VoIP.ms Wiki - User contributions [en]

10DLC Business Text Messaging Regulations and Requirements

2026-06-16T14:50:58Z

John:

{| class="wikitable"
|+
|-
! Article en Français !! Artículo en Español
|-
| [https://wiki.voip.ms/article/10DLC_R%C3%A9glementations_et_exigences_relatives_%C3%A0_la_messagerie_texte_professionnelle Français] || [https://wiki.voip.ms/article/Regulaciones_y_requisitos_de_mensajer%C3%ADa_de_texto_comercial_10DLC Español]
|}

As of early 2022, United Stated mobile carriers have been enforcing that any business who would like to send 10-digit long code (10DLC) messaging traffic for business purposes needs to subscribe to a database known as [https://www.campaignregistry.com/ The Campaign Registry]. In other words, business text messaging sent to “regular phone numbers” eg. +1 (431) 123-9999 need to preemptively register their messaging traffic. The purpose of this process is to reduce the quantity of SMShing (phishing by SMS) among many other types of fraud or undesired types of SMS/MMS. There are multiple upsides including better speed, better deliverability, and such.

VoIP.ms supports Application-to-Person (A2P) delivery for business text messaging and thus provides the possibility to register your messaging traffic with The Campaign Registry. The following article will provide all the information in relation to the process.

Please note that The Campaign Registry and mobile carriers both have pass-through fees to deliver your business text messaging. Note that VoIP.ms does not make any profit on top of those fees. As of the date of this article, Brand Registration and Campaign Registration with The Campaign Registry have a one-time fee of '''$4.50''', and '''$15.00''', respectively. Campaigns also have a monthly recurring fee anywhere from $2.00 to $30.00 per campaign. Mobile carriers have a per-message pass-through fees anywhere from '''$0.0020''' to '''$0.0325'''. All the related fees can be found below and may be subject to change based on industry changes.

__TOC__

=How can I register my DID Number?=

VoIP.ms provides a white glove service where a dedicated messaging team will help you register your messaging campaign directly with The Campaign Registry on your behalf. To get started, please send an email to messaging@voip.ms.

The process is two-fold, you first need to register your company (also known as “Brand”), then to register your campaign.

Don’t worry about all the technicals, our dedicated messaging team will be providing you with the required information that needs to be filled for each steps and provide you with all the subsequent updates to ensure that you can successfully start sending your business messages.

==Registering my Brand==

Registering your Brand is the first step to being able to start sending business SMS messages. Below are all the details that will be requested by our messaging team. These requirements are directly dictated by The Campaign Registry.

First Name

Last Name

Legal Company Name

DBA or Brand Name

Website

Vertical Type or Industry (Professional Services, Agriculture, Media and Communication, etc.)

DUNS, GIIIN or LEI Number

Address, street, city, state, ZIP code and country of the company

Email and phone number to reach the key contact

Current brand relationship with The Campaign Registry, if any.

As highlighted above, registering your Brand has a one-time pass-through fee from The Campaign Registry for an amount of $4.50.

===Information was filled for my Brand, now what?===

The Campaign Registry takes some time to analyze the information provided to then accept, or reject, the submission. As soon as our messaging team will have an update, they will let you know right away via the initial ticket that was opened.

Once your brand is accepted, you will get to the second process, which is creating your campaign.

==Registering my Campaign==

Your campaign consists of the messaging traffic that you will be sending to your audience. For this second step, our messaging team will be requesting the following information:

*DID number(s) will be related to the campaign ('''Maximum of 43 DIDs can be included in a campaign''')
*Type of campaign (see below or click [https://wiki.voip.ms/article/10DLC_Business_Text_Messaging_Regulations_and_Requirements#Brand_and_Campaign_Fees_from_The_Campaign_Registry here] to see all types of campaign)
*Attributes of the campaign (eg. subscriber can opt-in/opt-out, contains an embedded link, etc.)
*Sample(s) of the message(s) that will be sent to your audience

Similar to registering your Brand, registering a Campaign has an initial pass-through fee of $15.00 plus a monthly recurring charge based on the type of campaign that will be sent (see below or click [https://wiki.voip.ms/article/10DLC_Business_Text_Messaging_Regulations_and_Requirements#Brand_and_Campaign_Fees_from_The_Campaign_Registry here]).

Important: To avoid any sanctions from the US mobile carriers, make sure to provide the right information. The sanctions can be related to limiting/blocking your SMS/MMS traffic up to a fee penalty. If you are unsure, do not hesitate to reach out to our messaging team.

===Information was filled for my Campaign, now what?===

The Campaign Registry will analyze your request and will then accept, reject or potentially ask for more information. As soon as our messaging team has an update, they will let you know right away via the initial ticket that was opened. Note that the vetting process by The Campaign Registry can take up to 15 days.

Once your campaign is accepted, our team will be enabling your account for the requested business text messaging.

=Brand and Campaign Fees from The Campaign Registry=

Below is the breakdown for all the related fees.

==Fees for creating a brand==

There is a one-time fee of $4.50 for creating a brand. This is a pass-through fee from The Campaign Registry.

==Fees for creating a campaign==

There is a one-time fee of $15.00 for creating a campaign. More campaigns, more one-time fees. This is also a pass-through fee from The Campaign Registry.

There is also a monthly recurring fee for the type of campaign that is selected. This is also a pass-through fee from The Campaign Registry. See the table below for all the details.

{| class="wikitable"
|- style="text-align:center;"
! style="text-align:left;" | Use case
! Monthly Recurring Charge (MRC)
! Explanation
|-
| 2FA
| $10.00
| Two Factor authentication style of messages
|-
| Account Notification
| $10.00
| Anything in relation to notification of the audience account. Low balance, alerts, etc.
|-
| Customer Care
| $10.00
| Retention SMS messages, feedback on their experience, etc.
|-
| Delivery Notification
| $10.00
| Fedex/Amazon style of confirmation messages.
|-
| Fraud Alert Messaging
| $10.00
| Alerts related to fraud.
|-
| Higher Education
| $10.00
| Educational messages. “Did you know?” or anything similar.
|-
| Low Volume Mixed
| $1.50
| This is used if you truly cannot find the right category of SMS messages in the whole list.
|-
| Marketing
| $10.00
| All marketing-related content.
|-
| Mixed
| $10.00
| If this consists of Marketing, 2FA, Customer Care, etc.
|-
| Polling and Voting
| $10.00
| If you are inviting your customers on voting for a feature for example.
|-
| Public Safety Announcement
| $10.00
| Weather alert, accident notice, etc.
|-
| Security Alert
| $10.00
| Anything related to security which you want to notify via messages.
|-
| Agents and Franchises
| $30.00
| For example, collector agents.
|-
| Carrier Exemptions
| $10.00
| If you have a clear proof of having an exemption from mobile carriers.
|-
| Charity
| $3.00
| Any charity-related content.
|-
| Conversational Messaging
| $10.00
| Business conversation messages.
|-
| Emergency
| $5.00
| 911-related, catastrophic events alert, etc.
|-
| Political
| $10.00
| Any political messages.
|-
| Social
| $10.00
| For example: Social event invitation.
|-
| Sweepstake
| $10.00
| If you plan on doing Sweepstakes via messages.
|-
| Machine to Machine
| $10.00
| Machine-to-machine (M2M) is a process that implies wireless communication between two or more physical assets. There is no human interaction in the Machine to Machine campaign. Subscriber-facing campaigns are prohibited. This is a dedicated use case.
|-
| K-12 Education
| $10.00
| For faculty members to communicate with parents and students of schools from grades K-12, and distance learning centers, for educational support. This is not for Post-Secondary schools or for services outside of educational support such as PTA meetings, fundraising events, etc.
! Post-registration approval by MNO is required for this use case.
|-
| Proxy
| $10.00
| Peer-to-peer app-based group messaging with proxy/pooled numbers. Supporting personalized services and non-exposure of personal numbers for enterprise or A2P communications.
!Post-registration approval by MNO is required for this Use Case
|}

==Pass-through Fee from Mobile Carriers==

In order to deliver your messaging traffic, mobile carriers are charging a per-message pass-through fees. These fees mostly apply to outbound messaging, but for some cases, also apply to inbound messaging. Below is a breakdown of all per-message pass-through fees. Note that these pass-through fees are additional our retail pricing ($0.0075/SMS and $0.02/MMS).

{| class="wikitable"
|- style="text-align:center;"
! style="text-align:left;" | Carrier
! Fee per SMS
! Fee per MMS
|-
| Verizon Wireless
| $0.00255 to send <br />Free to receive
| $0.0052 to send <br />Free to receive
|-
| US Cellular
| $0.005 to send <br />Free to receive
| $0.01 to send <br />Free to receive
|-
| T-Mobile (incl. former Sprint network)
| $0.003 to send and receive
| $0.01 to send and receive
|-
| AT&T
| $0.002 to send <br />Free to receive
| $0.0035 to send <br />Free to receive
|}

'''Please note as of February 2023, VoIP.ms does not charge per-message pass-through fees from its upstream carriers. VoIP.ms currently absorbs the entirety of it. Note that this decision is subject to change in the future. '''

Panasonic KX-TGP 550

2025-08-07T17:58:45Z

John:

== Devices ==
* Panasonic KX-TGP-550 (cordless VoIP 'phone with one base handset, one remote handset)
* Panasonic KX-TGP-500 (cordless VoIP 'phone with one remote handset, no handset on base)
* Panasonic KX-TGP-551 (cordless VOIP 'phone with one base handset, one remote handset)
<center>
{|
||[[Image:Pana550b.jpg|300px|thumb|caption|Panasonic KX-TGP 550]]
||[[Image:Panasonic KX-TGP-500.jpg|175px|thumb|caption|Panasonic KX-TGP 500]]
|}</center>
<div style="width:100%;overflow:hidden;clear:both"></div>

== Register Handset(s). ==

*Remember to install batteries in your cordless handsets before you start the registration process.

* Plug the cordless handset in the base unit.
* Press the center button to select the '''Menu''' option, and once you enter the '''Menu''' select the '''Initial Settings''' option (the icon is a wrench).
* Once you enter the '''Initial Settings''' menu select the '''Registration''' option (it will be on the 2nd page), and after you enter the sub-menu select '''Register handset''' (should be highlighted)
* On the base unit, press the '''ALL''' button for 4 seconds, the unit may start beeping.
*On the TGP500 (black box), it will be located on the right side of the unit.
*On the TGP550 (phone base), it will be the fourth button on the top row.
Then press the OK soft button on the handset, it may take a few seconds for the unit to register. Repeat for every wireless handset you have.

== Enter the Panasonic KX-TGP 550 Configuration Panel. ==

* On the handset, press the center button to select '''Menu''' and once you enter the '''Menu''' select '''IP Service''' (looks like a toolbox).
* On the '''IP Service''' menu the '''Network Setting''' option should be highlighted, select it and on the sub-menu select '''IP Settings''' and the IP address of the base unit should be displayed, '''record this IP address''', as you will need it momentarily.
*If the IP address doesn't display you may need to repeat the handset registration procedure or check that it has a connection to the network and a DHCP server is available.
* Press the Back soft button to be returned to the menu, and select '''Embedded web'''. By default, '''Off''' is highlighted, press the middle button up once to highlight '''On''' and press it once to save the setting. The handset will beep if successful.
* Enter the IP address of the base unit into your web browser, the default username is '''admin''' and password is '''adminpass'''.

== Register the Panasonic KX-TGP 550 with Voip.ms ==

Complete the following fields in the VOIP Tab:

* '''Phone number:''' (Replace with your Main SIP account or Subaccount UserID, e.g. 100000 or 100000_sub)
* '''Line ID:''' (Replace with your Main SIP account or Subaccount UserID, e.g. 100000 or 100000_sub)
* '''Registrar Server Address:''' servername.voip.ms (Pick one of VoIP.ms multiple [http://wiki.voip.ms/article/Choosing_Server servers])
* '''Registrar Server port:''' 5060 (Default SIP Port)
* '''Proxy server Address:''' servername.voip.ms (Pick one of VoIP.ms multiple [http://wiki.voip.ms/article/Choosing_Server servers])
* '''Proxy server port:''' 5060
* '''Presence server port:''' 5060
* '''Service domain:''' servername.voip.ms (Pick one of VoIP.ms multiple [http://wiki.voip.ms/article/Choosing_Server servers])
Remember to match the '''Registrar Server Address''', '''Proxy server Address''' and the '''Service domain''' options with each other.
Also remember to use the same server that you set as '''Point of Presence''' (POP) to your DID.
* '''Source port:''' 5060 (You may choose from our supported ports: 5060, 5080 or 42872)
* '''**Authentication ID:''' (Replace with your Main SIP account or Subaccount UserID, e.g. 100000 or 100000_sub)
* '''**Authentication password:''' ********* (Use the SIP account password-By default the same as the Customer portal)
* '''Keep Alive Interval:''' Change from 0 to around 15

**You can also find this information by logging to your [https://www.voip.ms/m/accountinfo.php Customer portal].

[[File:Panasonic.png]]

==Additional Details==

On the TGP500 model currently it may or may not be possible to use sub accounts, as the device won't accept the underscore (_) for certain fields on which the SIP account userID needs to be entered. For the latest (2022) firmware update, it does seem to work correctly with sub accounts. Other issues to be aware of is ensuring the "Source Port" is correct on a multi-line setup.

The default registration expiry time on this device is 3600 seconds. You may have to reduce that if you lose registration. Unfortunately, that can not be set on the web interface. You need to load a provisioning configuration file from a web or FTP server (details [http://www.voicesonic.com/panasonic/manuals/panasonic-others/TGP500-550-Admin-Guide.pdf here], page 132 for file format). The parm is REG_EXPIRE_TIME_[n]="mmm", where n is line number, and mmm is the registration interval. e.g. REG_EXPIRE_TIME_1="180" to set line 1 registration time to 180 seconds. A symptom of this problem is that the telephone web interface shows that the line is registered, but the voip.ms control panel indicates that it is not registered. This device supports multiple provisioning files in a hierarchy.

The "Phone Number" field may contain only digits, and *must* contain at least one digit, or the base unit won't try to register it. Within those restrictions it can be any number you want. When using the handset (or base unit) to select a line from which to dial, this number will be displayed next to the entry for the associated line. I think Panasonic's idea is that this is supposed to be a DID associated with the SIP address, but it doesn't have to be.

Enter your SIP username in the "Line ID" field and again in the "Authentication ID" field. If you are registering a voip.ms main account your SIP username will be your six-digit account number. If you are registering a sub-account it will be your main account number followed by an underscore followed by the sub-account number.

You can register at any voip.ms server (e.g. "atlanta.voip.ms") but you must enter your chosen server name in the "Registrar Server Address" and "Proxy Server Address" and "Service Domain" fields.

After saving these settings, check the "VoIP Status" page in the "Status" section to see if the status of this line is "registered". This product can register up to eight SIP addresses at once. If any of the SIP addresses you configured into the device do not show their status on this page as being "registered", then the status light on the base unit will flash yellow. (This is not mentioned in either the user guide nor the admin guide for this product.)

The above information applies to both the TGP-550 (where the base unit has a handset, dialpad, LCD display, etc.) and the TGP-500 (where the base unit is a faceless black box that sits in your network closet).

==Dial Plans==

If you wish to have 7 digit dialing within a USA area code, go to "Telephone" "Call Control" "Line #" and in the "Dial Plan" box, enter: <:585>[23456789]xxxxxx
With the addition of immediate *11 calls (such as 411 or 911): [23456789]11|<:585>[23456789]xxxxxx

[[Category:SIP phones]]

Bria Desktop

2025-08-07T17:57:20Z

John:

'''
== Product Information ==
'''

<br>
Clients may enjoy the simplicity and use of the Bria softphone. For those requiring more features for use in a business environment the Bria softphone may be an option. The Bria softphone is available for both desktop (Windows, OSX and Linux) and mobile platforms (Android and iPhone/iPad) at this time. Note that this is not a Free Product.

'''Resources:'''<br>
Website:
[http://www.counterpath.com/bria.html Bria Product Page]

'''Help / Support:'''<br>
[http://www.counterpath.com/support.html Counterpath support page]

== '''Configuring the Bria softphone:''' ==
<br>
You will need a licensed version of Bria. If your version is not licensed then you will not be able to follow this guide.

Once Bria is open after installation, you'll see a window with 4 fields: Profile, Login server, Username and Password.

[[File:Bria_1st_image.jpg|thumb|none|600px]]

You'll need to select: "No login required" under the "Profile" field, then continue and enter your licence key.

[[File:Bria_2nd_image.jpg|thumb|none|600px]]

Once this is done press "Ok" and you'll be able to proceed with the steps below.

<br>We recommend that you read each step through in its entirety before performing the action indicated in the step.<br>
<br>
'''Launching Bria'''<br>

Launch Bria using the shortcut either on your desktop, your start/applications menu, or your applications folder.

If this is your first time launching Bria please choose the No login required option and enter your license.

You should now be taken to the Bria softphone's main window:

[[File:Bria_d_01.png‎]]

We will now need to access the account configuration menu.

For OSX you can click on the Bria menu at the top left of the screen. Then click on Preferences and Accounts to gain access to the accounts window.
For windows please click on the Softphone menu at the top left of the application. Then click on Account Settings to enter the account configuration dialog.

'''Configuring your Softphone with VoIP.ms Account Information'''<br><br>
After you have accessed the account configuration interface you will see a window with tabs across the top. Click on the plus button at the bottom of the Accounts tab to add a New SIP account. Please use the information below to configure your VoIP.ms account.
<br>
Within the Account tab fill in the fields as explained below:

'''Account Name:''' Please enter VoIP.ms or any identifiable name.
'''User ID:''' (Replace with your Main SIP account or Subaccount UserID, e.g. 100000 or 100000_sub)

'''Domain:''' servername.voip.ms (Pick one of VoIP.ms multiple [http://wiki.voip.ms/article/Choosing_Server VoIP Servers])
* You can also find this information by logging into your [https://www.voip.ms/m/accountinfo.php Customer portal].

'''Password:''' ****** (Use the SIP account password - By default this is the same as the Customer portal password)

'''Display Name:''' Blank or your name
Authorization name: (Replace with your Main SIP account or Subaccount UserID, e.g. 100000 or 100000_sub)

'''Domain Proxy'''
Register with domain and receive calls: Checked
Send outbound via: Select Domain

The dialing plan should be left as it is which is:
#1\a\a.T;match=1;prestrip=2;

<br>Your configuration screen should look similar to the one shown in the picture below:

[[File:Bria_d_02.png‎]]

'''Configuring Extra Settings'''<br>

Here we will configure options related to your network. For users who wish to use video this may be possible however both calling ends MUST support the same codec. In other words please make sure that at least H263 is enabled in your video codecs configuration.

Please make sure that the following options are configured:

'''Under Topology:'''

Firewall Traversal Method: None (use my local IP address)

[[File:Bria_d_03.png‎]]

'''Under Advanced:'''

Register Settings
Register every: 180 seconds
Minimum time: 20 seconds
Maximum time: 300 seconds

'''Advanced Options:'''

Send SIP Keep-Alives: Checked
Use rport: Checked

Click OK to save your settings and go back to the main phone screen. You should now be registered to your VoIP.ms account and it should show as being Available:

'''Congratulations!'''<br> You can now make a phone call.

You can make a test call to 4443 (Quality Assurance Echo Test), or if you have deposited funds into your account you can place a call to a traditional land line or mobile phone by dialing either:<br>
1 + the area code and number for calls to the US or Canada
Or
011 + the country code, area code and number for calls worldwide (you may also use 00 instead of 011).

Grandstream GXP2200

2025-08-07T17:56:56Z

John:

[[File:Grandstream_GXP2200.png|300px|thumb|left|Grandstream GXP2200]]
<div style="width:100%;overflow:hidden;clear:both"></div>

The Grandstream GXP2200 has a 480x272 pixel capacitive touch screen TFT LCD, 6 lines with up to 6 SIP accounts, HD wideband audio and Supports Android Operating System 2.3 and the vast number of 3rd party Android applications.

==Enter the Grandstream GXP2200 Configuration Panel==

Connect the GXP2200 to your network and power up the device. After 60 seconds on the LCD screen you'll see a window to add accounts, on the top of that window you'll see the IP address of your device (192.168.xx.xxx). Save that IP address.

On your computer open your preferred browser and put the IP address in the address bar.

Once the page is loaded it will ask you for a '''Username''' and '''Password''', the default is Username: '''admin''' Password: '''admin'''

==Register the Grandstream GXP2200 with Voip.ms==

Once you have entered the configuration panel, select '''General Settings''' on the panel on the left and select '''Account''' on the menu at the top of the page and complete the following fields:

*'''Account Active:''' Checked
*'''Account Name''' The outbound callerID Name '''(See the requirements below)'''
*'''SIP Server:''' atlanta.voip.ms (Pick any one of VoIP.ms multiple [[Choosing Server|servers]])
*'''SIP User ID:''' (Replace with your Main SIP account or Subaccount UserID, e.g. 100000 or 100000_sub)
*'''SIP Authenticate ID:''' (Replace with your Main SIP account or Subaccount UserID, e.g. 100000 or 100000_sub)
*'''SIP Authenticate Password:''' ********* (Use the SIP account password-By default the same as the Customer Portal)
*'''Voice Mail UserID:''' *97
*'''Name:''' Enter a name of your choice (it can be your name or company name)
: If you want to register your Main account, you can find the Username and Password information by logging into your [https://www.voip.ms/m/samples/voxalot.php Customer Portal].
: If you want to register a sub-account you will have to use the credentials that you set when you created that sub-account.

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

[[File:GXP2200.png|800px]]
<div style="width:100%;overflow:hidden;clear:both"></div>

Once you have '''Saved''' the Account information, click on the '''Network Settings''' option on the left and under the '''Account''' you configured complete the following fields:

*'''Outbound Proxy:''' atlanta.voip.ms (Pick any one of VoIP.ms multiple [[Choosing Server|servers]])
*'''NAT Traversal:''' Keep-alive

[[File:GPX2200_Network_Settings_N.png|800px]]

After saving the settings, you should now be able to start making calls. Please contact us via Live Chat or Ticket if you require any assistance.

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

User Manual : [http://www.grandstream.com/sites/default/files/Resources/gxp2200_usermanual_english.pdf Download PDF]

[[category:SIP phones]]

Grandstream HandyTone 702 - HT702

2025-08-07T17:56:29Z

John:

== GENERAL INFORMATION ==

The Grandstream HandyTone 702/7XX series is the latest in the HandyTone line. The 702 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/handytone-702/704 HT702 Product Page]

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

'''Documentation'''
[https://www.grandstream.com/sites/default/files/Resources/ht70x_usermanual_english.pdf Grandstream HandyTone 702 - User Manual]

[[File:Ht702.jpg‎]]

== Configuring the Handytone 702 ==

These instructions are based on HandyTone 702 software version 1.0.0.18; if you are running a different software version some menus and settings may be different.
<br>
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 1'''<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.

Finally plug the supplied power cable into the HandyTone.
<br>

'''Step 2'''<br>

: Wait 60 seconds after plugging your HT702 in.

Pick up the phone connected to the HT702 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 HT702 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.
<br>

'''Step 3'''<br>

: You should now see a page that looks like this:<br>
[[File:Grandstream_702_01.png‎]]

Enter the password for the HT702 in the password field. The default administrator password for the HT702 is "admin" (without quotes).

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

[[File:Grandstream_702_02.png‎]]
<br>

'''Step 4'''<br>

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

Please use the same server in the Failover SIP Server as your Primary SIP Server or leave the Failover SIP Server field Blank.

[[File:GrandstreamConfigCrctd.png]]

Please use the following settings to configure your VoIP.ms account:
'''
Configuration Page Settings:'''

Primary SIP Server: servername.voip.ms (Pick one of VoIP.ms multiple [http://wiki.voip.ms/article/Choosing_Server VoIP Servers])

* You can also find this information by logging into your [https://www.voip.ms/m/accountinfo.php Customer portal].

Failover SIP Server: (Please leave this Blank)

Outbound Proxy: servername.voip.ms (Pick one of VoIP.ms multiple [http://wiki.voip.ms/article/Choosing_Server VoIP Servers])
* You can also find this information by logging into your [https://www.voip.ms/m/accountinfo.php Customer portal].

NAT Traversal (STUN): No, but send keep-alive

SIP User ID: (Replace with your Main SIP account or Subaccount UserID, e.g. 100000 or 100000_sub)

Authenticate ID: (Replace with your Main SIP account or Subaccount UserID, e.g. 100000 or 100000_sub)

Authenticate Password: ****** (Use the SIP account password - By default this is the same as the Customer Portal)

DNS Mode: A Record

SIP Registration: Yes

Unregister On Reboot: Yes

Outgoing Call Without Registration: Yes

Register Expiration: 3

Allow Incoming SIP Messages from SIP Proxy Only: Yes

Preferred DTMF method: In-audio, RFC2833

Enable Call Features: No

Dial Plan: {[x*]+}

Preferred Vocoder: PCMU, G729, PCMA
<br>

'''Step 5'''<br>

: Once you have configured the settings above, click the Update button and then the Reboot button to save the configurations.

Your HT702 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 green color, then your unit is configured and ready to make calls.
<br>

'''Step 6'''<br>

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

1 + the area code and 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).
<br>

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

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

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

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

== Auto Provisioning ==

Some newer models of the HT702 now have Auto Provisioning and will delete the changes you make in setting up the device to use our service. Please go to your Graphical user interface and go to 'Advanced Settings' tab and look for "Firmware Upgrade and Provisioning" and disable it.

== Star codes ==
Here are the star codes available for HT701 and HT702:
* *02 Forcing a Codec (per call) *027110 (PCMU), *027111 (PCMA), *02723 (G723), *02729 (G729),*0272616 (G726-r16), *0272624 (G724-r24), *0272632 (G726-r32), *0272640 (G726-r40), *027201 (iLBC)

* *03 Disable LEC (per call) Dial “*03” + ” number ”. No dial tone is played in the middle.
* *16 Enable SRTP
* *17 Disable SRTP
* *30 Block Caller ID (for all subsequent calls)
* *31 Send Caller ID (for all subsequent calls)
* *47 Direct IP Calling. Dial “*47” + “IP address”. No dial tone is played in the middle. Detail see Direct IP Calling section on page 12.
* *50 Disable Call Waiting (for all subsequent calls)
* *51 Enable Call Waiting (for all subsequent calls)
* *67 Block Caller ID (per call). Dial “*67” + ” number ”. No dial tone is played in the middle.
* *82 Send Caller ID (per call). Dial “*82” + ” number ”. No dial tone is played in the middle.
* *69 Call Return Service: Dial *69 and the phone will dial the last incoming phone number received.
* *70 Disable Call Waiting (per call). Dial “*70” + ” number ”. No dial tone is played in the middle.
* *71 Enable Call Waiting (per call). Dial “*71” + ” number ”. No dial tone is played in the middle.
* *72 Unconditional Call Forward: Dial “*72” and then the forwarding number followed by “#”. Wait for dial tone and hang up. (dial tone indicates successful forward)
* *73 Cancel Unconditional Call Forward. To cancel “Unconditional Call Forward”, dial “*73”, wait for dial tone, then hang up.
* *74 Enable Paging Call: Dial “*74” and then the destination phone number you want to page.
* *78 Enable Do Not Disturb (DND): When enabled all incoming calls are rejected.
* *79 Disable Do Not Disturb (DND): When disabled, incoming calls are accepted.
* *87 Blind Transfer
* *90 Busy Call Forward: Dial “*90” and then the forwarding number followed by “#”. Wait for dial tone then hang up.
* *91 Cancel Busy Call Forward. To cancel “Busy Call Forward”, dial “*91”, wait for dial tone, then hang up.
* *92 Delayed Call Forward. Dial “*92” and then the forwarding number followed by “#”. Wait for dial tone then hang up
* *93 Cancel Delayed Call Forward. To cancel Delayed Call Forward, dial “*93”, wait for dial tone, then hang up. Flash/Hook Toggles between active call and incoming call (call waiting tone). If not in conversation,

*flash/hook will switch to a new channel for a new call.
*# Pressing pound sign will serve as Re-Dial key

''Keywords: Handytone 701, HT701, Grandstream''

[[Category:Analog Telephone Adapters]]

SFLphone

2025-08-07T17:55:33Z

John:

[[File:Main.jpg|thumb|left|SFLphone Softphone]]
<div style="width:100%;overflow:hidden;clear:both"></div>

SFLphone is an open-source SIP/IAX2 compatible softphone for Linux. SFLphone is free software released under the GNU General Public License. Packages are available for all major distributions including Debian, openSUSE, Fedora, Mandriva and the latest Ubuntu releases.

==Register the SFLphone Softphone with VoIP.ms==

Open the SFLphone Softphone and under the '''Edit''' menu select the '''Accounts''' option. Then on the Accounts window click on the '''Add''' button.

[[File:Sflphone_02.png|left]]
<div style="width:100%;overflow:hidden;clear:both"></div>

This option will lead you to the''' Account Settings''' window, once you're there under the '''Basic''' tab complete the following fields:

*'''Alias:''' Enter a name of your choice (it can be your name or your company name)
*'''Protocol:''' Choose the protocol you will use SIP or IAX2
*'''Host Name:''' servername.voip.ms (Pick one of VoIP.ms multiple [http://wiki.voip.ms/article/Choosing_Server Servers])
*'''User Name:''' (Replace with your Main SIP account or Subaccount UserID, e.g. 100000 or 100000_sub)
*'''Password:''' ********* (Use the SIP account password-By default the same as the Customer portal)
*'''Proxy:''' servername.voip.ms (Pick the same server as chosen before, VoIP.ms multiple [http://wiki.voip.ms/article/Choosing_Server Servers])
*'''User-Agent:''' (Replace with your Main SIP account or Subaccount UserID, e.g. 100000 or 100000_sub)
: If you want to register your Main account, you can find the Username and Password information by logging into your [https://www.voip.ms/m/samples/voxalot.php Customer Portal].
: If you want to register a sub-account you will have to use the credentials that you set when you created the sub-account.
[[File:Sflphone_account_settings.png]]
<div style="width:100%;overflow:hidden;clear:both"></div>

After you have entered the information in the '''Basic''' tab, click on the '''Apply''' button at the bottom and then click on the '''Audio''' tab, and make sure the only codecs enabled are the '''PCMU'''(G.711U), '''G.729a''' and '''GSM'''. Please also set the '''DTMF''' to '''RTP'''

[[File:Sflphone_04.png|left]]
<div style="width:100%;overflow:hidden;clear:both"></div>

After this, click on the '''Apply''' button at the bottom and go to the '''Advanced''' tab, once you're there set the '''Registration expires''' option to 300 seconds (5 minutes) and click on the Apply button at the bottom.

[[File:Sflphone_05.png|left]]
<div style="width:100%;overflow:hidden;clear:both"></div>

As a last step go to the '''Security''' tab and click on the '''Add''' button, this will lead you to the '''Credential settings''' window, once you're there complete the following fields with the information that you set on the '''Basic''' tab and '''Apply''' your changes.

*'''Authentication name:''' (Replace with your Main SIP account or Subaccount UserID, e.g. 100000 or 100000_sub)
*'''Password:''' ********* (Use the SIP account password-By default the same as the Customer portal)

[[File:Sflphone_06.png|left]]
<div style="width:100%;overflow:hidden;clear:both"></div>

Your SFLphone should be ready to make calls, as a test you can place a call to our Echo test (Just dial 4443).

[[category:softphones]]

Grandstream HandyTone 502 - HT502

2025-08-07T17:54:09Z

John:

[[File:Ht502.jpg|300px|thumb|left|Grandstream HandyTone 502 - HT502]]
The '''Grandstream HandyTone 502''' is a two-line analogue telephone adapter with a built-in single port wired 10/100Mbps LAN router and network address translation (NAT). It is reported to support both tone and [[pulse dial]] (a capability which was removed from the later [[Grandstream HandyTone 702 - HT702|HT702]]).

A [http://www.grandstream.com/products/gateways-and-atas/analog-telephone-adaptors/product/handytone-502 product description] and manual is available from the manufacturer. You can access your User Manual here and it includes your device's star codes on page 25: http://www.grandstream.com/sites/default/files/Resources/ht502_user_manual.pdf <br clear="all"/>

==Connecting the HandyTone==

1. Connect a standard touch-tone analog telephone to PHONE port (or PHONE1,
PHONE2 port for HT386/496/502).

2. Connect a PSTN telephone line to LINE port (optional, applies to HT386/486/488/503 only).

3. Insert the Ethernet cable into the Ethernet port (HT286/386) or WAN port (HT486/488/496/502/503) of
HandyTone and connect the other end of the Ethernet cable to an uplink port (a router or a modem, etc.).

