMERGENCIA TRADICIONALES, YA QUE EL REGISTRO EN LOS SERVICIOS DE EMERGENCIA ES OBLIGATORIO EN LA MAYORÍA DE LOS PAÍSES. ADEMÁS, EL CLIENTE ACEPTA QUE LA COMPAÑÍA NO TIENE RESPONSABILIDAD EN RELACIÓN CON LA CALIDAD DEL CONSEJO Y LOS SERVICIOS PROPORCIONADOS POR UN PSAP.

'''Precios'''
Se cobrará al Cliente una tarifa de configuración y una tarifa de recuperación regulatoria por mes por cada DID enviado a la base de datos de servicios de emergencia; consulte los precios. Esta tarifa no es reembolsable.

'''Regulación: Sección aplicable a los clientes que residen en los Estados Unidos o Canadá'''
Debido a recientes disposiciones y regulaciones de la FCC/CRTC, todos los Clientes que utilicen los servicios de la Compañía como su principal proveedor telefónico residencial o comercial deben activar los Servicios de Emergencia 911 en al menos uno de sus DIDs. De acuerdo con las regulaciones, los Clientes también deben asegurarse de que sus sistemas de comunicación permitan a todos los usuarios marcar siempre el 911, sin la necesidad de marcar un prefijo, y que, si se marca el 911 desde el sistema de comunicación del Cliente, el personal y las autoridades pertinentes sean informados de inmediato de la emergencia. La Compañía se esfuerza por proporcionar al Cliente los servicios necesarios para cumplir con las reglas y regulaciones, de acuerdo con estos Términos. No obstante, el Cliente acepta cumplir con dichas regulaciones y entiende que la Compañía no asumirá ninguna responsabilidad en caso de incumplimiento de dichas regulaciones por parte del Cliente.  

'''Regulación: Sección aplicable a los clientes que residen fuera de los Estados Unidos o Canadá'''
Los reguladores de telecomunicaciones fuera de Canadá y los Estados Unidos generalmente disponen que todos los Clientes que utilicen servicios VoIP como su principal proveedor telefónico residencial o comercial deben activar los Servicios de Emergencia en al menos uno de sus DIDs. Esta obligación se aplica a cualquier Cliente que resida en los siguientes países: Austria, Australia, Bélgica, Bulgaria, Croacia, República Checa, Chipre, Dinamarca, Estonia, Finlandia, Francia, Alemania, Grecia, Hong Kong, Hungría, Irlanda, Israel, Italia, Japón, Letonia, Lituania, Países Bajos, Nueva Zelanda, Noruega, Polonia, Portugal, Rumania, Singapur, Eslovaquia, Eslovenia, Corea del Sur, España, Suecia, Turquía y Reino Unido. Si un Cliente utiliza los Servicios para finalizar llamadas con un CallerID que no está registrado en su cuenta, el Cliente reconoce y acepta que la Compañía no podrá finalizar las llamadas de Servicios de Emergencia debido a la falta de la información necesaria y declina toda responsabilidad con respecto a este asunto. En este caso, el Cliente debe comunicarse con el operador con el que está registrado su número telefónico para activar los Servicios de Emergencia con ellos, ya que será su responsabilidad finalizar sus llamadas de Servicios de Emergencia.  

'''Uso de los Servicios de Emergencia fuera del país de residencia del Cliente'''
Los reguladores generalmente exigen que se registre una dirección local cuando se activan los Servicios de Emergencia para un número telefónico. Si un Cliente utiliza los Servicios para la terminación de llamadas locales en un país en el que no tiene una dirección de residencia, el Cliente puede no ser capaz de registrarse y activar los Servicios de Emergencia en dicho país y puede experimentar dificultades para comunicarse con los Servicios de Emergencia en su país de residencia. La Compañía declina toda responsabilidad con respecto a esta limitación de servicio, de acuerdo con esta sección de los Términos.  

==Precios de los servicios de emergencia==

{| class="wikitable" style="vertical-align:middle;"
|- style="font-weight:bold; text-align:center;"
! País
! Tarifa única
! Tarifa recurrente
|-
| Alemania
| 1,50 $
| 2,50 $
|-
| Australia
| 1,50 $
| 2,50 $
|-
| Austria
| 1,50 $
| 2,50 $
|-
| Bélgica
| 1,50 $
| 2,50 $
|-
| Bulgaria
| 1,50 $
| 2,50 $
|-
| Canadá
| 1,50 $
| 1,50 $
|-
| Chipre
| 1,50 $
| 2,50 $
|-
| Croacia
| 1,50 $
| 2,50 $
|-
| Dinamarca
| 1,50 $
| 2,50 $
|-
| Eslovaquia
| 1,50 $
| 2,50 $
|-
| Eslovenia
| 1,50 $
| 2,50 $
|-
| España
| 1,50 $
| 2,50 $
|-
| Estados Unidos
| 1,50 $
| 1,50 $
|-
| Estonia
| 1,50 $
| 2,50 $
|-
| Finlandia
| 1,50 $
| 2,50 $
|-
| Francia
| 1,50 $
| 2,50 $
|-
| Grecia
| 1,50 $
| 2,50 $
|-
| Hungría
| 1,50 $
| 2,50 $
|-
| Irlanda
| 1,50 $
| 2,50 $
|-
| Italia
| 1,50 $
| 2,50 $
|-
| Letonia
| 1,50 $
| 2,50 $
|-
| Lituania
| 1,50 $
| 2,50 $
|-
| Noruega
| 1,50 $
| 2,50 $
|-
| Nueva Zelanda
| 1,50 $
| 2,50 $
|-
| Países Bajos
| 1,50 $
| 2,50 $
|-
| Polonia
| 1,50 $
| 2,50 $
|-
| Portugal
| 1,50 $
| 2,50 $
|-
| Reino Unido
| 1,50 $
| 2,50 $
|-
| República Checa
| 4,50 $
| 2,50 $
|-
| Rumania
| 1,50 $
| 2,50 $
|-
| Suecia
| 1,50 $
| 2,50 $
|}

=Usando el 911 en la interfaz de revendedor =

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

Tenga en cuenta que el DID debe estar vinculado a su cliente. (Reseller> Manage client's accounts> Haga clic en Manage client donde está su cliente).

Vaya debajo de la barra de navegación en '''[Reseller]''' y luego haga clic en '''[Manage Rates and Packages]'''
: [[File:e911_Reseller_1.png|thumb|none|300px]]

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

: [[File:e911_Reseller_2.png|thumb|none|700px]]

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

: [[File:e911_Reseller_3.png|thumb|none|700px]]

Luego, desplácese hacia abajo y busque la función '''"e911"''' y habilítela.

: [[File:e911_Reseller_4.png|thumb|none|500px]]

1) Para agregar una entrada e911 para su cliente, o para ayudar a su cliente a agregar una. Vaya debajo de '''[Services]''' en la barra de navegación izquierda, luego en '''[e911]'''
: [[File:e911_Add.png|thumb|none|400px]]

2) En la fila donde se encuentra el DID deseado, haga clic en el botón''' [ENABLED]'''.
: [[File:e911_Add2.png|thumb|none|600px]]

3) Su cliente '''debe leer''' y '''aceptar los Términos y Condiciones'''.

4) Complete el formulario y haga clic en el botón '''[Validate]'''.

[[category:Guías]]

Service E911

2025-11-14T20:07:02Z

RP: /* Activation du service */

{| class="wikitable"
|+
|-
! Article in English !! Artículo en Español
|-
| [https://wiki.voip.ms/article/E911 English] || [https://wiki.voip.ms/article/Servicio_E911 Español]
|}

== Renseignements E911 ==
Le service 911 de base, fonctionne en localisant le lieu d'appel de l'appelant, par le numéro de téléphone de la ligne de terre. L'appel est automatiquement acheminé vers le point de sécurité publique le plus prêt, d'où le répartiteur de ce point, contacte le personnel du service d'urgence le plus prêt, pour ainsi faire face à l'appel.

Cependant, avec la venue des téléphones sans fil et téléphones IP, le système 911 d'origine est devenu incapable de localiser les appelants, et dans le cas où l'appelant n'est pas en mesure d'identifier ou de décrire le lieu où il se trouve, ceci représente une entrave pour le personnel d'urgence quant à l'assistance rendue. Les adresses de Puerto Rico peuvent maintenant être provisionnées. Par contre elles fonctionnent différemment du reste des États-Unis, mais similairement au Canada.

E911 est la solution à ce problème. Lorsqu'un appel d'urgence (E911) est placé à travers le réseau VoIP.ms, l'adresse physique que vous avez fournie au moment d'enregistrer le service pour un DID spécifique, sera transmise à votre PSAP local, fournissant ainsi au répartiteur l'emplacement exact où l'aide est requise.

== Activation du service ==

Vous pouvez activer ce service par votre portail client, Numéros DID > [https://www.voip.ms/m/me911.php E911].

<span style="color:red">
'''Note Important:'''<br/>
Notez que si votre DID avec VoIP.ms n'est pas enregistré au service, vous ne serez pas en mesure de composer 911.
Cela tombera dans une tonalité occupée, étant donné que le [[ID de l'appelant | numéro d'identification de l'appelant]] ne sera pas dans la base de données.
Cela n'atteindra pas la destination désirée. Assurez-vous aussi de lire les Termes d'utilisation à la fin de la page.
</span>

''' <span style="color:red"> Mise à jour octobre 2021 </span> '''

En raison de la nouvelle réglementation pour les services E911, les softphones sur les appareils mobiles commenceront à utiliser le numéroteur mobile natif pour passer des appels aux services d'urgence. '''Il est possible que tous les appareils de softphone ne fournissent pas cette fonctionnalité et il est fortement suggéré d'entrer en communication étroite avec le fournisseur de softphone pour d'autres questions.'''

[[File:911Menu.png|thumb|none|600px]]

L'utilisation du service engendre un frais d'activation de $1,50 ainsi qu'un frais régulier mensuel de $1,50 par numéro DID activé. VoIP.ms ne fait pas un sous sur l'utilisation de ce service. Ces coûts sont simplement ce qui doit être payé pour fournir ce service.

Veuillez noter que vous pouvez activer ce service uniquement pour les numéros canadiens ou US (y compris les numéros sans frais (Toll-Free) des États-Unis ou du Canada). Vous n'avez qu'à cliquer sur le bouton "Activer" afin de débuter le processus d'activation de ce service pour un numéro.

[[File:911nouveau.png|thumb|none|600px]]

Ensuite, vous pouvez vérifier les termes et conditions relatifs à ce service. Assurez-vous de les lire attentivement et cliquez sur "Je suis d'accord avec les conditions" si vous souhaitez activer le service.

Cliquez ici pour lire les Termes & conditions relatives à ce service. [[#Conditions d'utilisation|E911 TOS]]

Une nouvelle fenêtre demandera quelques informations nécessaires qui seront utilisées pour le service. Il suffit de remplir les champs requis et s'assurer d'entrer la bonne adresse. Enfin, cliquez sur "Valider" et si tout est conforme, vous recevrez alors un courriel de confirmation.

[[File:911Remplir.png|thumb|none|600px]]

== Appel d'urgence (e911) non-vérifié ==

VoIP.ms acheminera désormais les appels au 911 à partir de n’importe quel numéro aux États-Unis ou au Canada, même si le service E911 n’est pas configuré, afin d’assurer votre sécurité en cas d’urgence.

Des frais de 75 $ s’appliqueront pour chaque appel complété au 911, si le service E911 n’est pas activé sur votre numéro. Nous vous recommandons d’activer le service E911 pour éviter ces frais et pour que votre emplacement soit transmis avec précision aux services d’urgence.

== Utilisation du service ==

Une fois votre service activé, vous devrez définir votre numéro DID comme votre numéro d'identification de l'appelant (Caller ID) pour appels sortants, pour le compte principal ou sous-compte que vous utiliserez pour appeler.

Vous pouvez définir le numéro d'identification de l'appelant (Caller ID) de votre compte principal, dans votre portail client, Menu principal > Paramètres du compte > Généralités > Numéro d'identification de l'appelant.

Si vous prévoyez utiliser un compte secondaire pour composer, vous pouvez définir le numéro d'identification de l'appelant, dans votre portail client, Sous-comptes > [https://www.voip.ms/m/managesubaccount.php Gestion des sous-comptes] > Modifier > '''Numéro d'identification de l'appelant'''.

Pour tester que votre [[ID de l'appelant | numéro d'identification de l'appelant]] fonctionne correctement, vous pouvez composer 1-555-555-0911 à partir du réseau VoIP.ms. Le système vous répétera votre [[ID de l'appelant | numéro d'identification de l'appelant]], puis fera une courte pause, et jouera le résultat du test. Pour vous assurer que votre appareil / commutateur est correctement configuré pour le E911, vous devez vous assurer que votre [[ID de l'appelant | numéro d'identification de l'appelant]] correspond exactement au numéro DID qui est activé pour le service. C'est la seule façon de vous identifier correctement. Sinon, votre appel au E911 ne passera pas.

Le bon format du [[ID de l'appelant | numéro d'identification de l'appelant]] est composé de 10 chiffres, identiques à votre DID. Par exemple, si votre DID est 555.555.1234, votre [[ID de l'appelant | numéro d'identification de l'appelant]] devrait être 5555551234.

Si jamais vous deviez changer votre adresse, il vous suffirait de sélectionner l'option "Modifier", ensuite, vous seriez en mesure de modifier vos informations E911 et une fois l'information approuvée, vous recevriez une confirmation par courriel et le service serait mis à jour.

[[File:911Active.png|thumb|none|600px]]

== Conditions d'utilisation ==

La Société souhaite s'assurer que les clients sont conscients des différences importantes dans la manière dont les services d'urgence fonctionnent lorsqu'ils utilisent des services VoIP par rapport au service téléphonique traditionnel. Veuillez trouver ci-dessous ce que les clients doivent garder à l'esprit. Acheminement des appels d'urgence.  

'''Acheminement des appels d'urgence'''
Pour les résidents des États-Unis, lorsqu'un client effectue un appel d'urgence, la Société tentera de router automatiquement son appel via un fournisseur de services tiers vers le Point de Réponse de Sécurité Publique («PSAP») correspondant à l'adresse enregistrée du client sur son compte. Toutefois, la transmission de la localisation physique du client à son PSAP local n'est pas garantie. Il est possible que la localisation du client ne soit pas fournie au dispatcher du PSAP. Dans de tels cas, il sera de la seule responsabilité du client de fournir au dispatcher son nom, sa localisation (ou celle de l'urgence) et ses informations de contact pour recevoir l'assistance du service d'urgence. Pour les résidents canadiens, un appel d'urgence d'un client sera directement envoyé à un centre d'appels d'urgence confirmant son identité et sa localisation, puis immédiatement transféré au PSAP local.

'''Limitations dues aux réseaux VoIP'''
En raison des diverses dépendances des réseaux VoIP, la Société ne peut pas et ne garantit pas que l'appel d'urgence d'un client sera acheminé. De nombreuses conditions telles que la perte de courant, l'accès à Internet ou la connectivité, et/ou plusieurs autres conditions peuvent rendre les services d'urgence inopérants. La Société n'a pas de contrôle sur ces types de situations et ne peut donc pas être tenue responsable de cette inopérabilité. La Société prendra des mesures commercialement raisonnables pour prévenir les interruptions de service au sein de son réseau.  

'''Identification de l'appelant sortant (CallerID)'''
Pour que les informations d'adresse des services d'urgence soient transmises au dispatcher PSAP local du client, la valeur de l'ID de l'appelant sortant du client doit être définie sur le DID spécifique pour lequel il achète le service d'urgence. Par conséquent, en acceptant ces Termes, un client est considéré comme ayant défini le numéro de l’identifiant de l’appelant sortant sur le DID pour lequel il a activé les services d'urgence lorsqu'il effectue un appel d'urgence sortant. La Société a ajouté une extension à son réseau où tous les clients peuvent appeler pour tester leur valeur de CallerID. À tout moment, un client peut tester sa valeur de l’identifiant de l’appelant sortant en composant le ‘1-555-555-0911’ via le réseau de la Société.  

'''Limitations des services d'urgence'''
LES CLIENTS COMPRENNENT LES LIMITATIONS DES SERVICES D'URGENCE DE LA SOCIÉTÉ ET ASSUMENT TOUTE RESPONSABILITÉ ET RESPONSABILITÉ ET DÉGAGENT LA SOCIÉTÉ DANS CETTE MESURE POUR L'UTILISATION DES SERVICES D'URGENCE ET CONVIENNENT EN OUTRE DE DÉGAGER DE RESPONSABILITÉ LA SOCIÉTÉ, SES DIRIGEANTS, DIRECTEURS, EMPLOYÉS ET AGENTS POUR TOUT DOMMAGE, DIRECT OU INDIRECT, QUI POURRAIT RÉSULTER : (1) DES SERVICES D'URGENCE FOURNIS PAR LA SOCIÉTÉ (Y COMPRIS MAIS SANS S'Y LIMITER, LES SITUATIONS D'INDISPONIBILITÉ DES SERVICES D'URGENCE COMME DÉCRITES DANS CES TERMES ET LES INFORMATIONS DE LOCALISATION INCOMPLÈTES OU INCORRECTES FOURNIES PAR LE CLIENT) ; (2) L'ÉCHEC DU CLIENT À OBTENIR L'ACCÈS AUX SERVICES D'URGENCE CONVENTIONNELS DANS LE CADRE D'UN ABONNEMENT À UNE LIGNE TÉLÉPHONIQUE D'UNE AUTRE SOCIÉTÉ SOUS UN ACCORD SÉPARÉ ; (3) L'ÉCHEC OU LE RETARD DU CLIENT À UTILISER LES SERVICES D'URGENCE CONVENTIONNELS. LES CLIENTS QUI REVENT LES SERVICES CONVIENNENT QU'ILS SONT RESPONSABLES D'INFORMER, ET ACCEPTENT D'INFORMER, LEURS CLIENTS, CONTRACTANTS, AGENTS, EMPLOYÉS, ASSOCIÉS, ACTIONNAIRES, PARTENAIRES ET TOUT AUTRE UTILISATEUR POTENTIEL DES SERVICES DE LA SOCIÉTÉ SUR LA NATURE ET LES LIMITATIONS DES SERVICES D'URGENCE. SI UN CLIENT N'EST PAS À L'AISE AVEC LES LIMITATIONS DES APPELS D'URGENCE, IL DOIT ENVISAGER UN MOYEN ALTERNATIF POUR ACCÉDER AUX SERVICES D'URGENCE TRADITIONNELS, CAR L'INSCRIPTION AUX SERVICES D'URGENCE EST OBLIGATOIRE DANS LA PLUPART DES PAYS. DE PLUS, LE CLIENT ACCEPTE QUE LA SOCIÉTÉ N'AIT AUCUNE RESPONSABILITÉ EN RELATION AVEC LA QUALITÉ DES CONSEILS ET DES SERVICES FOURNIS PAR UN PSAP.

'''Tarification'''
Le Client sera facturé des frais de mise en place de récupération et des frais réglementaires de récupération par mois pour chaque DID soumis à la base de données des services d'urgence – voir tarification. Ces frais ne sont pas remboursables.

'''Réglementation : Section applicable aux clients résidant aux États-Unis ou au Canada'''
En raison des récentes décisions et réglementations de la FCC/CRTC, tous les Clients qui utilisent les services de la Société comme leur principal opérateur téléphonique résidentiel ou commercial doivent activer les services d'urgence 911 sur au moins l'un de leurs DIDs. Conformément aux réglementations, les Clients doivent également s'assurer que leurs systèmes de communication permettent à tous les utilisateurs de composer le 911 sans avoir besoin de composer un préfixe et que, si le 911 est composé à partir du système de communication du Client, le personnel compétent et les autorités soient rapidement informés de l'urgence. La Société s'efforce de fournir au Client les services nécessaires pour se conformer aux règles et réglementations conformément à ces Termes. Nonobstant ce qui précède, par les présents Termes, le Client accepte de se conformer à ces réglementations et comprend que la Société n'assumera aucune responsabilité en cas de violation de ces réglementations par le Client. 

'''Réglementation : Section applicable aux clients résidant en dehors des États-Unis ou du Canada'''
Les régulateurs de télécommunication hors du Canada et des États-Unis stipulent généralement que tous les Clients utilisant des services VoIP comme leur principal opérateur téléphonique résidentiel ou commercial doivent activer les Services d'urgence sur au moins l'un de leurs DIDs. Cette obligation s'applique à tout Client résidant dans les pays suivants : Autriche, Australie, Belgique, Bulgarie, Croatie, République tchèque, Chypre, Danemark, Estonie, Finlande, France, Allemagne, Grèce, Hong Kong, Hongrie, Irlande, Israël, Italie, Japon, Lettonie, Lituanie, Pays-Bas, Nouvelle-Zélande, Norvège, Pologne, Portugal, Roumanie, Singapour, Slovaquie, Slovénie, Corée du Sud, Espagne, Suède, Turquie et Royaume-Uni. Si un Client utilise les Services pour terminer des appels avec un identifiant de l’appelant qui n'est pas enregistré dans son compte, le Client reconnaît et accepte que la Société ne pourra pas terminer les appels des Services d'urgence en raison du manque des informations nécessaires et décline toute responsabilité à cet égard. Dans ce cas, le Client doit communiquer avec l'opérateur chez qui son numéro de téléphone est enregistré pour activer les Services d'urgence avec eux car il sera de leur responsabilité de terminer vos appels de Services d'urgence.  

'''Utilisation des services d'urgence en dehors du pays de résidence du client'''
Les régulateurs exigent généralement qu'une adresse locale soit enregistrée lors de l'activation des Services d'urgence pour un numéro de téléphone. Si un Client utilise les Services pour terminer des appels locaux dans un pays où il n'a pas d'adresse de résidence, il se peut qu'il ne puisse pas s'inscrire et activer les Services d'urgence dans ledit pays et qu'il rencontre des difficultés à communiquer avec les Services d'urgence dans son pays de résidence. La Société décline toute responsabilité concernant cette limitation de service conformément à cette section des Termes.  