4. Connect a PC to the LAN port of HandyTone (optional, applies to HT486/488/496/502/503 only).

5. Insert the power adapter into the HandyTone and connect it to an electrical outlet.

6. Using the HandyTone embedded web server or IVR (Interactive Voice Prompt) menu, you can further
configure the phone using either a static IP or DHCP.

[[File:502.jpg]]

==Accessing to the Web Configuration==

1. From the analog phone, press *** to get into the IVR menu. Enter option 02 to obtain the HandyTone’s IP
address.

2. For HT486/488/496/502/503, please enable the “WAN side HTTP access” option by entering IVR
option 12 and press 9. A reboot or power cycle of the HandyTone is required after this change. You can
also access the HandyTone’s web configuration from a PC connected to the LAN port via 192.168.2.1.

3. Type the HandyTone’s IP address in your PC browser.

4. Log in using password “admin” to configure the HandyTone.

==Settings==

'''Step 1:'''

* Click on '''FXS 1''' to configure your first line.

'''Step 2:'''

Fill the followings fields.

*'''Account Active:''' Yes
* '''Primary SIP Server:''' atlanta.voip.ms (Pick one of VoIP.ms multiple [http://wiki.voip.ms/article/Choosing_Server VoIP Servers])
*'''Outbound Proxy:''' Set the same server you've configured at the '''Primary SIP Server''' field
*'''NAT Traversal:''' No, But Keep-alive
* '''SIP User ID:''' (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:''' ********* (account password)

* '''SIP Registration:''' Yes
* '''Register Expiration:''' 5 minutes
*'''Preferred Vocoder:''' Select as primary option the PCMU codec and as secondary option the G729 codec
*'''Allow Incoming SIP Messages from SIP Proxy Only:''' Yes
[[File:HT502-02.png|600px]]

'''Step 3:'''
* (Optional) Select a customised dial plan.

'''Step 4:'''
* Click on '''Basic settings''' to configure your time zone.
* Set ''Time Zone'' to one of the predefined values (such as Eastern Time: GMT-5) or a self-defined time zone (for example: MTZ+6MDT+5,M4.1.0,M11.1.0) for your area.

: ''Don't forget to reboot your device after you've applied the recommended settings.''

== Dial plans ==
The use of a customised dial plan is optional.

A dial plan contains a series of digit sequences, separated by the | character, entirely enclosed within { curly brackets }. Each time a phone button is pressed, the ATA will attempt to match the digit sequence to the dial plan.

===Grandstream's default dial plan===

{ x+ | *x+ | *xx*x+ }

===VoIP.ms===

We recommend the use of the dial plan presented below, however, you can build your own dial plan depending on your needs
{911|[3469]11|*xx.|<=1>[2-9]xxxxxxxxx|1[2-9]xxxxxxxxx|<=1555>[2-9]xxxxxx|011xxxxxxxxxxxx.|011[2-9]x.| 4xxx | ***xx}
Replace the 555 with your area code if you want to be able to dial 7 digit numbers.

===Building your own dial plan===
In theory, a standard [[Dial Plan for Linksys ATAs|VoIP.ms dial plan]] could be adapted to Grandstream as:
: {911|310xxxx|<=1555>[2-9]xxxxxx|1[2-9]xx[2-9]xxxxxx|[2-9]xx[2-9]xxxxxx|*xx|***xxx|*xx.|[3468]11|822|0|00|4xxx|**275*x.|xxxxxxxxxxxx.}
This would attempt the following translations:
: 911 (the North American emergency number) is passed verbatim
: 310xxxx passes any seven-digit 310-xxxx numbers verbatim
: <=1555>[2-9]xxxxxx adds area code 1-555 to seven-digit local calls (which begin with anything other than 0 or 1)
: 1[2-9]xx[2-9]xxxxxx passes any full eleven-digit North American format (1-NXX-NXX-XXXX) telephone number verbatim
: [2-9]xx[2-9]xxxxxx passes any ten-digit North American format (NXX-NXX-XXXX) telephone number verbatim. This is not compatible with the 7-digit rule above.
: *xx passes any 'star' and two digit codes (like *73 to disable call forwarding)
: *xx. is similar but passes any code with 'star' followed by at least two digits
: ***xxx passes three stars followed by any three numeric digits
: [3468]11 passes 311, 411, 611 and 811 verbatim
: 822|0|00 passes 822, 0 or 00 verbatim
: 4xxx passes any four-digit number in the 4000-4999 range (such as 4443, an [[Dialing Codes|echo test]] for [[Troubleshooting Outgoing Calls|troubleshooting]])
: **275*x. passes any number with **275* followed by at least one digit (a [[Dialing Codes|dialling code]] to reach [http://www.sipbroker.com/sipbroker/action/providerWhitePages SIP Broker] numbers)
: xxxxxxxxxxxx. passes any number which consists of a dozen or more consecutive numeric digits

Unfortunately, in the Grandstream 502 dial plans, as soon as one rule is matched (except for x+ or x. which allow an arbitrary number of additional digits) the call is sent out immediately. (By contrast, the Linksys dial plan will wait for a short [typically five second] delay unless 'S0' is specified explicitly; later Grandstream models such as the [[Grandstream HandyTone 802 - HT802|HT802]] also support a three-second delay by placing a 'T' flag at the end of a rule). It's therefore not possible to deal with multiple conflicting alternatives on this ATA by waiting for the user to dial additional digits which could potentially match some other valid pattern.

For instance 444-3xxx (a seven-digit local call) and 4443 (an echo test) might both be desired as valid for some dial plans, but can't both be supported at once. Seven and ten-digit local calls also couldn't both be supported in the same dial plan.

For North American 7-digit dial (adds 1-555- to local calls):
: { 0x+ | *x+ | [2-9]11 | <=1555>[2-9]xxxxxx | 1[2-9]xx[2-9]xxxxxx |***xxx | **275*x+ | *xx*x+ }
For North American 10-digit dial:
: { 0x+ | *x+ | [2-9]11 | [2-9]xx[2-9]xxxxxx | 1[2-9]xx[2-9]xxxxxx |***xxx | **275*x+ | *xx*x+ }
For no translation of domestic numbers:
: { x+ | *x+ | ***xxx | **275*x+ | *xx*x+ }

Note: In the seven digit dial plan, replace 555 by the area code of your choice. The 4443 (Echo Test) and 4747 (DTMF Test) [[Dialing Codes|dialling codes]] will not be supported.

The dial plan structure (described on [http://grandstream.com/products/ht_series/ht502/documents/ht502_usermanual_english.pdf page 44 of the manual] is otherwise very similar to that described for the [[Dial Plan for Linksys ATAs]] with minor discrepancies:

{| class="wikitable"
|-
! Grandstream 502
! Cisco/Linksys ATA's
! Function
|-
| style="text-align:center;" colspan="2"|0 1 2 3 4 5 6 7 8 9 * # a b c d
| You can use any of these characters to represent a pressed phone digit.
|-
| style="text-align:center;" colspan="2"| x
| Any phone digit, [0-9]
|-
| style="text-align:center;" colspan="2"|[sequence]
| You can enter characters between brackets to create a list of acceptable digits. <br>For example, if you enter the range [1-5], the user may only press the digits from 1 to 5. <br>You can also use individual numbers, and certain other characters, in combination. For example [35-8*] allows the user to press 3, 5, 6, 7, 8 or *. [369]11 allows calls to 311, 611 or 911.
|-
| style="text-align:center;"|+ (plus) or . (period)
| style="text-align:center;"|. (period)
| You can use a period to accept zero or more entries of a give digit. <br>For example, '''01.''' allows the user to enter 0, 01, 011 and so on, while '''xx+''' allows any two or more numeric digits to be dialled.
|-
| style="text-align:center;"|<dialed=substituted>
| style="text-align:center;"|<dialed:substituted>
| This is used for sequence substitution, you can use this to indicate that certain numbers dialled are replaced by other characters. The ''dialled'' digits can be zero or more characters. <br>For example, with '''<=1555>xxxxxxx''' if the user dials a 7 digit number, the number 1555 is added to the beginning of the sequence. If the user presses 6782345, the system transmits 15556782345. ''<2=011>'' replaces digit 2 with 011 when dialling and ''< =1>'' adds a leading 1 to all numbers dialled
|-
| style="text-align:center;"|^ (leading carat)
| style="text-align:center;"|! (trailing exclamation point)
| You can use this character to prohibit a dial sequence. <br/>For example '''^1900x+''' will reject any sequence that starts with 1900.
|-
| style="text-align:center;"|{} (curly brackets)
| style="text-align:center;"|() (parentheses)
| These enclose the entire dial plan text
|-
| style="text-align:center;" colspan="2"| | (pipe character)
| These separate each of the individual rules within the dial plan
|}

There are no supported Grandstream 502 equivalents to these codes:
{| class="wikitable"
|-
! Cisco/Linksys ATA's
! Function
|-
| style="text-align:center;"|, (comma)
| No supported equivalent. Used between digits to play an “outside line” dial tone after a user-entered sequence.
|-
| style="text-align:center;"|S0 or L0
| No supported equivalent. Overrides the Short or Long inter-digit timer to 0 seconds
|-
| style="text-align:center;"|P#
| No supported equivalent. Pauses # seconds.
|}

The factory default Grandstream dial plan is: { x+ | *x+ | *xx*x+ } where:
: x matches any numeric digit
: x+ matches one or more digits
: 0 1 2 3 4 5 6 7 8 9 * # a b c d match those specific, individual digits only
Each rule is separated from the others by | and the full dial plan is enclosed in { } curly braces.

To ensure that user-dial strings such as *123 are passed through to activate features provided by VoIP.ms (as service provider) instead of being handled locally by the Grandstream ATA, the * sequence should be defined inside the dial plan; the resulting dial plan rule would contain: { *x+ }

== Known Issues and Resolutions ==

'''Hearing an Echo on the Line:'''
: Please go to your FXS port setting screen, the one you are using with our service, and verify that the option 'Disable Line Echo Canceller' is set to NO.
: You can also adjust the setting Gain, where the Rx (Other Person) is a gain level for signals transmitted by FXS and Tx (You) is a gain level for signals received by FXS.

'''Receiving Weird Calls such as from CallerID 100 or in the middle of the Night not showing in your CDR:'''
: These calls are not going through our Network but rather through the internet directly to your ATA Device.
: Please check in the Web GUI under each FXS port tab that this option is enabled: Allow Incoming SIP Messages from SIP Proxy Only: Set to YES.
: This will make sure you can only receive calls from VoIP.ms.

==Guide Links==
In the event where you need the guide directly from Grandstream, you may find the company guide below:

User Manual : [http://www.grandstream.com/sites/default/files/Resources/ht502_user_manual.pdf Download PDF]

[[category: Analog Telephone Adapters]]

Call Forwarding

2025-08-03T19:11:06Z

John:

{| 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>

=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]]

File:CFWD320250803.png

2025-08-03T19:10:46Z

John:

File:CFWD220250803.png

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John:

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2025-08-03T19:09:01Z

John:

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John: uploaded a new version of "File:CFWD120250803.png"

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John:

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John: uploaded a new version of "File:Callfwd.JPG": Reverted to version as of 18:05, 3 August 2025

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John: uploaded a new version of "File:Callfwd.JPG"

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John: uploaded a new version of "File:Callfwd.JPG"

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John: uploaded a new version of "File:Callfwd.JPG": Reverted to version as of 17:52, 3 August 2025

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John: uploaded a new version of "File:Callfwd.JPG"

File:Callfwd.JPG

2025-08-03T17:52:15Z

John: uploaded a new version of "File:Callfwd.JPG": Reverted to version as of 16:36, 24 May 2017

File:Callfwd.JPG

2025-08-03T17:50:54Z

John: uploaded a new version of "File:Callfwd.JPG"

Softphones

2025-06-26T14:22:05Z

John: /* Most Popular Softphones */

A softphone is a software program for making telephone calls over the Internet using a general purpose computer, tablet or smartphone, rather than using dedicated hardware. Often a softphone is designed to behave like a traditional telephone, sometimes appearing as an image of a phone with a display panel and buttons with which the user can interact. A softphone is usually used with a headset connected to the sound card of the PC, or as an installed app on your phone. See also: [[Smartphone]]

Take a peek at VoIP.ms Blog Article : [https://wiki.voip.ms/article/Back_to_Basics,_What_is_a_Softphone_and_How_to_Leverage_It%3F Back to Basics, What is a Softphone and How to Leverage It?]

__NOTOC__

==Most Popular Softphones==
Not sure what to use? Here's our top used softphones across our network:
* [[VoIP.ms_Softphone | VoIP.ms Softphone]]
* [[Softphones#ZoiPer | Zoiper]]
* [[Softphones#Bria_Teams | Bria]]
* [[Softphones#Linphone | Linphone]]

==List of Softphones==
{| 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" | Softphone Name
! scope="col" | Android<br />Compatible
! scope="col" | iOS<br />Compatible
! scope="col" | Windows<br />Compatible
! scope="col" | MAC<br />Compatible
! scope="col" | Linux<br />Compatible
! scope="col" | Other<br />Support
! scope="col" | SMS<br />Support
! scope="col" | SIP TLS<br />Support

|-
| style="text-align:left;" | [https://wiki.voip.ms/article/VoIP.ms_Softphone VoIP.ms Softphone]
| style="text-align:center;" | X
| style="text-align:center;" | Coming soon
|
|
|
|
| style="text-align:center;" | X
| style="text-align:center;" | X
|-
| style="text-align:left;" | [https://wiki.voip.ms/w/index.php?title=Softphones&stable=1#Acrobits Acrobits]
| style="text-align:center;" | X
| style="text-align:center;" | X
|
|
|
|
| style="text-align:center;" | X
| style="text-align:center;" | X
|-
| style="text-align:left;" | [https://wiki.voip.ms/w/index.php?title=Softphones&stable=1#AntiSIP_.28Mobile.29 AntiSIP (Mobile)]
| style="text-align:center;" | X
|
|
|
|
|
| style="text-align:center;" | X
| style="text-align:center;" | X
|-
| style="text-align:left;" | [https://wiki.voip.ms/w/index.php?title=Softphones&stable=1#Blink_.28Desktop.29 Blink (Desktop)]
|
|
|
| style="text-align:center;" | X
| style="text-align:center;" | X
|
| style="text-align:center;" | X
|
|-
| style="text-align:left;" | [https://wiki.voip.ms/w/index.php?title=Softphones&stable=1#Bria_Mobile Bria Mobile]
| style="text-align:center;" | X
| style="text-align:center;" | X
|
|
|
|
| style="text-align:center;" | X
| style="text-align:center;" | X
|-
| style="text-align:left;" | [https://wiki.voip.ms/w/index.php?title=Softphones&stable=1#Bria_Solo Bria Solo]
| style="text-align:center;" | X
| style="text-align:center;" | X
| style="text-align:center;" | X
| style="text-align:center;" | X
|
|
| style="text-align:center;" | X
| style="text-align:center;" | X
|-
| style="text-align:left;" | [https://wiki.voip.ms/w/index.php?title=Softphones&stable=1#Bria_Teams Bria Teams]
| style="text-align:center;" | X
| style="text-align:center;" | X
| style="text-align:center;" | X
| style="text-align:center;" | X
|
|
| style="text-align:center;" | X
| style="text-align:center;" | X
|-
| style="text-align:left;" | [https://wiki.voip.ms/w/index.php?title=Softphones&stable=1#Dialer.2B Dialer+]
|
| style="text-align:center;" | X
|
|
|
|
| style="text-align:center;" | X
| style="text-align:center;" | X
|-
| style="text-align:left;" | [https://wiki.voip.ms/w/index.php?title=Softphones&stable=1#Ekiga Ekiga]
|
|
| style="text-align:center;" | X
|
| style="text-align:center;" | X
|
| style="text-align:center;" | X
|
|-
| style="text-align:left;" | [https://wiki.voip.ms/w/index.php?title=Softphones&stable=1#FgVoIP FgVoIP]
|
|
|
|
|
| style="text-align:center;" | Blackberry
|
|
|-
| style="text-align:left;" | [https://wiki.voip.ms/w/index.php?title=Softphones&stable=1#Fring Fring]
| style="text-align:center;" | X
| style="text-align:center;" | X
| style="text-align:center;" | X
| style="text-align:center;" | X
| style="text-align:center;" | X
|
| style="text-align:center;" | X
| style="text-align:center;" | X
|-
| style="text-align:left;" | [https://wiki.voip.ms/w/index.php?title=Softphones&stable=1#Jami Jami]
| style="text-align:center;" | X
| style="text-align:center;" | X
| style="text-align:center;" | X
| style="text-align:center;" | X
| style="text-align:center;" | X
| style="text-align:center;" |
| style="text-align:center;" | X
| style="text-align:center;" | X
|-
|-
| style="text-align:left;" | [https://wiki.voip.ms/w/index.php?title=Softphones&stable=1#Jitsi Jitsi]
|
|
|
| style="text-align:center;" | X
| style="text-align:center;" | X
| style="text-align:center;" |
| style="text-align:center;" | X
| style="text-align:center;" | X
|-
| style="text-align:left;" | [https://wiki.voip.ms/w/index.php?title=Softphones&stable=1#Kiax Kiax]
|
|
| style="text-align:center;" | X
| style="text-align:center;" | X
| style="text-align:center;" | X
|
|
|
|-
| style="text-align:left;" | [https://wiki.voip.ms/w/index.php?title=Softphones&stable=1#Linphone Linphone]
| style="text-align:center;" | X
| style="text-align:center;" | X
| style="text-align:center;" | X
| style="text-align:center;" | X
| style="text-align:center;" | X
| style="text-align:center;" |
| style="text-align:center;" | X
| style="text-align:center;" | X
|-
| style="text-align:left;" | [https://wiki.voip.ms/w/index.php?title=Softphones&stable=1#NetDial_Sip_Phone NetDial Sip Phone]
| style="text-align:center;" | X
| style="text-align:center;" | X
|
|
|
|
| style="text-align:center;" | X
|
|-
| style="text-align:left;" | [https://wiki.voip.ms/w/index.php?title=Softphones&stable=1#MicroSIP MicroSIP]
|
|
| style="text-align:center;" | X
| style="text-align:center;" | X
| style="text-align:center;" | X
|
| style="text-align:center;" | X
| style="text-align:center;" | X
|-
| style="text-align:left;" | [https://wiki.voip.ms/w/index.php?title=Softphones&stable=1#MizuPhone MizuPhone]
|
|
| style="text-align:center;" | X
|
|
|
| style="text-align:center;" | X
| style="text-align:center;" | X
|-
| style="text-align:left;" | [https://wiki.voip.ms/w/index.php?title=Softphones&stable=1#MizuDroid MizuDroid]
| style="text-align:center;" | X
|
|
|
|
|
| style="text-align:center;" | X
| style="text-align:center;" | X
|-
| style="text-align:left;" | [https://wiki.voip.ms/w/index.php?title=Softphones&stable=1#NinjaLite NinjaLite]
|
|
| style="text-align:center;" | X
|
|
|
|
|
|-
| style="text-align:left;" | [https://wiki.voip.ms/w/index.php?title=Softphones&stable=1#PhonerLite PhonerLite]
|
|
| style="text-align:center;" | X
|
|
|
| style="text-align:center;" | X
| style="text-align:center;" | X
|-
| style="text-align:left;" | [https://wiki.voip.ms/w/index.php?title=Softphones&stable=1#PortGo PortGo]
|
|
| style="text-align:center;" | X
|
|
|
|
|
|-
| style="text-align:left;" | [https://wiki.voip.ms/w/index.php?title=Softphones&stable=1#QuteCom QuteCom]
|
|
| style="text-align:center;" | X
| style="text-align:center;" | X
| style="text-align:center;" | X
|
|
|
|-
| style="text-align:left;" | [https://wiki.voip.ms/w/index.php?title=Softphones&stable=1#SessionTalk SessionTalk]
| style="text-align:center;" | X
| style="text-align:center;" | X
|
|
|
|
|
| style="text-align:center;" | X
|-
| style="text-align:left;" | [https://wiki.voip.ms/w/index.php?title=Softphones&stable=1#SIP_Droid SIP Droid]
| style="text-align:center;" | X
|
|
|
|
|
|
|
|-
| style="text-align:left;" | [https://wiki.voip.ms/w/index.php?title=Softphones&stable=1#SIPnetic SIPnetic]
| style="text-align:center;" | X
|
|
|
|
|
|
|
|-
| style="text-align:left;" | [https://wiki.voip.ms/w/index.php?title=Softphones&stable=1#SJPhone SJPhone]
|
|
| style="text-align:center;" | X
|
|
| style="text-align:center;" | Windows Mobile
|
|
|-
| style="text-align:left;" | [https://wiki.voip.ms/w/index.php?title=Softphones&stable=1#Softphone.Pro Softphone.Pro]
|
|
| style="text-align:center;" | X
| style="text-align:center;" | X
|
|
| style="text-align:center;" | X
|
|-
| style="text-align:left;" | [https://wiki.voip.ms/w/index.php?title=Softphones&stable=1#Taki Taki]
|
|
|
|
|
| style="text-align:center;" | Blackberry
|
|
|-
| style="text-align:left;" | [https://wiki.voip.ms/w/index.php?title=Softphones&stable=1#Telephone Telephone]
|
|
|
| style="text-align:center;" | X
|
|
|
|
|-
| style="text-align:left;" | [https://wiki.voip.ms/w/index.php?title=Softphones&stable=1#tSIP tSIP]
|
|
| style="text-align:center;" | X
| style="text-align:center;" | X
|
|
|
| style="text-align:center;" | X
|-
| style="text-align:left;" | [https://wiki.voip.ms/w/index.php?title=Softphones&stable=1#X-Lite X-Lite]
|
|
| style="text-align:center;" | X
| style="text-align:center;" | X
| style="text-align:center;" | X
|
|
|
|-
| style="text-align:left;" | [https://wiki.voip.ms/w/index.php?title=Softphones&stable=1#ZoiPer ZoiPer]
| style="text-align:center;" | X
| style="text-align:center;" | X
| style="text-align:center;" | X
| style="text-align:center;" | X
| style="text-align:center;" | X
|
| style="text-align:center;" | X
| style="text-align:center;" | X
|-
| style="text-align:left;" | [https://wiki.voip.ms/w/index.php?title=Softphones&stable=1#ZoiPer_Communicator ZoiPer Communicator]
|
|
| style="text-align:center;" | X
| style="text-align:center;" | X
| style="text-align:center;" | X
|
|
|
|}

==VoIP.ms Softphone==

[[File:VoipmsLogoSymbolSquare.png|300px|left|VoIP.ms Softphone]]

'''Product:''' VoIP.ms Softphone

'''Company:''' [https://www.voip.ms VoIP.ms]

'''OS:''' Android, iOS (coming soon)

'''SMS Compatible''': Yes

'''Overview:'''
Our very own mobile app designed to bring even more flexibility to your experience as our customer, so you can enjoy seamless, secure calling with push notifications at no extra cost.

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

==Acrobits==

[[File:Acrobits_logo_new.png|300px|left|Acrobits]]

[[File:Acrobitscert.png|180px|right|Acrobits|link=https://www.acrobits.net//?utm_campaign=itsp-partners&utm_medium=cp-certified-logo&utm_source=voipms]]

'''Product:''' Acrobits

'''Company:''' [https://www.acrobits.net//?utm_campaign=itsp-partners&utm_medium=cp-certified-logo&utm_source=voipms Acrobits]

'''OS:''' IOS, Android

'''SMS Compatible''': Yes

'''Overview:'''
Acrobits is a mobile software development company with a focus on SIP Clients for mobile devices; their SIP client, Acrobits Softphone, is the leading app of it's type on the iTunes App Store.

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

==AntiSIP (Mobile)==

[[File:Antisip-logo.png|300px|left|Acrobits]]

'''Product:''' AntiSIP

'''Company:''' [https://www.antisip.com/antisip-company.html AntiSIP]

'''OS:''' Android

'''Overview:'''
AntiSIP is a mobile software SIP, WebRTC, Audio, Video on all platforms!

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

==Blink (Desktop)==

[[File:Blink-logo.png|300px|left|Blink]]

'''Product:''' Blink

'''Company:''' [https://icanblink.com/ Blink]

'''OS:''' MAC & Linux

'''Overview:'''
Blink is a softphone available for MAC and Linux users, offering a variety of features free to use like the possibility to integrate System Address Book, Answering machine, Call: recording, transfer, waiting, Message Waiting Indicator (MWI) among many other options.

For a list of all the features available with Blink, consult the following link

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

==Bria Mobile==

[[File:Softphone Clients - Smartphones.jpg|350px|left|Bria Mobile|link=https://www.counterpath.com/?utm_campaign=itsp-partners&utm_medium=cp-certified-logo&utm_source=voipms]]

[[File:CounterPath ITSP Certified Partner 2017.png|250px|right|CounterPath|link=https://www.counterpath.com/?utm_campaign=itsp-partners&utm_medium=cp-certified-logo&utm_source=voipms]]

'''Product:''' Bria Mobile

'''Company:''' [https://www.counterpath.com/?utm_campaign=itsp-partners&utm_medium=cp-certified-logo&utm_source=voipms CounterPath]

'''OS:''' Windows, MacOS, Linux (Ubuntu), iOS, Android

'''SMS Compatible''': Yes

'''Overview:'''
Bria 5 is a carrier-grade next generation softphone application that enables you to manage your communications easily and efficiently
– all from your computer desktop. Replacing or complementing your hard phone, the Bria softphone allows you to make VoIP and Video calls over IP, see when your contacts are available, send Instant Messages and transfer files with ease and efficiency.

Bria Mobile for Andoid and IOS is a highly secure, standards-based mobile VoIP softphone that works over both 3G and Wi-Fi networks.

Using the device’s existing contact list, Bria Android Edition facilitates easy and effective communication management with an intuitive interface. Deskphone-class calling functionality includes the ability to swap between two calls, merge calls and perform attended and unattended transfers.

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

==Bria Solo==

[[File:Briasolologo.jpg|350px|left|Bria Solo|link=https://www.counterpath.com/?utm_campaign=itsp-partners&utm_medium=cp-certified-logo&utm_source=voipms]]

[[File:CounterPath ITSP Certified Partner 2017.png|250px|right|CounterPath|link=https://www.counterpath.com/?utm_campaign=itsp-partners&utm_medium=cp-certified-logo&utm_source=voipms]]

'''Product:''' Bria Solo

'''Company:''' [https://www.counterpath.com/?utm_campaign=itsp-partners&utm_medium=cp-certified-logo&utm_source=voipms CounterPath]

'''OS:''' Windows, MacOS, iOS, Android

'''SMS Compatible''': Yes

'''Overview:'''
Bria Solo provides desktop and mobile softphone applications that facilitate best-in-class calling and robust call control features.

Whether used for inbound or outbound calls, Bria makes it easy to integrate contacts from various sources, streamline communications, and manage calls from one easy-to-use interface.

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

==Bria Teams==

[[File:Bria-teams-thumb.png|350px|left|Bria Teams|link=https://www.counterpath.com/?utm_campaign=itsp-partners&utm_medium=cp-certified-logo&utm_source=voipms]]

[[File:CounterPath ITSP Certified Partner 2017.png|250px|right|CounterPath|link=https://www.counterpath.com/?utm_campaign=itsp-partners&utm_medium=cp-certified-logo&utm_source=voipms]]

'''Product:''' [https://www.softphone.com/ Bria Teams]

'''Company:''' [https://www.counterpath.com/?utm_campaign=itsp-partners&utm_medium=cp-certified-logo&utm_source=voipms CounterPath]

'''OS:''' Windows, MacOS, iOS, Android

'''SMS Compatible''': Yes

'''Overview:'''
Team Communication made Easy
Bria Teams helps you regain productivity by streamlining all team communications into one application across your devices. With Bria's provisioning, it's easy to get your team talking.

It offers a softphone that helps you gain productivity by streamlining all team communications into one application. It has all the tools you need in one interface, across your devices and can be managed by one simple dashboard.

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

==Dialer+==

[[File:Dialerplus.png|200px|thumb|left|Dialer+]]

'''Product:''' cd26

'''Company:''' cd26

'''OS:''' IOS

'''Overview:'''
Dialer+ is a SIP-based softphone for iPhone , iPod touch and iPad

Standard Phone Features
* Fast Contact Search – Use phone pad to find contacts by names, initials or any part of the numbers.
• iOS 4.0+ Background Support
• Supports GSM calls when the network is unavailable or slowly
• Multiple Account Support
• Auto switch to best service provider when use multiple account
• Pre-configured VoIP providers list
• Call display
• Speakerphone, Mute and Hold
• Call history
• Contact List and Contact Favorites - leveraging the iPhone Contacts
• Ringtones and contact avatars
• Multiple incoming Call Support - swap between two active calls; merge and split calls
• Audio codecs include G.711,G.722, iLBC ,Speex and GSM, Make an in app purchase to add G.729 for great performance over 3G networks
• Support for RFC 2833 DTMF
• VPN Support
• uPNP Support
• iPhone 4 Retina Display Support

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

==Ekiga==

[[File:Ekigaw0.png|200px|thumb|left|Ekiga Softphone]]

'''Product:''' Ekiga

'''Company:''' Ekiga

'''OS:''' Windows, Linux

'''Overview:'''Ekiga is a software phone and video conferencing application. It allows you to hear and see your friends for free using your computer and Internet. You can also chat with them and see if they are online or not. It works on GNU/Linux and Windows.

More specifically, Ekiga is a VoIP, IP Telephony, and Video Conferencing application that allows you to make audio and video calls to remote users with SIP or H.323 compatible hardware and software.

It supports many audio and video codecs and all modern VoIP features for both SIP and H.323. Ekiga is the first Open Source application to support both H.323 and SIP, as well as audio and video. Ekiga was formerly known as GnomeMeeting.

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

==FgVoIP==

[[File:Fgvoip0.jpeg|200px|thumb|left|FgVoIP]]

'''Product:''' FgVoIP

'''Company:''' FG Microtec

'''OS:''' BlackberryOS

'''Overview:''' fg's BlackBerry SIP VoIP Client is a feature rich soft phone application for BlackBerry devices. It can register to any SIP VoIP provider or IP/SIP capable PBX using a WiFi connection in office, campus, home, hotel rooms, public hot-spots etc.

Its much like having a BlackBerry desktop phone in your pocket, providing significantly higher overall productivity while at the same time lowering telephony costs.

Experience the freedom of choosing your own VoIP provider whenever you want, wherever you want. All you need is a WiFi network and a VoIP account.

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

==Jami==

[[File:JamiLogo.png|left|Fring]]

'''Product:''' Jami

'''Company:''' Jami

'''OS:''' Windows, Mac, Linux, iOS and Android

Jami is an open source universal communication software that respects the freedom and privacy of its users. At its core, Jami aims to be a complete, decentralized, secure and robust means of communication for people.

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

==Fring==

[[File:Fring00.jpg|left|Fring]]

'''Product:''' Fring

'''Company:''' Fring

'''OS:''' Windows, MacOS, IOS, Linux, Android, Symbian

'''Overview:'''
Fring mobile communication service that gives users internet-rich communication from their mobile phones.

Fring members make FREE video calls, voice calls, live chat to other fringsters and to friends on other social networks.

You can now use your SIP account even from non-SIP enabled phones.

Choose whichever SIP provider you have and in just few clicks, start using this account to make calls to landlines, GSM phones, or to other SIP providers.

Supports Symbian S60, iPhone/ iPod touch, Android, Windows Mobile, MeeGo, Linux, J2MEand proprietary OS.

Fring allows you to easily switch between them, without having to reconfigure your handset through a nightmarish process.

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

==Jitsi==

[[File:Logo2.png|200px|left|Jitsi]]

'''Product:''' Jitsi

'''Company:''' Jitsi

'''OS:''' Windows, MacOS, Linux

'''SMS Compatible''': Yes

'''Overview:''' Jitsi (formerly SIP Communicator) is an audio/video and chat communicator that supports protocols such as SIP, XMPP/Jabber, AIM/ICQ, Windows Live, Yahoo! and many other useful features.

Jitsi is an Open Source / Free Software, and is available under the terms of the LGPL.

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

==Kiax==

[[File:Kiax3.png|left|200px|Kiax]]

'''Product:''' Kiax

'''Company:''' Kiax

'''OS:''' Windows, MacOS, Linux

'''Overview:'''
Kiax is a softphone (soft phone, VoIP client) with a simple and comfortable user interface for making VoIP calls to Asterisk PBX. It depends on the iaxclient library to use Asterisk's IAX2 protocol for easy call configuration and audio setup.

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

==Linphone==

[[File:Linphone_Icon.png|left|200px|Linphone]]

'''Product''': Linphone

'''Company''': Linphone

'''OS:''' Windows, Linux, MacOSX, IPhone, Android.

'''SMS Compatible''': Yes

'''App Storage Consumption:''' 8.09mb (ability to move app to SD card. Data uses an additional ~0.89mb approx.)

'''Overview:''' Linphone is an internet phone or Voice Over IP phone (VoIP). With linphone you can communicate freely with people over the internet, with voice, video, and text instant messaging. Linphone makes use of the SIP protocol, an open standard for internet telephony. You can use Linphone with any SIP VoIP operator.

[[Linphone|See Configuration Details for PC]]
<br>
[[Linphone_iOS|See Configuration Details for iOS]]
<br>
[[Linphone_Android|See Configuration Details for Android]]
<div style="width:100%;overflow:hidden;clear:both">

</div>

==NetDial Sip Phone==

[[File:NetDial.jpg|left|200px|Linphone]]

'''Product''': NetDial Sip Phone

'''Company''': NeoMecca

'''OS:''' iOS, Android.

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

==MicroSIP==

[[File:MicroSIPlogo.png|left|200px|MicroSIP]]

'''Product:''' MicroSIP

'''Company:''' MicroSIP

'''OS:''' Windows XP/Vista/7/8/8.1/10, Linux, macOS

'''SMS Compatible''': Yes

'''Overview:'''
MicroSIP is a softphone that works on Windows Operating Systems. It is a SIP Client (softphone) that enables users to make and receive VoIP calls, enabling them to use their devices as an IP-PBX phone extension in their office or anywhere else in the world as well as being really easy to configure.

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

==MizuPhone==

[[File:MizuLogo.png|left|200px|MizuPhone]]

'''Product:''' MizuPhone

'''Company:''' Mizutech

'''OS:''' Windows

'''Overview:'''
Mizu SoftPhone is our award winner professional VoIP client application based on the open standard SIP protocol with an easy to use modern interface. With Mizu SoftPhone you can connect to any SIP server on the public internet or on your local area network.

Multiple accounts and multiple SIP server registrations combined with a powerful dial plan can minimize your telecom bills while using the greatest features from the VoIP industry.

Features include built-in encryption, IM, presence, HD Video, history with voice records, skype like voice quality using ultra-wideband codecs and much more.

A free edition is also available based on the same engine as our corporate sip softphone but it has some features disabled (like G.729 and wideband codecs, multiple accounts and file transfer over SIP).

[[MizuPhone_Classic_Edition|See Configuration Details for MizuPhone Classic Edition]]
<br>
[[MizuPhone_Modern_Edition|See Configuration Details for MizuPhone Modern Edition]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==MizuDroid==

[[File:MizuDroid_logo.png|left|200px]]

'''Product:''' MizuDroid

'''Company:''' Mizutech

'''OS:''' Android

'''Overview:'''
MizuDroid is an unlocked VoIP softphone for Android mobile phones and tablets based on open standards, compatible with all VoIP providers, software and devices using the SIP protocol.

[[MizuDroid|See Configuration Details for MizuDroid]]

==NinjaLite==

[[File:Ninja6.png|left|200px|NinjaLite]]

'''Product:''' NinjaLite

'''Company:''' Global IP Telecommunications

'''OS:''' Windows

'''Overview:'''
Ninja Lite offers Audio- and video telephony for everybody, Ninja features individual skins and ringtones!, Ninja is natively Vista 32bit /64 bit and Windows7 compatible.

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

==PhonerLite==

[[File:PhonerLiteLogo2.0.png|left|200px|PhonerLite]]

'''Product:''' PhonerLite

'''Company:''' PhonerLite

'''OS:''' Windows

'''Overview:'''
PhonerLite is a clearly arranged application for Windows. PhonerLite enables your PC to use it for internet telephony (VoIP , Voice over IP ). Pre-conditions are a full-duplex sound card , a microphone and speakers (alternativelya headset), an internet connection and a registration at a provider supporting the protocol SIP .PhonerLite supports several SIP profiles, each configurable independently. In thesame way the integrated phone book and call log are easy to use.

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

==PortGo==

[[File:Portgo3.png|left|200px|PortGo]]

'''Product:''' PortGo

'''Company:''' PortSIP Solutions, Inc

'''OS:''' Windows (desktop and mobile)

'''Overview:'''
PortGo is the newest SIP softphone from PortSIP, it's built base on PortSIP VoIP SDK, allowing users to enjoy multimedia communications in a dynamic way.

Featuring an intuitive interface, PortGo is expanding the softphone experience by making it even easier to make VoIP and Video over IP calls, see when your contacts are available and send Instant Messages. PortGo features an IM interface which focuses on your contacts and friends.

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

==QuteCom==

[[File:Qutecom3.png|left|200px|QuteCom]]

'''Product:''' QuteCom

'''Company:''' QuteCom

'''OS:''' Windows,MacOS,Linux

'''Overview:'''
QuteCom is a community of enthusiasts and developers, creating free software products related to communication over IP. The flagship product of the QuteCom project is a softphone which allows you to make free PC to PC video and voice calls, and to integrate all your IM contacts in one place.

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

==SessionTalk==

[[File:SessionTalk logo.png|left|200px|SessionTalk]]

'''Product:''' SessionTalk

'''Company:''' SessionTalk

'''OS:''' Android, iOS

'''Overview:'''
SessionTalk Mobile is a feature rich mobile SIP Client for iOS and Android handsets Voice calls and SMS Messaging* support.

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

==SIP Droid==

[[File:Sipdroid_00.png|200px|thumb|left|SIP Droid]]

'''Product:''' SIP Droid

'''Company:''' SIP Droid

'''OS:''' Android

'''App Storage Consumption:''' 1.85mb (ability to move app to SD card. Data uses an additional ~4kb approx.)

'''Overview:''' SIPDroid is a java based, open source SIP client that has recently been developed for use with mobile devices based on Google’s Android platform.

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

==SIPnetic==

[[File:Sipnetic-VoIPms.png|200px|thumb|left|SIPnetic]]

'''Product:''' SIP Droid

'''Company:''' SIP Droid

'''OS:''' Android

'''Overview:''' Sipnetic is a free VoIP softphone based on the SIP protocol. It allows you to connect to your VoIP provider, cloud PBX, or an enterprise telephony server. Voice over IP (VoIP) technology offers many attractive advantages over the legacy telephony. It is often cheaper and provides some advanced features, such as video calls, instant messages, or data encryption.

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

==SJPhone==

[[File:Sjphone0.png|200px|thumb|left|SJPhone]]

'''Product:''' SJPhone

'''Company:''' SJ Labs, Inc.

'''OS:''' Windows,Windows Mobile

'''Overview:'''
SJphone is a VOIP softphone that allows you to speak with any other softphone, any stand-alone IP-phone, or using ITSP with any traditional wired or mobile phone.

It supports both SIP and H.323 standards and is fully inter-operable with most major VOIP vendors and ITSP.

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

==Softphone.Pro==

[[File:SoftphoneProStats.png|350px|thumb|left|Softphone.Pro]]

'''Product:''' Softphone.Pro

'''Company:''' Vedisoft

'''OS:''' Windows, macOS

'''Overview:'''
Softphone.Prop is a great solution for call center agents, sales pros and support teams.

'''Main features:'''

• Easy integration with CRM
• 32 SIP accounts supported
• Call recording and screen capture
• Click-to-call and auto and easy 2-click transfer
• Voicemail Drop and Prerecorded Greeting on Answer
• ACW and BLF status control
• Professional headsets support
• Listen, whisper, barge
• Customizable interface
• Hotkeys

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

==Taki==

[[File:blackberry_z30_taki_oncall.png|200px|thumb|left|Taki]]

'''Product:''' Taki

'''Company:''' Taki

'''OS:''' BlackBerry 10, BlackBerry PlayBook

'''Project site:''' http://taki.sourceforge.net/

'''App in the AppWorld:''' http://appworld.blackberry.com/webstore/content/130295

'''Overview:''' Taki is a native SIP softphone for BlackBerry® PlayBook™ and
BlackBerry® 10 platforms.

'''Main Features:'''
- Multiple SIP accounts support
- Multiple simultaneous calls
- Call recording
- Conference calls
- Advanced call control: Transfer, Hold, Mute, Reject, Redial, switch
between multiple active calls
- Speaker phone support
- Address book. Integration with local contacts on BlackBerry 10 phones
- Comprehensive SIP settings (port binding, outbound proxy, registrar,
codecs priority, NAT traversal, STUN, Presence, and more)
- Transports: UDP/TCP
- Audio codecs: G.711a/u, GSM, iLBC, G.722 (HD), Speex (NB, WB, UWB)
- Built for BlackBerry

[[Taki|See Configuration Details]]

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

==Telephone==

[[File:telephone-user-interface.png|200px|thumb|left|Telephone]]

'''Product:''' Telephone

'''Company:''' 64 Characters

'''OS:''' OS X 10.9 or later

'''Project site:''' http://www.64characters.com/telephone/

'''Mac App Store:''' https://itunes.apple.com/us/app/telephone/id406825478

'''Overview:''' Telephone is a fully-featured SIP client with a minimalist user interface.

<br>

'''Main Features:'''

* Multiple SIP accounts
* Multiple simultaneous calls w/conferencing
* Call control: Transfer, Hold, Mute, Reject, Redial, Call Waiting, multiple active calls
* Speaker phone/headset support (via Mac sound device)
* Integration with Contacts app, dial by name, autocomplete, click to dial
* Send DTMF tones
* Languages: English, German, Russian

[[Telephone|See Configuration Details]]

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

==tSIP==

[[File:TSIP_logo.jpg|200px|thumb|left|Telephone]]

'''Product:''' tSIP

'''Project site:''' https://tomeko.net/software/SIPclient/

'''Overview:''' tSIP is a one-man dedicated project throughout multiple years of development and updates, free for anyone to use.

<br>

'''Main Features:'''

* Call control: Transfer, Hold, Mute, Reject, Redial
* Send DTMF tones
* Languages: English, German, Spanish, French, Polish, Portuguese, Russian

[[TSIP|See Configuration Details]]

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

==X-Lite==

[[File:Xlite_00.png|200px|thumb|left|X-Lite]]

'''Product:''' X-Lite

'''Company:''' CounterPath

'''OS:''' Windows, MacOS, Linux

'''SMS Compatible''': Yes

'''Overview:''' CounterPath's X-Lite helps you seamlessly transition from a traditional phone environment into the world of Voice over IP.

The latest release of X-Lite provides a completely redesigned interface that allows for a contact-centric or dialpad-centric user experience, or a combination of the two. It also provides you with some of the most popular features of our fully loaded Bria and eyeBeam softphones so you can take them for a test drive before you make your purchase.

Having a simple voice conversation, you’ll soon see why having a softphone on your desktop or laptop is the ultimate communications experience.

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

==ZoiPer==

[[File:Zoiper0.png|300px|thumb|left|Zoiper Classic]]
[[File:zlogo.jpg|300px|thumb|right|https://www.zoiper.com/en]]

'''Product:''' ZoiPer

'''Company:''' ZoiPer

'''OS:''' Windows, Mac OS, Android, iOS

'''SMS Compatible''': Yes

'''Overview:''' IAX & SIP multilanguage softphone is a VoIP soft client, meant to work with any IP-based communications systems and infrastructure.

Zoiper is available for Windows, Mac OS X and Linux and supports the following languages: English, German, Spanish, French, Dutch, Portuguese, Russian, Chinese, Japanese, Italian, Polish, Magyar.

*[[Zoiper_Classic|See Configuration Details for Zoiper Classic for PC/MAC]]<br />
*[[Zoiper_5| See Configuration Details for ZoiPer 5 for PC/MAC]]<br />
*[[Zoiper_5_Pro| See Configuration Details for ZoiPer 5 PRO for PC`MAC]]
*[[ZoIPer_for_Android|See Configuration Details for android]]<br/>
*[[ZoIPer_for_iOS|See Configuration Details for iOS]]<br/>
<div style="width:100%;overflow:hidden;clear:both"></div>
[[category:softphones]]

==ZoiPer Communicator==

[[File:Zoiperco1.png|200px|thumb|left|Zoiper Communicator]]
[[File:zlogo.jpg|300px|thumb|right|https://www.zoiper.com/en]]

'''Product:''' Zoiper Communicator

'''Company:''' Zoiper

'''OS:''' Windows, MacOS, Linux, Solaris

'''SMS Compatible''': Yes

'''Overview:''' Zoiper Communicator IAX & SIP softphone is a converged Internet communication tool combining high-quality voice and video calls, fax, instant messaging and presence through a contact-centric intuitive interface.

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

SMS-MMS-ES

2024-11-15T18:04:56Z

John: /* Limitaciones del Servicio */

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

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

= Artículos del Blog =
:* [https://wiki.voip.ms/article/Benefits_of_SMS/MMS_Campaigns_vs._Email_Marketing Beneficios de las campañas de SMS/MMS vs. Email Marketing]
:* [https://wiki.voip.ms/article/How_to_Generate_Leads_with_SMS/MMS%3F ¿Cómo generar clientes potenciales con SMS/MMS?]
:* [https://wiki.voip.ms/article/Instantly_Boost_Your_Customer_Experience,_Allow_Your_Customers_To_Text_You Aumente al instante su experiencia como cliente, permita que sus clientes le envíen mensajes de texto.]

= Limitaciones del Servicio =

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

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

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

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

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

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

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

</div>

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

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

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

</div>

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

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

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

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

==Configuración del servicio SMS/MMS==

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

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

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

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

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

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

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

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

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

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

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

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

</div>

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

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

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

'''Paso 1'''

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

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

'''Paso 2'''

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

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

'''Paso 3'''

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

'''Paso 4'''

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

== Historial de SMS / MMS ==

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

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

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

= SMS utilizando un servidor SMPP. =

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

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

Por ejemplo, si tiene una promoción para su negocio que ofrece un descuento durante un determinado lapso de tiempo y que sus clientes han dado su conformidad para recibir este tipo de promociones vía SMS, podría enviar su promoción a sus clientes de una manera más rápida que si lo hiciera desde su portal de clientes, aumentando así la eficiencia.
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'''NOTA''' 💬: ➜ Tenga en cuenta que el envío y la recepción de mensajes SMS se cobrará a nuestra tarifa habitual de 0.0075 dólares por mensaje.'''
</div>

==Requisitos==

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

== Configuración en su servidor SMPP ==

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

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

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

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

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

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

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

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

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

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

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

==Centro de mensajes SMS para móviles==

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

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

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

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

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

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

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

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

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

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

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

== Enviando ==

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

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

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

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

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

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

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

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

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

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

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

== Enviar SIP / SMS con FreePBX ==

<div style="margin-left:50px;
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'''NOTES''' 💬: ➜ AEL ENVÍO DE MMS UTILIZANDO LA CUENTA SIP NO ES POSIBLE EN ESTE MOMENTO.'''
</div>

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

[[category:guides]]

SMS-MMS-FR

2024-11-15T18:04:33Z

John: /* Limites du service */

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

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

= Articles de blog =
:* [https://wiki.voip.ms/article/Benefits_of_SMS/MMS_Campaigns_vs._Email_Marketing Avantages des campagnes SMS/MMS par rapport au marketing par courriel]
:* [https://wiki.voip.ms/article/How_to_Generate_Leads_with_SMS/MMS%3F Comment engendrer des prospects avec les SMS/MMS ?]
:* [https://wiki.voip.ms/article/Instantly_Boost_Your_Customer_Experience,_Allow_Your_Customers_To_Text_You Améliorez instantanément votre expérience client, permettez à vos clients de vous envoyer des SMS!]

= Limites du service =

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

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width:70vw;">

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

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

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

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

</div>

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

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

</div>

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

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

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

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

==Configuration du service SMS/MMS==

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

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

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

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

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

:*'''SMS/MMS Forward''': Vous pouvez faire transférer vos messages SMS vers votre téléphone portable ou tout autre numéro qui supporte les SMS. Avec cette option, vous recevrez les messages SMS de notre système vers le numéro de téléphone configuré.
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'''NOTE''' 💬: ➜ Le numéro d'identification de l'appelant envoyée à un numéro mobile renvoyé sera le numéro DID qui a reçu le message SMS. Il n'est pas possible de renvoyer l'ID de l'appelant original pour le moment.
</div>

:*'''SMS/MMS SIP Account''': Vous pouvez faire transférer vos messages SMS vers le compte SIP de votre choix. Ils seront envoyés par notre passerelle SMS/SIP en tant que SIP MESSAGE.
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'''NOTE''' 💬: ➜ La destination (champ d'en-tête "To") du SIP MESSAGE sera votre nom de compte, au lieu du numéro de destination du SMS original.
</div>

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

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

</div>

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

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

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

'''Step 1'''

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

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

'''Step 2'''

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

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

'''Step 3'''

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

'''Step 4'''

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

== Historique SMS/MMS ==

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

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

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

= SMS à l'aide d'un serveur SMPP =

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

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

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

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

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

==Exigences==

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

== Configuration on your SMPP Server ==

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

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

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

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

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

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

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

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

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

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

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

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

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

==Centre de messages SMS pour mobile==

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

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

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

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

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

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

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

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

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

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

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

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

== Envoi ==

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

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

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

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

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

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

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

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

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

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

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

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

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

== Envoyer des SIP/SMS avec FreePBX ==

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

[[category:guides]]

SMS-MMS-ES

2024-11-15T17:49:27Z

John: /* Limitaciones del Servicio */

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

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

= Artículos del Blog =
:* [https://wiki.voip.ms/article/Benefits_of_SMS/MMS_Campaigns_vs._Email_Marketing Beneficios de las campañas de SMS/MMS vs. Email Marketing]
:* [https://wiki.voip.ms/article/How_to_Generate_Leads_with_SMS/MMS%3F ¿Cómo generar clientes potenciales con SMS/MMS?]
:* [https://wiki.voip.ms/article/Instantly_Boost_Your_Customer_Experience,_Allow_Your_Customers_To_Text_You Aumente al instante su experiencia como cliente, permita que sus clientes le envíen mensajes de texto.]

= Limitaciones del Servicio =

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

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

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

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

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

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

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

</div>

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

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

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

</div>

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

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

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

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

==Configuración del servicio SMS/MMS==

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

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

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

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

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

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

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

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

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

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

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

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

</div>

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

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

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

'''Paso 1'''

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

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

'''Paso 2'''

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

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

'''Paso 3'''

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

'''Paso 4'''

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

== Historial de SMS / MMS ==

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

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

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

= SMS utilizando un servidor SMPP. =

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

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

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

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

==Requisitos==

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

== Configuración en su servidor SMPP ==

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

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

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

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

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

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

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

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

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

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

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

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

==Centro de mensajes SMS para móviles==

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

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

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

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

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

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

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

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

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

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

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

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

== Enviando ==

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

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

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

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

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

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

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

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

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

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

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

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

== Enviar SIP / SMS con FreePBX ==

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

[[category:guides]]

SMS-MMS-FR

2024-11-15T17:48:36Z

John: /* Limites du service */

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

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

= Articles de blog =
:* [https://wiki.voip.ms/article/Benefits_of_SMS/MMS_Campaigns_vs._Email_Marketing Avantages des campagnes SMS/MMS par rapport au marketing par courriel]
:* [https://wiki.voip.ms/article/How_to_Generate_Leads_with_SMS/MMS%3F Comment engendrer des prospects avec les SMS/MMS ?]
:* [https://wiki.voip.ms/article/Instantly_Boost_Your_Customer_Experience,_Allow_Your_Customers_To_Text_You Améliorez instantanément votre expérience client, permettez à vos clients de vous envoyer des SMS!]

= Limites du service =

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

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

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

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

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

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

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

</div>

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

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

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

</div>

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

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

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

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

==Configuration du service SMS/MMS==

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

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

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

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

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

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

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

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

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

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

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

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

</div>

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

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

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

'''Step 1'''

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

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

'''Step 2'''

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

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

'''Step 3'''

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

'''Step 4'''

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

== Historique SMS/MMS ==

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

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

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

= SMS à l'aide d'un serveur SMPP =

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

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

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

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

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

==Exigences==

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

== Configuration on your SMPP Server ==

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

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

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

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

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

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

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

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

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

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

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

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

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

==Centre de messages SMS pour mobile==

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

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

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

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

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

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

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

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

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

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

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

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

== Envoi ==

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

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

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

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

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

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

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

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

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

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

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

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

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

== Envoyer des SIP/SMS avec FreePBX ==

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

[[category:guides]]

SMS-MMS-FR

2024-11-15T17:47:05Z

John: /* Limites du service */

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

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

= Articles de blog =
:* [https://wiki.voip.ms/article/Benefits_of_SMS/MMS_Campaigns_vs._Email_Marketing Avantages des campagnes SMS/MMS par rapport au marketing par courriel]
:* [https://wiki.voip.ms/article/How_to_Generate_Leads_with_SMS/MMS%3F Comment engendrer des prospects avec les SMS/MMS ?]
:* [https://wiki.voip.ms/article/Instantly_Boost_Your_Customer_Experience,_Allow_Your_Customers_To_Text_You Améliorez instantanément votre expérience client, permettez à vos clients de vous envoyer des SMS!]

= Limites du service =

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

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

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

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

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

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

</div>

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

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

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

</div>

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

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

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

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

==Configuration du service SMS/MMS==

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

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

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

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

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

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

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

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

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

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

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

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

</div>

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

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

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

'''Step 1'''

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

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

'''Step 2'''

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

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

'''Step 3'''

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

'''Step 4'''

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

== Historique SMS/MMS ==

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

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

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

= SMS à l'aide d'un serveur SMPP =

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

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

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

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

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

==Exigences==

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

== Configuration on your SMPP Server ==

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

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

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

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

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

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

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

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

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

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

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

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

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

==Centre de messages SMS pour mobile==

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

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

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

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

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

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

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

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

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

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

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

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

== Envoi ==

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

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

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

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

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

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

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

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

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

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

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

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

== Envoyer des SIP/SMS avec FreePBX ==

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

[[category:guides]]

SMS-MMS

2024-11-15T17:22:11Z

John: /* Service Limitations */

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

SMS means '''S'''hort '''M'''essage '''S'''ervice ('''SMS'''). With the addition of '''M'''ultimedia '''M'''essage '''S'''ervice ('''MMS''') to the service, users are now capable of sending and receiving media files ('''photos, videos, and audio''') as well as sending text messages up to 160 characters (which is the limit for an SMS). The MMS service also allows the user to send a message to multiple recipients at the same time.

= Blog Articles =
:* [https://wiki.voip.ms/article/Benefits_of_SMS/MMS_Campaigns_vs._Email_Marketing Benefits of SMS/MMS Campaigns vs. Email Marketing]
:* [https://wiki.voip.ms/article/How_to_Generate_Leads_with_SMS/MMS%3F How to Generate Leads with SMS/MMS?]
:* [https://wiki.voip.ms/article/Instantly_Boost_Your_Customer_Experience,_Allow_Your_Customers_To_Text_You Instantly Boost Your Customer Experience, Allow Your Customers to Text You!]

= Service Limitations =

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🏗️ ''' Please notice that the MMS service is in the Beta phase.'''
: It is important to us that you report any issues with this service by sending an email to [mailto:support@voip.ms support] so that the developers can get involved if necessary.
</div>

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⚠️ '''LIMITATIONS'''
: ➜ '''The limit of the capacity to send SMS/MMS via the portal is unlimited.'''
: ➜ The limit of sending SMS/MMS via the API is limited to '''<u>100 per day</u>'''. For any modification to the limit, please contact support or open a ticket via the customer portal.
: ➜ At the moment, receiving and reading MMS is only possible through the message center or by using the SMS/MMS Forward option. '''MMS through SIP protocol is not available.'''

: ➜ The SMS/MMS Service is only available for <u>'''US and Canadian DID and Toll-Free Numbers'''</u> marked with the distinctive '''SMS/MMS Icon'''.
: ➜ The SMS/MMS Service can only be used to send SMS/MMS Messages to <u>'''Canadian and US'''</u> 10 digit numbers at this time.
: ➜ SIP SMS messages will only be delivered to and accepted from SIP trunks using username/password authentication.
: ➜ SIP SMS messages use the SIP MESSAGE method. See [https://tools.ietf.org/html/rfc3428 RFC 3428] for details.
: ➜ At this time we '''<u>cannot guarantee</u> that Short Code SMS Messages''', which are usually 6 digits or less ''(e.g.: Skype, Bank Codes, TV Commercials etc...)'' will work.
: ➜ At this time we '''<u>cannot guarantee</u> that 2FA SMS Messages''', which are usually to confirm your identity and log into your service portal ''(e.g.: Microsoft, among other providers offering this feature)'' will work.
: ➜ At this time SMS messages from/to short code numbers '''<u>aren't supported</u> for Toll-Free numbers'''.
: ➜ We cannot guarantee that '''accents or special characters''' including non-Latin letters will be properly delivered.

: ➜ Public URL shorteners such as '''''bit.ly, tinyurl.com''''' etc, <font style="color:red">'''are widely restricted'''</font> by mobile providers as a response from the industry to combat SPAM campaigns and protect the consumer. Including this type of links into your messages '''<u>may result in a blocked or undelivered message</u>'''.
: ➜ For '''Ported In Numbers''' compatible with the feature: The SMS/MMS Functionality will be available up to '''48 hours after the porting process is marked as Completed'''.

: ➜ VoIP.ms reserves the right, at its sole discretion, to limit or disable the SMS/MMS service from any account that may present marketing patterns/automation patterns/bulk sending or receiving.

</div>

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'''💬 NOTE ''' If you have further questions don't hesitate to contact the Support Staff on the Live Chat or Ticket System.
:➜ '''SMS Messages have a cost of $0.0075 per incoming and outgoing message.'''
:➜ '''MMS Messages have a cost of $0.02 per incoming and outgoing message.'''

: ➜ '''Additionally, please note of the following restrictions for sending MMS:'''
:: ➜ Permitted attachment types are JPG, GIF, JPEG, PNG, MP3, WAV, MIDI, MP4 and 3GP.
:: ➜ The maximum size for each attachment is 1300 KB. ''(1.3 MB)''
:: ➜ You can attach up to 3 files.
:: ➜ Maximum characters in a text message is 2048.

</div>

=Using the SMS/MMS=
==Identifying a DID Number SMS/MMS capable==

The first thing you need to know is that this feature is only available for local US '''(incl. Alaska, Hawaii and Puerto Rico)''' and Canadian numbers at this time. You will note that some numbers have a little [[File:Sms-mms-icon.png|15px]] icon, this indicates that the number supports this feature.

You can start ordering a DID Number from your portal >> DID Numbers >> Order DID >> Local Numbers >> (US or Canada) Numbers and you will be able to purchase your desired number, just make sure it supports the MMS feature.

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

==Configuring the SMS/MMS service==

Once purchased the SMS/MMS Service needs to be activated on the DID settings, from your portal >> DID Numbers >> Manage DID >> Edit Selection >> SMS/MMS configuration section.

In that section, you can activate and configure some forward options for the SMS/MMS service.

: [[File:SMSenable20240520.png|thumb|none|600px]]

:* '''Message Service (SMS/MMS)''': Check the box to enable the service.
:* '''VoIP.ms SMS/MMS Portal''': This is enabled by default and it can't be modified, that means that you will be able to create and send messages from the SMS Message Center.
:* '''SMS/MMS Email Address''': By activating this option all the SMS Messages will be sent to the email address you configure in this field. You can receive them to more than 1 email address by separating each emails with a comma. The advantage of this option is that you will receive your SMS Messages directly to your email and you will be able to Reply to these messages from your email too. You just need to click on '''reply''' to the email. Please note that e-mail responses are as well limited to 160 characters and if they exceed this limit, they'll be split in two (or more messages if applies) when sent.

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'''NOTE''' 💬: ➜ To reply the SMS Message via your email it is important to click on '''Reply''' and '''DON'T change or modify''' the destination mail "TO" (sms@voip.ms) and the subject E.G ''[#USXXXXXX] Message sent to 5555555555''. Make sure that your email address inbox isn't full to avoid loops when receiving an SMS/MMS message via email from an automated service.
</div>

:*'''SMS/MMS Forward''': You can have your SMS/MMS Messages forwarded to your cellphone or any other number that supports SMS/MMS, with this option you will receive the SMS/MMS Messages from our system to the configured phone number.
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'''NOTE''' 💬: ➜ The CallerID sent to the Call Forwarded Cell Phone will be the DID Number that received the SMS Message. Is not possible to forward the original Caller ID at this time. When using SMS/MMS Forward there is a charge for the incoming message received by the DID and the outgoing message sent to the number set for the forward.
</div>

:*'''SMS/MMS SIP Account''': You can have your SMS Messages forwarded to your selected SIP account. They will be sent through our SMS/SIP gateway as a SIP MESSAGE.
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'''NOTE''' 💬: ➜ The destination ("To" header field) of the SIP MESSAGE will be your account name, instead of the destination number of the original SMS.
</div>

:'''SMS/MMS URL Callback''': By enabling this option you will be able to send the messages to another destination e.g. your own server. If selected SMS Messages received by your DID will Send a GET request to the URL Callback provided. Available variables for your URL:
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:: '''Example''': <nowiki>https://YourDomainName.tld/sms.php?to={TO}&from={FROM}&message={MESSAGE}&id={ID}&date={TIMESTAMP}</nowiki>

: '''{ID}''' = The ID of the SMS message.
: '''{TIMESTAMP}''' = The date and time the message was received.
: '''{FROM}''' = The phone number that sent you the message.
: '''{TO}''' = The DID Number that received the message.
: '''{MESSAGE}''' = The content of the message.
: '''{MEDIA}''' = Comma-separated list of media files.

</div>

:*'''SMS/MMS Webhook URL (3CX)''': If you have a 3CX system, you will need to fill this information. For more information, [[3CX_Phone_System#Configuring_VoIP.ms_SMS.2FMMS_service_with_your_3CX | please review this article]].

:*'''Dialing Mode for SMS (3CX/API)''': If you are using a 3CX system or VoIP.ms API, you can choose to send and receive SMS messages with the NANPA or E164 format.
::* '''E164''': To send a SMS or MMS with this format, you will need to include '''+''' sign and the country code for every messages
::* '''NANPA''': To send a SMS or MMS with this format, you only need to enter the 10-digit number without the country code.

:*'''URL Callback Retry''': When selected, we will be expecting an "ok" output (without quotes) from your URL callback page as an indicator that you have received the message correctly. If we don't received the "ok" (without quotes) from your callback page, we will keep sending you the same message every 30 minutes.

Once you configure the desired option, click on Apply Changes.

==Sending a SMS/MMS (Web Interface)==

'''Step 1'''

: Click the "send New Message" button to get a list of contacts displayed or directly type in the number into the contact text field. If this number is registered in the phonebook, it will display its name instead.

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

'''Step 2'''

:Click to display the list of available DID numbers. Choose the number to send the message from.

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

'''Step 3'''

:Write your message on the text field and click the red arrow to send it. Text messages up to 160 characters will be considered SMS. Media messages or text messages over 160 characters will be considered MMS and you will see the icon turning from SMS to MMS indicating of this.
:: [[File:MMS-Webportal4.png|border|400px]]

'''Step 4'''

: In order to send a media file over an MMS, click the clip button to browse and choose a media file.

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

: Click the browse button to search for the media file in your computer or drag and drop the file directly into the box.

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

: The attached files will show below the text box.

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

: Once the message is sent, a new conversation will be opened on the left. Selecting this conversation will show the messages history on the box.

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

==Receiving an SMS/MMS (Web Interface)==

: While SMS messages can be received directly in an App through the SIP Protocol and correct configuration, at the moment, '''receiving and reading MMS is only possible through the message center.'''

: You can access the communication center from your customer portal at DID numbers > SMS/MMS Message Center. Any message received will appear here on the left side of the panel.

: Use the DID Filter to choose and display messages for a specific DID number or for all DID numbers.

: You can also filter the messages by date and time zone by clicking on the down arrow, which will show the search parameters

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

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

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

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

== SMS/MMS History ==

You can now also gain access to a general history of SMS/MMS that was sent and received for all of your DIDs, or for a specific one by going to Customer Portal >> DID Numbers >> SMS/MMS Message Center >> History.

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

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

= SMS using a SMPP Server =

An SMPP server, short of '''Short Message Peer to Peer''' is a method designed to send and receive high volumes of SMS messages through a dedicated server.

In other words, this allows you to better control the quantity of messages that will be sent to your target audience while avoiding to pass via a softphone application or your VoIP.ms portal to send/receive these SMS, thus eliminating multiple repetitive tasks and saving you time.