==Tarification des services d'urgence==

{| class="wikitable" style="vertical-align:middle;"
|- style="font-weight:bold; text-align:center;"
! Pays
! Frais uniques
! Frais récurrents
|-
| Allemagne
| 1,50 $
| 2,50 $
|-
| Australie
| 1,50 $
| 2,50 $
|-
| Autriche
| 1,50 $
| 2,50 $
|-
| Belgique
| 1,50 $
| 2,50 $
|-
| Bulgarie
| 1,50 $
| 2,50 $
|-
| Canada
| 1,50 $
| 1,50 $
|-
| Chypre
| 1,50 $
| 2,50 $
|-
| Croatie
| 1,50 $
| 2,50 $
|-
| Danemark
| 1,50 $
| 2,50 $
|-
| Espagne
| 1,50 $
| 2,50 $
|-
| Estonie
| 1,50 $
| 2,50 $
|-
| États-Unis
| 1,50 $
| 1,50 $
|-
| Finlande
| 1,50 $
| 2,50 $
|-
| France
| 1,50 $
| 2,50 $
|-
| Grèce
| 1,50 $
| 2,50 $
|-
| Hongrie
| 1,50 $
| 2,50 $
|-
| Irlande
| 1,50 $
| 2,50 $
|-
| Italie
| 1,50 $
| 2,50 $
|-
| Lettonie
| 1,50 $
| 2,50 $
|-
| Lituanie
| 1,50 $
| 2,50 $
|-
| Norvège
| 1,50 $
| 2,50 $
|-
| Nouvelle-Zélande
| 1,50 $
| 2,50 $
|-
| Pays-Bas
| 1,50 $
| 2,50 $
|-
| Pologne
| 1,50 $
| 2,50 $
|-
| Portugal
| 1,50 $
| 2,50 $
|-
| République tchèque
| 4,50 $
| 2,50 $
|-
| Roumanie
| 1,50 $
| 2,50 $
|-
| Royaume-Uni
| 1,50 $
| 2,50 $
|-
| Slovaquie
| 1,50 $
| 2,50 $
|-
| Slovénie
| 1,50 $
| 2,50 $
|-
| Suède
| 1,50 $
| 2,50 $
|}

=e911 via l'interface revendeur=

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

Notez que le DID doit être lié à votre client. (Revendeur > Gestion des comptes clients > Cliquez sur '''Gérer''' où se trouve votre client.)

Dirigez-vous sous la barre de navigation sur [Revendeur] puis cliquez sur [Gestion des tarifs et des forfaits]
: [[File:e911_Reseller_1_FR.png|thumb|none|300px]]

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

: [[File:e911_Reseller_2_FR.png|thumb|none|700px]]

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

: [[File:e911_Reseller_3_FR.png|thumb|none|700px]]

Puis défiler vers le bas et trouvez la fonction '''"e911"''', et activez-la.

: [[File:e911_Reseller_4_FR.png|thumb|none|500px]]

1) Pour ajouter une IVR à votre client, ou pour aider votre client à en ajouter une. Allez sous le '''[Services]''' de la barre de navigation de gauche, puis sur '''[e911]'''
: [[File:e911_Add.png|thumb|none|400px]]

2) Dans la ligne où le DID souhaité, cliquez sur le bouton ACTIVÉ [ENABLED].
: [[File:e911_Add2.png|thumb|none|600px]]

3) Votre client doit '''lire''' et '''accepter les conditions générales'''.

4)Remplissez le formulaire et cliquez sur le bouton Valider [Validate].

[[category:Guides en français]]

Emergency Services

2025-11-14T20:05:27Z

RP: /* Emergency Services */

{| class="wikitable"
|+
|-
! Article en Français !! Artículo en Español
|-
| [https://wiki.voip.ms/article/Service_E911 Français] ||
[https://wiki.voip.ms/article/Servicios_de_Emergencia Español]
|}

== Article blog ==
[https://wiki.voip.ms/article/Enhanced_911_and_VoIP Enhanced 911 and VoIP]

== Emergency Services ==

For customers residing in <b>United States</b>, when you make an emergency call, VoIP.ms will attempt to automatically route your call through a third-party service provider to the Public Safety Answering Point ("PSAP") corresponding to your address of record on your account.

For customers residing in <b>Canada</b>, your emergency call will be directly sent to an emergency call center confirming your identity and location, and then immediately transferred to the local PSAP.

For customers residing <b>outside United States or Canada</b>, telecommunication regulators generally provide that all Customers who are using VoIP services as their primary residential or business telephone carrier must activate Emergency Services on at least one of their DIDs.

For emergency services address information to be passed to your local PSAP dispatcher, you must set your outbound CallerID value to the specific DID you are purchasing emergency service for.

Due to the various dependencies of VoIP networks, VoIP.ms cannot and does not guarantee your emergency call will go through. Many conditions such as loss of power, Internet access or connectivity and/or several other conditions may cause emergency services to be inoperable.

Emergency Services have a one-time fee <b>starting at $ 1.50 and a monthly fee starting at $ 1.50 per month per DID number activated per month</b>. For complete pricing, visit: https://www.voip.ms/residential/pricing

VoIP.ms has added an extension to its network where all VoIP.ms users may call to test their CallerID. At any time, you may test your outbound CallerID by dialing ’1-555-555-0911’ through VoIP.ms’ network and the system will playback your CallerID, then make a short pause, and play the test result. <b>As a reminder, if it is not correctly configured, your call will not go through</b>.

For all the details, refer to our Terms of Service: https://www.voip.ms/terms-of-service



You can activate this service at your Customer Portal >> DID Numbers >> Emergency Services.

'''Important Note:'''
<span style="color:red">'''Note that if your DID from Voip.ms is not subscribed to the Emergency Services, attempting to make a call to 911 will result in a
'''busy signal, since your callerID (which should be a valid DID or number) won't be in Voip.ms Database. This will result by not'''
'''reaching the desired destination. Also make sure to read the Terms and Condition on this page.'''</span>

''' <span style="color:red"> Update as of October 2021 </span> '''

Due to new regulations for Emergency Services, softphones on mobile devices will start using the native mobile dialer to place calls to the emergency services. '''It is possible that not all softphone devices will provide this functionality and it is highly suggested to come into close communication with the softphone provider for further questions.'''

[[File:Emergency Services option.png|thumb|none|600px]]

Use of Emergency Services costs a recovery setup fee of $ 1.50 on activation and a regulatory recovery fee of $ 1.50 per DID number activated per month. VoIP.MS does not make a cent on this charge, it is simply what must be paid to provide this service.

Please note you can enable this service only for Canadian or US numbers (including USA or Canadian toll free numbers).
You just have to click on the checkbox button in order to start the process to enable the service for this number.

[[File:Enable emergency services.png|thumb|none|600px]]

After this you can verify the Terms & conditions regarding of this service. Just please read them carefully and type "I agree" if you wish to enable the emergency services.

Click here to read Terms & conditions regarding this service. [[E911#VoIP.ms_911.2Fe911_Emergency_Service:_Terms_of_Service|Emergency services TOS]]

A new window will request some required information that will be used for the emergency services. Just fill the required fields and be sure to use the correct address. After this, click on the Validate button and if everything is fine, you will receive an email confirmation.

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

Once your emergency service is active, you have to set your DID number as your outgoing [[Caller ID]] number for your main account or [[Sub Accounts]] you are going to use to dial out.

You can set the [[Caller ID]] number for your main account at your Customer Portal >> Main Menu >> [[Account Settings]] >> General tab >> "[[Caller ID]] number" option.

If you are going to use a [[Sub Accounts|sub account]] to dial out, you can set the [[Caller ID]] for it inside your Customer Portal >> [[Sub Accounts|sub account]] >> Manage [[Sub Accounts|sub account]] >> Edit >> [[Caller ID]] number.

To test that your [[Caller ID]] is working properly, you can Dial 1-555-555-0911 from VoIP.ms Network. The system will playback your [[Caller ID]], then make a short pause, and play the test result. To make sure your device/switch is correctly configured for emergency services, you must ensure that your [[Caller ID]] number matches exactly the DID number that is activated for emergency services. That is the only way to identify you correctly. Otherwise your 911 call will not go through. Correctly formatted [[Caller ID]] consist of a 10 digits number, identical to your DID. For example, if your DID is 555.555.1234, your [[Caller ID]] number should be 5555551234.

If at any time you need to change your address, you just need to select the "Modify" icon (The one with the pencil and paper). After this you will be able to change your emergency services information and after the information has been approved, you will be confirmed by email and the emergency services will be updated.

[[File:E911Enable.png|thumb|none|600px]]

== Unverified Emergency Calls (E911) ==

We’ve introduced an important update to help keep you safe. From now on, 911 calls will connect from any number within the U.S. or Canada, even if E911 service isn’t configured.

This change ensures that you can reach emergency services in critical situations, no matter your setup.

Please note that a $75 fee applies per completed 911 call if E911 is not active on your number. We strongly recommend enabling E911 to avoid this fee and ensure your location is automatically shared with emergency responders.

== VoIP.ms 911/e911 Emergency Service: Terms of Service ==

EMERGENCY COMMUNICATIONS

The Company wants to make sure that Customers are aware of important differences in the way Emergency Services operate when using VoIP services when compared with traditional telephone service. Please find below what Customers need to keep in mind.  

'''Routing of Emergency Calls'''
For United States residents, when a Customer makes an emergency call, the Company will attempt to automatically route its call through a third-party service provider to the Public Safety Answering Point (“PSAP”) corresponding to the Customer’s address of record on its account. However, the delivery of the Customer’s physical location to its local PSAP is not guaranteed. It is possible that the Customer’s location will not be provided to the PSAP dispatcher On such occasions, it will be the Customer’s sole responsibility to give the dispatcher its name, location (or location of the emergency) and contact information to receive emergency service assistance. For Canadian residents, a Customer’s emergency call will be directly sent to an emergency call center confirming its identity and location, and then immediately transferred to the local PSAP.

'''Limitations Due to VoIP Networks'''
Due to the various dependencies of VoIP networks, the Company cannot and does not guarantee a Customer’s emergency call will go through. Many conditions such as loss of power, Internet access or connectivity and/or several other conditions may cause emergency services to be inoperable. The Company does not have control over those types of situations and therefore cannot be held liable of such inoperability. The Company will take commercially reasonable measures to prevent service outages within its network.  

'''Outbound CallerID'''
For emergency services address information to be passed to a Cusomter’s local PSAP dispatcher, the Customer’s outbound CallerID value must be set to the specific DID it is purchasing emergency service for. Therefore, by agreeing to these Terms, a Customer is deemed to have set the outbound CallerID number to the DID it has enabled emergency services for when making an outbound emergency call.  The Company has added an extension to its network where all Customers may call to test their CallerID value. At any time, a Customer may test its outbound CallerID value by dialing ‘1-555-555-0911’ through the Company’s network.  