For example, if you have a promotion for your business that offers a discount for a certain lapse of time and that your customers have provided their agreement to receive these types of promotions via SMS, you could send your promotion to your customers in a faster manner than if you would do it from your customer portal, thus increasing efficiency.
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'''NOTES''' 💬: ➜ Note that sending and receiving SMS messages will be charged our usual rate at $0.0075 per messages'''
</div>

==Requirements==

: '''This will require you to have your own SMPP server.'''
: Once configured properly, you will then be able to send and receive SMS messages to your own server.

== Configuration on your SMPP Server ==

: In order for your server to communicate and authenticate accordingly with VoIP.ms SMPP server, you will need to configure the following:

::* A '''username''' and '''password''' of your choice in order for VoIP.ms SMPP server to authenticate you.
::* Have all your communications be sent to '''smpp.voip.ms''' via port '''2775''', for regular SMPP, or port '''3550''' which is our encrypted SMPP port.
::* Open a connection using the '''bind_transceiver''' command. You can keep this connection open for as long as you need to and inquire about it using the '''enquire_link''' command.
::* Send your messages with either '''deliver_sm''' or '''submit_sm''' commands. Make sure to include your PDU's '''source_addr''' as one of your SMS enabled DIDs from your account. Your destination SMS number should be set as '''destination_addr''' while your text message will be included in '''short_message'''.

== Configuration SMPP on your VoIP.ms Portal ==

Once you have set up the above on your server, you will need to enable and specify which username and password will be used by VoIP.ms to authenticate your SMPP requests. In order to do so, head into your VoIP.ms portal, DID Numbers, Manage DIDs, edit your DID that supports SMS messages and by heading down the page, you will see the following options:

:* '''SMPP Enabled:''' If selected, SMS Messages will be allowed to be sent and received using SMPP.
:: You may specify the URL to be used to send a copy of inbound messages to your SMPP server. Leave empty if you only wish to send messages. See the example below. The '''scheme''' part is mandatory and must be set to either of '''smpp''' for unencrypted SMPP, or '''ssmpp''' for encrypted SMPP using TLS. The '''user:password@''' part is optional and will default to the same username and password defined for authentication on our side as defined below. The ''':port''' part is also optional and will default to 2775 for '''smpp''' and to '''3550''' for ssmpp.

:* '''SMPP Username:''' The username that you will use to authenticate to our SMPP server for sending messages
:* '''SMPP Password:''' The password that you will use to authenticate to our SMPP server for sending messages

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'''NOTES''' 💬
: ➜ In order to help avoid any confusion while configuring the username on your server, you can use your VoIP.ms account ID as a username (the 6 digits of your account).
: ➜ Make sure to have the option '''Message Service (SMS/MMS)''' enabled.
</div>

Enable the option '''SMPP Enabled''', fill the username and password per what was configured on your SMPP server and once done, apply the changes by pressing '''Click here to apply changes'''.

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

You may now start sending and receiving SMS messages from your SMPP server.

= Send and Receive Text Messages (Web Interface)=

==SMS Message Center for mobile==

To start using the service you will need to use the SMS Message Center, from your portal >> DID Numbers >> [https://www.voip.ms/m/sms.php SMS Message Center].

For Mobile Devices please use [https://sms.voip.ms/ sms.voip.ms] as a complete all in one solution.

From the SMS Message Center you can check your SMS History, send and receive SMS messages.

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

:* '''Search Range''': Select a Date Range of your SMS History. You can select a time period by marking the "Show Details" field and the text messages will be displayed.
:* '''Search Filter''': You can filter your Search by DID, Contact and Type (Sent, Received and Both).
:* '''Send New SMS Messages''': Another window will open by clicking here in order to create a new SMS Message.
:* '''Delete Selected Messages''': This action will delete any messages that you have selected, by clicking on the box next to them. '''This action cannot be undone.'''
:* '''Delete All Messages''': Delete ALL your SMS Messages, both sent and received. '''This action cannot be undone.'''
:* '''Checking a Message''': To check a received SMS Message just click it directly in order to display the information.

==Create a New SMS Message (Legacy)==

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

You can create a new '''SMS Message''' we will explain you all the parameters you can find on this section:

* '''DID''': If you have more than one DID Number you can select any of the numbers available to send your message.
* '''Contact''': The destination phone number or if you have a [[Phone book|Phone book entry]] start typing the name and the system will display the information if available.
* '''Message''': You can create your message on that field, please note that you have up to 160 characters.

Then click on 'Send Message' to send the SMS.

= Send and Receive Messages (via SIP MESSAGE Protocol) =
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'''NOTES''' 💬
: ➜ '''SMS via SIP''' service is only available through '''SIP Protocol and user/password authentication'''. '''IP Auth and IAX2 are not available at this time.'''
: ➜ '''Please refer to our guide [[SIP/SMS with FreePBX]] for instructions on how to configure the service with a FreePBX'''
</div>

== Sending ==

:* DID must have the '''"Short Message Service (SMS)"''' service option enabled. This option can be enabled from the Manage DIDs > Edit DID page.

:* To send an SMS from your SIP account, it is required that you set your Caller ID number to one of your SMS enabled DIDs. This will be the number you will be sending the message from. You can configure the Caller ID number from your customer portal for the specific subaccount if you are using a softphone or directly from your extension or trunk if you are using an Asterisk or PBX server.

:* It's important to note that if you configure your caller ID name in your SIP client to be a 10 digits number, this will override your caller ID number. If the caller ID name is anything different than a 10 digits number this will be discarded and the Caller ID number will be used.

:* If you are an asterisk or PBX user, please make sure to use the latest version of '''Asterisk (v12 or higher)''' and use '''chan_pjsip''' for the trunk. This is an asterisk limitation.

:* The Desktop version of Zoiper requires the user to have the '''PRO version''' activated to be able to send SMS messages as well as to enable '''SIP Presence'''. Please contact Technical Support to request SIP presence to be enabled for your account for the VoIP.ms POP you are using. While normally not required, there may be other cases not documented of softphones or apps requiring SIP Presence to be enabled. Please contact technical support if you encounter any issue with your specific software.

:* The '''free Desktop version''' of Zoiper cannot send SMS messages and is only capable of receiving SMS.

:* The '''mobile version''' of Zoiper is completely free of these limitations. Free version already works for sending and receiving without the need of SIP presence.

:* When sending an SMS message, the recipient number must be 10 digits. Avoid using the prefix '"1"' since is not supported at the moment. The system will recognize the numbers as in 10 digits.

:* The current characters limit for a '''single SMS is 160'''. If a message is bigger than 160 characters, it will not be sent.

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

'''NOTES''' 💬: ➜ AT THE MOMENT SENDING MMS USING SIP ACCOUNT IS NOT POSSIBLE AT THIS TIME.'''
</div>

:*DID must have the '''"Message Service (SMS/MMS)"''' service option enabled. This option can be enabled from the Manage DIDs > Edit DID page.
:*Enable the option '''"SMS SIP Account"''' and select the account destination for your SMS messages to this number.
:*The account receiving SMS must be registered successfully in any of our POPs.
:*To ensure the best results receiving the SMS messages, do not register the same SIP account on more than one softphone or app at the same time, this can lead to unexpected results.
:*For SIP devices that require this information, such as a trunk or PBX, we encode the original destination number in a custom SIP header field named "X-SMS-To".
:*3CX softphone currently does not support any third party SMS service. It will only work by having a direct connection to the 3CX PBX system. '''Please contact 3CX for more details'''.

== Send SIP/SMS with FreePBX ==

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

'''NOTES''' 💬: ➜ AT THE MOMENT SENDING MMS USING SIP ACCOUNT IS NOT POSSIBLE AT THIS TIME.'''
</div>

Yes, it is possible to configure a FreePBX system to send SMS, please refer [[SIP/SMS with FreePBX | here]] for detailed instructions.

=SMS/MMS using the Reseller Interface=

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

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

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

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

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

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

Then scroll down and find the feature "’’’SMS/MMS’’’", and enable it.

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

= Enabling/Disabling SMS/MMS on a DID via the client's interface =
For your client to enable or disable SMS/MMS from his portal, your customer will need to head under Services, My Services.
: [[File:SMSMMSreseller5.png|thumb|none|500px]]

Then proceed by heading into ‘’’DID Lists’’’ tab and edit DID.

: [[File:SMSMMSreseller6.png|thumb|none|500px]]
You will then be able to enable SMS/MMS, receive via email and more.

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

= Accessing their SMS/MMS via the clients interface =
1) To gain access to their SMS/MMS messages, head under Services and then SMS/MMS.
: [[File:SMSMMSreseller8.png|thumb|none|500px]]

= Configuring VoIP.ms SMS/MMS service with your 3CX =

==Prerequisites==
: 1. Ensure that you have the latest version of your 3CX PBX, including the updated version of the VoIP.ms trunk template.
: 2. Have a DID that supports SMS/MMS with the service enabled. ''(Navigate to Main Menu > DID Number > Edit DID)''
: 3. Generate a copy of your API Key, known as the "Bearer Token." [[#Generate_a_Bearer_Token|See this section]]
: 4. Follow the rest of the guide [[3CX_Phone_System#Configuring_VoIP.ms_SMS.2FMMS_service_with_your_3CX | Here]].

Cisco SPA112

2024-02-15T18:40:58Z

John: /* CA Certificate */

[[File:SPA112.jpg|300px|thumb|left|Cisco SPA112]]
<div style="width:100%;overflow:hidden;clear:both"></div>

Please note '''these models are EOL''' (End of life, discontinued mid-2020) and are '''not supported by CISCO anymore'''. Their successors are the (very-similar) [[Cisco ATA191 and ATA192]]. https://www.cisco.com/c/en/us/products/collateral/unified-communications/small-business-voice-gateways-ata/eos-eol-notice-c51-743206.html

: '''There have been some reports of issues with this device from customers of both VoIP.ms and other providers.'''
: '''Make sure to install the latest firmware from [https://software.cisco.com/download/release.html?mdfid=283998771&softwareid=282463187&release=1.4.1%20SR3&relind=AVAILABLE&rellifecycle=&reltype=latest Cisco Software].'''
: '''Version 1.1 or later should be used for proper Caller ID support. '''
: '''Some People have reported issues using Firefox to Configure this device; please try Chrome or IE. '''

==Documentation==
[https://www.cisco.com/c/en/us/support/unified-communications/spa112-2-port-phone-adapter/model.html#~tab-documents Official Configuration guide / User guide / Data Sheets]

== Configuration Details ==

=== Getting the IP address of your device ===

There are two ways to retrieve the IP address of your Cisco SPA112: via analog phone menu, and via your internet router.

{| class="wikitable"
! style="text-align: center;" | Analog phone interface
! style="text-align: center;" | Internet Router
|-
|
# Attach the Cisco SPA112 to your network and attach an analog phone to one of the phone ports, then do the following:
## Dial **** from the phone, even though there is no dial tone.
## When you hear "System Configuration Menu," dial 1 1 0 # slowly. The current IP address will be read back. (e.g. 192.168.X.X)

'''If you hear 0.0.0.0, check your network connection and DHCP server. If necessary, a static IP address'''
'''can be assigned by using option 111# at the IVR, then entering the IP address with your phone's keypad'''
'''(for example, 10*1*27*2 for 10.1.27.2). The network mask can be set with option 121# and the default'''
'''gateway can be sent with option 131#'''
Learn more about the IVR menu options from the https://supportforums.cisco.com/docs/DOC-9900 document.

Be sure to allow at least a minute or two for the box to initialize; even a correctly configured and installed SPA112/122 will give no power or dialtone to the phone until initialization is complete.

Note that the SPA122 is basically a SPA112 with a second network port, intended for installation between a local network hub (LAN) and an upstream Internet (WAN) connection. The SPA122 may be configured as either a "NAT" or "bridge". Depending on configuration, this leaves the SPA122 with two addresses; a local area network address (such as 192.168.15.1) and an outside Internet address. Dialing ****110# will give one address, ****210# will give the other.
|
# Attach the Cisco SPA112 to your network
# Access your router's remote administration interface via your web browser (typical addresses may be <code>192.168.0.1</code> or <code>192.168.1.1</code>). Refer to your router instructions for more information.
# Enter your username/password if asked. If you have not set one, then it is likely the unchanged default password.
# In the router's menu, there should be a page showing a list of connected clients, with their internal IP address. Find the entry corresponding to the Cisco SPA112. It should identify itself in the list as "SPA112"
# Navigate to this IP address via your web browser
|}

=== Accessing to the device's settings page ===

Open your web browser and go to the IP address you obtained in step 1 (for example, http://192.168.2.1).
The default username is admin, and the default password is also admin.

For the SPA122, if one address does not return the web interface (or has some functions greyed/disabled), try the other.

=== Configuring the Quick Setup screen ===

Go to Quick Setup and configure Line 1 as follows:

'''Proxy:''' atlanta.voip.ms (You can choose any of our multiple VoIP.ms [http://wiki.voip.ms/article/Choosing_Server servers])

'''Display Name:''' Your Outbound callerID Name'''(See the requirements below)'''

'''User ID:''' 100000 (Replace with your 6 digit Main SIP account UserID or Sub Account user name, i.e. 123456 or 123456_sub)

'''Password:''' Your VoIP.MS SIP Password

'''Dial Plan:''' (911S0|310xxxx|<:1555>[2-9]xxxxxx|1[2-9]xx[2-9]xxxxxxS0|[2-9]xx[2-9]xxxxxxS0|*xx|***xxx|*xx.|[3468]11|822|0|00|4xxx|**275*x.|xxxxxxxxxxxx.)

('''''Note''''': Replace 555 in the dial plan with your area code, See [[Dial Plan for Linksys ATAs]] for details.)

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

Click Submit to save settings.

[[File:quick_setup_test.png|800px|thumb|left|Quick Setup Page - Click to enlarge]]
<br clear="all" />

=== Configuring the Voice Line ===
==== Nat Settings ====

Click on Voice, then Line 1

Set '''NAT Mapping Enable''' to Yes, then set '''NAT Keep Alive Enable''' to Yes. If your environment does not use NAT, you can leave these settings disabled. These features can usually be disabled on the SPA122 if it is connected directly to your modem since its traffic will not be subject to NAT in this configuration.

If using the second phone line on an SPA122 device, change the SIP Port for one of the lines to e.g. 5080.

[[File:VL_1_nat_settings.png|800px|thumb|left|NAT Settings - Click to enlarge]]
<br clear="all" />

==== Proxy and Registration ====

Under '''Proxy and Registration''', set the server you will use as registration server and the proper values for the Register Expires and Proxy Fallback Intvl:

'''Proxy''': atlanta.voip.ms (one of VoIP.ms multiple [http://wiki.voip.ms/article/Choosing_Server servers], you can choose the one closest to your location)
'''Register Expires''' to 300
'''Proxy Fallback Intvl''' to 300

Also confirm the following settings:<br />
'''Register:''' YES <br />
'''Use DNS SRV:''' NO<br />
'''DNS SRV Auto Prefix:''' NO

[[File:VL_2_proxyAndRegistration.png|800px|thumb|left|Proxy and Registration - Click to enlarge]]
<br clear="all" />

Click Submit to save these changes

==== Subscriber Information ====

In this section please confirm that you have the proper account information:

'''Display Name''': Your name (that will be shown as callerID name)
'''User ID''': 100000 (Replace with your 6 digit Main SIP account UserID or Sub Account user name, i.e. 123456 or 123456_sub)
'''Password''': Your VoIP.ms SIP Password

[[File:VL_3_subscriberInformation.png|800px|thumb|left|Subscriber Information - Click to enlarge]]
<br clear="all" />

==== Audio Configuration ====

You can verify or change the audio codec that will be used with the calls. Please verify that you have the same codec selected in your SIP account's settings.

Preferred codec: g711u (or G729)

[[File:VL_4_audioConfig.png|800px|thumb|left|Audio configuration - Click to enlarge]]
<br clear="all" />

==== Dial Plan ====

We recommend to use this dial plan.

(911S0|310xxxx|<:1555>[2-9]xxxxxx|1[2-9]xx[2-9]xxxxxxS0|[2-9]xx[2-9]xxxxxxS0|*xx|***xxx|*xx.|[3468]11|822|0|00|4xxx|**275*x.|xxxxxxxxxxxx.)

('''''Note''''': Replace 555 in the dial plan with your area code, See [[Dial Plan for Linksys ATAs]] for details.)

[[File:VL_5_dialPlan.png|800px|thumb|left|Dial Plan - Click to enlarge]]
<br clear="all" />

You can create your own dial plan if you need it. See [[Dial Plan for Linksys ATAs]] and also [https://www.cisco.com/c/en/us/support/docs/collaboration-endpoints/spa901-1-line-ip-phone/108747-pqa-108747.html here] you'll find a great explanation about dial plans for Cisco devices

=== Optional settings ===

==== Outbound audio "breaking up". ====

Cisco's defaults (SIP T1 = 0.5 sec, RTP packet size 0.030 on most Sipura adapters) respectively may cause unnecessary retransmission of commands over connections with high latency and create issues with outbound audio "breaking up".

Click '''Voice''', then go to '''SIP'''.

Set SIP Timer Values (sec)

SIP T1: 1

Set RTP Parameters

RTP Packet Size: 0.02
RTP Port Min: 10000
RTP Port Max: 20000

Click Submit to save the changes

[[File:VS_sipAndRTP.png|800px|thumb|left|SIP Values - Click to enlarge]]
<br clear="all" />

==== Caller ID display showing incorrect time ====

Sometimes the hour shown in your caller ID is incorrect. Following this suggestion usually solves the issue:

Enter your device's settings and click '''Network Setup''', then go to '''Basic Setup''', then click '''Time Settings'''

Set your time zone and (optional) NTP settings. A good time server choice is 0.pool.ntp.org. Setting the proper time zone will ensure that the time which appears on your Caller ID display is correct.

===Configuring a Voice line using TLS===

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

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

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

====Verifying the device's Firmware version====

First, check your firmware's version. This is, from your Device's Configuration Utility at ''Status >> System Information''

We '''strongly recommend''' to use the latest firmware version available, up today is the ''1.4.1 (SR3) Apr 3 2019''. If you do not have this version, you may consider its upgrade.

[[File:SPA_FW_Ver.png|800px|thumb|left|Click to enlarge]]
<br clear="all" />

==== Enabling TLS for the line ====

Go to the User's line you will use (If you use the Line 1 go to User 1) and navigate to the '''''Supplementary Service Settings''''', there set:
:''Secure Call Setting'': '''''yes'''''

[[File:SPA_User_SCS.png|800px|thumb|left|Click to enlarge]]
<br clear="all" />

==== Configuring the transport and port ====

Go to the line you will be using with TLS and navigate to the section ''SIP Settings'', then set:
:''SIP Transport'' : '''''TLS'''''
:''SIP Port'' : '''''5061'''''

[[File:SPA_Voice_Line.png|800px|thumb|left|Click to enlarge]]
<br clear="all" />

==== CA Certificate ====

As per CISCO's requirements, a CA certificate is needed to use Secure calls with the SPA112's device. To achieve this you will need to import the CA Cert.

Go to ''Voice >> Provisioning'' and once there navigate to '''''CA Settings''''', at ''Custom CA URL'' enter the following:

https://spa1xx.voip.ms/cca.pem
If your device has firmware version 1.4.1 SR5 you'll need to enter: http://spa1xx.voip.ms/cca.pem

[[File:SPA_Prov_CA.png|800px|thumb|left|Click to enlarge]]
<br clear="all" />

Click ''Submit'', the device will reboot and after that will register and you will be ready to use it with TLS

==== Secure Call Indication Tone ====

Once the secure call feature is enabled, during all the duration of your calls you will hear a couple of tones (beeps), this is normal and beyond ''VoIP.ms'' control, however you can disable this notification going to: ''Voice >> Regional >> Secure Call Indication Tone''

'''''Note:''''' Please notice that this setting is not an on/off, you will need to remove all the line
In case you need to set it back, the default value is
397@-19,507@-19;15(0/2/0,.2/.1/1,.1/2.1/2)

[[File:SPA_Regional_Tone.png|800px|thumb|left|Click to enlarge]]
<br clear="all" />

Click Submit to save the changes

==Known Issues==

=== '''Phone will not ring on handset''' ===

Sometimes the Phone you are using is designed for a certain voltage and ring waveform. If someone tries to call you and the phone appears to be ringing for the caller but your phone never rings, please follow these steps to hopefully resolve this issue for you.

Step 1: First access the SPA web interface.

Step 2: Click on the '''Admin Login''' and then click on '''(switch to advanced view)'''

Step 3: Click on your Regional tab on the top menu.

Step 4: Go halfway down the page until you see the heading '''Ring and Call Waiting Tone Spec'''

[[File:Pap2Ring.jpg|800px|thumb|left| Ring and Call Waiting - Click to enlarge]]
<br clear="all" />

Step 5: Change the Ring Waveform setting to Sinusoid or Trapezoid, the opposite of what you have set. You can also change the Ring Voltage in increments of 5 to 90 or 95.

Step 6: Save settings and test an incoming call.

=== Receiving Unwanted Calls in the middle of the Night (i.e. CallerID 100) that do not appear in your CDR: ===

These calls are not going through our Network but rather through the internet directly to your ATA Device.

Please look under the Voice>> Line 1 page in your SPA device for the following setting: Restrict Source IP and make sure it's enabled.

This way the ATA device will block any traffic not coming from our servers.

[[File:VL_1_restrictSourceIP.png|800px|thumb|left|Restrict IP - Click to enlarge]]
<br clear="all" />

=== Firmware Upgrade ===

SPA112 and SPA122 adapters were distributed with outdated (1.0.x) firmware at least as late as 2012; affected boxes will not show Caller ID on any inbound call, even though the caller names and numbers are visible in the call detail record on VoIP.ms (or other provider's) web interface.

Updated firmware is available from the Cisco site [https://software.cisco.com/download/release.html?mdfid=283998771&softwareid=282463187&release=1.4.1%20SR3&relind=AVAILABLE&rellifecycle=&reltype=latest Cisco Firmware] as a .ZIP archive which contains two files (a .BIN with the actual firmware and a .PDF with documentation). Download and unzip this file. Go to the 'administration' tab on the web interface (on the SPA122, this needs to be done from the LAN side with SPA122's built-in networking set to NAT mode). On the left sidebar, click 'update firmware' (as most of the administration menu does not appear for Firefox users, downgrade to MS IE or another browser temporarily). Click the 'upload' button and indicate the location of the unzipped .BIN file. A box will appear with a progress indicator and a warning not to interrupt the upgrade. When the upgrade is completed, the SPA112/122 will reset and will likely take a minute or more to reinitialize, reconnect to the network and restore dial tone. SPA122 users who have installed the device in-line between the local PCs and the Internet will be disconnected from the Internet until reinitialization is complete.

Once the new firmware is deployed, call display will operate normally and the configuration web page will display in Firefox without missing options in the administration menu.

A manual for Cisco's SPA100 series adapters is online at http://www.cisco.com/en/US/docs/voice_ip_comm/csbpvga/spa100-200/admin_guide_SPA100/spa100_ag.html

* You can check the most commonly used [[Cisco/Linksys Star Codes]]

=== SPA Star Codes ===

{| class="wikitable" style="margin-left: center; margin-right: auto; border: none;"
! colspan="3" | SPA Star Codes
|-
! scope="col" | Star Code
! scope="col" | Name
! scope="col" | Result
|-
| *69 || Call Return Code || This code calls the last caller.
|-
| *07 || Call Redial Code || Redials the last number called. (Not in pap2t)
|-
| *98 || Blind Transfer Code || Begins a blind transfer of the current call to the extension specified after the activation code.
|-
| *66 || Call Back Act Code || Starts a callback when the last outbound call is not busy.
|-
| *86 || Call Back Dea t Code || Cancels a callback.
|-
| *05 || Call Back Busy Act Code || Starts a callback when the last outbound call is busy. (Not in pap2t)
|-
| *72 || Cfwd All Act Code || Forwards all calls to the extension specified after the activation code.
|-
| *73 || Cfwd All Deact Code || Cancels call forwarding of all calls.
|-
| *90 || Cfwd Busy Act Code || Forwards busy calls to the extension specified after the activation code.
|-
| *91 || Cfwd Busy Deact Code || Cancels call forwarding of busy calls.
|-
| *92 || Cfwd No Ans Act Code || Forwards no-answer calls to the extension specified after the activation code.
|-
| *93 || Cfwd No Ans Deact Code || Cancels call forwarding of no-answer calls.
|-
| *63 || Cfwd Last Act Code || Forwards the last inbound or outbound calls to the extension specified after the activation code.
|-
| *83 || Cfwd Last Deact Code || Cancels call forwarding of the last inbound or outbound calls.
|-
| *60 || Block Last Act Code || Blocks the last inbound call.
|-
| *80 || Block Last Deact Code || Cancels blocking of the last inbound call.
|-
| *64 || Accept Last Act Code || Accepts the last outbound call. It lets the call ring through when do not disturb or call forwarding of all calls are enabled.
|-
| *84 || Accept Last Deact Code || Cancels the code to accept the last outbound call.
|-
| *56 || CW Act Code || Enables call waiting on all calls.
|-
| *57 || CW Deact Code || Disables call waiting on all calls.
|-
| *71 || CW Per Call Act Code || Enables call waiting for the next call.
|-
| *70 || CW Per Call Deact Code || Disables call waiting for the next call.
|-
| *67 || Block CID Act Code || Blocks caller ID on all outbound calls.
|-
| *68 || Block CID Deact Code || Removes caller ID blocking on all outbound calls.
|-
| *81 || Block CID Per Call Act Code || Blocks caller ID on the next outbound call.
|-
| *82 || Block CID Per Call Deact Code || Removes caller ID blocking on the next inbound call.
|-
| *77 || Block ANC Act Code || Blocks all anonymous calls.
|-
| *87 || Block ANC Deact Code || Removes blocking of all anonymous calls.
|-
| *78 || DND Act Code || Enables the do not disturb feature.
|-
| *79 || DND Deact Code || Disables the do not disturb feature.
|-
| *65 || CID Act Code || Enables caller ID generation.
|-
| *85 || CID Deact Code || Disables caller ID generation.
|-
| *25 || CWCID Act Code || Enables call waiting, caller ID generation.
|-
| *45 || CWCID Deact Code || Disables call waiting, caller ID generation.
|-
| *26 || Dist Ring Act Code || Enables the distinctive ringing feature.
|-
| *46 || Dist Ring Deact Code || Enables the distinctive ringing feature. The default is *46 ||.
|-
| *74 || Speed Dial Act Code || Assigns a speed dial number.
|-
| *16 || Secure All Call Act Code || Makes all outbound calls secure.
|-
| *17 || Secure No Call Act Code || Makes all outbound calls not secure.
|-
| *18 || Secure One Call Act Code || Makes the next outbound call secure. (It is redundant if all outbound calls are secure by default.)
|-
| *19 || Secure One Call Deact Code || Secure One Call Deact Code Makes the next outbound call not secure. (It is redundant if all outbound calls are not secure by default.)
|}

==Guide Links==
In the event where you need the guide directly from Cisco, you may find the admin manual guide below:

Admin Manual : [https://www.cisco.com/c/dam/en/us/td/docs/voice_ip_comm/csbpvga/spa100-200/admin_guide_SPA100/spa100_ag_1_3_5.pdf Download PDF]

[[category:Analog Telephone Adapters]]

Softphones

2024-01-12T22:06:28Z

John:

A softphone is a software program for making telephone calls over the Internet using a general purpose computer, tablet or smartphone, rather than using dedicated hardware. Often a softphone is designed to behave like a traditional telephone, sometimes appearing as an image of a phone with a display panel and buttons with which the user can interact. A softphone is usually used with a headset connected to the sound card of the PC, or as an installed app on your phone. See also: [[Smartphone]]

Take a peek at VoIP.ms Blog Article : [https://wiki.voip.ms/article/Back_to_Basics,_What_is_a_Softphone_and_How_to_Leverage_It%3F Back to Basics, What is a Softphone and How to Leverage It?]

==Acrobits==

[[File:Acrobits_logo_new.png|300px|left|Acrobits]]

[[File:Acrobitscert.png|180px|right|Acrobits|link=https://www.acrobits.net//?utm_campaign=itsp-partners&utm_medium=cp-certified-logo&utm_source=voipms]]

'''Product:''' Acrobits

'''Company:''' [https://www.acrobits.net//?utm_campaign=itsp-partners&utm_medium=cp-certified-logo&utm_source=voipms Acrobits]

'''OS:''' IOS, Android

'''Overview:'''
Acrobits is a mobile software development company with a focus on SIP Clients for mobile devices; their SIP client, Acrobits Softphone, is the leading app of it's type on the iTunes App Store.

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

==AntiSIP (Mobile)==

[[File:Antisip-logo.png|300px|left|Acrobits]]

'''Product:''' AntiSIP

'''Company:''' [https://www.antisip.com/antisip-company.html AntiSIP]

'''OS:''' Android

'''Overview:'''
AntiSIP is a mobile software SIP, WebRTC, Audio, Video on all platforms!

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

==Bria Mobile==

[[File:Softphone Clients - Smartphones.jpg|350px|left|Bria Mobile|link=https://www.counterpath.com/?utm_campaign=itsp-partners&utm_medium=cp-certified-logo&utm_source=voipms]]

[[File:CounterPath ITSP Certified Partner 2017.png|250px|right|CounterPath|link=https://www.counterpath.com/?utm_campaign=itsp-partners&utm_medium=cp-certified-logo&utm_source=voipms]]

'''Product:''' Bria Mobile

'''Company:''' [https://www.counterpath.com/?utm_campaign=itsp-partners&utm_medium=cp-certified-logo&utm_source=voipms CounterPath]

'''OS:''' Windows, MacOS, Linux (Ubuntu), iOS, Android

'''Overview:'''
Bria 5 is a carrier-grade next generation softphone application that enables you to manage your communications easily and efficiently
– all from your computer desktop. Replacing or complementing your hard phone, the Bria softphone allows you to make VoIP and Video calls over IP, see when your contacts are available, send Instant Messages and transfer files with ease and efficiency.

Bria Mobile for Andoid and IOS is a highly secure, standards-based mobile VoIP softphone that works over both 3G and Wi-Fi networks.

Using the device’s existing contact list, Bria Android Edition facilitates easy and effective communication management with an intuitive interface. Deskphone-class calling functionality includes the ability to swap between two calls, merge calls and perform attended and unattended transfers.

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

==Bria Solo==

[[File:Briasolologo.jpg|350px|left|Bria Solo|link=https://www.counterpath.com/?utm_campaign=itsp-partners&utm_medium=cp-certified-logo&utm_source=voipms]]

[[File:CounterPath ITSP Certified Partner 2017.png|250px|right|CounterPath|link=https://www.counterpath.com/?utm_campaign=itsp-partners&utm_medium=cp-certified-logo&utm_source=voipms]]

'''Product:''' Bria Solo

'''Company:''' [https://www.counterpath.com/?utm_campaign=itsp-partners&utm_medium=cp-certified-logo&utm_source=voipms CounterPath]

'''OS:''' Windows, MacOS, iOS, Android

'''Overview:'''
Bria Solo provides desktop and mobile softphone applications that facilitate best-in-class calling and robust call control features.

Whether used for inbound or outbound calls, Bria makes it easy to integrate contacts from various sources, streamline communications, and manage calls from one easy-to-use interface.

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

==Bria Teams==

[[File:Bria-teams-thumb.png|350px|left|Bria Teams|link=https://www.counterpath.com/?utm_campaign=itsp-partners&utm_medium=cp-certified-logo&utm_source=voipms]]

[[File:CounterPath ITSP Certified Partner 2017.png|250px|right|CounterPath|link=https://www.counterpath.com/?utm_campaign=itsp-partners&utm_medium=cp-certified-logo&utm_source=voipms]]

'''Product:''' [https://www.softphone.com/ Bria Teams]

'''Company:''' [https://www.counterpath.com/?utm_campaign=itsp-partners&utm_medium=cp-certified-logo&utm_source=voipms CounterPath]

'''OS:''' Windows, MacOS, iOS, Android

'''Overview:'''
Team Communication made Easy
Bria Teams helps you regain productivity by streamlining all team communications into one application across your devices. With Bria's provisioning, it's easy to get your team talking.

It offers a softphone that helps you gain productivity by streamlining all team communications into one application. It has all the tools you need in one interface, across your devices and can be managed by one simple dashboard.

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

==Dialer+==

[[File:Dialerplus.png|200px|thumb|left|Dialer+]]

'''Product:''' cd26

'''Company:''' cd26

'''OS:''' IOS

'''Overview:'''
Dialer+ is a SIP-based softphone for iPhone , iPod touch and iPad

Standard Phone Features
* Fast Contact Search – Use phone pad to find contacts by names, initials or any part of the numbers.
• iOS 4.0+ Background Support
• Supports GSM calls when the network is unavailable or slowly
• Multiple Account Support
• Auto switch to best service provider when use multiple account
• Pre-configured VoIP providers list
• Call display
• Speakerphone, Mute and Hold
• Call history
• Contact List and Contact Favorites - leveraging the iPhone Contacts
• Ringtones and contact avatars
• Multiple incoming Call Support - swap between two active calls; merge and split calls
• Audio codecs include G.711,G.722, iLBC ,Speex and GSM, Make an in app purchase to add G.729 for great performance over 3G networks
• Support for RFC 2833 DTMF
• VPN Support
• uPNP Support
• iPhone 4 Retina Display Support

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

==Ekiga==

[[File:Ekigaw0.png|200px|thumb|left|Ekiga Softphone]]

'''Product:''' Ekiga

'''Company:''' Ekiga

'''OS:''' Windows, Linux

'''Overview:'''Ekiga is a software phone and video conferencing application. It allows you to hear and see your friends for free using your computer and Internet. You can also chat with them and see if they are online or not. It works on GNU/Linux and Windows.

More specifically, Ekiga is a VoIP, IP Telephony, and Video Conferencing application that allows you to make audio and video calls to remote users with SIP or H.323 compatible hardware and software.

It supports many audio and video codecs and all modern VoIP features for both SIP and H.323. Ekiga is the first Open Source application to support both H.323 and SIP, as well as audio and video. Ekiga was formerly known as GnomeMeeting.

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

==Express Talk==

[[File:Express0.png|200px|thumb|left|Express Talk]]

'''Product:''' Express Talk

'''Company:''' NCH Software

'''OS:''' Windows, MacOS

'''Overview:''' Express Talk works like a telephone to let you make calls through your computer. Call anyone via the internet who also has a softphone installed and if you sign up with a VoIP gateway service company you call regular telephone numbers as well.

With Express Talk you could potentially call someone using their internet IP (eg. "bob@1.2.3.4") but this is not usually practical. It is much easier to use a friendly address like "bob@myphoneco.com". To have this you need to sign up with a SIP proxy service. This is almost always free to signup for and usually requires nothing more than an email address. See our Recommended SIP Service Providers List for a list of options.

Some of the companies also provide gateway services. This means connecting phone calls to ordinary (non internet or analog PSTN) phone lines. These calls tend to be much less expensive particularly for international calls compared to regular long distance charges since they can offer local call rates.

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

==FgVoIP==

[[File:Fgvoip0.jpeg|200px|thumb|left|FgVoIP]]

'''Product:''' FgVoIP

'''Company:''' FG Microtec

'''OS:''' BlackberryOS

'''Overview:''' fg's BlackBerry SIP VoIP Client is a feature rich soft phone application for BlackBerry devices. It can register to any SIP VoIP provider or IP/SIP capable PBX using a WiFi connection in office, campus, home, hotel rooms, public hot-spots etc.

Its much like having a BlackBerry desktop phone in your pocket, providing significantly higher overall productivity while at the same time lowering telephony costs.

Experience the freedom of choosing your own VoIP provider whenever you want, wherever you want. All you need is a WiFi network and a VoIP account.

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

==Fring==

[[File:Fring00.jpg|left|Fring]]

'''Product:''' Fring

'''Company:''' Fring

'''OS:''' Windows, MacOS, IOS, Linux, Android, Symbian

'''Overview:'''
Fring mobile communication service that gives users internet-rich communication from their mobile phones.

Fring members make FREE video calls, voice calls, live chat to other fringsters and to friends on other social networks.

You can now use your SIP account even from non-SIP enabled phones.

Choose whichever SIP provider you have and in just few clicks, start using this account to make calls to landlines, GSM phones, or to other SIP providers.

Supports Symbian S60, iPhone/ iPod touch, Android, Windows Mobile, MeeGo, Linux, J2MEand proprietary OS.

Fring allows you to easily switch between them, without having to reconfigure your handset through a nightmarish process.

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

==Jitsi==

[[File:Logo2.png|200px|left|Jitsi]]

'''Product:''' Jitsi

'''Company:''' Jitsi

'''OS:''' Windows, MacOS, Linux

'''Overview:''' Jitsi (formerly SIP Communicator) is an audio/video and chat communicator that supports protocols such as SIP, XMPP/Jabber, AIM/ICQ, Windows Live, Yahoo! and many other useful features.

Jitsi is an Open Source / Free Software, and is available under the terms of the LGPL.

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

==Kiax==

[[File:Kiax3.png|left|200px|Kiax]]

'''Product:''' Kiax

'''Company:''' Kiax

'''OS:''' Windows, MacOS, Linux

'''Overview:'''
Kiax is a softphone (soft phone, VoIP client) with a simple and comfortable user interface for making VoIP calls to Asterisk PBX. It depends on the iaxclient library to use Asterisk's IAX2 protocol for easy call configuration and audio setup.

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

==Linphone==

[[File:Linphone_Icon.png|left|200px|Linphone]]

'''Product''': Linphone

'''Company''': Linphone

'''OS:''' Windows, Linux, MacOSX, IPhone, Android.

'''App Storage Consumption:''' 8.09mb (ability to move app to SD card. Data uses an additional ~0.89mb approx.)

'''Overview:''' Linphone is an internet phone or Voice Over IP phone (VoIP). With linphone you can communicate freely with people over the internet, with voice, video, and text instant messaging. Linphone makes use of the SIP protocol, an open standard for internet telephony. You can use Linphone with any SIP VoIP operator.

[[Linphone|See Configuration Details for PC]]
<br>
[[Linphone_iOS|See Configuration Details for iOS]]
<br>
[[Linphone_Android|See Configuration Details for Android]]
<div style="width:100%;overflow:hidden;clear:both">

</div>

==NetDial Sip Phone==

[[File:NetDial.jpg|left|200px|Linphone]]

'''Product''': NetDial Sip Phone

'''Company''': NeoMecca

'''OS:''' iOS, Android.

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

==Media5-fone==

[[File:Media5Android.jpg|left|Media5Android]]

'''Product:''' Media5-fone

'''Company:''' Media5 Corporation

'''OS:''' Android, iOS

'''Overview:'''
The Media5-fone is a softphone application that runs on the Android and the iOS Operating System. It is a SIP Client (softphone) that enables users to make and receive VoIP calls, enabling them to use their devices as an IP-PBX phone extension in their office or anywhere else in the world.

[[Media5-fone|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==MicroSIP==

[[File:MicroSIPlogo.png|left|200px|MicroSIP]]

'''Product:''' MicroSIP

'''Company:''' MicroSIP

'''OS:''' Windows XP/Vista/7/8/8.1/10, Linux, macOS

'''Overview:'''
MicroSIP is a softphone that works on Windows Operating Systems. It is a SIP Client (softphone) that enables users to make and receive VoIP calls, enabling them to use their devices as an IP-PBX phone extension in their office or anywhere else in the world as well as being really easy to configure.

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

==MizuPhone==

[[File:MizuLogo.png|left|200px|MizuPhone]]

'''Product:''' MizuPhone

'''Company:''' Mizutech

'''OS:''' Windows

'''Overview:'''
Mizu SoftPhone is our award winner professional VoIP client application based on the open standard SIP protocol with an easy to use modern interface. With Mizu SoftPhone you can connect to any SIP server on the public internet or on your local area network.

Multiple accounts and multiple SIP server registrations combined with a powerful dial plan can minimize your telecom bills while using the greatest features from the VoIP industry.

Features include built-in encryption, IM, presence, HD Video, history with voice records, skype like voice quality using ultra-wideband codecs and much more.

A free edition is also available based on the same engine as our corporate sip softphone but it has some features disabled (like G.729 and wideband codecs, multiple accounts and file transfer over SIP).

[[MizuPhone_Classic_Edition|See Configuration Details for MizuPhone Classic Edition]]
<br>
[[MizuPhone_Modern_Edition|See Configuration Details for MizuPhone Modern Edition]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==MizuDroid==

[[File:MizuDroid_logo.png|left|200px]]

'''Product:''' MizuDroid

'''Company:''' Mizutech

'''OS:''' Android

'''Overview:'''
MizuDroid is an unlocked VoIP softphone for Android mobile phones and tablets based on open standards, compatible with all VoIP providers, software and devices using the SIP protocol.

[[MizuDroid|See Configuration Details for MizuDroid]]

==NinjaLite==

[[File:Ninja6.png|left|200px|NinjaLite]]

'''Product:''' NinjaLite

'''Company:''' Global IP Telecommunications

'''OS:''' Windows

'''Overview:'''
Ninja Lite offers Audio- and video telephony for everybody, Ninja features individual skins and ringtones!, Ninja is natively Vista 32bit /64 bit and Windows7 compatible.

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

==PhonerLite==

[[File:PhonerLite.png|left|300px|PhonerLite]]

'''Product:''' PhonerLite

'''Company:''' PhonerLite

'''OS:''' Windows

'''Overview:'''
PhonerLite is a clearly arranged application for Windows. PhonerLite enables your PC to use it for internet telephony (VoIP , Voice over IP ). Pre-conditions are a full-duplex sound card , a microphone and speakers (alternativelya headset), an internet connection and a registration at a provider supporting the protocol SIP .PhonerLite supports several SIP profiles, each configurable independently. In thesame way the integrated phone book and call log are easy to use.

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

==PortGo==

[[File:Portgo3.png|left|200px|PortGo]]

'''Product:''' PortGo

'''Company:''' PortSIP Solutions, Inc

'''OS:''' Windows (desktop and mobile)

'''Overview:'''
PortGo is the newest SIP softphone from PortSIP, it's built base on PortSIP VoIP SDK, allowing users to enjoy multimedia communications in a dynamic way.

Featuring an intuitive interface, PortGo is expanding the softphone experience by making it even easier to make VoIP and Video over IP calls, see when your contacts are available and send Instant Messages. PortGo features an IM interface which focuses on your contacts and friends.

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

==QuteCom==

[[File:Qutecom3.png|left|200px|QuteCom]]

'''Product:''' QuteCom

'''Company:''' QuteCom

'''OS:''' Windows,MacOS,Linux

'''Overview:'''
QuteCom is a community of enthusiasts and developers, creating free software products related to communication over IP. The flagship product of the QuteCom project is a softphone which allows you to make free PC to PC video and voice calls, and to integrate all your IM contacts in one place.

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

==SessionTalk==

[[File:SessionTalk logo.png|left|200px|SessionTalk]]

'''Product:''' SessionTalk

'''Company:''' SessionTalk

'''OS:''' Android, iOS

'''Overview:'''
SessionTalk Mobile is a feature rich mobile SIP Client for iOS and Android handsets Voice calls and SMS Messaging* support.

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

==SIP Droid==

[[File:Sipdroid_00.png|200px|thumb|left|SIP Droid]]

'''Product:''' SIP Droid

'''Company:''' SIP Droid

'''OS:''' Android

'''App Storage Consumption:''' 1.85mb (ability to move app to SD card. Data uses an additional ~4kb approx.)

'''Overview:''' SIPDroid is a java based, open source SIP client that has recently been developed for use with mobile devices based on Google’s Android platform.

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

==SJPhone==

[[File:Sjphone0.png|200px|thumb|left|SJPhone]]

'''Product:''' SJPhone

'''Company:''' SJ Labs, Inc.

'''OS:''' Windows,Windows Mobile

'''Overview:'''
SJphone is a VOIP softphone that allows you to speak with any other softphone, any stand-alone IP-phone, or using ITSP with any traditional wired or mobile phone.

It supports both SIP and H.323 standards and is fully inter-operable with most major VOIP vendors and ITSP.

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

==Softphone.Pro==

[[File:SoftphoneProStats.png|350px|thumb|left|Softphone.Pro]]

'''Product:''' Softphone.Pro

'''Company:''' Vedisoft

'''OS:''' Windows, macOS

'''Overview:'''
Softphone.Prop is a great solution for call center agents, sales pros and support teams.

'''Main features:'''

• Easy integration with CRM
• 32 SIP accounts supported
• Call recording and screen capture
• Click-to-call and auto and easy 2-click transfer
• Voicemail Drop and Prerecorded Greeting on Answer
• ACW and BLF status control
• Professional headsets support
• Listen, whisper, barge
• Customizable interface
• Hotkeys

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

==Taki==

[[File:blackberry_z30_taki_oncall.png|200px|thumb|left|Taki]]

'''Product:''' Taki

'''Company:''' Taki

'''OS:''' BlackBerry 10, BlackBerry PlayBook

'''Project site:''' http://taki.sourceforge.net/

'''App in the AppWorld:''' http://appworld.blackberry.com/webstore/content/130295

'''Overview:''' Taki is a native SIP softphone for BlackBerry® PlayBook™ and
BlackBerry® 10 platforms.

'''Main Features:'''
- Multiple SIP accounts support
- Multiple simultaneous calls
- Call recording
- Conference calls
- Advanced call control: Transfer, Hold, Mute, Reject, Redial, switch
between multiple active calls
- Speaker phone support
- Address book. Integration with local contacts on BlackBerry 10 phones
- Comprehensive SIP settings (port binding, outbound proxy, registrar,
codecs priority, NAT traversal, STUN, Presence, and more)
- Transports: UDP/TCP
- Audio codecs: G.711a/u, GSM, iLBC, G.722 (HD), Speex (NB, WB, UWB)
- Built for BlackBerry

[[Taki|See Configuration Details]]

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

==Telephone==

[[File:telephone-user-interface.png|200px|thumb|left|Telephone]]

'''Product:''' Telephone

'''Company:''' 64 Characters

'''OS:''' OS X 10.9 or later

'''Project site:''' http://www.64characters.com/telephone/

'''Mac App Store:''' https://itunes.apple.com/us/app/telephone/id406825478

'''Overview:''' Telephone is a fully-featured SIP client with a minimalist user interface.

<br>

'''Main Features:'''

* Multiple SIP accounts
* Multiple simultaneous calls w/conferencing
* Call control: Transfer, Hold, Mute, Reject, Redial, Call Waiting, multiple active calls
* Speaker phone/headset support (via Mac sound device)
* Integration with Contacts app, dial by name, autocomplete, click to dial
* Send DTMF tones
* Languages: English, German, Russian

[[Telephone|See Configuration Details]]

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

==X-Lite==

[[File:Xlite_00.png|200px|thumb|left|X-Lite]]

'''Product:''' X-Lite

'''Company:''' CounterPath

'''OS:''' Windows, MacOS, Linux

'''Overview:''' CounterPath's X-Lite helps you seamlessly transition from a traditional phone environment into the world of Voice over IP.

The latest release of X-Lite provides a completely redesigned interface that allows for a contact-centric or dialpad-centric user experience, or a combination of the two. It also provides you with some of the most popular features of our fully loaded Bria and eyeBeam softphones so you can take them for a test drive before you make your purchase.

Having a simple voice conversation, you’ll soon see why having a softphone on your desktop or laptop is the ultimate communications experience.

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

==ZoiPer==

[[File:Zoiper0.png|300px|thumb|left|Zoiper Classic]]
[[File:zlogo.jpg|300px|thumb|right|https://www.zoiper.com/en]]

'''Product:''' ZoiPer

'''Company:''' ZoiPer

'''OS:''' Windows, Mac OS, Android, iOS

'''Overview:''' IAX & SIP multilanguage softphone is a VoIP soft client, meant to work with any IP-based communications systems and infrastructure.

Zoiper is available for Windows, Mac OS X and Linux and supports the following languages: English, German, Spanish, French, Dutch, Portuguese, Russian, Chinese, Japanese, Italian, Polish, Magyar.

*[[Zoiper_Classic|See Configuration Details for Zoiper Classic for PC/MAC]]<br />
*[[Zoiper_5| See Configuration Details for ZoiPer 5 for PC/MAC]]<br />
*[[Zoiper_5_Pro| See Configuration Details for ZoiPer 5 PRO for PC`MAC]]
*[[ZoIPer_for_Android|See Configuration Details for android]]<br/>
*[[ZoIPer_for_iOS|See Configuration Details for iOS]]<br/>
<div style="width:100%;overflow:hidden;clear:both"></div>
[[category:softphones]]

==ZoiPer Communicator==

[[File:Zoiperco1.png|200px|thumb|left|Zoiper Communicator]]
[[File:zlogo.jpg|300px|thumb|right|https://www.zoiper.com/en]]

'''Product:''' Zoiper Communicator

'''Company:''' Zoiper

'''OS:''' Windows, MacOS, Linux, Solaris

'''Overview:''' Zoiper Communicator IAX & SIP softphone is a converged Internet communication tool combining high-quality voice and video calls, fax, instant messaging and presence through a contact-centric intuitive interface.

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

Cisco Linksys PAP2T

2023-10-02T16:10:10Z

John:

[[File:Pap2t.jpg|300px|thumb|left|Cisco Linksys PAP2T]]
<div style="width:100%;overflow:hidden;clear:both"></div>

Haga clic aquí para la versión en Español:: [[Cisco Linksys PAP2T (Español)]]
__TOC__

'''If you are running the PAP2t with a tomato firmware router, this may represent a persistent issue after'''
'''rebooting the device, generally it won't register back as usual, in order to fix this you only need to set'''
'''Tomato UDP Unreplied timeout down to 10 (from default of 30).'''

==Configuration Details==

'''Step 1'''

The first step is to find out what IP Address your adapter is currently using. To do this, pick a phone connected on Line 1 and do the following:

'''Dial: **** (That is 4 asterisks)'''<br>
'''Once this is done, dial: 110# (110 followed by a square)'''

The system should now playback the IP Address your device has been assigned.<br>
(Example: 192.168.1.2)

'''Step 2'''

'''Using your favorite web browser from a computer on the same network, point the address to the IP address of your adapter.'''<br>
<nowiki>(example http://192.168.1.2)</nowiki> Replace 192.168.1.2 by the IP Address your device is currently using.

'''Step 3'''

You should now see the web interface of your Linksys/Sipura.

'''At the top of the page below the tabs, click on the link "Admin", and once the page has reloaded, click again on the link "Advanced View".'''

'''Step 4'''

'''Under the LINE 1 Tab, Find the following fields and fill them with the following information'''

'''Nat Keep Alive:''' Yes<br>
'''Nat Mapping/Traversal:''' Yes<br>
'''Proxy:''' atlanta.voip.ms (You can choose any of our multiple VoIP.ms servers)<br>
'''Register Expires:''' 180<br>
'''Proxy Fallback Intvl:''' 180<br>
'''Display Name:''' JOHN SMITH (Replace with your name or company name) ''' (See the requirements below)''' <br>
'''User ID:''' 100000 (Replace with your VoIP.ms account number or sub account ID)<br>
'''Password:''' ******** (Type Main account password, or sub account password)<br>
'''Use DNS SRV:''' NO<br>
'''DNS SRV Auto Prefix:''' NO

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

'''Step 5'''

You can configure your adapter with a better dial plan, allowing faster dialing of 10 digits number (Local US/Canada) and also enable 7 digits dialing in one area code of your choice.

'''At the bottom of Line 1 TAB, you will find a field called Dial Plan'''

Replace the 555 digits in the following line by the area code of your choice and copy the line, including parenthesis, in the Dialplan field in Line 1 Tab at the bottom of the page:

(911S0|310xxxx|<:1555>[2-9]xxxxxx|1[2-9]xx[2-9]xxxxxxS0|[2-9]xx[2-9]xxxxxxS0|*xx|***xxx|*xx.|[3468]11|822|0|00|4xxx|**275*x.|xxxxxxxxxxxx.)

[[Dial Plan for Linksys ATAs|Click here for more information on Linksys dial plans]]

'''Step 6'''

Click on the "Save Settings " button at the bottom of the form.

'''Configuration Screens'''

[[File:PAP2T001.JPG|PAP2T Configuration Screen|frame|left]]

[[File:PAP2T002.JPG|PAP2T Configuration Screen|frame|left]]

[[File:PAP2T003.JPG|PAP2T Configuration Screen|frame|left]]

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

'''Customer Submitted Information:''' <br>
For North America:<br>
Found this link on configuring the PAP2-NA hardware to work better in North America and specifically with VOIP.ms.<br>
Read the article called: [https://toao.net/VoIP/linksys-ata-configuration.html Configure your Linksys VoIP ATA the right way!]

To upgrade a firmware version from a Windows system, the PAP2T-NA documentation nor Cisco's web site does not say how to do this.
Go to [http://www.cisco.com/en/US/products/ps10029/index.html Cisco PAP2T Internet Phone Adapter with 2 VoIP Ports]. Click on the link "Download Firmware", which downloads the .zip file. Run the .exe file, then enter the IP number of the ATA device (called “SPA” in the program). It then upgrades the device. This is provided for your information: the author is not saying you have to upgrade the firmware.

== How to avoid the long delay to hear the ringtone ==

If you ever experience some delay to hear the ringtone when you make outgoing calls with your PAP2T. Changing the PAP2T's Interdigit Long Timer value can help resolve the issue. Follow the next steps in order to change that setting:

'''Note''': However before changing that option, test if calling the number with an # at the end of the number works(e.g. 5554441234#).
If that doesn't work you need to contact the support staff in voip.ms.

*1- First access the PAP2T's web interface.
*2- Click on the '''Admin Login''' and then click on '''(switch to advanced view)'''

[[File:Pap2admlog.jpg|800px]]

*3- Click on the '''Regional''' tab and look for the '''Control Timer Values (sec)''' section.

[[File:Regional.jpg|800px]]

*4- Enter the desire value in the '''Interdigit Long Timer''' field (for example lower this value to 4).

[[File:Ctrl timer values.jpg|800px]]

== '''Phone will not ring on handset''' ==

Sometimes the Phone you are using is designed for a certain Voltage and Ring Waveform. If someone tries to call you and the phone appears to be ringing for the caller but your phone never rings please follow these steps to hopefully resolve this issue for you.

Step 1: First access the PAP2's web interface.

Step 2: Click on the '''Admin Login''' and then click on '''(switch to advanced view)'''

Step 3: Click on your Regional Tab on the Top Menu.

Step 4: Go Halfway Down the Page until you see the Heading '''Ring and Call Waiting Tone Spec'''

[[File:Pap2Ring.jpg|800px]]

Step 5: Change the Ring Waveform setting to Sinusoid or Trapezoid, the opposite of what you have set. You can also change the Ring Voltage in increments of 5 to 90 or 95.

Step 6: Save Settings and Test an Incoming Call

== Receiving Unwanted Calls in the middle of the Night ( i.e. CallerID 100) that do not appear in your CDR: ==

These calls are not going through our Network but rather through the internet directly to your ATA Device.

Please look under the Voice>> Line 1 page in your Linksys device for the following setting: Restrict Source IP and make sure it's enabled.

This way the ATA device will block any traffic not coming from our servers.

[[File:RestrictSourceIP.png|800px]]

==Linksys PAP2T ATA Adapter Reset Procedure==

Sometimes it will be very helpful to reset your linksys ATA adapter to factory default settings.

*Connect a telephone to line 1 of the PAP2T unit and power it on.
*Disconnect your PAP2T adapter from the internet connection(unplug the Ethernet cable from the PAP2T hardware unit). Resetting with internet connection may mess up the unit making it completely useless.
*Dial ****, and wait for the Interactive Voice Menu (IVM) to get activated.
*Type in the following number including the # symbol.

73738#

(This number spells RESET.)

*Confirm this by pressing 1.

Your linksys ATA unit will now go back to it factory default settings.

'''Note:''' There are some PAP2/PAP2T devices in circulation which were originally 'locked' to one provider and subsequently unlocked by end users. Do *not* use the RESET$ command with these boxes.

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

User Manual : [https://www.cisco.com/c/dam/en/us/td/docs/voice_ip_comm/csbpvga/pap2t/administration/guide/pap2t_user.pdf Download PDF]

Admin Manual : [https://www.cisco.com/c/dam/en/us/td/docs/voice_ip_comm/csbpvga/ata/administration/guide/ATA_AG_v3_NC-WEB.pdf Download PDF]

== See also ==
* [[Dial Plan for Linksys ATAs]]
* You can check the most commonly used Star Codes from [[Cisco/Linksys Star Codes]]

[[Category:Analog Telephone Adapters]]

=== SPA Star Codes ===

* *69 Call Return Code
This code calls the last caller.

* *07 Call Redial Code
Redials the last number called. (Not in pap2t)

* *98 Blind Transfer Code
Begins a blind transfer of the current call to the extension specified after the activation code.

* *66 Call Back Act Code
Starts a callback when the last outbound call is not busy.

* *86 Call Back Deact Code
Cancels a callback.

* *05 Call Back Busy Act Code
Starts a callback when the last outbound call is busy. (Not in pap2t)

* *72 Cfwd All Act Code
Forwards all calls to the extension specified after the activation code.

* *73 Cfwd All Deact Code
Cancels call forwarding of all calls.

* *90 Cfwd Busy Act Code
Forwards busy calls to the extension specified after the activation code.

* *91 Cfwd Busy Deact Code
Cancels call forwarding of busy calls.

* *92 Cfwd No Ans Act Code
Forwards no-answer calls to the extension specified after the activation code.

* *93 Cfwd No Ans Deact Code
Cancels call forwarding of no-answer calls.

* *63 Cfwd Last Act Code
Forwards the last inbound or outbound calls to the extension specified after the activation code.

* *83 Cfwd Last Deact Code
Cancels call forwarding of the last inbound or outbound calls.

* *60 Block Last Act Code
Blocks the last inbound call.

* *80 Block Last Deact Code
Cancels blocking of the last inbound call.

* *64 Accept Last Act Code
Accepts the last outbound call. It lets the call ring through when do not disturb or call forwarding of all calls are enabled.

* *84 Accept Last Deact Code
Cancels the code to accept the last outbound call.

* *56 CW Act Code
Enables call waiting on all calls.

* *57 CW Deact Code
Disables call waiting on all calls.

* *71 CW Per Call Act Code
Enables call waiting for the next call.

* *70 CW Per Call Deact Code
Disables call waiting for the next call.

* *67 Block CID Act Code
Blocks caller ID on all outbound calls.

* *68 Block CID Deact Code
Removes caller ID blocking on all outbound calls.

* *81 Block CID Per Call Act Code
Blocks caller ID on the next outbound call.

* *82 Block CID Per Call Deact Code
Removes caller ID blocking on the next inbound call.

* *77 Block ANC Act Code
Blocks all anonymous calls.

* *87 Block ANC Deact Code
Removes blocking of all anonymous calls.

* *78 DND Act Code
Enables the do not disturb feature.

* *79 DND Deact Code
Disables the do not disturb feature.

* *65 CID Act Code
Enables caller ID generation.

* *85 CID Deact Code
Disables caller ID generation.

* *25 CWCID Act Code
Enables call waiting, caller ID generation.

* *45 CWCID Deact Code
Disables call waiting, caller ID generation.

* *26 Dist Ring Act Code
Enables the distinctive ringing feature.

* *46 Dist Ring Deact Code
Disables the distinctive ringing feature. The default is *46.

* *74 Speed Dial Act Code
Assigns a speed dial number.

* *16 Secure All Call Act Code
Makes all outbound calls secure.

* *17 Secure No Call Act Code
Makes all outbound calls not secure.

* *18 Secure One Call Act Code
Makes the next outbound call secure. (It is redundant if all outbound calls are secure by default.)

* *19 Secure One Call Deact Code
Secure One Call Deact Code Makes the next outbound call not secure. (It is redundant if all outbound calls are not secure by default.)

SMS-MMS

2023-09-05T17:12:50Z

John:

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

= What is SMS or MMS =

SMS means '''S'''hort '''M'''essage '''S'''ervice ('''SMS'''). With the addition of '''M'''ultimedia '''M'''essage '''S'''ervice ('''MMS''') to the service, users are now capable of sending and receiving media files ('''photos, videos, and audio''') as well as sending text messages with more than 160 characters (which is the limit for an SMS). The MMS service also allows the user to send a message to multiple recipients at the same time.

= Blog Articles =
:* [https://wiki.voip.ms/article/Benefits_of_SMS/MMS_Campaigns_vs._Email_Marketing Benefits of SMS/MMS Campaigns vs. Email Marketing]
:* [https://wiki.voip.ms/article/How_to_Generate_Leads_with_SMS/MMS%3F How to Generate Leads with SMS/MMS?]
:* [https://wiki.voip.ms/article/Instantly_Boost_Your_Customer_Experience,_Allow_Your_Customers_To_Text_You Instantly Boost Your Customer Experience, Allow Your Customers to Text You!]

= Service Limitations =

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🏗️ ''' Please notice that the MMS service is in the Beta phase.'''
: It is important to us that you report any issues with this service by sending an email to [mailto:support@voip.ms support] so that the developers can get involved if necessary.
</div>

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⚠️ '''LIMITATIONS'''
: ➜ '''The limit of the capacity to send SMS/MMS via the portal is unlimited.'''
: ➜ The limit of sending SMS/MMS via the API is limited to '''<u>100 per day</u>'''. For any modification to the limit, please contact support or open a ticket via the customer portal.
: ➜ At the moment, receiving and reading MMS is only possible through the message center or by using the SMS/MMS Forward option. '''MMS through SIP protocol is not available.'''

: ➜ The SMS/MMS Service is only available for <u>'''US and Canadian DID and Toll-Free Numbers'''</u> marked with the distinctive '''SMS/MMS Icon'''.
: ➜ The SMS/MMS Service can only be used to send SMS/MMS Messages to <u>'''Canadian and US'''</u> 10 digit numbers at this time.
: ➜ SIP SMS messages will only be delivered to and accepted from SIP trunks using username/password authentication.
: ➜ SIP SMS messages use the SIP MESSAGE method. See [https://tools.ietf.org/html/rfc3428 RFC 3428] for details.
: ➜ At this time we '''<u>cannot guarantee</u> that Short Code SMS Messages''', which are usually 6 digits or less ''(e.g.: Skype, Bank Codes, TV Commercials etc...)'' will work.
: ➜ We cannot guarantee that '''accents or special characters''' including non-Latin letters will be properly delivered.

: ➜ Public URL shorteners such as '''''bit.ly, tinyurl.com''''' etc, <font style="color:red">'''are widely restricted'''</font> by mobile providers as a response from the industry to combat SPAM campaigns and protect the consumer. Including this type of links into your messages '''<u>may result in a blocked or undelivered message</u>'''.
: ➜ For '''Ported In Numbers''' compatible with the feature: The SMS/MMS Functionality will be available up to '''48 hours after the porting process is marked as Completed'''.

: ➜ VoIP.ms reserves the right, at its sole discretion, to limit or disable the SMS/MMS service from any account that may present marketing patterns/automation patterns/bulk sending or receiving.

</div>

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'''💬 NOTE ''' If you have further questions don't hesitate to contact the Support Staff on the Live Chat or Ticket System.
:➜ '''SMS Messages have a cost of $0.0075 per incoming and outgoing message.'''
:➜ '''MMS Messages have a cost of $0.02 per incoming and outgoing message.'''

: ➜ '''Additionally, please note of the following restrictions for sending MMS:'''
:: ➜ Permitted attachment types are JPG, GIF, JPEG, PNG, MP3, WAV, MIDI, MP4 and 3GP.
:: ➜ The maximum size for each attachment is 1300 KB. ''(1.3 MB)''
:: ➜ You can attach up to 3 files.
:: ➜ Maximum characters in a text message is 2048.

</div>

=Using the SMS/MMS=
==Identifying a DID Number SMS/MMS capable==

The first thing you need to know is that this feature is only available for local US '''(incl. Alaska, Hawaii and Puerto Rico)''' and Canadian numbers at this time. You will note that some numbers have a little [[File:Sms-mms-icon.png|15px]] icon, this indicates that the number supports this feature.

You can start ordering a DID Number from your portal >> DID Numbers >> Order DID >> Local Numbers >> (US or Canada) Numbers and you will be able to purchase your desired number, just make sure it supports the MMS feature.

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

==Configuring the SMS/MMS service==

Once purchased the SMS/MMS Service needs to be activated on the DID settings, from your portal >> DID Numbers >> Manage DID >> Edit Selection >> SMS/MMS configuration section.

In that section, you can activate and configure some forward options for the SMS/MMS service.

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

:* '''Message Service (SMS/MMS)''': Check the box to enable the service.
:* '''VoIP.ms SMS/MMS Portal''': This is enabled by default and it can't be modified, that means that you will be able to create and send messages from the SMS Message Center.
:* '''SMS/MMS Email Address''': By activating this option all the SMS Messages will be sent to the email address you configure in this field. You can receive them to more than 1 email address by separating each emails with a comma. The advantage of this option is that you will receive your SMS Messages directly to your email and you will be able to Reply to these messages from your email too. You just need to click on '''reply''' to the email. Please note that e-mail responses are as well limited to 160 characters and if they exceed this limit, they'll be split in two (or more messages if applies) when sent.