'''Limitations on Emergency Services'''
CUSTOMERS UNDERSTAND THE LIMITATIONS OF THE COMPANY’S EMERGENCY SERVICES AND ASSUMES ALL LIABILITY AND RESPONSIBILITY, AND RELEASES THE COMPANY TO SUCH EXTENT, FOR THE USE OF EMERGENCY SERVICES,AND FURTHER AGREES TO HOLD THE COMPANY, ITS OFFICERS, DIRECTORS, EMPLOYEES AND AGENTS HARMLESS FOR ANY DAMAGE, WHETHER DIRECT OR INDIRECT THAT MAY RESULT FROM : (1) THE EMERGENCY SERVICES PROVIDED BY THE COMPANY (INCLUDING BUT NOT LIMITED TO SITUATIONS OF UNAVAILABILITY OF EMERGENCY SERVICES AS DESCRIBED IN THESE TERMS AND INCOMPLETE OR INCORRECT LOCATION INFORMATION PROVIDED BY THE CUSTOMER); (2) CUSTOMER’S FAILURE TO OBTAIN ACCESS TO CONVENTIONAL EMERGENCY SERVICE AS PART OF A TELEPHONE LINE SUBSCRIPTION FROM ANOTHER COMPANY UNDER SEPARATE AGREEMENT; (3) CUSTOMER’S FAILURE OR DELAY IN UTILIZING CONVENTIONAL EMERGENCY SERVICE. CUSTOMERS WHO RESELL THE SERVICES FURTHER AGREE THAT THEY ARE RESPONSIBLE FOR NOTIFYING, AND AGREE TO NOTIFY, THEIR CUSTOMERS, CONTRACTORS, AGENTS, EMPLOYEES, ASSOCIATES, SHAREHOLDERS, PARTNERS, AND ANY OTHER POTENTIAL USER OF THE COMPANY’S SERVICES OF THE NATURE AND LIMITATIONS OF THE EMERGENCY SERVICES.  IF A CUSTOMER IS NOT COMFORTABLE WITH THE LIMITATIONS OF EMERGENCY CALLS, THE CUSTOMER MUST CONSIDER AN ALTERNATE MEANS FOR ACCESSING TRADITIONAL EMERGENCY SERVICES, AS REGISTRATION TO EMERGENCY SERVICES IS MANDATORY IN MOST COUNTRIES. FURTHERMORE, THE CUSTOMER AGREES THAT THE COMPANY HAS NO LIABILITY IN RELATION TO THE QUALITY OF THE ADVICE AND SERVICES PROVIDED BY A PSAP.

==Pricing for Emergency Services==

{| class="wikitable"
|- style="font-weight:bold; vertical-align:bottom;"
! Country
! One-time fee
! Recurring Fee
|-
| style="vertical-align:bottom;" | Australia
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | Austria
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | Belgium
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | Bulgaria
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | Canada
| $ 1.50
| $ 1.50
|-
| style="vertical-align:bottom;" | Croatia
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | Cyprus
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | Czech Republic
| $ 4.50
| $ 2.50
|-
| style="vertical-align:bottom;" | Denmark
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | Estonia
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | Finland
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | France
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | Germany
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | Greece
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | Hungary
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | Ireland
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | Italy
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | Latvia
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | Lithuania
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | Netherlands
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | New Zealand
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | Norway
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | Poland
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | Portugal
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | Romania
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | Slovakia
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | Slovenia
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | Spain
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | Sweden
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | United Kingdom
| $ 1.50
| $ 2.50
|-
| style="vertical-align:bottom;" | United States
| $ 1.50
| $ 1.50
|}

=Using 911 using the Reseller Interface=

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

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

Go under the navigation bar on '''[Reseller]''' then click on '''[Manage Rates & Packages]'''
: [[File:e911_Reseller_1.png|thumb|none|300px]]

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

: [[File:e911_Reseller_2.png|thumb|none|700px]]

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

: [[File:e911_Reseller_3.png|thumb|none|700px]]

Then scroll down and find the feature "'''e911'''", and enable it.

: [[File:e911_Reseller_4.png|thumb|none|500px]]

1) To add a e911 entry for your client, or to help your client adding one. Go under the '''[Services]''' at the left navigation bar, then on '''[e911]'''
: [[File:e911_Add.png|thumb|none|400px]]

2) In the row where the wished DID, click on the [ENABLED] button.
: [[File:e911_Add2.png|thumb|none|600px]]

3) Your client '''must''' read and '''Agreed the Terms and Conditions'''.

4) Complete the form and click the '''[Validate]''' button.

[[category:guides]]

ATA Devices

2025-10-15T14:34:47Z

RP:

== Blog Articles ==

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

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

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

=Most Popular ATA Devices=
Not sure what to use? Here's our top used device across our network:
* [[ATA_Devices#Grandstream_HandyTone_802_-_HT802 | Grandstream HT801 and HT802]]
* [[ATA_Devices#Grandstream_HandyTone_802V2_-_HT802V2 | Grandstream HT802V2]]

__NOTOC__

== Devices and what is supported ==

{| class="wikitable sortable static-row-numbers sticky-header sort-under" style="margin-left:30px;"
|+ Devices and what is supported
|-
! data-sort-type=text scope="col" | Device Name
! scope="col" | T.38 Faxing
! scope="col" | SIP TLS
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#Atcom_AG188N Atcom AG188N]
| X
|
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#Auerswald_COMpact_5010 Auerswald COMpact 5010]
| X
|
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#Cisco_Linksys_PAP2 Cisco Linksys PAP2]
| X
|
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#Cisco_Linksys_PAP2T Cisco Linksys PAP2T]
| X
|
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#Cisco_SPA2100_Phone_Adapter Cisco SPA2100 Phone Adapter]
| X
|
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#Cisco_SPA2102_Phone_Adapter_with_Router Cisco SPA2102 Phone Adapter with Router]
| X
| X
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#Cisco_WRP400_and_WRP500 Cisco WRP400 and WRP500]
| X
| X
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#Grandstream_HandyTone_486 Grandstream HandyTone 486]
| X
|
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#Grandstream_HandyTone_502_-_HT502 Grandstream HandyTone 502 - HT502]
| X
| X
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#Grandstream_HandyTone_702_-_HT702 Grandstream HandyTone 702 - HT702]
| X
| X
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#Grandstream_HandyTone_802_-_HT802 Grandstream HandyTone 802 - HT802]
| X
| X
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#Grandstream_HandyTone_802V2_-_HT802V2 Grandstream HandyTone 802V2 - HT802V2]
| X
| X
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#Mediatrix_C7_and_Mediatrix_4100_Series Mediatrix C7 and Mediatrix 4100 Series]
| X
| X
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#Netgear_WGR615V Netgear WGR615V]
| X
|
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#OBi_100.2F110_.26_OBi_200 OBi 100/110 & OBi 200]
| X
| X
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#Polycom_OBi300 Polycom OBi300]
| X
| X
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#ReadyNet_AC1000MS_and_AC1300MS ReadyNet AC1000MS and AC1300MS]
| X
|
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#Telco_AC-211 Telco AC-211]
| X
|
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#TP-Link_TD-VG3631 TP-Link TD-VG3631]
| X
|
|-
|}

==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 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.

Both units were discontinued by the manufacturer in 2017.

[[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" />

====Grandstream HandyTone 802V2 - HT802V2====

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

'''Product:''' Grandstream HandyTone 802V2 - HT802V2

'''Company:''' Grandstream

'''Overview:''' The HT802V2 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 HT802V2 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_HT802v2|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 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]]
<div style="width:100%;overflow:hidden;clear:both"></div>

===ReadyNet AC1000MS and AC1300MS===

[[File:readynet-ac1000ms.jpg|300px|thumb|left|ReadyNet AC1000MS]]

'''Product:''' ReadyNet AC1000MS (two lines) or AC1300MS (one line)

'''Company:''' ReadyNet Solutions (Phonex Broadband Corporation)

'''Overview:''' The ReadyNet AC1000MS is a full-featured 1200 megabit-per-second dual-band Wi-Fi router with a built in two-line SIP ATA, eliminating the need for two separate boxes. Both lines may be configured independently.

[[ReadyNet AC1000MS and AC1300MS|See Configuration Details]]

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

===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]]

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

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

ATA Devices

2025-10-15T14:34:07Z

RP:

== Blog Articles ==

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

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

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

=Most Popular ATA Devices=
Not sure what to use? Here's our top used device across our network:
* [[ATA_Devices#Grandstream_HandyTone_802_-_HT802 | Grandstream HT801 and HT802]]
* [[ATA_Devices#Grandstream_HandyTone_802V2_-_HT802V2 | Grandstream HT802V2]]

__NOTOC__

== Devices and what is supported ==

{| class="wikitable sortable static-row-numbers sticky-header sort-under" style="margin-left:30px;"
|+ Devices and what is supported
|-
! data-sort-type=text scope="col" | Device Name
! scope="col" | T.38 Faxing
! scope="col" | SIP TLS
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#Atcom_AG188N Atcom AG188N]
| X
|
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#Auerswald_COMpact_5010 Auerswald COMpact 5010]
| X
|
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#Cisco_Linksys_PAP2 Cisco Linksys PAP2]
| X
|
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#Cisco_Linksys_PAP2T Cisco Linksys PAP2T]
| X
|
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#Cisco_SPA2100_Phone_Adapter Cisco SPA2100 Phone Adapter]
| X
|
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#Cisco_SPA2102_Phone_Adapter_with_Router Cisco SPA2102 Phone Adapter with Router]
| X
| X
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#Cisco_WRP400_and_WRP500 Cisco WRP400 and WRP500]
| X
| X
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#Grandstream_HandyTone_486 Grandstream HandyTone 486]
| X
|
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#Grandstream_HandyTone_502_-_HT502 Grandstream HandyTone 502 - HT502]
| X
| X
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#Grandstream_HandyTone_702_-_HT702 Grandstream HandyTone 702 - HT702]
| X
| X
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#Grandstream_HandyTone_802_-_HT802 Grandstream HandyTone 802 - HT802]
| X
| X
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#Grandstream_HandyTone_802_-_HT802V2 Grandstream HandyTone 802V2 - HT802V2]
| X
| X
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#Mediatrix_C7_and_Mediatrix_4100_Series Mediatrix C7 and Mediatrix 4100 Series]
| X
| X
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#Netgear_WGR615V Netgear WGR615V]
| X
|
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#OBi_100.2F110_.26_OBi_200 OBi 100/110 & OBi 200]
| X
| X
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#Polycom_OBi300 Polycom OBi300]
| X
| X
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#ReadyNet_AC1000MS_and_AC1300MS ReadyNet AC1000MS and AC1300MS]
| X
|
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#Telco_AC-211 Telco AC-211]
| X
|
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#TP-Link_TD-VG3631 TP-Link TD-VG3631]
| X
|
|-
|}

==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 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.

Both units were discontinued by the manufacturer in 2017.

[[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" />

====Grandstream HandyTone 802V2 - HT802V2====

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

'''Product:''' Grandstream HandyTone 802V2 - HT802V2

'''Company:''' Grandstream

'''Overview:''' The HT802V2 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 HT802V2 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_HT802v2|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 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]]
<div style="width:100%;overflow:hidden;clear:both"></div>

===ReadyNet AC1000MS and AC1300MS===

[[File:readynet-ac1000ms.jpg|300px|thumb|left|ReadyNet AC1000MS]]

'''Product:''' ReadyNet AC1000MS (two lines) or AC1300MS (one line)

'''Company:''' ReadyNet Solutions (Phonex Broadband Corporation)

'''Overview:''' The ReadyNet AC1000MS is a full-featured 1200 megabit-per-second dual-band Wi-Fi router with a built in two-line SIP ATA, eliminating the need for two separate boxes. Both lines may be configured independently.

[[ReadyNet AC1000MS and AC1300MS|See Configuration Details]]

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

===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]]

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

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

ATA Devices

2025-10-15T14:33:16Z

RP: /* Most Popular ATA Devices */

== Blog Articles ==

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

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

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

=Most Popular ATA Devices=
Not sure what to use? Here's our top used device across our network:
* [[ATA_Devices#Grandstream_HandyTone_802_-_HT802 | Grandstream HT801 and HT802]]
* [[ATA_Devices#Grandstream_HandyTone_802_-_HT802V2 | Grandstream HT802V2]]

__NOTOC__

== Devices and what is supported ==

{| class="wikitable sortable static-row-numbers sticky-header sort-under" style="margin-left:30px;"
|+ Devices and what is supported
|-
! data-sort-type=text scope="col" | Device Name
! scope="col" | T.38 Faxing
! scope="col" | SIP TLS
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#Atcom_AG188N Atcom AG188N]
| X
|
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#Auerswald_COMpact_5010 Auerswald COMpact 5010]
| X
|
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#Cisco_Linksys_PAP2 Cisco Linksys PAP2]
| X
|
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#Cisco_Linksys_PAP2T Cisco Linksys PAP2T]
| X
|
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#Cisco_SPA2100_Phone_Adapter Cisco SPA2100 Phone Adapter]
| X
|
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#Cisco_SPA2102_Phone_Adapter_with_Router Cisco SPA2102 Phone Adapter with Router]
| X
| X
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#Cisco_WRP400_and_WRP500 Cisco WRP400 and WRP500]
| X
| X
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#Grandstream_HandyTone_486 Grandstream HandyTone 486]
| X
|
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#Grandstream_HandyTone_502_-_HT502 Grandstream HandyTone 502 - HT502]
| X
| X
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#Grandstream_HandyTone_702_-_HT702 Grandstream HandyTone 702 - HT702]
| X
| X
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#Grandstream_HandyTone_802_-_HT802 Grandstream HandyTone 802 - HT802]
| X
| X
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#Grandstream_HandyTone_802_-_HT802V2 Grandstream HandyTone 802V2 - HT802V2]
| X
| X
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#Mediatrix_C7_and_Mediatrix_4100_Series Mediatrix C7 and Mediatrix 4100 Series]
| X
| X
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#Netgear_WGR615V Netgear WGR615V]
| X
|
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#OBi_100.2F110_.26_OBi_200 OBi 100/110 & OBi 200]
| X
| X
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#Polycom_OBi300 Polycom OBi300]
| X
| X
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#ReadyNet_AC1000MS_and_AC1300MS ReadyNet AC1000MS and AC1300MS]
| X
|
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#Telco_AC-211 Telco AC-211]
| X
|
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#TP-Link_TD-VG3631 TP-Link TD-VG3631]
| X
|
|-
|}

==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 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.

Both units were discontinued by the manufacturer in 2017.

[[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" />

====Grandstream HandyTone 802V2 - HT802V2====

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

'''Product:''' Grandstream HandyTone 802V2 - HT802V2

'''Company:''' Grandstream

'''Overview:''' The HT802V2 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 HT802V2 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_HT802v2|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 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]]
<div style="width:100%;overflow:hidden;clear:both"></div>

===ReadyNet AC1000MS and AC1300MS===

[[File:readynet-ac1000ms.jpg|300px|thumb|left|ReadyNet AC1000MS]]

'''Product:''' ReadyNet AC1000MS (two lines) or AC1300MS (one line)

'''Company:''' ReadyNet Solutions (Phonex Broadband Corporation)

'''Overview:''' The ReadyNet AC1000MS is a full-featured 1200 megabit-per-second dual-band Wi-Fi router with a built in two-line SIP ATA, eliminating the need for two separate boxes. Both lines may be configured independently.

[[ReadyNet AC1000MS and AC1300MS|See Configuration Details]]

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

===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]]

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

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

ATA Devices

2025-10-15T14:31:38Z

RP: /* Most Popular ATA Devices */

== Blog Articles ==

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

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

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

=Most Popular ATA Devices=
Not sure what to use? Here's our top used device across our network:
* [[ATA_Devices#Grandstream_HandyTone_802_-_HT802 | Grandstream HT801 and HT802]]
* [[ATA_Devices#Grandstream_HandyTone_802_-_HT802v2 | Grandstream HT802V2]]

__NOTOC__

== Devices and what is supported ==

{| class="wikitable sortable static-row-numbers sticky-header sort-under" style="margin-left:30px;"
|+ Devices and what is supported
|-
! data-sort-type=text scope="col" | Device Name
! scope="col" | T.38 Faxing
! scope="col" | SIP TLS
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#Atcom_AG188N Atcom AG188N]
| X
|
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#Auerswald_COMpact_5010 Auerswald COMpact 5010]
| X
|
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#Cisco_Linksys_PAP2 Cisco Linksys PAP2]
| X
|
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#Cisco_Linksys_PAP2T Cisco Linksys PAP2T]
| X
|
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#Cisco_SPA2100_Phone_Adapter Cisco SPA2100 Phone Adapter]
| X
|
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#Cisco_SPA2102_Phone_Adapter_with_Router Cisco SPA2102 Phone Adapter with Router]
| X
| X
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#Cisco_WRP400_and_WRP500 Cisco WRP400 and WRP500]
| X
| X
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#Grandstream_HandyTone_486 Grandstream HandyTone 486]
| X
|
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#Grandstream_HandyTone_502_-_HT502 Grandstream HandyTone 502 - HT502]
| X
| X
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#Grandstream_HandyTone_702_-_HT702 Grandstream HandyTone 702 - HT702]
| X
| X
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#Grandstream_HandyTone_802_-_HT802 Grandstream HandyTone 802 - HT802]
| X
| X
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#Grandstream_HandyTone_802_-_HT802V2 Grandstream HandyTone 802V2 - HT802V2]
| X
| X
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#Mediatrix_C7_and_Mediatrix_4100_Series Mediatrix C7 and Mediatrix 4100 Series]
| X
| X
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#Netgear_WGR615V Netgear WGR615V]
| X
|
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#OBi_100.2F110_.26_OBi_200 OBi 100/110 & OBi 200]
| X
| X
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#Polycom_OBi300 Polycom OBi300]
| X
| X
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#ReadyNet_AC1000MS_and_AC1300MS ReadyNet AC1000MS and AC1300MS]
| X
|
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#Telco_AC-211 Telco AC-211]
| X
|
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#TP-Link_TD-VG3631 TP-Link TD-VG3631]
| X
|
|-
|}

==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 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.

Both units were discontinued by the manufacturer in 2017.

[[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" />

====Grandstream HandyTone 802V2 - HT802V2====

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

'''Product:''' Grandstream HandyTone 802V2 - HT802V2

'''Company:''' Grandstream

'''Overview:''' The HT802V2 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 HT802V2 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_HT802v2|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 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]]
<div style="width:100%;overflow:hidden;clear:both"></div>

===ReadyNet AC1000MS and AC1300MS===

[[File:readynet-ac1000ms.jpg|300px|thumb|left|ReadyNet AC1000MS]]

'''Product:''' ReadyNet AC1000MS (two lines) or AC1300MS (one line)

'''Company:''' ReadyNet Solutions (Phonex Broadband Corporation)

'''Overview:''' The ReadyNet AC1000MS is a full-featured 1200 megabit-per-second dual-band Wi-Fi router with a built in two-line SIP ATA, eliminating the need for two separate boxes. Both lines may be configured independently.

[[ReadyNet AC1000MS and AC1300MS|See Configuration Details]]

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

===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]]

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

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

ATA Devices

2025-10-15T14:28:39Z

RP:

== Blog Articles ==

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

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

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

=Most Popular ATA Devices=
Not sure what to use? Here's our top used device across our network:
* [[ATA_Devices#Grandstream_HandyTone_802_-_HT802 | Grandstream HT801 and HT802]]
* [[ATA_Devices#Grandstream_HT802v2 | Grandstream HT802V2]]

__NOTOC__

== Devices and what is supported ==

{| class="wikitable sortable static-row-numbers sticky-header sort-under" style="margin-left:30px;"
|+ Devices and what is supported
|-
! data-sort-type=text scope="col" | Device Name
! scope="col" | T.38 Faxing
! scope="col" | SIP TLS
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#Atcom_AG188N Atcom AG188N]
| X
|
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#Auerswald_COMpact_5010 Auerswald COMpact 5010]
| X
|
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#Cisco_Linksys_PAP2 Cisco Linksys PAP2]
| X
|
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#Cisco_Linksys_PAP2T Cisco Linksys PAP2T]
| X
|
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#Cisco_SPA2100_Phone_Adapter Cisco SPA2100 Phone Adapter]
| X
|
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#Cisco_SPA2102_Phone_Adapter_with_Router Cisco SPA2102 Phone Adapter with Router]
| X
| X
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#Cisco_WRP400_and_WRP500 Cisco WRP400 and WRP500]
| X
| X
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#Grandstream_HandyTone_486 Grandstream HandyTone 486]
| X
|
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#Grandstream_HandyTone_502_-_HT502 Grandstream HandyTone 502 - HT502]
| X
| X
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#Grandstream_HandyTone_702_-_HT702 Grandstream HandyTone 702 - HT702]
| X
| X
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#Grandstream_HandyTone_802_-_HT802 Grandstream HandyTone 802 - HT802]
| X
| X
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#Grandstream_HandyTone_802_-_HT802V2 Grandstream HandyTone 802V2 - HT802V2]
| X
| X
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#Mediatrix_C7_and_Mediatrix_4100_Series Mediatrix C7 and Mediatrix 4100 Series]
| X
| X
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#Netgear_WGR615V Netgear WGR615V]
| X
|
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#OBi_100.2F110_.26_OBi_200 OBi 100/110 & OBi 200]
| X
| X
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#Polycom_OBi300 Polycom OBi300]
| X
| X
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#ReadyNet_AC1000MS_and_AC1300MS ReadyNet AC1000MS and AC1300MS]
| X
|
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#Telco_AC-211 Telco AC-211]
| X
|
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices#TP-Link_TD-VG3631 TP-Link TD-VG3631]
| X
|
|-
|}

==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 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.

Both units were discontinued by the manufacturer in 2017.