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'''NOTE''' 💬: ➜ To reply the SMS Message via your email it is important to click on '''Reply''' and '''DON'T change or modify''' the destination mail "TO" (sms@voip.ms) and the subject E.G ''[#USXXXXXX] Message sent to 5555555555''. Make sure that your email address inbox isn't full to avoid loops when receiving an SMS/MMS message via email from an automated service.
</div>

:*'''SMS/MMS Forward''': You can have your SMS/MMS Messages forwarded to your cellphone or any other number that supports SMS/MMS, with this option you will receive the SMS/MMS Messages from our system to the configured phone number.
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'''NOTE''' 💬: ➜ The CallerID sent to the Call Forwarded Cell Phone will be the DID Number that received the SMS Message. Is not possible to forward the original Caller ID at this time. When using SMS/MMS Forward there is a charge for the incoming message received by the DID and the outgoing message sent to the number set for the forward.
</div>

:*'''SMS/MMS SIP Account''': You can have your SMS Messages forwarded to your selected SIP account. They will be sent through our SMS/SIP gateway as a SIP MESSAGE.
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'''NOTE''' 💬: ➜ The destination ("To" header field) of the SIP MESSAGE will be your account name, instead of the destination number of the original SMS.
</div>

:'''SMS/MMS URL Callback''': By enabling this option you will be able to send the messages to another destination e.g. your own server. If selected SMS Messages received by your DID will Send a GET request to the URL Callback provided. Available variables for your URL:
<div style="margin-left:50px;
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:: '''Example''': <nowiki>https://YourDomainName.tld/sms.php?to={TO}&from={FROM}&message={MESSAGE}&id={ID}&date={TIMESTAMP}</nowiki>

: '''{ID}''' = The ID of the SMS message.
: '''{TIMESTAMP}''' = The date and time the message was received.
: '''{FROM}''' = The phone number that sent you the message.
: '''{TO}''' = The DID Number that received the message.
: '''{MESSAGE}''' = The content of the message.

</div>

:*'''URL Callback Retry''': When selected, we will be expecting an "ok" output (without quotes) from your URL callback page as an indicator that you have received the message correctly. If we don't received the "ok" (without quotes) from your callback page, we will keep sending you the same message every 30 minutes.

Once you configure the desired option, click on Apply Changes.

==Sending a SMS/MMS (Web Interface)==

'''Step 1'''

: Click the "send New Message" button to get a list of contacts displayed or directly type in the number into the contact text field. If this number is registered in the phonebook, it will display its name instead.

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

'''Step 2'''

:Click to display the list of available DID numbers. Choose the number to send the message from.

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

'''Step 3'''

:Write your message on the text field and click the red arrow to send it. Text messages up to 160 characters will be considered SMS. Media messages or text messages over 160 characters will be considered MMS and you will see the icon turning from SMS to MMS indicating of this.
:: [[File:MMS-Webportal4.png|border|400px]]

'''Step 4'''

: In order to send a media file over an MMS, click the clip button to browse and choose a media file.

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

: Click the browse button to search for the media file in your computer or drag and drop the file directly into the box.

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

: The attached files will show below the text box.

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

: Once the message is sent, a new conversation will be opened on the left. Selecting this conversation will show the messages history on the box.

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

==Receiving an SMS/MMS (Web Interface)==

: While SMS messages can be received directly in an App through the SIP Protocol and correct configuration, at the moment, '''receiving and reading MMS is only possible through the message center.'''

: You can access the communication center from your customer portal at DID numbers > SMS/MMS Message Center. Any message received will appear here on the left side of the panel.

: Use the DID Filter to choose and display messages for a specific DID number or for all DID numbers.

: You can also filter the messages by date and time zone by clicking on the down arrow, which will show the search parameters

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

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

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

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

== SMS/MMS History ==

You can now also gain access to a general history of SMS/MMS that was sent and received for all of your DIDs, or for a specific one by going to Customer Portal >> DID Numbers >> SMS/MMS Message Center >> History.

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

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

= SMS using a SMPP Server =

An SMPP server, short of '''Short Message Peer to Peer''' is a method designed to send and receive high volumes of SMS messages through a dedicated server.

In other words, this allows you to better control the quantity of messages that will be sent to your target audience while avoiding to pass via a softphone application or your VoIP.ms portal to send/receive these SMS, thus eliminating multiple repetitive tasks and saving you time.

For example, if you have a promotion for your business that offers a discount for a certain lapse of time and that your customers have provided their agreement to receive these types of promotions via SMS, you could send your promotion to your customers in a faster manner than if you would do it from your customer portal, thus increasing efficiency.
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'''NOTES''' 💬: ➜ Note that sending and receiving SMS messages will be charged our usual rate at $0.0075 per messages'''
</div>

==Requirements==

: '''This will require you to have your own SMPP server.'''
: Once configured properly, you will then be able to send and receive SMS messages to your own server.

== Configuration on your SMPP Server ==

: In order for your server to communicate and authenticate accordingly with VoIP.ms SMPP server, you will need to configure the following:

::* A '''username''' and '''password''' of your choice in order for VoIP.ms SMPP server to authenticate you.
::* Have all your communications be sent to '''smpp.voip.ms''' via port '''2775''', for regular SMPP, or port '''3550''' which is our encrypted SMPP port.
::* Open a connection using the '''bind_transceiver''' command. You can keep this connection open for as long as you need to and inquire about it using the '''enquire_link''' command.
::* Send your messages with either '''deliver_sm''' or '''submit_sm''' commands. Make sure to include your PDU's '''source_addr''' as one of your SMS enabled DIDs from your account. Your destination SMS number should be set as '''destination_addr''' while your text message will be included in '''short_message'''.

== Configuration SMPP on your VoIP.ms Portal ==

Once you have set up the above on your server, you will need to enable and specify which username and password will be used by VoIP.ms to authenticate your SMPP requests. In order to do so, head into your VoIP.ms portal, DID Numbers, Manage DIDs, edit your DID that supports SMS messages and by heading down the page, you will see the following options:

:* '''SMPP Enabled:''' If selected, SMS Messages will be allowed to be sent and received using SMPP.
:: You may specify the URL to be used to send a copy of inbound messages to your SMPP server. Leave empty if you only wish to send messages. See the example below. The '''scheme''' part is mandatory and must be set to either of '''smpp''' for unencrypted SMPP, or '''ssmpp''' for encrypted SMPP using TLS. The '''user:password@''' part is optional and will default to the same username and password defined for authentication on our side as defined below. The ''':port''' part is also optional and will default to 2775 for '''smpp''' and to '''3550''' for ssmpp.

:* '''SMPP Username:''' The username that you will use to authenticate to our SMPP server for sending messages
:* '''SMPP Password:''' The password that you will use to authenticate to our SMPP server for sending messages

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'''NOTES''' 💬
: ➜ In order to help avoid any confusion while configuring the username on your server, you can use your VoIP.ms account ID as a username (the 6 digits of your account).
: ➜ Make sure to have the option '''Message Service (SMS/MMS)''' enabled.
</div>

Enable the option '''SMPP Enabled''', fill the username and password per what was configured on your SMPP server and once done, apply the changes by pressing '''Click here to apply changes'''.

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

You may now start sending and receiving SMS messages from your SMPP server.

= Send and Receive Text Messages (Web Interface)=

==SMS Message Center for mobile==

To start using the service you will need to use the SMS Message Center, from your portal >> DID Numbers >> [https://www.voip.ms/m/sms.php SMS Message Center].

For Mobile Devices please use [https://sms.voip.ms/ sms.voip.ms] as a complete all in one solution.

From the SMS Message Center you can check your SMS History, send and receive SMS messages.

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

:* '''Search Range''': Select a Date Range of your SMS History. You can select a time period by marking the "Show Details" field and the text messages will be displayed.
:* '''Search Filter''': You can filter your Search by DID, Contact and Type (Sent, Received and Both).
:* '''Send New SMS Messages''': Another window will open by clicking here in order to create a new SMS Message.
:* '''Delete Selected Messages''': This action will delete any messages that you have selected, by clicking on the box next to them. '''This action cannot be undone.'''
:* '''Delete All Messages''': Delete ALL your SMS Messages, both sent and received. '''This action cannot be undone.'''
:* '''Checking a Message''': To check a received SMS Message just click it directly in order to display the information.

==Create a New SMS Message (Legacy)==

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

You can create a new '''SMS Message''' we will explain you all the parameters you can find on this section:

* '''DID''': If you have more than one DID Number you can select any of the numbers available to send your message.
* '''Contact''': The destination phone number or if you have a [[Phone book|Phone book entry]] start typing the name and the system will display the information if available.
* '''Message''': You can create your message on that field, please note that you have up to 160 characters.

Then click on 'Send Message' to send the SMS.

= Send and Receive Messages (via SIP MESSAGE Protocol) =
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'''NOTES''' 💬
: ➜ '''SMS via SIP''' service is only available through '''SIP Protocol and user/password authentication'''. '''IP Auth and IAX2 are not available at this time.'''
: ➜ '''Please refer to our guide [[SIP/SMS with FreePBX]] for instructions on how to configure the service with a FreePBX'''
</div>

== Sending ==

:* DID must have the '''"Short Message Service (SMS)"''' service option enabled. This option can be enabled from the Manage DIDs > Edit DID page.

:* To send an SMS from your SIP account, it is required that you set your Caller ID number to one of your SMS enabled DIDs. This will be the number you will be sending the message from. You can configure the Caller ID number from your customer portal for the specific subaccount if you are using a softphone or directly from your extension or trunk if you are using an Asterisk or PBX server.

:* It's important to note that if you configure your caller ID name in your SIP client to be a 10 digits number, this will override your caller ID number. If the caller ID name is anything different than a 10 digits number this will be discarded and the Caller ID number will be used.

:* If you are an asterisk or PBX user, please make sure to use the latest version of '''Asterisk (v12 or higher)''' and use '''chan_pjsip''' for the trunk. This is an asterisk limitation.

:* The Desktop version of Zoiper requires the user to have the '''PRO version''' activated to be able to send SMS messages as well as to enable '''SIP Presence'''. Please contact Technical Support to request SIP presence to be enabled for your account for the VoIP.ms POP you are using. While normally not required, there may be other cases not documented of softphones or apps requiring SIP Presence to be enabled. Please contact technical support if you encounter any issue with your specific software.

:* The '''free Desktop version''' of Zoiper cannot send SMS messages and is only capable of receiving SMS.

:* The '''mobile version''' of Zoiper is completely free of these limitations. Free version already works for sending and receiving without the need of SIP presence.

:* When sending an SMS message, the recipient number must be 10 digits. Avoid using the prefix '"1"' since is not supported at the moment. The system will recognize the numbers as in 10 digits.

:* The current characters limit for a '''single SMS is 160'''. If a message is bigger than 160 characters it will be split into blocks of up to 160 characters each and be sent as independent messages. The maximum total message length for SMS via SIP is 2000 characters.

== Receiving ==
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'''NOTES''' 💬: ➜ AT THE MOMENT SENDING MMS USING SIP ACCOUNT IS NOT POSSIBLE AT THIS TIME.'''
</div>

:*DID must have the '''"Message Service (SMS/MMS)"''' service option enabled. This option can be enabled from the Manage DIDs > Edit DID page.
:*Enable the option '''"SMS SIP Account"''' and select the account destination for your SMS messages to this number.
:*The account receiving SMS must be registered successfully in any of our POPs.
:*To ensure the best results receiving the SMS messages, do not register the same SIP account on more than one softphone or app at the same time, this can lead to unexpected results.
:*For SIP devices that require this information, such as a trunk or PBX, we encode the original destination number in a custom SIP header field named "X-SMS-To".
:*3CX softphone currently does not support any third party SMS service. It will only work by having a direct connection to the 3CX PBX system. '''Please contact 3CX for more details'''.

== Send SIP/SMS with FreePBX ==

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

'''NOTES''' 💬: ➜ AT THE MOMENT SENDING MMS USING SIP ACCOUNT IS NOT POSSIBLE AT THIS TIME.'''
</div>

Yes, it is possible to configure a FreePBX system to send SMS, please refer [[SIP/SMS with FreePBX | here]] for detailed instructions.

=SMS/MMS using the Reseller Interface=

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

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

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

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

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

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

Then scroll down and find the feature "’’’SMS/MMS’’’", and enable it.

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

= Enabling/Disabling SMS/MMS on a DID via the client's interface =
For your client to enable or disable SMS/MMS from his portal, your customer will need to head under Services, My Services.
: [[File:SMSMMSreseller5.png|thumb|none|500px]]

Then proceed by heading into ‘’’DID Lists’’’ tab and edit DID.

: [[File:SMSMMSreseller6.png|thumb|none|500px]]
You will then be able to enable SMS/MMS, receive via email and more.

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

= Accessing their SMS/MMS via the clients interface =
1) To gain access to their SMS/MMS messages, head under Services and then SMS/MMS.
: [[File:SMSMMSreseller8.png|thumb|none|500px]]

[[category:guides]]

File:MicroSIPTLS.png

2023-08-08T14:22:25Z

John:

Microsip

2023-08-08T14:21:51Z

John:

==Tutorial Video==

[[Image:MicroSIPthumbnail.jpg|200px|link=https://www.youtube.com/watch?v=2UJm2J0j-2k]]

This is the very basic set up for MicroSIP, there are only a few steps needed to have your device up and running. If you wish to improve your softphone performance or enable some features, refer to the advanced section.

==Basic setup==
1. Download and install '''MicroSIP''' from this [https://www.microsip.org/downloads/current link.]

2. Once downloaded and installed, start MicroSIP and on the welcome screen head onto the arrow pointing down on the top right of the softphone application > Add Account.

[[File:Microsip.png|300px|border]]
<br>
<br>

3. Fill in the following
<br>
<br>
'''Account name:''' <b style="color:#b30000;">Type your phone number or write any name, it's just an identifier</b><br>
'''SIP Server:''' <b style="color:#b30000;">Type the server of your number, check it out from your portal, i.e. denver1.voip.ms</b><br>
'''Username:''' <b style="color:#b30000;">Type your account number or subaccount name</b><br>
'''Domain:''' <b style="color:#b30000;">Type the server of your number, check it out from your portal, i.e. denver1.voip.ms</b><br>
'''Password:''' <b style="color:#b30000;">Type your password</b><br>
'''Display name:''' <b style="color:#b30000;">Type your name or company name (NOT guaranteed to be displayed)</b><br>
<br>

Finally, press '''Save''' and you are done configuring MicroSIP.

[[File:Microsip3.png|300px|border]] [[File:Popserver.png|600px|border]]
<br>
<br>

4. Go to the dialpad and test the audio dialing 4443

[[File:Microsip4.png|300px|border]] [[File:Microsip5.png|300px|border]]
<br>
<br>

5. Finally, define a caller ID for your username so you can start making outgoing calls:
<br>
<br>
<b>For the main username: </b><b style=color:#b30000>Go to Main Menu -> Account settings -> General tab and fill the "Caller ID number"</b><br>
<b>For a subaccount/extension: </b><b style=color:#b30000>Go to Subaccounts -> Manage subaccount -> Edit subaccount and fill the "Caller ID number"</b><br>

In both cases, make sure to wait 3 minutes after setting the caller ID number and trying to dial out <br>

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

==Additional Settings (Optional)==

===TLS SRTP===

In order to enable TLS SRTP (call encryption), you will first need to enable this feature by modifying your main account under Main Menu > Account Settings > Advanced tab and setting '''Encrypted SIP Traffic''' to '''Yes'''

Once done, go into the menu option (arrow pointing down) on the top right of MicroSIP > Edit Account and set the following options:
:- Media Encryption: Mandatory SRTP
:- Transport: TLS

[[File:microSIPTLS.png|thumb|none|600px|Click to enlarge]]

Save and then head once again on the top right of MicroSIP > Settings and once there, modify '''Source Port''' option to 5061 and RTP to 10001 to 20000.

Save these settings and you will now

[[File:microSIP4.png|thumb|none|600px|Click to enlarge]]

===Some additional settings===
By heading into the menu option (arrow pointing down) on the top right of MicroSIP > Settings there are some settings that can be improved in order to avoid any issues while on a call:

:- Enabled Codecs: Only have G711u allowed (or GSM if you have a slow internet connection)
:- Source port: Set it to 5060 (5080 or 42872 for alternate ports) or 5061 (5081 or 42873 for alternate ports) for call encryption
:- RTP Ports: Set the minimum to 10001 and maximum to 20000
:- EC: Make sure this option is enabled. This is an Echo Cancellation option

Otherwise, there are many other options that you can customize like the ring tone you will hear when someone calls among many other options you can customize or enable that you can have a look into. You can also consult [https://www.microsip.org/help/this link] if you have doubts on the options MicroSIP offers.

File:MicroSIP3.png

2023-08-07T18:26:26Z

John: uploaded a new version of "File:MicroSIP3.png": Reverted to version as of 18:21, 7 August 2023

File:MicroSIP3.png

2023-08-07T18:25:47Z

John: uploaded a new version of "File:MicroSIP3.png": Reverted to version as of 18:20, 7 August 2023

File:MicroSIP3.png

2023-08-07T18:21:58Z

John: uploaded a new version of "File:MicroSIP3.png"

File:MicroSIP3.png

2023-08-07T18:20:21Z

John: uploaded a new version of "File:MicroSIP3.png"

File:MicroSIP3.png

2023-08-07T18:19:07Z

John: uploaded a new version of "File:MicroSIP3.png"

File:MicroSIP3.png

2023-08-07T18:16:30Z

John: uploaded a new version of "File:MicroSIP3.png": Reverted to version as of 14:38, 8 July 2020

File:MicroSIP3.png

2023-08-07T18:15:29Z

John: uploaded a new version of "File:MicroSIP3.png"

Microsip

2023-08-07T17:50:06Z

John:

==Tutorial Video==

[[Image:MicroSIPthumbnail.jpg|200px|link=https://www.youtube.com/watch?v=2UJm2J0j-2k]]

This is the very basic set up for MicroSIP, there are only a few steps needed to have your device up and running. If you wish to improve your softphone performance or enable some features, refer to the advanced section.

==Basic setup==
1. Download and install '''MicroSIP''' from this [https://www.microsip.org/downloads/current link.]

2. Once downloaded and installed, start MicroSIP and on the welcome screen head onto the arrow pointing down on the top right of the softphone application > Add Account.

[[File:Microsip.png|300px|border]]
<br>
<br>

3. Fill in the following
<br>
<br>
'''Account name:''' <b style="color:#b30000;">Type your phone number or write any name, it's just an identifier</b><br>
'''SIP Server:''' <b style="color:#b30000;">Type the server of your number, check it out from your portal, i.e. denver1.voip.ms</b><br>
'''Username:''' <b style="color:#b30000;">Type your account number or subaccount name</b><br>
'''Domain:''' <b style="color:#b30000;">Type the server of your number, check it out from your portal, i.e. denver1.voip.ms</b><br>
'''Password:''' <b style="color:#b30000;">Type your password</b><br>
'''Display name:''' <b style="color:#b30000;">Type your name or company name (NOT guaranteed to be displayed)</b><br>
<br>

Finally, press '''Save''' and you are done configuring MicroSIP.

[[File:Microsip3.png|300px|border]] [[File:Popserver.png|600px|border]]
<br>
<br>

4. Go to the dialpad and test the audio dialing 4443

[[File:Microsip4.png|300px|border]] [[File:Microsip5.png|300px|border]]
<br>
<br>

5. Finally, define a caller ID for your username so you can start making outgoing calls:
<br>
<br>
<b>For the main username: </b><b style=color:#b30000>Go to Main Menu -> Account settings -> General tab and fill the "Caller ID number"</b><br>
<b>For a subaccount/extension: </b><b style=color:#b30000>Go to Subaccounts -> Manage subaccount -> Edit subaccount and fill the "Caller ID number"</b><br>

In both cases, make sure to wait 3 minutes after setting the caller ID number and trying to dial out <br>

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

==Additional Settings (Optional)==

===TLS SRTP===

In order to enable TLS SRTP (call encryption), you will first need to enable this feature by modifying your main account under Main Menu > Account Settings > Advanced tab and setting '''Encrypted SIP Traffic''' to '''Yes'''

Once done, go into the menu option (arrow pointing down) on the top right of MicroSIP > Edit Account and set the following options:
:- Media Encryption: Mandatory SRTP
:- Transport: TLS

[[File:microSIP3.png|thumb|none|600px|Click to enlarge]]

Save and then head once again on the top right of MicroSIP > Settings and once there, modify '''Source Port''' option to 5061 and RTP to 10001 to 20000.

Save these settings and you will now

[[File:microSIP4.png|thumb|none|600px|Click to enlarge]]

===Some additional settings===
By heading into the menu option (arrow pointing down) on the top right of MicroSIP > Settings there are some settings that can be improved in order to avoid any issues while on a call:

:- Enabled Codecs: Only have G711u allowed (or GSM if you have a slow internet connection)
:- Source port: Set it to 5060 (5080 or 42872 for alternate ports) or 5061 (5081 or 42873 for alternate ports) for call encryption
:- RTP Ports: Set the minimum to 10001 and maximum to 20000
:- EC: Make sure this option is enabled. This is an Echo Cancellation option

Otherwise, there are many other options that you can customize like the ring tone you will hear when someone calls among many other options you can customize or enable that you can have a look into. You can also consult [https://www.microsip.org/help/this link] if you have doubts on the options MicroSIP offers.

SMS-MMS

2023-05-16T19:36:50Z

John:

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

= What is SMS or MMS =

SMS means '''S'''hort '''M'''essage '''S'''ervice ('''SMS'''). With the addition of '''M'''ultimedia '''M'''essage '''S'''ervice ('''MMS''') to the service, users are now capable of sending and receiving media files ('''photos, videos, and audio''') as well as sending text messages with more than 160 characters (which is the limit for an SMS). The MMS service also allows the user to send a message to multiple recipients at the same time.

= Blog Articles =
:* [https://wiki.voip.ms/article/Benefits_of_SMS/MMS_Campaigns_vs._Email_Marketing Benefits of SMS/MMS Campaigns vs. Email Marketing]
:* [https://wiki.voip.ms/article/How_to_Generate_Leads_with_SMS/MMS%3F How to Generate Leads with SMS/MMS?]
:* [https://wiki.voip.ms/article/Instantly_Boost_Your_Customer_Experience,_Allow_Your_Customers_To_Text_You Instantly Boost Your Customer Experience, Allow Your Customers to Text You!]

= Service Limitations =

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

🏗️ ''' Please notice that the MMS service is in the Beta phase.'''
: It is important to us that you report any issues with this service by sending an email to [mailto:support@voip.ms support] so that the developers can get involved if necessary.
</div>

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

⚠️ '''LIMITATIONS'''
: ➜ '''The limit of the capacity to send SMS/MMS via the portal is unlimited.'''
: ➜ The limit of sending SMS/MMS via the API is limited to '''<u>100 per day</u>'''. For any modification to the limit, please contact support or open a ticket via the customer portal.
: ➜ At the moment, receiving and reading MMS is only possible through the message center or by using the SMS/MMS Forward option. '''MMS through SIP protocol is not available.'''

: ➜ The SMS/MMS Service is only available for <u>'''US and Canadian local DID Numbers'''</u> marked with the distinctive '''SMS/MMS Icon'''.
: ➜ The SMS/MMS Service can only be used to send SMS/MMS Messages to <u>'''Canadian and US'''</u> 10 digit numbers at this time.
: ➜ SIP SMS messages will only be delivered to and accepted from SIP trunks using username/password authentication.
: ➜ SIP SMS messages use the SIP MESSAGE method. See [https://tools.ietf.org/html/rfc3428 RFC 3428] for details.
: ➜ At this time we '''<u>cannot guarantee</u> that Short Code SMS Messages''', which are usually 6 digits or less ''(e.g.: Skype, Bank Codes, TV Commercials etc...)'' will work.
: ➜ We cannot guarantee that '''accents or special characters''' including non-Latin letters will be properly delivered.

: ➜ Public URL shorteners such as '''''bit.ly, tinyurl.com''''' etc, <font style="color:red">'''are widely restricted'''</font> by mobile providers as a response from the industry to combat SPAM campaigns and protect the consumer. Including this type of links into your messages '''<u>may result in a blocked or undelivered message</u>'''.
: ➜ For '''Ported In Numbers''' compatible with the feature: The SMS/MMS Functionality will be available up to '''48 hours after the porting process is marked as Completed'''.

: ➜ VoIP.ms reserves the right, at its sole discretion, to limit or disable the SMS/MMS service from any account that may present marketing patterns/automation patterns/bulk sending or receiving.

</div>

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

'''💬 NOTE ''' If you have further questions don't hesitate to contact the Support Staff on the Live Chat or Ticket System.
:➜ '''SMS Messages have a cost of $0.0075 per incoming and outgoing message.'''
:➜ '''MMS Messages have a cost of $0.02 per incoming and outgoing message.'''

: ➜ '''Additionally, please note of the following restrictions for sending MMS:'''
:: ➜ Permitted attachment types are JPG, GIF, JPEG, PNG, MP3, WAV, MIDI, MP4 and 3GP.
:: ➜ The maximum size for each attachment is 1300 KB. ''(1.3 MB)''
:: ➜ You can attach up to 3 files.
:: ➜ Maximum characters in a text message is 2048.

</div>

=Using the SMS/MMS=
==Identifying a DID Number SMS/MMS capable==

The first thing you need to know is that this feature is only available for local US and Canadian numbers at this time. You will note that some numbers have a little [[File:Sms-mms-icon.png|15px]] icon, this indicates that the number supports this feature.

You can start ordering a DID Number from your portal >> DID Numbers >> Order DID >> Local Numbers >> (US or Canada) Numbers and you will be able to purchase your desired number, just make sure it supports the MMS feature.

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

==Configuring the SMS/MMS service==

Once purchased the SMS/MMS Service needs to be activated on the DID settings, from your portal >> DID Numbers >> Manage DID >> Edit Selection >> SMS/MMS configuration section.

In that section, you can activate and configure some forward options for the SMS/MMS service.

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

:* '''Message Service (SMS/MMS)''': Check the box to enable the service.
:* '''VoIP.ms SMS/MMS Portal''': This is enabled by default and it can't be modified, that means that you will be able to create and send messages from the SMS Message Center.
:* '''SMS/MMS Email Address''': By activating this option all the SMS Messages will be sent to the email address you configure in this field. You can receive them to more than 1 email address by separating each emails with a comma. The advantage of this option is that you will receive your SMS Messages directly to your email and you will be able to Reply to these messages from your email too. You just need to click on '''reply''' to the email. Please note that e-mail responses are as well limited to 160 characters and if they exceed this limit, they'll be split in two (or more messages if applies) when sent.

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

'''NOTE''' 💬: ➜ To reply the SMS Message via your email it is important to click on '''Reply''' and '''DON'T change or modify''' the destination mail "TO" (sms@voip.ms) and the subject E.G ''[#USXXXXXX] Message sent to 5555555555''. Make sure that your email address inbox isn't full to avoid loops when receiving an SMS/MMS message via email from an automated service.
</div>

:*'''SMS/MMS Forward''': You can have your SMS/MMS Messages forwarded to your cellphone or any other number that supports SMS/MMS, with this option you will receive the SMS/MMS Messages from our system to the configured phone number.
<div style="margin-left:50px;
padding:5px;
border-left: 6px solid rgb(208, 56, 45, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(51, 51, 51, 0.08);
width:70vw;">

'''NOTE''' 💬: ➜ The CallerID sent to the Call Forwarded Cell Phone will be the DID Number that received the SMS Message. Is not possible to forward the original Caller ID at this time. When using SMS/MMS Forward there is a charge for the incoming message received by the DID and the outgoing message sent to the number set for the forward.
</div>

:*'''SMS/MMS SIP Account''': You can have your SMS Messages forwarded to your selected SIP account. They will be sent through our SMS/SIP gateway as a SIP MESSAGE.
<div style="margin-left:50px;
padding:5px;
border-left: 6px solid rgb(208, 56, 45, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(51, 51, 51, 0.08);
width:70vw;">

'''NOTE''' 💬: ➜ The destination ("To" header field) of the SIP MESSAGE will be your account name, instead of the destination number of the original SMS.
</div>

:'''SMS/MMS URL Callback''': By enabling this option you will be able to send the messages to another destination e.g. your own server. If selected SMS Messages received by your DID will Send a GET request to the URL Callback provided. Available variables for your URL:
<div style="margin-left:50px;
padding:5px;
font-family: Consolas, Monaco, Lucida Console, Liberation Mono, DejaVu Sans Mono, Bitstream Vera Sans Mono, Courier New, monospace;
color: silver;
border-left: 6px solid rgb(208, 56, 45, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: black;
width:58vw;">
:: '''Example''': <nowiki>https://YourDomainName.tld/sms.php?to={TO}&from={FROM}&message={MESSAGE}&id={ID}&date={TIMESTAMP}</nowiki>

: '''{ID}''' = The ID of the SMS message.
: '''{TIMESTAMP}''' = The date and time the message was received.
: '''{FROM}''' = The phone number that sent you the message.
: '''{TO}''' = The DID Number that received the message.
: '''{MESSAGE}''' = The content of the message.

</div>

:*'''URL Callback Retry''': When selected, we will be expecting an "ok" output (without quotes) from your URL callback page as an indicator that you have received the message correctly. If we don't received the "ok" (without quotes) from your callback page, we will keep sending you the same message every 30 minutes.

Once you configure the desired option, click on Apply Changes.

==Sending a SMS/MMS (Web Interface)==

'''Step 1'''

: Click the "send New Message" button to get a list of contacts displayed or directly type in the number into the contact text field. If this number is registered in the phonebook, it will display its name instead.

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

'''Step 2'''

:Click to display the list of available DID numbers. Choose the number to send the message from.

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

'''Step 3'''

:Write your message on the text field and click the red arrow to send it. Text messages up to 160 characters will be considered SMS. Media messages or text messages over 160 characters will be considered MMS and you will see the icon turning from SMS to MMS indicating of this.
:: [[File:MMS-Webportal4.png|border|400px]]

'''Step 4'''

: In order to send a media file over an MMS, click the clip button to browse and choose a media file.

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

: Click the browse button to search for the media file in your computer or drag and drop the file directly into the box.

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

: The attached files will show below the text box.

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

: Once the message is sent, a new conversation will be opened on the left. Selecting this conversation will show the messages history on the box.

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

==Receiving an SMS/MMS (Web Interface)==

: While SMS messages can be received directly in an App through the SIP Protocol and correct configuration, at the moment, '''receiving and reading MMS is only possible through the message center.'''

: You can access the communication center from your customer portal at DID numbers > SMS/MMS Message Center. Any message received will appear here on the left side of the panel.

: Use the DID Filter to choose and display messages for a specific DID number or for all DID numbers.

: You can also filter the messages by date and time zone by clicking on the down arrow, which will show the search parameters

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

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

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

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

== SMS/MMS History ==

You can now also gain access to a general history of SMS/MMS that was sent and received for all of your DIDs, or for a specific one by going to Customer Portal >> DID Numbers >> SMS/MMS Message Center >> History.

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

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

= SMS using a SMPP Server =

An SMPP server, short of '''Short Message Peer to Peer''' is a method designed to send and receive high volumes of SMS messages through a dedicated server.

In other words, this allows you to better control the quantity of messages that will be sent to your target audience while avoiding to pass via a softphone application or your VoIP.ms portal to send/receive these SMS, thus eliminating multiple repetitive tasks and saving you time.

For example, if you have a promotion for your business that offers a discount for a certain lapse of time and that your customers have provided their agreement to receive these types of promotions via SMS, you could send your promotion to your customers in a faster manner than if you would do it from your customer portal, thus increasing efficiency.
<div style="margin-left:50px;
padding:5px;
border-left: 6px solid rgb(208, 56, 45, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(51, 51, 51, 0.08);
width:70vw;">

'''NOTES''' 💬: ➜ Note that sending and receiving SMS messages will be charged our usual rate at $0.0075 per messages'''
</div>

==Requirements==

: '''This will require you to have your own SMPP server.'''
: Once configured properly, you will then be able to send and receive SMS messages to your own server.

== Configuration on your SMPP Server ==

: In order for your server to communicate and authenticate accordingly with VoIP.ms SMPP server, you will need to configure the following:

::* A '''username''' and '''password''' of your choice in order for VoIP.ms SMPP server to authenticate you.
::* Have all your communications be sent to '''smpp.voip.ms''' via port '''2775''', for regular SMPP, or port '''3550''' which is our encrypted SMPP port.
::* Open a connection using the '''bind_transceiver''' command. You can keep this connection open for as long as you need to and inquire about it using the '''enquire_link''' command.
::* Send your messages with either '''deliver_sm''' or '''submit_sm''' commands. Make sure to include your PDU's '''source_addr''' as one of your SMS enabled DIDs from your account. Your destination SMS number should be set as '''destination_addr''' while your text message will be included in '''short_message'''.

== Configuration SMPP on your VoIP.ms Portal ==

Once you have set up the above on your server, you will need to enable and specify which username and password will be used by VoIP.ms to authenticate your SMPP requests. In order to do so, head into your VoIP.ms portal, DID Numbers, Manage DIDs, edit your DID that supports SMS messages and by heading down the page, you will see the following options:

:* '''SMPP Enabled:''' If selected, SMS Messages will be allowed to be sent and received using SMPP.
:: You may specify the URL to be used to send a copy of inbound messages to your SMPP server. Leave empty if you only wish to send messages. See the example below. The '''scheme''' part is mandatory and must be set to either of '''smpp''' for unencrypted SMPP, or '''ssmpp''' for encrypted SMPP using TLS. The '''user:password@''' part is optional and will default to the same username and password defined for authentication on our side as defined below. The ''':port''' part is also optional and will default to 2775 for '''smpp''' and to '''3550''' for ssmpp.

:* '''SMPP Username:''' The username that you will use to authenticate to our SMPP server for sending messages
:* '''SMPP Password:''' The password that you will use to authenticate to our SMPP server for sending messages

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'''NOTES''' 💬
: ➜ In order to help avoid any confusion while configuring the username on your server, you can use your VoIP.ms account ID as a username (the 6 digits of your account).
: ➜ Make sure to have the option '''Message Service (SMS/MMS)''' enabled.
</div>

Enable the option '''SMPP Enabled''', fill the username and password per what was configured on your SMPP server and once done, apply the changes by pressing '''Click here to apply changes'''.

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

You may now start sending and receiving SMS messages from your SMPP server.

= Send and Receive Text Messages (Web Interface)=

==SMS Message Center for mobile==

To start using the service you will need to use the SMS Message Center, from your portal >> DID Numbers >> [https://www.voip.ms/m/sms.php SMS Message Center].

For Mobile Devices please use [https://sms.voip.ms/ sms.voip.ms] as a complete all in one solution.

From the SMS Message Center you can check your SMS History, send and receive SMS messages.

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

:* '''Search Range''': Select a Date Range of your SMS History. You can select a time period by marking the "Show Details" field and the text messages will be displayed.
:* '''Search Filter''': You can filter your Search by DID, Contact and Type (Sent, Received and Both).
:* '''Send New SMS Messages''': Another window will open by clicking here in order to create a new SMS Message.
:* '''Delete Selected Messages''': This action will delete any messages that you have selected, by clicking on the box next to them. '''This action cannot be undone.'''
:* '''Delete All Messages''': Delete ALL your SMS Messages, both sent and received. '''This action cannot be undone.'''
:* '''Checking a Message''': To check a received SMS Message just click it directly in order to display the information.

==Create a New SMS Message (Legacy)==

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

You can create a new '''SMS Message''' we will explain you all the parameters you can find on this section:

* '''DID''': If you have more than one DID Number you can select any of the numbers available to send your message.
* '''Contact''': The destination phone number or if you have a [[Phone book|Phone book entry]] start typing the name and the system will display the information if available.
* '''Message''': You can create your message on that field, please note that you have up to 160 characters.

Then click on 'Send Message' to send the SMS.

= Send and Receive Messages (via SIP MESSAGE Protocol) =
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'''NOTES''' 💬
: ➜ '''SMS via SIP''' service is only available through '''SIP Protocol and user/password authentication'''. '''IP Auth and IAX2 are not available at this time.'''
: ➜ '''Please refer to our guide [[SIP/SMS with FreePBX]] for instructions on how to configure the service with a FreePBX'''
</div>

== Sending ==

:* DID must have the '''"Short Message Service (SMS)"''' service option enabled. This option can be enabled from the Manage DIDs > Edit DID page.

:* To send an SMS from your SIP account, it is required that you set your Caller ID number to one of your SMS enabled DIDs. This will be the number you will be sending the message from. You can configure the Caller ID number from your customer portal for the specific subaccount if you are using a softphone or directly from your extension or trunk if you are using an Asterisk or PBX server.

:* It's important to note that if you configure your caller ID name in your SIP client to be a 10 digits number, this will override your caller ID number. If the caller ID name is anything different than a 10 digits number this will be discarded and the Caller ID number will be used.

:* If you are an asterisk or PBX user, please make sure to use the latest version of '''Asterisk (v12 or higher)''' and use '''chan_pjsip''' for the trunk. This is an asterisk limitation.

:* The Desktop version of Zoiper requires the user to have the '''PRO version''' activated to be able to send SMS messages as well as to enable '''SIP Presence'''. Please contact Technical Support to request SIP presence to be enabled for your account for the VoIP.ms POP you are using. While normally not required, there may be other cases not documented of softphones or apps requiring SIP Presence to be enabled. Please contact technical support if you encounter any issue with your specific software.

:* The '''free Desktop version''' of Zoiper cannot send SMS messages and is only capable of receiving SMS.

:* The '''mobile version''' of Zoiper is completely free of these limitations. Free version already works for sending and receiving without the need of SIP presence.

:* When sending an SMS message, the recipient number must be 10 digits. Avoid using the prefix '"1"' since is not supported at the moment. The system will recognize the numbers as in 10 digits.

:* The current characters limit for a '''single SMS is 160'''. If a message is bigger than 160 characters it will be split into blocks of up to 160 characters each and be sent as independent messages. The maximum total message length for SMS via SIP is 2000 characters.

== Receiving ==
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'''NOTES''' 💬: ➜ AT THE MOMENT SENDING MMS USING SIP ACCOUNT IS NOT POSSIBLE AT THIS TIME.'''
</div>

:*DID must have the '''"Message Service (SMS/MMS)"''' service option enabled. This option can be enabled from the Manage DIDs > Edit DID page.
:*Enable the option '''"SMS SIP Account"''' and select the account destination for your SMS messages to this number.
:*The account receiving SMS must be registered successfully in any of our POPs.
:*To ensure the best results receiving the SMS messages, do not register the same SIP account on more than one softphone or app at the same time, this can lead to unexpected results.
:*For SIP devices that require this information, such as a trunk or PBX, we encode the original destination number in a custom SIP header field named "X-SMS-To".
:*3CX softphone currently does not support any third party SMS service. It will only work by having a direct connection to the 3CX PBX system. '''Please contact 3CX for more details'''.

== Send SIP/SMS with FreePBX ==

<div style="margin-left:50px;
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box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(51, 51, 51, 0.08);
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'''NOTES''' 💬: ➜ AT THE MOMENT SENDING MMS USING SIP ACCOUNT IS NOT POSSIBLE AT THIS TIME.'''
</div>

Yes, it is possible to configure a FreePBX system to send SMS, please refer [[SIP/SMS with FreePBX | here]] for detailed instructions.

=SMS/MMS using the Reseller Interface=

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

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

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

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

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

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

Then scroll down and find the feature "’’’SMS/MMS’’’", and enable it.

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

= Enabling/Disabling SMS/MMS on a DID via the client's interface =
For your client to enable or disable SMS/MMS from his portal, your customer will need to head under Services, My Services.
: [[File:SMSMMSreseller5.png|thumb|none|500px]]

Then proceed by heading into ‘’’DID Lists’’’ tab and edit DID.

: [[File:SMSMMSreseller6.png|thumb|none|500px]]
You will then be able to enable SMS/MMS, receive via email and more.

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

= Accessing their SMS/MMS via the clients interface =
1) To gain access to their SMS/MMS messages, head under Services and then SMS/MMS.
: [[File:SMSMMSreseller8.png|thumb|none|500px]]

[[category:guides]]

SMS-MMS

2023-05-16T15:31:11Z

John:

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

= What is SMS or MMS =

SMS means '''S'''hort '''M'''essage '''S'''ervice ('''SMS'''). With the addition of '''M'''ultimedia '''M'''essage '''S'''ervice ('''MMS''') to the service, users are now capable of sending and receiving media files ('''photos, videos, and audio''') as well as sending text messages with more than 160 characters (which is the limit for an SMS). The MMS service also allows the user to send a message to multiple recipients at the same time.

= Blog Articles =
:* [https://wiki.voip.ms/article/Benefits_of_SMS/MMS_Campaigns_vs._Email_Marketing Benefits of SMS/MMS Campaigns vs. Email Marketing]
:* [https://wiki.voip.ms/article/How_to_Generate_Leads_with_SMS/MMS%3F How to Generate Leads with SMS/MMS?]
:* [https://wiki.voip.ms/article/Instantly_Boost_Your_Customer_Experience,_Allow_Your_Customers_To_Text_You Instantly Boost Your Customer Experience, Allow Your Customers to Text You!]

= Service Limitations =

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🏗️ ''' Please notice that the MMS service is in the Beta phase.'''
: It is important to us that you report any issues with this service by sending an email to [mailto:support@voip.ms support] so that the developers can get involved if necessary.
</div>

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⚠️ '''LIMITATIONS'''
: ➜ '''The limit of the capacity to send SMS/MMS via the portal is unlimited.'''
: ➜ The limit of sending SMS/MMS via the API is limited to '''<u>100 per day</u>'''. For any modification to the limit, please contact support or open a ticket via the customer portal.
: ➜ At the moment, receiving and reading MMS is only possible through the message center or by using the SMS/MMS Forward option. '''MMS through SIP protocol is not available.'''

: ➜ The SMS/MMS Service is only available for <u>'''US and Canadian local DID Numbers'''</u> marked with the distinctive '''SMS/MMS Icon'''.
: ➜ The SMS/MMS Service can only be used to send SMS/MMS Messages to <u>'''Canadian and US'''</u> 10 digit numbers at this time.
: ➜ SIP SMS messages will only be delivered to and accepted from SIP trunks using username/password authentication.
: ➜ SIP SMS messages use the SIP MESSAGE method. See [https://tools.ietf.org/html/rfc3428 RFC 3428] for details.
: ➜ At this time we '''<u>cannot guarantee</u> that Short Code SMS Messages''', which are usually 6 digits or less ''(e.g.: Skype, Bank Codes, TV Commercials etc...)'' will work.
: ➜ We cannot guarantee that '''accents or special characters''' including non-Latin letters will be properly delivered.

: ➜ Public URL shorteners such as '''''bit.ly, tinyurl.com''''' etc, <font style="color:red">'''are widely restricted'''</font> by mobile providers as a response from the industry to combat SPAM campaigns and protect the consumer. Including this type of links into your messages '''<u>may result in a blocked or undelivered message</u>'''.
: ➜ For '''Ported In Numbers''' compatible with the feature: The SMS/MMS Functionality will be available up to '''48 hours after the porting process is marked as Completed'''.

: ➜ VoIP.ms reserves the right, at its sole discretion, to limit or disable the SMS/MMS service from any account that may present marketing patterns/automation patterns/bulk sending or receiving.

</div>

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'''💬 NOTE ''' If you have further questions don't hesitate to contact the Support Staff on the Live Chat or Ticket System.
:➜ '''SMS Messages have a cost of $0.0075 per incoming and outgoing message.'''
:➜ '''MMS Messages have a cost of $0.02 per incoming and outgoing message.'''

: ➜ '''Additionally, please note of the following restrictions for sending MMS:'''
:: ➜ Permitted attachment types are JPG, GIF, JPEG, PNG, MP3, WAV, MIDI, MP4 and 3GP.
:: ➜ The maximum size for each attachment is 1300 KB. ''(1.3 MB)''
:: ➜ You can attach up to 3 files.
:: ➜ Maximum characters in a text message is 2048.

</div>

=Using the SMS/MMS=
==Identifying a DID Number SMS/MMS capable==

The first thing you need to know is that this feature is only available for local US and Canadian numbers at this time. You will note that some numbers have a little [[File:Sms-mms-icon.png|15px]] icon, this indicates that the number supports this feature.

You can start ordering a DID Number from your portal >> DID Numbers >> Order DID >> Local Numbers >> (US or Canada) Numbers and you will be able to purchase your desired number, just make sure it supports the MMS feature.

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

==Configuring the SMS/MMS service==

Once purchased the SMS/MMS Service needs to be activated on the DID settings, from your portal >> DID Numbers >> Manage DID >> Edit Selection >> SMS/MMS configuration section.

In that section, you can activate and configure some forward options for the SMS/MMS service.

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

:* '''Message Service (SMS/MMS)''': Check the box to enable the service.
:* '''VoIP.ms SMS/MMS Portal''': This is enabled by default and it can't be modified, that means that you will be able to create and send messages from the SMS Message Center.
:* '''SMS/MMS Email Address''': By activating this option all the SMS Messages will be sent to the email address you configure in this field. You can receive them to more than 1 email address by separating each emails with a comma. The advantage of this option is that you will receive your SMS Messages directly to your email and you will be able to Reply to these messages from your email too. You just need to click on '''reply''' to the email. Please note that e-mail responses are as well limited to 160 characters and if they exceed this limit, they'll be split in two (or more messages if applies) when sent.

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'''NOTE''' 💬: ➜ To reply the SMS Message via your email it is important to click on '''Reply''' and '''DON'T change or modify''' the destination mail "TO" (sms@voip.ms) and the subject E.G ''[#USXXXXXX] Message sent to 5555555555''.
</div>

:*'''SMS/MMS Forward''': You can have your SMS/MMS Messages forwarded to your cellphone or any other number that supports SMS/MMS, with this option you will receive the SMS/MMS Messages from our system to the configured phone number.
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'''NOTE''' 💬: ➜ The CallerID sent to the Call Forwarded Cell Phone will be the DID Number that received the SMS Message. Is not possible to forward the original Caller ID at this time. When using SMS/MMS Forward there is a charge for the incoming message received by the DID and the outgoing message sent to the number set for the forward.
</div>

:*'''SMS/MMS SIP Account''': You can have your SMS Messages forwarded to your selected SIP account. They will be sent through our SMS/SIP gateway as a SIP MESSAGE.
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'''NOTE''' 💬: ➜ The destination ("To" header field) of the SIP MESSAGE will be your account name, instead of the destination number of the original SMS.
</div>

:'''SMS/MMS URL Callback''': By enabling this option you will be able to send the messages to another destination e.g. your own server. If selected SMS Messages received by your DID will Send a GET request to the URL Callback provided. Available variables for your URL:
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:: '''Example''': <nowiki>https://YourDomainName.tld/sms.php?to={TO}&from={FROM}&message={MESSAGE}&id={ID}&date={TIMESTAMP}</nowiki>

: '''{ID}''' = The ID of the SMS message.
: '''{TIMESTAMP}''' = The date and time the message was received.
: '''{FROM}''' = The phone number that sent you the message.
: '''{TO}''' = The DID Number that received the message.
: '''{MESSAGE}''' = The content of the message.

</div>

:*'''URL Callback Retry''': When selected, we will be expecting an "ok" output (without quotes) from your URL callback page as an indicator that you have received the message correctly. If we don't received the "ok" (without quotes) from your callback page, we will keep sending you the same message every 30 minutes.

Once you configure the desired option, click on Apply Changes.

==Sending a SMS/MMS (Web Interface)==

'''Step 1'''

: Click the "send New Message" button to get a list of contacts displayed or directly type in the number into the contact text field. If this number is registered in the phonebook, it will display its name instead.

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

'''Step 2'''

:Click to display the list of available DID numbers. Choose the number to send the message from.

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

'''Step 3'''

:Write your message on the text field and click the red arrow to send it. Text messages up to 160 characters will be considered SMS. Media messages or text messages over 160 characters will be considered MMS and you will see the icon turning from SMS to MMS indicating of this.
:: [[File:MMS-Webportal4.png|border|400px]]

'''Step 4'''

: In order to send a media file over an MMS, click the clip button to browse and choose a media file.

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

: Click the browse button to search for the media file in your computer or drag and drop the file directly into the box.

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

: The attached files will show below the text box.

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

: Once the message is sent, a new conversation will be opened on the left. Selecting this conversation will show the messages history on the box.

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

==Receiving an SMS/MMS (Web Interface)==

: While SMS messages can be received directly in an App through the SIP Protocol and correct configuration, at the moment, '''receiving and reading MMS is only possible through the message center.'''

: You can access the communication center from your customer portal at DID numbers > SMS/MMS Message Center. Any message received will appear here on the left side of the panel.

: Use the DID Filter to choose and display messages for a specific DID number or for all DID numbers.

: You can also filter the messages by date and time zone by clicking on the down arrow, which will show the search parameters

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

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

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

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

== SMS/MMS History ==

You can now also gain access to a general history of SMS/MMS that was sent and received for all of your DIDs, or for a specific one by going to Customer Portal >> DID Numbers >> SMS/MMS Message Center >> History.

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

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

= SMS using a SMPP Server =

An SMPP server, short of '''Short Message Peer to Peer''' is a method designed to send and receive high volumes of SMS messages through a dedicated server.

In other words, this allows you to better control the quantity of messages that will be sent to your target audience while avoiding to pass via a softphone application or your VoIP.ms portal to send/receive these SMS, thus eliminating multiple repetitive tasks and saving you time.

For example, if you have a promotion for your business that offers a discount for a certain lapse of time and that your customers have provided their agreement to receive these types of promotions via SMS, you could send your promotion to your customers in a faster manner than if you would do it from your customer portal, thus increasing efficiency.
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'''NOTES''' 💬: ➜ Note that sending and receiving SMS messages will be charged our usual rate at $0.0075 per messages'''
</div>

==Requirements==

: '''This will require you to have your own SMPP server.'''
: Once configured properly, you will then be able to send and receive SMS messages to your own server.

== Configuration on your SMPP Server ==

: In order for your server to communicate and authenticate accordingly with VoIP.ms SMPP server, you will need to configure the following:

::* A '''username''' and '''password''' of your choice in order for VoIP.ms SMPP server to authenticate you.
::* Have all your communications be sent to '''smpp.voip.ms''' via port '''2775''', for regular SMPP, or port '''3550''' which is our encrypted SMPP port.
::* Open a connection using the '''bind_transceiver''' command. You can keep this connection open for as long as you need to and inquire about it using the '''enquire_link''' command.
::* Send your messages with either '''deliver_sm''' or '''submit_sm''' commands. Make sure to include your PDU's '''source_addr''' as one of your SMS enabled DIDs from your account. Your destination SMS number should be set as '''destination_addr''' while your text message will be included in '''short_message'''.

== Configuration SMPP on your VoIP.ms Portal ==

Once you have set up the above on your server, you will need to enable and specify which username and password will be used by VoIP.ms to authenticate your SMPP requests. In order to do so, head into your VoIP.ms portal, DID Numbers, Manage DIDs, edit your DID that supports SMS messages and by heading down the page, you will see the following options:

:* '''SMPP Enabled:''' If selected, SMS Messages will be allowed to be sent and received using SMPP.
:: You may specify the URL to be used to send a copy of inbound messages to your SMPP server. Leave empty if you only wish to send messages. See the example below. The '''scheme''' part is mandatory and must be set to either of '''smpp''' for unencrypted SMPP, or '''ssmpp''' for encrypted SMPP using TLS. The '''user:password@''' part is optional and will default to the same username and password defined for authentication on our side as defined below. The ''':port''' part is also optional and will default to 2775 for '''smpp''' and to '''3550''' for ssmpp.

:* '''SMPP Username:''' The username that you will use to authenticate to our SMPP server for sending messages
:* '''SMPP Password:''' The password that you will use to authenticate to our SMPP server for sending messages

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'''NOTES''' 💬
: ➜ In order to help avoid any confusion while configuring the username on your server, you can use your VoIP.ms account ID as a username (the 6 digits of your account).
: ➜ Make sure to have the option '''Message Service (SMS/MMS)''' enabled.
</div>

Enable the option '''SMPP Enabled''', fill the username and password per what was configured on your SMPP server and once done, apply the changes by pressing '''Click here to apply changes'''.

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

You may now start sending and receiving SMS messages from your SMPP server.

= Send and Receive Text Messages (Web Interface)=

==SMS Message Center for mobile==

To start using the service you will need to use the SMS Message Center, from your portal >> DID Numbers >> [https://www.voip.ms/m/sms.php SMS Message Center].

For Mobile Devices please use [https://sms.voip.ms/ sms.voip.ms] as a complete all in one solution.

From the SMS Message Center you can check your SMS History, send and receive SMS messages.

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

:* '''Search Range''': Select a Date Range of your SMS History. You can select a time period by marking the "Show Details" field and the text messages will be displayed.
:* '''Search Filter''': You can filter your Search by DID, Contact and Type (Sent, Received and Both).
:* '''Send New SMS Messages''': Another window will open by clicking here in order to create a new SMS Message.
:* '''Delete Selected Messages''': This action will delete any messages that you have selected, by clicking on the box next to them. '''This action cannot be undone.'''
:* '''Delete All Messages''': Delete ALL your SMS Messages, both sent and received. '''This action cannot be undone.'''
:* '''Checking a Message''': To check a received SMS Message just click it directly in order to display the information.

==Create a New SMS Message (Legacy)==

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

You can create a new '''SMS Message''' we will explain you all the parameters you can find on this section:

* '''DID''': If you have more than one DID Number you can select any of the numbers available to send your message.
* '''Contact''': The destination phone number or if you have a [[Phone book|Phone book entry]] start typing the name and the system will display the information if available.
* '''Message''': You can create your message on that field, please note that you have up to 160 characters.

Then click on 'Send Message' to send the SMS.

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

'''NOTES''' 💬
: ➜ '''SMS via SIP''' service is only available through '''SIP Protocol and user/password authentication'''. '''IP Auth and IAX2 are not available at this time.'''
: ➜ '''Please refer to our guide [[SIP/SMS with FreePBX]] for instructions on how to configure the service with a FreePBX'''
</div>

== Sending ==

:* DID must have the '''"Short Message Service (SMS)"''' service option enabled. This option can be enabled from the Manage DIDs > Edit DID page.

:* To send an SMS from your SIP account, it is required that you set your Caller ID number to one of your SMS enabled DIDs. This will be the number you will be sending the message from. You can configure the Caller ID number from your customer portal for the specific subaccount if you are using a softphone or directly from your extension or trunk if you are using an Asterisk or PBX server.

:* It's important to note that if you configure your caller ID name in your SIP client to be a 10 digits number, this will override your caller ID number. If the caller ID name is anything different than a 10 digits number this will be discarded and the Caller ID number will be used.

:* If you are an asterisk or PBX user, please make sure to use the latest version of '''Asterisk (v12 or higher)''' and use '''chan_pjsip''' for the trunk. This is an asterisk limitation.

:* The Desktop version of Zoiper requires the user to have the '''PRO version''' activated to be able to send SMS messages as well as to enable '''SIP Presence'''. Please contact Technical Support to request SIP presence to be enabled for your account for the VoIP.ms POP you are using. While normally not required, there may be other cases not documented of softphones or apps requiring SIP Presence to be enabled. Please contact technical support if you encounter any issue with your specific software.

:* The '''free Desktop version''' of Zoiper cannot send SMS messages and is only capable of receiving SMS.

:* The '''mobile version''' of Zoiper is completely free of these limitations. Free version already works for sending and receiving without the need of SIP presence.

:* When sending an SMS message, the recipient number must be 10 digits. Avoid using the prefix '"1"' since is not supported at the moment. The system will recognize the numbers as in 10 digits.

:* The current characters limit for a '''single SMS is 160'''. If a message is bigger than 160 characters it will be split into blocks of up to 160 characters each and be sent as independent messages. The maximum total message length for SMS via SIP is 2000 characters.

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

'''NOTES''' 💬: ➜ AT THE MOMENT SENDING MMS USING SIP ACCOUNT IS NOT POSSIBLE AT THIS TIME.'''
</div>

:*DID must have the '''"Message Service (SMS/MMS)"''' service option enabled. This option can be enabled from the Manage DIDs > Edit DID page.
:*Enable the option '''"SMS SIP Account"''' and select the account destination for your SMS messages to this number.
:*The account receiving SMS must be registered successfully in any of our POPs.
:*To ensure the best results receiving the SMS messages, do not register the same SIP account on more than one softphone or app at the same time, this can lead to unexpected results.
:*For SIP devices that require this information, such as a trunk or PBX, we encode the original destination number in a custom SIP header field named "X-SMS-To".
:*3CX softphone currently does not support any third party SMS service. It will only work by having a direct connection to the 3CX PBX system. '''Please contact 3CX for more details'''.

== Send SIP/SMS with FreePBX ==

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

'''NOTES''' 💬: ➜ AT THE MOMENT SENDING MMS USING SIP ACCOUNT IS NOT POSSIBLE AT THIS TIME.'''
</div>

Yes, it is possible to configure a FreePBX system to send SMS, please refer [[SIP/SMS with FreePBX | here]] for detailed instructions.

=SMS/MMS using the Reseller Interface=

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

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

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

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

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

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

Then scroll down and find the feature "’’’SMS/MMS’’’", and enable it.

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

= Enabling/Disabling SMS/MMS on a DID via the client's interface =
For your client to enable or disable SMS/MMS from his portal, your customer will need to head under Services, My Services.
: [[File:SMSMMSreseller5.png|thumb|none|500px]]

Then proceed by heading into ‘’’DID Lists’’’ tab and edit DID.

: [[File:SMSMMSreseller6.png|thumb|none|500px]]
You will then be able to enable SMS/MMS, receive via email and more.

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

= Accessing their SMS/MMS via the clients interface =
1) To gain access to their SMS/MMS messages, head under Services and then SMS/MMS.
: [[File:SMSMMSreseller8.png|thumb|none|500px]]

[[category:guides]]

Cisco SPA112

2023-02-24T15:46:07Z

John:

[[File:SPA112.jpg|300px|thumb|left|Cisco SPA112]]
<div style="width:100%;overflow:hidden;clear:both"></div>

'''Please note these models are EOL (End of life) & aren't supported by CISCO anymore''' https://www.cisco.com/c/en/us/products/collateral/unified-communications/small-business-voice-gateways-ata/eos-eol-notice-c51-743206.html

: '''There have been some reports of issues with this device from customers of both VoIP.ms and other providers.'''
: '''Make sure to install the latest firmware from [https://software.cisco.com/download/release.html?mdfid=283998771&softwareid=282463187&release=1.4.1%20SR3&relind=AVAILABLE&rellifecycle=&reltype=latest Cisco Software].'''
: '''Version 1.1 or later should be used for proper Caller ID support. '''
: '''Some People have reported issues using Firefox to Configure this device; please try Chrome or IE. '''

==Documentation==
[https://www.cisco.com/c/en/us/support/unified-communications/spa112-2-port-phone-adapter/model.html#~tab-documents Official Configuration guide / User guide / Data Sheets]

== Configuration Details ==

=== Getting the IP address of your device ===

There are two ways to retrieve the IP address of your Cisco SPA112: via analog phone menu, and via your internet router.

{| class="wikitable"
! style="text-align: center;" | Analog phone interface
! style="text-align: center;" | Internet Router
|-
|
# Attach the Cisco SPA112 to your network and attach an analog phone to one of the phone ports, then do the following:
## Dial **** from the phone, even though there is no dial tone.
## When you hear "System Configuration Menu," dial 1 1 0 # slowly. The current IP address will be read back. (e.g. 192.168.X.X)

'''If you hear 0.0.0.0, check your network connection and DHCP server. If necessary, a static IP address'''
'''can be assigned by using option 111# at the IVR, then entering the IP address with your phone's keypad'''
'''(for example, 10*1*27*2 for 10.1.27.2). The network mask can be set with option 121# and the default'''
'''gateway can be sent with option 131#'''
Learn more about the IVR menu options from the https://supportforums.cisco.com/docs/DOC-9900 document.

Be sure to allow at least a minute or two for the box to initialize; even a correctly configured and installed SPA112/122 will give no power or dialtone to the phone until initialization is complete.

Note that the SPA122 is basically a SPA112 with a second network port, intended for installation between a local network hub (LAN) and an upstream Internet (WAN) connection. The SPA122 may be configured as either a "NAT" or "bridge". Depending on configuration, this leaves the SPA122 with two addresses; a local area network address (such as 192.168.15.1) and an outside Internet address. Dialing ****110# will give one address, ****210# will give the other.
|
# Attach the Cisco SPA112 to your network
# Access your router's remote administration interface via your web browser (typical addresses may be <code>192.168.0.1</code> or <code>192.168.1.1</code>). Refer to your router instructions for more information.
# Enter your username/password if asked. If you have not set one, then it is likely the unchanged default password.
# In the router's menu, there should be a page showing a list of connected clients, with their internal IP address. Find the entry corresponding to the Cisco SPA112. It should identify itself in the list as "SPA112"
# Navigate to this IP address via your web browser
|}

=== Accessing to the device's settings page ===

Open your web browser and go to the IP address you obtained in step 1 (for example, http://192.168.2.1).
The default username is admin, and the default password is also admin.

For the SPA122, if one address does not return the web interface (or has some functions greyed/disabled), try the other.

=== Configuring the Quick Setup screen ===

Go to Quick Setup and configure Line 1 as follows:

'''Proxy:''' atlanta.voip.ms (You can choose any of our multiple VoIP.ms [http://wiki.voip.ms/article/Choosing_Server servers])

'''Display Name:''' Your Outbound callerID Name'''(See the requirements below)'''

'''User ID:''' 100000 (Replace with your 6 digit Main SIP account UserID or Sub Account user name, i.e. 123456 or 123456_sub)

'''Password:''' Your VoIP.MS SIP Password

'''Dial Plan:''' (911S0|310xxxx|<:1555>[2-9]xxxxxx|1[2-9]xx[2-9]xxxxxxS0|[2-9]xx[2-9]xxxxxxS0|*xx|***xxx|*xx.|[3468]11|822|0|00|4xxx|**275*x.|xxxxxxxxxxxx.)

('''''Note''''': Replace 555 in the dial plan with your area code, See [[Dial Plan for Linksys ATAs]] for details.)

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

Click Submit to save settings.

[[File:quick_setup_test.png|800px|thumb|left|Quick Setup Page - Click to enlarge]]
<br clear="all" />

=== Configuring the Voice Line ===
==== Nat Settings ====

Click on Voice, then Line 1

Set '''NAT Mapping Enable''' to Yes, then set '''NAT Keep Alive Enable''' to Yes. If your environment does not use NAT, you can leave these settings disabled. These features can usually be disabled on the SPA122 if it is connected directly to your modem since its traffic will not be subject to NAT in this configuration.

If using the second phone line on an SPA122 device, change the SIP Port for one of the lines to e.g. 5080.

[[File:VL_1_nat_settings.png|800px|thumb|left|NAT Settings - Click to enlarge]]
<br clear="all" />

==== Proxy and Registration ====

Under '''Proxy and Registration''', set the server you will use as registration server and the proper values for the Register Expires and Proxy Fallback Intvl:

'''Proxy''': atlanta.voip.ms (one of VoIP.ms multiple [http://wiki.voip.ms/article/Choosing_Server servers], you can choose the one closest to your location)
'''Register Expires''' to 300
'''Proxy Fallback Intvl''' to 300

Also confirm the following settings:<br />
'''Register:''' YES <br />
'''Use DNS SRV:''' NO<br />
'''DNS SRV Auto Prefix:''' NO

[[File:VL_2_proxyAndRegistration.png|800px|thumb|left|Proxy and Registration - Click to enlarge]]
<br clear="all" />

Click Submit to save these changes

==== Subscriber Information ====

In this section please confirm that you have the proper account information:

'''Display Name''': Your name (that will be shown as callerID name)
'''User ID''': 100000 (Replace with your 6 digit Main SIP account UserID or Sub Account user name, i.e. 123456 or 123456_sub)
'''Password''': Your VoIP.ms SIP Password

[[File:VL_3_subscriberInformation.png|800px|thumb|left|Subscriber Information - Click to enlarge]]
<br clear="all" />

==== Audio Configuration ====

You can verify or change the audio codec that will be used with the calls. Please verify that you have the same codec selected in your SIP account's settings.

Preferred codec: g711u (or G729)

[[File:VL_4_audioConfig.png|800px|thumb|left|Audio configuration - Click to enlarge]]
<br clear="all" />

==== Dial Plan ====

We recommend to use this dial plan.

(911S0|310xxxx|<:1555>[2-9]xxxxxx|1[2-9]xx[2-9]xxxxxxS0|[2-9]xx[2-9]xxxxxxS0|*xx|***xxx|*xx.|[3468]11|822|0|00|4xxx|**275*x.|xxxxxxxxxxxx.)

('''''Note''''': Replace 555 in the dial plan with your area code, See [[Dial Plan for Linksys ATAs]] for details.)