[[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" />

====Grandstream HandyTone 802V2 - HT802V2====

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

'''Product:''' Grandstream HandyTone 802V2 - HT802V2

'''Company:''' Grandstream

'''Overview:''' The HT802V2 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 HT802V2 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_HT802v2|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 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]]
<div style="width:100%;overflow:hidden;clear:both"></div>

===ReadyNet AC1000MS and AC1300MS===

[[File:readynet-ac1000ms.jpg|300px|thumb|left|ReadyNet AC1000MS]]

'''Product:''' ReadyNet AC1000MS (two lines) or AC1300MS (one line)

'''Company:''' ReadyNet Solutions (Phonex Broadband Corporation)

'''Overview:''' The ReadyNet AC1000MS is a full-featured 1200 megabit-per-second dual-band Wi-Fi router with a built in two-line SIP ATA, eliminating the need for two separate boxes. Both lines may be configured independently.

[[ReadyNet AC1000MS and AC1300MS|See Configuration Details]]

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

===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]]

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

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

ATA Devices

2025-10-01T21:05:16Z

RP:

== Blog Articles ==

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

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

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

=Most Popular ATA Devices=
Not sure what to use? Here's our top used device across our network:
* [[ATA_Devices#Grandstream_HandyTone_802_-_HT802 | Grandstream HT801 and HT802]]
* [[ATA_Devices#Grandstream_HT802v2 | Grandstream HT802V2]]

__NOTOC__

== Devices and what is supported ==

{| class="wikitable sortable static-row-numbers sticky-header sort-under" style="margin-left:30px;"
|+ Devices and what is supported
|-
! data-sort-type=text scope="col" | Device Name
! scope="col" | T.38 Faxing
! scope="col" | SIP TLS
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices&oldid=18187#Atcom_AG188N Atcom AG188N]
| X
|
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices&oldid=18187#Auerswald_COMpact_5010 Auerswald COMpact 5010]
| X
|
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices&oldid=18187#Cisco_Linksys_PAP2 Cisco Linksys PAP2]
| X
|
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices&oldid=18187#Cisco_Linksys_PAP2T Cisco Linksys PAP2T]
| X
|
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices&oldid=18187#Cisco_SPA2100_Phone_Adapter Cisco SPA2100 Phone Adapter]
| X
|
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices&oldid=18187#Cisco_SPA2102_Phone_Adapter_with_Router Cisco SPA2102 Phone Adapter with Router]
| X
| X
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices&oldid=18187#Cisco_WRP400_and_WRP500 Cisco WRP400 and WRP500]
| X
| X
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices&oldid=18187#Grandstream_HandyTone_486 Grandstream HandyTone 486]
| X
|
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices&oldid=18187#Grandstream_HandyTone_502_-_HT502 Grandstream HandyTone 502 - HT502]
| X
| X
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices&oldid=18187#Grandstream_HandyTone_702_-_HT702 Grandstream HandyTone 702 - HT702]
| X
| X
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices&oldid=18187#Grandstream_HandyTone_802_-_HT802 Grandstream HandyTone 802 - HT802]
| X
| X
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices&oldid=18187#Grandstream_HandyTone_802_-_HT802V2 Grandstream HandyTone 802V2 - HT802V2]
| X
| X
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices&oldid=18187#Mediatrix_C7_and_Mediatrix_4100_Series Mediatrix C7 and Mediatrix 4100 Series]
| X
| X
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices&oldid=18187#Netgear_WGR615V Netgear WGR615V]
| X
|
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices&oldid=18187#OBi_100.2F110_.26_OBi_200 OBi 100/110 & OBi 200]
| X
| X
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices&oldid=18187#Polycom_OBi300 Polycom OBi300]
| X
| X
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices&oldid=18187#ReadyNet_AC1000MS_and_AC1300MS ReadyNet AC1000MS and AC1300MS]
| X
|
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices&oldid=18187#Telco_AC-211 Telco AC-211]
| X
|
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices&oldid=18187#TP-Link_TD-VG3631 TP-Link TD-VG3631]
| X
|
|-
|}

==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 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.

Both units were discontinued by the manufacturer in 2017.

[[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" />

====Grandstream HandyTone 802V2 - HT802V2====

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

'''Product:''' Grandstream HandyTone 802V2 - HT802V2

'''Company:''' Grandstream

'''Overview:''' The HT802V2 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 HT802V2 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_HT802v2|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 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]]
<div style="width:100%;overflow:hidden;clear:both"></div>

===ReadyNet AC1000MS and AC1300MS===

[[File:readynet-ac1000ms.jpg|300px|thumb|left|ReadyNet AC1000MS]]

'''Product:''' ReadyNet AC1000MS (two lines) or AC1300MS (one line)

'''Company:''' ReadyNet Solutions (Phonex Broadband Corporation)

'''Overview:''' The ReadyNet AC1000MS is a full-featured 1200 megabit-per-second dual-band Wi-Fi router with a built in two-line SIP ATA, eliminating the need for two separate boxes. Both lines may be configured independently.

[[ReadyNet AC1000MS and AC1300MS|See Configuration Details]]

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

===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]]

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

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

ATA Devices

2025-10-01T21:01:46Z

RP: /* Grandstream HandyTone 802 - HT802 */

== Blog Articles ==

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

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

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

=Most Popular ATA Devices=
Not sure what to use? Here's our top used device across our network:
* [[ATA_Devices#Grandstream_HandyTone_802_-_HT802 | Grandstream HT801 and HT802]]

__NOTOC__

== Devices and what is supported ==

{| class="wikitable sortable static-row-numbers sticky-header sort-under" style="margin-left:30px;"
|+ Devices and what is supported
|-
! data-sort-type=text scope="col" | Device Name
! scope="col" | T.38 Faxing
! scope="col" | SIP TLS
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices&oldid=18187#Atcom_AG188N Atcom AG188N]
| X
|
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices&oldid=18187#Auerswald_COMpact_5010 Auerswald COMpact 5010]
| X
|
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices&oldid=18187#Cisco_Linksys_PAP2 Cisco Linksys PAP2]
| X
|
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices&oldid=18187#Cisco_Linksys_PAP2T Cisco Linksys PAP2T]
| X
|
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices&oldid=18187#Cisco_SPA2100_Phone_Adapter Cisco SPA2100 Phone Adapter]
| X
|
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices&oldid=18187#Cisco_SPA2102_Phone_Adapter_with_Router Cisco SPA2102 Phone Adapter with Router]
| X
| X
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices&oldid=18187#Cisco_WRP400_and_WRP500 Cisco WRP400 and WRP500]
| X
| X
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices&oldid=18187#Grandstream_HandyTone_486 Grandstream HandyTone 486]
| X
|
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices&oldid=18187#Grandstream_HandyTone_502_-_HT502 Grandstream HandyTone 502 - HT502]
| X
| X
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices&oldid=18187#Grandstream_HandyTone_702_-_HT702 Grandstream HandyTone 702 - HT702]
| X
| X
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices&oldid=18187#Grandstream_HandyTone_802_-_HT802 Grandstream HandyTone 802 - HT802]
| X
| X
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices&oldid=18187#Mediatrix_C7_and_Mediatrix_4100_Series Mediatrix C7 and Mediatrix 4100 Series]
| X
| X
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices&oldid=18187#Netgear_WGR615V Netgear WGR615V]
| X
|
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices&oldid=18187#OBi_100.2F110_.26_OBi_200 OBi 100/110 & OBi 200]
| X
| X
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices&oldid=18187#Polycom_OBi300 Polycom OBi300]
| X
| X
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices&oldid=18187#ReadyNet_AC1000MS_and_AC1300MS ReadyNet AC1000MS and AC1300MS]
| X
|
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices&oldid=18187#Telco_AC-211 Telco AC-211]
| X
|
|-
| [https://wiki.voip.ms/w/index.php?title=ATA_Devices&oldid=18187#TP-Link_TD-VG3631 TP-Link TD-VG3631]
| X
|
|-
|}

==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 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.

Both units were discontinued by the manufacturer in 2017.

[[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" />

====Grandstream HandyTone 802V2 - HT802V2====

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

'''Product:''' Grandstream HandyTone 802V2 - HT802V2

'''Company:''' Grandstream

'''Overview:''' The HT802V2 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 HT802V2 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_HT802v2|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 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]]
<div style="width:100%;overflow:hidden;clear:both"></div>

===ReadyNet AC1000MS and AC1300MS===

[[File:readynet-ac1000ms.jpg|300px|thumb|left|ReadyNet AC1000MS]]

'''Product:''' ReadyNet AC1000MS (two lines) or AC1300MS (one line)

'''Company:''' ReadyNet Solutions (Phonex Broadband Corporation)

'''Overview:''' The ReadyNet AC1000MS is a full-featured 1200 megabit-per-second dual-band Wi-Fi router with a built in two-line SIP ATA, eliminating the need for two separate boxes. Both lines may be configured independently.

[[ReadyNet AC1000MS and AC1300MS|See Configuration Details]]

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

===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]]

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

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

MediaWiki:Sidebar

2025-09-29T18:46:52Z

RP:

* 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
** ATA_Devices|ATA Devices
** IP_Phones|IP Phones
** VoIP.ms_Teams_Connector|Microsoft Teams
** PBXs|PBXs
** Session Border Controllers|SBCs
** softphones|Softphones
** Other_Devices|Other Devices
** Other_Services|Other Services

* 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 Parking | Call Parking
** Call quality issues|Call quality issues
** Call Recordings|Call Recordings
** Call Transcription|Call Transcription
** 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
** Custom Music on Hold|Custom Music on Hold
** 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
** Emergency_Services|Emergency Services
** Extra_Services_Costs|Extra Services Costs
** False_Answer_Supervision_FAS|FAS (False Answer Supervision)
** Finances|Finances
** Firewall|Firewall
** General_Security|General Security
** Finances#Generate_Invoice | Generate invoice
** International_Calls|International Calls
** Manage DID|Manage DID
** Multi-Tenant_Locations|Multi-Tenant Locations
** 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
** Sequence|Sequence
** Service Cost|Service Cost
** SIP Requests|SIP Requests
** SIP_Responses|SIP Responses
** SIP URI|SIP URI
** Smartphone|Smartphone
** SMPP|SMPP
** SMS-MMS | SMS-MMS
** Sub Accounts|Sub Accounts
** Time Conditions|Time Conditions
** Transaction_history|Transaction history
** Troubleshooting Outgoing Calls|Troubleshooting Outgoing Calls
** Toll-free number|Toll-free number
** Two-step Verification|Two-step Verification
** TOTP Authentication|TOTP Authentication
** Value vs Premium|Value vs Premium
** Virtual Fax|Virtual Fax
** Voicemail|Voicemail
** Whitelabeling_your_SMS/MMS_and_fax_services|SMS/MMS & Fax Whitelabel

*Guides (Français)
** Bienvenue|Bienvenue
** Mise en Marche|Mise en Marche
** Ajouter_un_Article| Ajouter un Article
**Appels internationaux|Appels internationaux
** Audioconférence|Audioconférence
**Authentification TOTP|Authentification TOTP
** 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
**Composition par impulsión|Composition par impulsion
** Conditions Temporelles | Conditions Temporelles
**Coût des appels|Coût des appels
** Coût des Services Rendus|Coût des Services Rendus
** Cryptage des appels - TLS/SRTP|Cryptage des appels - TLS/SRTP
** Détails des appels|Détails des appels
** Accès direct en entrée au système - DISA | DISA
** Effacer un Numéro DID|Effacer un Numéro DID
**Emplacement_à_site_multiple | Emplacement à site multiple
** Enregistrements | Enregistrements
** Enregistrements d'appels|Enregistrements d'appels
** File d'attente | File d'attente
** Identification de l'appelant | Identification de l'appelant
** Finances_Fr|Finances
** Finances_Fr#G.C3.A9n.C3.A9rer_une_facture|Générer une facture
** Fonction de Rappel | Fonction de Rappel
** Garde d'appels (Call Parking) | Garde d'appels
** Gérer les numéros DID|Gérer les numéros DID
** Groupe de Sonnerie|Groupe de Sonnerie
**Historique des transactions|Historique des transactions
** ID de l'appelant | ID de l'appelant
** Messagerie vocale | Messagerie vocale
**Musique d'attente personnalisée|Musique d'attente personnalisée
**Numéros sans frais|Numéros sans frais
** Paramètres du compte|Paramètres du compte
**Pare-feu|Pare-feu
**Personnalisation_en_marque_blanche_de_vos_services_SMS/MMS_et_fax| Personnalisation SMS/MMS et fax
**Plan de composition pour les ATA Linksys|Plan de composition pour les ATA Linksys
**Problème de qualité sonore|Problème de qualité sonore
** Programme de référencement|Programme de référencement
** Problèmes_de_Connexion|Problèmes de Connexion
** Problème d'enregistrement|Problème d'enregistrement
**Questions fréquentes sur la transférabilité|Questions fréquentes sur la transférabilité
** Questions Les Plus Fréquentes | Questions Les Plus Fréquentes
** Recherche d’Appel | Recherche d'Appel
**Règles de composition | Règles de composition
**Règles et motifs de composition |Règles et motifs de composition
** Renvoi d'appel | Renvoi d'appel
** Répertoire téléphonique | Répertoire téléphonique
** Réceptionniste virtuelle IVR | Réceptionniste virtuelle IVR
**Réponses SIP | Réponses SIP
**Requêtes SIP | Requêtes SIP
** Revendeur_Guide_Elementaire|Revendeur Guide Elementaire
**Sécurité Générale | Sécurité Générale
**Sécurité PBX | Sécurité PBX
** Sequence Fr|Sequence Fr
** Service_E911|Service E911
** SIP_URI_FR|SIP URI
**SMPP (FR) |SMPP (FR)
** SMS-MMS-FR | SMS-MMS
**Solutions de problèmes des appels sortants | Solutions de problèmes des appels sortants
** Solutions de problèmes d'un DID|Solutions de problèmes d'un DID
** Sous Comptes|Sous Comptes
**Statut d'enregistrement sur le bureau | Statut d'enregistrement sur le bureau
**Supervision des fausses réponses | Supervision des fausses réponses
** Télécopieur virtuel | Télécopieur virtuel
**Téléphone intelligent | Téléphone intelligent
**Transcription d'appels|Transcription d'appels
** Transférabilité des DID | Transférabilité des DID
**Usurpation du numéro d'identification de l'appelant | Usurpation du numéro d'identification de l'appelant
** Value vs Premium (Français)|Value vs Premium
**Vérification en deux étapes | Vérification en deux étapes

* Guías (Español)
** Bienvenido|Bienvenido
** Primeros Pasos|Primeros Pasos
** Agregar artículos|Agregar artículos
** Audioconferencia|Audioconferencia
** Preguntas_frecuentes_LNP|Preguntas frecuentes LNP
** Acceso directo al sistema interno (DISA)|Acceso directo al sistema interno
** Autentificación TOTP|Autentificación TOTP
** Buzón de voz (Voicemail)|Buzón de voz
**Caceria de llamadas|Caceria de llamadas
** Calculando mis gastos|Calculando mis gastos
** Cambiar Plan de DID (Changing DID Billing Plan)|Cambiar Plan de DID
**Cancelar/Borrar un DID |Cancelar un DID
** Cancelar/Borrar_un_DID| Cancelar/Borrar un DID
** Códigos de Marcación|Códigos de Marcación
** Comprar un número DID|Comprar un número DID
** 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
**Cortafuego|Cortafuego
** Costo del servicio|Costo del servicio
** Costo por llamada|Costo por llamada
** 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
** Servicio_E911|Servicio E911
** Elegir servidor|Elegir servidor
** Eliminar un DID|Eliminar un DID
** Encriptado de llamadas- TLS/SRTP|Encriptado de llamadas- TLS/SRTP
** Estatus_de_registro_en_escritorio|Estatus de registro en escritorio
** Fax Virtual|Fax Virtual
** Filtro de Llamadas (CallerID Filtering)|Filtro de Llamadas
** Finanzas|Finanzas
** Finanzas#Generar_Factura | Generar una Factura
** FAS_(supervisión_de_respuesta_falsa)|FAS (supervisión de respuesta falsa)
** Guía Básica de Reseller|Guía Básica de Reseller
** Grabaciones (Recordings)|Grabaciones
** Grabaciones de llamadas|Grabaciones de llamadas
** Grupo de timbre / Ring Groups|Grupo de timbre
** Historial de transacciones|Historial de transacciones
** Llamadas en Cola (Calling Queues)|Llamadas en Cola
** Llamadas internacionales|Llamadas internacionales
**Marcacion por pulsos | Marcacion por pulsos
**Música personalizada en espera|Música personalizada en espera
** Numero Identificador (Caller ID)|Numero Identificador
** Números de Lada sin costo|Números de Lada sin costo
** Parqueo de llamadas | Parqueo de llamadas
**Personalizacion_en_marca_blanca_de_sus_servicios_SMS/MMS_y_fax|Personalizacion SMS/MMS y fax
** 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
** Programa de referencia|Programa de referencia
** Recepcionista_Digital|Recepcionista_Digital (IVR)
** Registro de Llamadas (CDR)|Registro de Llamadas
** Reglas de marcación|Reglas de marcación
** Reglas_y_Patrones_de_Marcado|Reglas y Patrones de Marcado
** Regreso de LLamada (Callback)|Regreso de LLamada
** Respuestas_SIP|Respuestas SIP
** Secuencia|Secuencia
** Seguridad en PBX|Seguridad en PBX
**Seguridad General | Seguridad General
** SIP_URI_español|SIP URI(Español)
**SMPP (ES) | SMPP (ES)
**SMS-MMS-ES | SMS-MMS
**Solicitudes SIP | Solicitudes SIP
** Solución de problemas de llamadas salientes|Solución de problemas de llamadas salientes
**Solución de problemas de un DID | Solución de problemas de un DID
** Sub Cuentas (Sub Accounts)|Sub Cuentas
** Suplantación de identidad (CallerID spoofing)|Suplantación de identidad
**Transcripción de llamadas|Transcripción de llamadas
** Teléfono inteligente|Teléfono inteligente
** Value vs. Premium|Value vs. Premium
** Verificación de dos pasos|Verificación de dos pasos
**Ubicaciones_para_múltiples_inquilinos | Ubicaciones para múltiples inquilinos

* SEARCH
* TOOLBOX
* LANGUAGES

Emplacement à site multiple

2025-09-25T22:44:18Z

RP: /* SIP URI */

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

__TOC__

=Qu’est-ce qu’un emplacement à site multiple=

Cette fonctionnalité vous permet d’organiser les extensions par emplacement plutôt que par compte, ce qui rend possible la réutilisation des mêmes numéros d’extension sur différents sites sans conflit.

'''Fonctionnement:'''
* '''Emplacements:''' Chaque site ou bureau fonctionne comme son propre espace de travail.
* '''Sous-comptes:''' Les utilisateurs au sein d’un emplacement.
* '''Extensions:''' Chaque sous-compte peut avoir des extensions (ex. : 101, 102, 103). Les mêmes numéros peuvent également exister dans d’autres emplacements de manière indépendante.

'''Avantages:'''
* Garde l’organisation multi-sites claire et facile à gérer.
* Élimine les conflits de numéros d’extension entre sites.
* Simplifie l’attribution, le suivi et l’identification des extensions par emplacement.

En résumé, l’emplacement à site multiple avec extensions basées sur l’emplacement permet une organisation claire, une gestion simplifiée et la réutilisation sécurisée des numéros d’extension à travers plusieurs sites.

=Configuration des emplacements à site multiple=

1. Pour créer votre emplacement à site multiple, allez dans le menu Sous-comptes et sélectionnez Emplacements à site multiple pour commencer la configuration.

[[File:MultitenantFR1.png|thumb|none|500px]]

2. Cliquez sur Ajouter un emplacement pour créer un nouvel emplacement.

[[File:MultitenantFR2.png|thumb|none|500px]]

3. Renseignez les informations requises :

* Description de l’emplacement: Entrez un nom descriptif pour l’emplacement (ex. : Bureau de Floride, 2e bâtiment, Cottage).
* Rechercher un sous-compte / Numéro d’extension: Sélectionnez le sous-compte à inclure dans le groupe, puis attribuez le numéro d’extension (ex. : 1, 100, 101). Répétez cette étape pour chaque sous-compte devant faire partie de l’emplacement à site multiple.

[[File:MultitenantFR3.png|thumb|none|500px]]

4. Après avoir ajouté tous les sous-comptes et leurs extensions, cliquez sur Créer pour finaliser la configuration de l’emplacement à site multiple.

Vos sous-comptes peuvent désormais se joindre facilement en utilisant les extensions internes attribuées. Vous pouvez également créer d’autres groupes pour attribuer des extensions internes à des sous-ensembles spécifiques de sous-comptes, selon vos besoins.

=Paramètres supplémentaires=

==Extension accessible par tous==

Si vous souhaitez que certains sous-comptes soient accessibles par tous (au lieu d’appartenir à un groupe spécifique), ou si vous voulez ajuster leurs paramètres d’emplacement à site multiple :

:'''1.''' Allez dans le menu Sous-comptes et sélectionnez Gérer les sous-comptes.
:'''2.''' Modifiez le sous-compte que vous souhaitez consulter.
:'''3.''' Faites défiler la page jusqu’en bas pour trouver l’option Extension interne.

À cet endroit, vous pouvez changer le groupe attribué au sous-compte ou le rendre accessible par tous les utilisateurs de votre compte.

[[File:MultitenantFR4.png|thumb|none|500px]]

==SIP URI==

Pour appeler un sous-compte qui n’est pas dans un groupe spécifique et qui est accessible par tous, utilisez :{accountID}{extension}@{VoIP.ms server} '''Exemple: 123456101@montreal1.voip.ms'''

* '''123456:''' Votre identifiant principal de compte
* '''101:''' L’extension du sous-compte
* '''@montreal1.voip.ms:''' Le serveur où le sous-compte est actuellement enregistré

Pour les sous-comptes faisant partie d’un emplacement à site multiple, le format inclut l’ID de l’emplacement :
{accountID}#{locationID}#{extension}@{VoIP.ms server} '''Exemple: 123456#33#1@montreal1.voip.ms'''

* '''123456:''' Votre identifiant principal de compte
* '''#33:''' L’identifiant unique de l’emplacement
* '''#1:''' L’extension du sous-compte
* '''@montreal1.voip.ms:''' Le serveur où le sous-compte est actuellement enregistré

Pour trouver l’identifiant unique de l’emplacement, il suffit d’aller dans Sous-compte, Emplacements multi-locataires. Vous verrez alors le Numéro d’emplacement à côté de vos entrées actuelles. C’est l’emplacement que vous devrez utiliser dans votre chaîne SIP URI.

[[File:MultiTenantSIPURIFR1.png|thumb|none|500px]]

Vous pouvez également retrouver cette information en allant dans Sous-comptes, Gérer les sous-comptes et en éditant votre sous-compte tout en bas de la page.

[[File:MultiTenantSIPURIFR2.png|thumb|none|500px]]

File:MultiTenantSIPURIFR2.png

2025-09-25T22:44:02Z

RP:

Ubicaciones para múltiples inquilinos

2025-09-25T22:43:38Z

RP: /* SIP URIs */

{| class="wikitable"
|+
|-
! Article in English !! Article en Français
|-
| [https://wiki.voip.ms/article/Multi-Tenant_Locations English] || [https://wiki.voip.ms/article/Emplacement_%C3%A0_site_multiple Français]
|}

__TOC__

=¿Qué son las Ubicaciones para múltiples inquilinos?=

Esta función te permite organizar las extensiones por ubicación en lugar de por cuenta, lo que hace posible reutilizar los mismos números de extensión en diferentes sitios sin generar conflictos.

'''Cómo funciona:'''
*'''Ubicaciones:''' Cada sitio u oficina funciona como su propio espacio de trabajo.
*'''Subcuentas:''' Usuarios dentro de una ubicación.
*'''Extensiones:''' Cada subcuenta puede tener extensiones (por ejemplo: 101, 102, 103). Los mismos números también pueden existir en otras ubicaciones de manera independiente.

'''Beneficios:'''
* Mantiene las configuraciones de múltiples oficinas estructuradas y fáciles de gestionar.
* Elimina los conflictos de números de extensión entre sitios.
* Simplifica la asignación, el seguimiento y la identificación de extensiones por ubicación.

En resumen, las Ubicaciones para múltiples inquilinos con extensiones basadas en la ubicación permiten una organización clara, una gestión sencilla y la reutilización segura de números de extensión en múltiples sitios.

=Configuración de Ubicaciones para múltiples inquilinos=

1. Para crear tu ubicación para múltiples inquilinos, ve al menú Subcuentas y selecciona Ubicaciones para múltiples inquilinos para comenzar la configuración.

[[File:MultitenantSP1.png|thumb|none|500px]]

2. Haz clic en Agregar ubicación para crear una nueva ubicación.

[[File:MultitenantSP2.png|thumb|none|500px]]

3. Completa la información requerida:

Descripción de la ubicación: Ingresa un nombre descriptivo para la ubicación (ej.: Oficina Florida, 2.º edificio, Cabaña).

Buscar subcuenta / Número de extensión: Selecciona la subcuenta que deseas incluir en el grupo y luego asigna el número de extensión (ej.: 1, 100, 101). Repite este paso para cada subcuenta que deba formar parte de la ubicación para múltiples inquilinos.

[[File:MultitenantSP3.png|thumb|none|500px]]

4. Una vez que hayas agregado todas las subcuentas necesarias y sus extensiones, haz clic en Crear para finalizar la configuración de la ubicación para múltiples inquilinos.

=Configuraciones adicionales=

==Extensión accesible por cualquiera==

Si deseas que ciertas subcuentas sean accesibles por cualquiera (en lugar de pertenecer a un grupo específico), o si quieres ajustar sus configuraciones de Ubicaciones para múltiples inquilinos:

: '''1.''' Ve al menú Subcuentas y selecciona Administrar subcuentas.
: '''2.''' Edita la subcuenta que deseas revisar.
: '''3.''' Desplázate hasta la parte inferior de la página para encontrar la opción Extensión interna.

Aquí puedes cambiar el grupo asignado a la subcuenta o hacer que sea accesible por cualquiera dentro de tu cuenta.

[[File:MultitenantSP4.png|thumb|none|500px]]

==SIP URIs==

Para llamar a una subcuenta que no está en un grupo específico y que es accesible por cualquiera, utiliza: {accountID}{extension}@{VoIP.ms server} '''Ejemplo: 123456101@montreal1.voip.ms'''

* '''123456:''' Es tu ID principal de cuenta
* '''101:''' Es la extensión de la subcuenta
* '''@montreal1.voip.ms:''' Es el servidor donde la subcuenta está registrada actualmente

Para las subcuentas dentro de una Ubicación para múltiples inquilinos, el formato incluye el ID de la ubicación: {accountID}#{locationID}#{extension}@{VoIP.ms server} '''Ejemplo: 123456#33#1@montreal1.voip.ms'''

* '''123456:''' Es tu ID principal de cuenta
* '''#33:''' Es el identificador único de la ubicación
* '''#1:''' Es la extensión de la subcuenta
* '''@montreal1.voip.ms:''' Es el servidor donde la subcuenta está registrada actualmente

Para encontrar el identificador único de la ubicación, simplemente dirígete a Subcuenta, Ubicaciones multiinquilino. Allí verás el Número de ubicación junto a tus entradas actuales. Esa es la ubicación que necesitarás usar en tu cadena SIP URI.

[[File:MultiTenantSIPURISP1.png|thumb|none|500px]]

También puedes encontrar esta información ingresando en Subcuentas, Administrar subcuentas y editando tu subcuenta al final de la página.

[[File:MultiTenantSIPURISP2.png|thumb|none|500px]]

File:MultiTenantSIPURISP1.png

2025-09-25T22:42:33Z

RP:

Emplacement à site multiple

2025-09-25T22:41:49Z

RP: /* SIP URI */

File:MultiTenantSIPURISP2.png

2025-09-25T22:41:30Z

RP:

File:MultiTenantSIPURIFR1.png

2025-09-25T22:40:45Z

RP:

Ubicaciones para múltiples inquilinos

2025-09-24T17:47:47Z

RP:

{| class="wikitable"
|+
|-
! Article in English !! Article en Français
|-
| [https://wiki.voip.ms/article/Multi-Tenant_Locations English] || [https://wiki.voip.ms/article/Emplacement_%C3%A0_site_multiple Français]
|}

__TOC__

=¿Qué son las Ubicaciones para múltiples inquilinos?=

Esta función te permite organizar las extensiones por ubicación en lugar de por cuenta, lo que hace posible reutilizar los mismos números de extensión en diferentes sitios sin generar conflictos.

'''Cómo funciona:'''
*'''Ubicaciones:''' Cada sitio u oficina funciona como su propio espacio de trabajo.
*'''Subcuentas:''' Usuarios dentro de una ubicación.
*'''Extensiones:''' Cada subcuenta puede tener extensiones (por ejemplo: 101, 102, 103). Los mismos números también pueden existir en otras ubicaciones de manera independiente.

'''Beneficios:'''
* Mantiene las configuraciones de múltiples oficinas estructuradas y fáciles de gestionar.
* Elimina los conflictos de números de extensión entre sitios.
* Simplifica la asignación, el seguimiento y la identificación de extensiones por ubicación.

En resumen, las Ubicaciones para múltiples inquilinos con extensiones basadas en la ubicación permiten una organización clara, una gestión sencilla y la reutilización segura de números de extensión en múltiples sitios.

=Configuración de Ubicaciones para múltiples inquilinos=

1. Para crear tu ubicación para múltiples inquilinos, ve al menú Subcuentas y selecciona Ubicaciones para múltiples inquilinos para comenzar la configuración.

[[File:MultitenantSP1.png|thumb|none|500px]]

2. Haz clic en Agregar ubicación para crear una nueva ubicación.

[[File:MultitenantSP2.png|thumb|none|500px]]

3. Completa la información requerida:

Descripción de la ubicación: Ingresa un nombre descriptivo para la ubicación (ej.: Oficina Florida, 2.º edificio, Cabaña).

Buscar subcuenta / Número de extensión: Selecciona la subcuenta que deseas incluir en el grupo y luego asigna el número de extensión (ej.: 1, 100, 101). Repite este paso para cada subcuenta que deba formar parte de la ubicación para múltiples inquilinos.

[[File:MultitenantSP3.png|thumb|none|500px]]

4. Una vez que hayas agregado todas las subcuentas necesarias y sus extensiones, haz clic en Crear para finalizar la configuración de la ubicación para múltiples inquilinos.

=Configuraciones adicionales=

==Extensión accesible por cualquiera==

Si deseas que ciertas subcuentas sean accesibles por cualquiera (en lugar de pertenecer a un grupo específico), o si quieres ajustar sus configuraciones de Ubicaciones para múltiples inquilinos:

: '''1.''' Ve al menú Subcuentas y selecciona Administrar subcuentas.
: '''2.''' Edita la subcuenta que deseas revisar.
: '''3.''' Desplázate hasta la parte inferior de la página para encontrar la opción Extensión interna.

Aquí puedes cambiar el grupo asignado a la subcuenta o hacer que sea accesible por cualquiera dentro de tu cuenta.

[[File:MultitenantSP4.png|thumb|none|500px]]

==SIP URIs==

Para llamar a una subcuenta que no está en un grupo específico y que es accesible por cualquiera, utiliza: {accountID}{extension}@{VoIP.ms server} '''Ejemplo: 123456101@montreal1.voip.ms'''

* '''123456:''' Es tu ID principal de cuenta
* '''101:''' Es la extensión de la subcuenta
* '''@montreal1.voip.ms:''' Es el servidor donde la subcuenta está registrada actualmente

Para las subcuentas dentro de una Ubicación para múltiples inquilinos, el formato incluye el ID de la ubicación: {accountID}#{locationID}#{extension}@{VoIP.ms server} '''Ejemplo: 123456#33#1@montreal1.voip.ms'''

* '''123456:''' Es tu ID principal de cuenta
* '''#33:''' Es el identificador único de la ubicación
* '''#1:''' Es la extensión de la subcuenta
* '''@montreal1.voip.ms:''' Es el servidor donde la subcuenta está registrada actualmente

Para encontrar el identificador único de la ubicación, simplemente dirígete a Subcuenta, Ubicaciones multiinquilino. Allí verás el Número de ubicación junto a tus entradas actuales. Esa es la ubicación que necesitarás usar en tu cadena SIP URI.

[[File:MTSIPURI1.png|thumb|none|500px]]

También puedes encontrar esta información ingresando en Subcuentas, Administrar subcuentas y editando tu subcuenta al final de la página.

[[File:MTSIPURI2.png|thumb|none|500px]]