[[File:VL_5_dialPlan.png|800px|thumb|left|Dial Plan - Click to enlarge]]
<br clear="all" />

You can create your own dial plan if you need it. See [[Dial Plan for Linksys ATAs]] and also [https://www.cisco.com/c/en/us/support/docs/collaboration-endpoints/spa901-1-line-ip-phone/108747-pqa-108747.html here] you'll find a great explanation about dial plans for Cisco devices

=== Optional settings ===

==== Outbound audio "breaking up". ====

Cisco's defaults (SIP T1 = 0.5 sec, RTP packet size 0.030 on most Sipura adapters) respectively may cause unnecessary retransmission of commands over connections with high latency and create issues with outbound audio "breaking up".

Click '''Voice''', then go to '''SIP'''.

Set SIP Timer Values (sec)

SIP T1: 1

Set RTP Parameters

RTP Packet Size: 0.02
RTP Port Min: 10000
RTP Port Max: 20000

Click Submit to save the changes

[[File:VS_sipAndRTP.png|800px|thumb|left|SIP Values - Click to enlarge]]
<br clear="all" />

==== Caller ID display showing incorrect time ====

Sometimes the hour shown in your caller ID is incorrect. Following this suggestion usually solves the issue:

Enter your device's settings and click '''Network Setup''', then go to '''Basic Setup''', then click '''Time Settings'''

Set your time zone and (optional) NTP settings. A good time server choice is 0.pool.ntp.org. Setting the proper time zone will ensure that the time which appears on your Caller ID display is correct.

===Configuring a Voice line using TLS===

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

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

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

====Verifying the device's Firmware version====

First, check your firmware's version. This is, from your Device's Configuration Utility at ''Status >> System Information''

We '''strongly recommend''' to use the latest firmware version available, up today is the ''1.4.1 (SR3) Apr 3 2019''. If you do not have this version, you may consider its upgrade.

[[File:SPA_FW_Ver.png|800px|thumb|left|Click to enlarge]]
<br clear="all" />

==== Enabling TLS for the line ====

Go to the User's line you will use (If you use the Line 1 go to User 1) and navigate to the '''''Supplementary Service Settings''''', there set:
:''Secure Call Setting'': '''''yes'''''

[[File:SPA_User_SCS.png|800px|thumb|left|Click to enlarge]]
<br clear="all" />

==== Configuring the transport and port ====

Go to the line you will be using with TLS and navigate to the section ''SIP Settings'', then set:
:''SIP Transport'' : '''''TLS'''''
:''SIP Port'' : '''''5061'''''

[[File:SPA_Voice_Line.png|800px|thumb|left|Click to enlarge]]
<br clear="all" />

==== CA Certificate ====

As per CISCO's requirements, a CA certificate is needed to use Secure calls with the SPA112's device. To achieve this you will need to import the CA Cert.

Go to ''Voice >> Provisioning'' and once there navigate to '''''CA Settings''''', at ''Custom CA URL'' enter the following:

https://spa1xx.voip.ms/cca.pem

[[File:SPA_Prov_CA.png|800px|thumb|left|Click to enlarge]]
<br clear="all" />

Click ''Submit'', the device will reboot and after that will register and you will be ready to use it with TLS

==== Secure Call Indication Tone ====

Once the secure call feature is enabled, during all the duration of your calls you will hear a couple of tones (beeps), this is normal and beyond ''VoIP.ms'' control, however you can disable this notification going to: ''Voice >> Regional >> Secure Call Indication Tone''

'''''Note:''''' Please notice that this setting is not an on/off, you will need to remove all the line
In case you need to set it back, the default value is
397@-19,507@-19;15(0/2/0,.2/.1/1,.1/2.1/2)

[[File:SPA_Regional_Tone.png|800px|thumb|left|Click to enlarge]]
<br clear="all" />

Click Submit to save the changes

==Known Issues==

=== '''Phone will not ring on handset''' ===

Sometimes the Phone you are using is designed for a certain voltage and ring waveform. If someone tries to call you and the phone appears to be ringing for the caller but your phone never rings, please follow these steps to hopefully resolve this issue for you.

Step 1: First access the SPA web interface.

Step 2: Click on the '''Admin Login''' and then click on '''(switch to advanced view)'''

Step 3: Click on your Regional tab on the top menu.

Step 4: Go halfway down the page until you see the heading '''Ring and Call Waiting Tone Spec'''

[[File:Pap2Ring.jpg|800px|thumb|left| Ring and Call Waiting - Click to enlarge]]
<br clear="all" />

Step 5: Change the Ring Waveform setting to Sinusoid or Trapezoid, the opposite of what you have set. You can also change the Ring Voltage in increments of 5 to 90 or 95.

Step 6: Save settings and test an incoming call.

=== Receiving Unwanted Calls in the middle of the Night (i.e. CallerID 100) that do not appear in your CDR: ===

These calls are not going through our Network but rather through the internet directly to your ATA Device.

Please look under the Voice>> Line 1 page in your SPA device for the following setting: Restrict Source IP and make sure it's enabled.

This way the ATA device will block any traffic not coming from our servers.

[[File:VL_1_restrictSourceIP.png|800px|thumb|left|Restrict IP - Click to enlarge]]
<br clear="all" />

=== Firmware Upgrade ===

SPA112 and SPA122 adapters were distributed with outdated (1.0.x) firmware at least as late as 2012; affected boxes will not show Caller ID on any inbound call, even though the caller names and numbers are visible in the call detail record on VoIP.ms (or other provider's) web interface.

Updated firmware is available from the Cisco site [https://software.cisco.com/download/release.html?mdfid=283998771&softwareid=282463187&release=1.4.1%20SR3&relind=AVAILABLE&rellifecycle=&reltype=latest Cisco Firmware] as a .ZIP archive which contains two files (a .BIN with the actual firmware and a .PDF with documentation). Download and unzip this file. Go to the 'administration' tab on the web interface (on the SPA122, this needs to be done from the LAN side with SPA122's built-in networking set to NAT mode). On the left sidebar, click 'update firmware' (as most of the administration menu does not appear for Firefox users, downgrade to MS IE or another browser temporarily). Click the 'upload' button and indicate the location of the unzipped .BIN file. A box will appear with a progress indicator and a warning not to interrupt the upgrade. When the upgrade is completed, the SPA112/122 will reset and will likely take a minute or more to reinitialize, reconnect to the network and restore dial tone. SPA122 users who have installed the device in-line between the local PCs and the Internet will be disconnected from the Internet until reinitialization is complete.

Once the new firmware is deployed, call display will operate normally and the configuration web page will display in Firefox without missing options in the administration menu.

A manual for Cisco's SPA100 series adapters is online at http://www.cisco.com/en/US/docs/voice_ip_comm/csbpvga/spa100-200/admin_guide_SPA100/spa100_ag.html

* You can check the most commonly used [[Cisco/Linksys Star Codes]]

=== SPA Star Codes ===

{| class="wikitable" style="margin-left: center; margin-right: auto; border: none;"
! colspan="3" | SPA Star Codes
|-
! scope="col" | Star Code
! scope="col" | Name
! scope="col" | Result
|-
| *69 || Call Return Code || This code calls the last caller.
|-
| *07 || Call Redial Code || Redials the last number called. (Not in pap2t)
|-
| *98 || Blind Transfer Code || Begins a blind transfer of the current call to the extension specified after the activation code.
|-
| *66 || Call Back Act Code || Starts a callback when the last outbound call is not busy.
|-
| *86 || Call Back Dea t Code || Cancels a callback.
|-
| *05 || Call Back Busy Act Code || Starts a callback when the last outbound call is busy. (Not in pap2t)
|-
| *72 || Cfwd All Act Code || Forwards all calls to the extension specified after the activation code.
|-
| *73 || Cfwd All Deact Code || Cancels call forwarding of all calls.
|-
| *90 || Cfwd Busy Act Code || Forwards busy calls to the extension specified after the activation code.
|-
| *91 || Cfwd Busy Deact Code || Cancels call forwarding of busy calls.
|-
| *92 || Cfwd No Ans Act Code || Forwards no-answer calls to the extension specified after the activation code.
|-
| *93 || Cfwd No Ans Deact Code || Cancels call forwarding of no-answer calls.
|-
| *63 || Cfwd Last Act Code || Forwards the last inbound or outbound calls to the extension specified after the activation code.
|-
| *83 || Cfwd Last Deact Code || Cancels call forwarding of the last inbound or outbound calls.
|-
| *60 || Block Last Act Code || Blocks the last inbound call.
|-
| *80 || Block Last Deact Code || Cancels blocking of the last inbound call.
|-
| *64 || Accept Last Act Code || Accepts the last outbound call. It lets the call ring through when do not disturb or call forwarding of all calls are enabled.
|-
| *84 || Accept Last Deact Code || Cancels the code to accept the last outbound call.
|-
| *56 || CW Act Code || Enables call waiting on all calls.
|-
| *57 || CW Deact Code || Disables call waiting on all calls.
|-
| *71 || CW Per Call Act Code || Enables call waiting for the next call.
|-
| *70 || CW Per Call Deact Code || Disables call waiting for the next call.
|-
| *67 || Block CID Act Code || Blocks caller ID on all outbound calls.
|-
| *68 || Block CID Deact Code || Removes caller ID blocking on all outbound calls.
|-
| *81 || Block CID Per Call Act Code || Blocks caller ID on the next outbound call.
|-
| *82 || Block CID Per Call Deact Code || Removes caller ID blocking on the next inbound call.
|-
| *77 || Block ANC Act Code || Blocks all anonymous calls.
|-
| *87 || Block ANC Deact Code || Removes blocking of all anonymous calls.
|-
| *78 || DND Act Code || Enables the do not disturb feature.
|-
| *79 || DND Deact Code || Disables the do not disturb feature.
|-
| *65 || CID Act Code || Enables caller ID generation.
|-
| *85 || CID Deact Code || Disables caller ID generation.
|-
| *25 || CWCID Act Code || Enables call waiting, caller ID generation.
|-
| *45 || CWCID Deact Code || Disables call waiting, caller ID generation.
|-
| *26 || Dist Ring Act Code || Enables the distinctive ringing feature.
|-
| *46 || Dist Ring Deact Code || Enables the distinctive ringing feature. The default is *46 ||.
|-
| *74 || Speed Dial Act Code || Assigns a speed dial number.
|-
| *16 || Secure All Call Act Code || Makes all outbound calls secure.
|-
| *17 || Secure No Call Act Code || Makes all outbound calls not secure.
|-
| *18 || Secure One Call Act Code || Makes the next outbound call secure. (It is redundant if all outbound calls are secure by default.)
|-
| *19 || Secure One Call Deact Code || Secure One Call Deact Code Makes the next outbound call not secure. (It is redundant if all outbound calls are not secure by default.)
|}

==Guide Links==
In the event where you need the guide directly from Cisco, you may find the admin manual guide below:

Admin Manual : [https://www.cisco.com/c/dam/en/us/td/docs/voice_ip_comm/csbpvga/spa100-200/admin_guide_SPA100/spa100_ag_1_3_5.pdf Download PDF]

[[category:Analog Telephone Adapters]]

SMS-MMS

2023-02-20T19:23:30Z

John:

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

= What is SMS or MMS =

SMS means '''S'''hort '''M'''essage '''S'''ervice ('''SMS'''). With the addition of '''M'''ultimedia '''M'''essage '''S'''ervice ('''MMS''') to the service, users are now capable of sending and receiving media files ('''photos, videos, and audio''') as well as sending text messages with more than 160 characters (which is the limit for an SMS). The MMS service also allows the user to send a message to multiple recipients at the same time.

= Blog Articles =
:* [https://wiki.voip.ms/article/Benefits_of_SMS/MMS_Campaigns_vs._Email_Marketing Benefits of SMS/MMS Campaigns vs. Email Marketing]
:* [https://wiki.voip.ms/article/How_to_Generate_Leads_with_SMS/MMS%3F How to Generate Leads with SMS/MMS?]
:* [https://wiki.voip.ms/article/Instantly_Boost_Your_Customer_Experience,_Allow_Your_Customers_To_Text_You Instantly Boost Your Customer Experience, Allow Your Customers to Text You!]

= Service Limitations =

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

🏗️ ''' Please notice that the MMS service is in the Beta phase.'''
: It is important to us that you report any issues with this service by sending an email to [mailto:support@voip.ms support] so that the developers can get involved if necessary.
</div>

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

⚠️ '''LIMITATIONS'''
: ➜ '''The limit of the capacity to send SMS/MMS via the portal is unlimited.'''
: ➜ The limit of sending SMS/MMS via the API is limited to '''<u>100 per day</u>'''. For any modification to the limit, please contact support or open a ticket via the customer portal.
: ➜ At the moment, receiving and reading MMS is only possible through the message center or by using the SMS/MMS Forward option. '''MMS through SIP protocol is not available.'''

: ➜ The SMS/MMS Service is only available for <u>'''US and Canadian local DID Numbers'''</u> marked with the distinctive '''SMS/MMS Icon'''.
: ➜ The SMS/MMS Service can only be used to send SMS/MMS Messages to <u>'''Canadian and US'''</u> 10 digit numbers at this time.
: ➜ SIP SMS messages will only be delivered to and accepted from SIP trunks using username/password authentication.
: ➜ SIP SMS messages use the SIP MESSAGE method. See [https://tools.ietf.org/html/rfc3428 RFC 3428] for details.
: ➜ At this time we '''<u>cannot guarantee</u> that Short Code SMS Messages''', which are usually 6 digits or less ''(e.g.: Skype, Bank Codes, TV Commercials etc...)'' will work.
: ➜ We cannot guarantee that '''accents or special characters''' including non-Latin letters will be properly delivered.

: ➜ Public URL shorteners such as '''''bit.ly, tinyurl.com''''' etc, <font style="color:red">'''are widely restricted'''</font> by mobile providers as a response from the industry to combat SPAM campaigns and protect the consumer. Including this type of links into your messages '''<u>may result in a blocked or undelivered message</u>'''.
: ➜ For '''Ported In Numbers''' compatible with the feature: The SMS/MMS Functionality will be available up to '''48 hours after the porting process is marked as Completed'''.

: ➜ VoIP.ms reserves the right, at its sole discretion, to limit or disable the SMS/MMS service from any account that may present marketing patterns/automation patterns/bulk sending or receiving.

</div>

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

'''💬 NOTE ''' If you have further questions don't hesitate to contact the Support Staff on the Live Chat or Ticket System.
:➜ '''MMS Messages have a cost of $0.02 per incoming and outgoing message.'''

: ➜ '''Additionally, please note of the following restrictions for sending MMS:'''
:: ➜ Permitted attachment types are JPG, GIF, JPEG, PNG, MP3, WAV, MIDI, MP4 and 3GP.
:: ➜ The maximum size for each attachment is 1300 KB. ''(1.3 MB)''
:: ➜ You can attach up to 3 files.
:: ➜ Maximum characters in a text message is 2048.

</div>

=Using the SMS/MMS=
==Identifying a DID Number SMS/MMS capable==

The first thing you need to know is that this feature is only available for local US and Canadian numbers at this time. You will note that some numbers have a little [[File:Sms-mms-icon.png|15px]] icon, this indicates that the number supports this feature.

You can start ordering a DID Number from your portal >> DID Numbers >> Order DID >> Local Numbers >> (US or Canada) Numbers and you will be able to purchase your desired number, just make sure it supports the MMS feature.

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

==Configuring the SMS/MMS service==

Once purchased the SMS/MMS Service needs to be activated on the DID settings, from your portal >> DID Numbers >> Manage DID >> Edit Selection >> SMS/MMS configuration section.

In that section, you can activate and configure some forward options for the SMS/MMS service.

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

:* '''Message Service (SMS/MMS)''': Check the box to enable the service.
:* '''VoIP.ms SMS/MMS Portal''': This is enabled by default and it can't be modified, that means that you will be able to create and send messages from the SMS Message Center.
:* '''SMS/MMS Email Address''': By activating this option all the SMS Messages will be sent to the email address you configure in this field. You can receive them to more than 1 email address by separating each emails with a comma. The advantage of this option is that you will receive your SMS Messages directly to your email and you will be able to Reply to these messages from your email too. You just need to click on '''reply''' to the email. Please note that e-mail responses are as well limited to 160 characters and if they exceed this limit, they'll be split in two (or more messages if applies) when sent.

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

'''NOTE''' 💬: ➜ To reply the SMS Message via your email it is important to click on '''Reply''' and '''DON'T change or modify''' the destination mail "TO" (sms@voip.ms) and the subject E.G ''[#USXXXXXX] Message sent to 5555555555''.
</div>

:*'''SMS/MMS Forward''': You can have your SMS Messages forwarded to your cellphone or any other number that supports SMS, with this option you will receive the SMS Messages from our system to the configured phone number.
<div style="margin-left:50px;
padding:5px;
border-left: 6px solid rgb(208, 56, 45, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(51, 51, 51, 0.08);
width:70vw;">

'''NOTE''' 💬: ➜ The CallerID sent to the Call Forwarded Cell Phone will be the DID Number that received the SMS Message. Is not possible to forward the original Caller ID at this time.
</div>

:*'''SMS/MMS SIP Account''': You can have your SMS Messages forwarded to your selected SIP account. They will be sent through our SMS/SIP gateway as a SIP MESSAGE.
<div style="margin-left:50px;
padding:5px;
border-left: 6px solid rgb(208, 56, 45, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(51, 51, 51, 0.08);
width:70vw;">

'''NOTE''' 💬: ➜ The destination ("To" header field) of the SIP MESSAGE will be your account name, instead of the destination number of the original SMS.
</div>

:'''SMS/MMS URL Callback''': By enabling this option you will be able to send the messages to another destination e.g. your own server. If selected SMS Messages received by your DID will Send a GET request to the URL Callback provided. Available variables for your URL:
<div style="margin-left:50px;
padding:5px;
font-family: Consolas, Monaco, Lucida Console, Liberation Mono, DejaVu Sans Mono, Bitstream Vera Sans Mono, Courier New, monospace;
color: silver;
border-left: 6px solid rgb(208, 56, 45, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: black;
width:58vw;">
:: '''Example''': <nowiki>https://YourDomainName.tld/sms.php?to={TO}&from={FROM}&message={MESSAGE}&id={ID}&date={TIMESTAMP}</nowiki>

: '''{ID}''' = The ID of the SMS message.
: '''{TIMESTAMP}''' = The date and time the message was received.
: '''{FROM}''' = The phone number that sent you the message.
: '''{TO}''' = The DID Number that received the message.
: '''{MESSAGE}''' = The content of the message.

</div>

:*'''URL Callback Retry''': When selected, we will be expecting an "ok" output (without quotes) from your URL callback page as an indicator that you have received the message correctly. If we don't received the "ok" (without quotes) from your callback page, we will keep sending you the same message every 30 minutes.

Once you configure the desired option, click on Apply Changes.

==Sending a SMS/MMS (Web Interface)==

'''Step 1'''

: Click the "send New Message" button to get a list of contacts displayed or directly type in the number into the contact text field. If this number is registered in the phonebook, it will display its name instead.

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

'''Step 2'''

:Click to display the list of available DID numbers. Choose the number to send the message from.

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

'''Step 3'''

:Write your message on the text field and click the red arrow to send it. Text messages up to 160 characters will be considered SMS. Media messages or text messages over 160 characters will be considered MMS and you will see the icon turning from SMS to MMS indicating of this.
:: [[File:MMS-Webportal4.png|border|400px]]

'''Step 4'''

: In order to send a media file over an MMS, click the clip button to browse and choose a media file.

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

: Click the browse button to search for the media file in your computer or drag and drop the file directly into the box.

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

: The attached files will show below the text box.

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

: Once the message is sent, a new conversation will be opened on the left. Selecting this conversation will show the messages history on the box.

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

==Receiving an SMS/MMS (Web Interface)==

: While SMS messages can be received directly in an App through the SIP Protocol and correct configuration, at the moment, '''receiving and reading MMS is only possible through the message center.'''

: You can access the communication center from your customer portal at DID numbers > SMS/MMS Message Center. Any message received will appear here on the left side of the panel.

: Use the DID Filter to choose and display messages for a specific DID number or for all DID numbers.

: You can also filter the messages by date and time zone by clicking on the down arrow, which will show the search parameters

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

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

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

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

== SMS/MMS History ==

You can now also gain access to a general history of SMS/MMS that was sent and received for all of your DIDs, or for a specific one by going to Customer Portal >> DID Numbers >> SMS/MMS Message Center >> History.

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

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

= SMS using a SMPP Server =

An SMPP server, short of '''Short Message Peer to Peer''' is a method designed to send and receive high volumes of SMS messages through a dedicated server.

In other words, this allows you to better control the quantity of messages that will be sent to your target audience while avoiding to pass via a softphone application or your VoIP.ms portal to send/receive these SMS, thus eliminating multiple repetitive tasks and saving you time.

For example, if you have a promotion for your business that offers a discount for a certain lapse of time and that your customers have provided their agreement to receive these types of promotions via SMS, you could send your promotion to your customers in a faster manner than if you would do it from your customer portal, thus increasing efficiency.
<div style="margin-left:50px;
padding:5px;
border-left: 6px solid rgb(208, 56, 45, 0.8);
box-shadow: 8px 6px 9px -2px #f2f2f2;
background-color: rgba(51, 51, 51, 0.08);
width:70vw;">

'''NOTES''' 💬: ➜ Note that sending and receiving SMS messages will be charged our usual rate at $0.0075 per messages'''
</div>

==Requirements==

: '''This will require you to have your own SMPP server.'''
: Once configured properly, you will then be able to send and receive SMS messages to your own server.

== Configuration on your SMPP Server ==

: In order for your server to communicate and authenticate accordingly with VoIP.ms SMPP server, you will need to configure the following:

::* A '''username''' and '''password''' of your choice in order for VoIP.ms SMPP server to authenticate you.
::* Have all your communications be sent to '''smpp.voip.ms''' via port '''2775''', for regular SMPP, or port '''3550''' which is our encrypted SMPP port.
::* Open a connection using the '''bind_transceiver''' command. You can keep this connection open for as long as you need to and inquire about it using the '''enquire_link''' command.
::* Send your messages with either '''deliver_sm''' or '''submit_sm''' commands. Make sure to include your PDU's '''source_addr''' as one of your SMS enabled DIDs from your account. Your destination SMS number should be set as '''destination_addr''' while your text message will be included in '''short_message'''.

== Configuration SMPP on your VoIP.ms Portal ==

Once you have set up the above on your server, you will need to enable and specify which username and password will be used by VoIP.ms to authenticate your SMPP requests. In order to do so, head into your VoIP.ms portal, DID Numbers, Manage DIDs, edit your DID that supports SMS messages and by heading down the page, you will see the following options:

:* '''SMPP Enabled:''' If selected, SMS Messages will be allowed to be sent and received using SMPP.
:: You may specify the URL to be used to send a copy of inbound messages to your SMPP server. Leave empty if you only wish to send messages. See the example below. The '''scheme''' part is mandatory and must be set to either of '''smpp''' for unencrypted SMPP, or '''ssmpp''' for encrypted SMPP using TLS. The '''user:password@''' part is optional and will default to the same username and password defined for authentication on our side as defined below. The ''':port''' part is also optional and will default to 2775 for '''smpp''' and to '''3550''' for ssmpp.

:* '''SMPP Username:''' The username that you will use to authenticate to our SMPP server for sending messages
:* '''SMPP Password:''' The password that you will use to authenticate to our SMPP server for sending messages

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

'''NOTES''' 💬
: ➜ In order to help avoid any confusion while configuring the username on your server, you can use your VoIP.ms account ID as a username (the 6 digits of your account).
: ➜ Make sure to have the option '''Message Service (SMS/MMS)''' enabled.
</div>

Enable the option '''SMPP Enabled''', fill the username and password per what was configured on your SMPP server and once done, apply the changes by pressing '''Click here to apply changes'''.

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

You may now start sending and receiving SMS messages from your SMPP server.

= Send and Receive Text Messages (Web Interface)=

==SMS Message Center for mobile==

To start using the service you will need to use the SMS Message Center, from your portal >> DID Numbers >> [https://www.voip.ms/m/sms.php SMS Message Center].

For Mobile Devices please use [https://sms.voip.ms/ sms.voip.ms] as a complete all in one solution.

From the SMS Message Center you can check your SMS History, send and receive SMS messages.

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

:* '''Search Range''': Select a Date Range of your SMS History. You can select a time period by marking the "Show Details" field and the text messages will be displayed.
:* '''Search Filter''': You can filter your Search by DID, Contact and Type (Sent, Received and Both).
:* '''Send New SMS Messages''': Another window will open by clicking here in order to create a new SMS Message.
:* '''Delete Selected Messages''': This action will delete any messages that you have selected, by clicking on the box next to them. '''This action cannot be undone.'''
:* '''Delete All Messages''': Delete ALL your SMS Messages, both sent and received. '''This action cannot be undone.'''
:* '''Checking a Message''': To check a received SMS Message just click it directly in order to display the information.

==Create a New SMS Message (Legacy)==

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

You can create a new '''SMS Message''' we will explain you all the parameters you can find on this section:

* '''DID''': If you have more than one DID Number you can select any of the numbers available to send your message.
* '''Contact''': The destination phone number or if you have a [[Phone book|Phone book entry]] start typing the name and the system will display the information if available.
* '''Message''': You can create your message on that field, please note that you have up to 160 characters.

Then click on 'Send Message' to send the SMS.

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

'''NOTES''' 💬
: ➜ '''SMS via SIP''' service is only available through '''SIP Protocol and user/password authentication'''. '''IP Auth and IAX2 are not available at this time.'''
: ➜ '''Please refer to our guide [[SIP/SMS with FreePBX]] for instructions on how to configure the service with a FreePBX'''
</div>

== Sending ==

:* DID must have the '''"Short Message Service (SMS)"''' service option enabled. This option can be enabled from the Manage DIDs > Edit DID page.

:* To send an SMS from your SIP account, it is required that you set your Caller ID number to one of your SMS enabled DIDs. This will be the number you will be sending the message from. You can configure the Caller ID number from your customer portal for the specific subaccount if you are using a softphone or directly from your extension or trunk if you are using an Asterisk or PBX server.

:* It's important to note that if you configure your caller ID name in your SIP client to be a 10 digits number, this will override your caller ID number. If the caller ID name is anything different than a 10 digits number this will be discarded and the Caller ID number will be used.

:* If you are an asterisk or PBX user, please make sure to use the latest version of '''Asterisk (v12 or higher)''' and use '''chan_pjsip''' for the trunk. This is an asterisk limitation.

:* The Desktop version of Zoiper requires the user to have the '''PRO version''' activated to be able to send SMS messages as well as to enable '''SIP Presence'''. Please contact Technical Support to request SIP presence to be enabled for your account for the VoIP.ms POP you are using. While normally not required, there may be other cases not documented of softphones or apps requiring SIP Presence to be enabled. Please contact technical support if you encounter any issue with your specific software.

:* The '''free Desktop version''' of Zoiper cannot send SMS messages and is only capable of receiving SMS.

:* The '''mobile version''' of Zoiper is completely free of these limitations. Free version already works for sending and receiving without the need of SIP presence.

:* When sending an SMS message, the recipient number must be 10 digits. Avoid using the prefix '"1"' since is not supported at the moment. The system will recognize the numbers as in 10 digits.

:* The current characters limit for a '''single SMS is 160'''. If a message is bigger than 160 characters it will be split into blocks of up to 160 characters each and be sent as independent messages. The maximum total message length for SMS via SIP is 2000 characters.

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

'''NOTES''' 💬: ➜ AT THE MOMENT SENDING MMS USING SIP ACCOUNT IS NOT POSSIBLE AT THIS TIME.'''
</div>

:*DID must have the '''"Message Service (SMS/MMS)"''' service option enabled. This option can be enabled from the Manage DIDs > Edit DID page.
:*Enable the option '''"SMS SIP Account"''' and select the account destination for your SMS messages to this number.
:*The account receiving SMS must be registered successfully in any of our POPs.
:*To ensure the best results receiving the SMS messages, do not register the same SIP account on more than one softphone or app at the same time, this can lead to unexpected results.
:*For SIP devices that require this information, such as a trunk or PBX, we encode the original destination number in a custom SIP header field named "X-SMS-To".
:*3CX softphone currently does not support any third party SMS service. It will only work by having a direct connection to the 3CX PBX system. '''Please contact 3CX for more details'''.

== Send SIP/SMS with FreePBX ==

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

'''NOTES''' 💬: ➜ AT THE MOMENT SENDING MMS USING SIP ACCOUNT IS NOT POSSIBLE AT THIS TIME.'''
</div>

Yes, it is possible to configure a FreePBX system to send SMS, please refer [[SIP/SMS with FreePBX | here]] for detailed instructions.

=SMS/MMS using the Reseller Interface=

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

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

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

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

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

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

Then scroll down and find the feature "’’’SMS/MMS’’’", and enable it.

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

= Enabling/Disabling SMS/MMS on a DID via the client's interface =
For your client to enable or disable SMS/MMS from his portal, your customer will need to head under Services, My Services.
: [[File:SMSMMSreseller5.png|thumb|none|500px]]

Then proceed by heading into ‘’’DID Lists’’’ tab and edit DID.

: [[File:SMSMMSreseller6.png|thumb|none|500px]]
You will then be able to enable SMS/MMS, receive via email and more.

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

= Accessing their SMS/MMS via the clients interface =
1) To gain access to their SMS/MMS messages, head under Services and then SMS/MMS.
: [[File:SMSMMSreseller8.png|thumb|none|500px]]

[[category:guides]]

Servers

2023-01-18T19:28:55Z

John:

[[File:ChooseServerImg.png|thumb|none|1280px|VoIP.ms servers]]

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

= Choosing a Server =

[http://www.voip.ms VoIP.ms] offers many different servers, but which one should you choose? To put it simple, you can use the server closest to your location. Although that doesn't mean that a server further from your location will not work accordingly. For example, if you are located in Chicago, but you wish to use our Washington server, you will usually get good results. This entirely depends on the path taken by your internet service provider among many other reasons. Although the right rule of thumb is to use the server closer to you.

'''Note:''' Make sure that your SIP/IAX device and your phone number are pointing to the same server.

=== IPs ===

<div><ul>
<li style="display: inline-block; vertical-align:top;">
'''Canada'''
*Montreal 1, QC ('''montreal.voip.ms''') 208.100.60.19
*Montreal 2, QC ('''montreal2.voip.ms''') 208.100.60.20
*Montreal 3, QC ('''montreal3.voip.ms''') 208.100.60.21
*Montreal 4, QC ('''montreal4.voip.ms''') 208.100.60.22
*Montreal 5, QC ('''montreal5.voip.ms''') 208.100.60.23
*Montreal 6, QC ('''montreal6.voip.ms''') 208.100.60.24
*Montreal 7, QC ('''montreal7.voip.ms''') 208.100.60.25
*Montreal 8, QC ('''montreal8.voip.ms''') 208.100.60.26
*Montreal 9, QC ('''montreal9.voip.ms''') 208.100.60.27
*Montreal 10, QC ('''montreal10.voip.ms''') 208.100.60.28
*Toronto 1, ON ('''toronto.voip.ms''') 208.100.60.50
*Toronto 2, ON ('''toronto2.voip.ms''') 208.100.60.51
*Toronto 3, ON ('''toronto3.voip.ms''') 208.100.60.52
*Toronto 4, ON ('''toronto4.voip.ms''') 208.100.60.53
*Toronto 5, ON ('''toronto5.voip.ms''') 208.100.60.54
*Toronto 6, ON ('''toronto6.voip.ms''') 208.100.60.55
*Toronto 7, ON ('''toronto7.voip.ms''') 208.100.60.56
*Toronto 8, ON ('''toronto8.voip.ms''') 208.100.60.57
*Toronto 9, ON ('''toronto9.voip.ms''') 208.100.60.58
*Toronto 10, ON ('''toronto10.voip.ms''') 208.100.60.59
*Vancouver 1, BC ('''vancouver.voip.ms''') 208.100.60.60
*Vancouver 2, BC ('''vancouver2.voip.ms''') 208.100.60.61
*Vancouver 3, BC ('''vancouver3.voip.ms''') 208.100.60.62
</li>

<li style="display: inline-block; vertical-align:top;">
'''United States'''
*Atlanta 1, GA ('''atlanta.voip.ms''') 208.100.60.17
*Atlanta 2, GA ('''atlanta2.voip.ms''') 208.100.60.18
*Chicago 1, IL ('''chicago.voip.ms''') 208.100.60.8
*Chicago 2, IL ('''chicago2.voip.ms''') 208.100.60.9
*Chicago 3, IL ('''chicago3.voip.ms''') 208.100.60.10
*Chicago 4, IL ('''chicago4.voip.ms''') 208.100.60.6
*Dallas, TX ('''dallas.voip.ms''') 208.100.60.29
*Dallas 2, TX ('''dallas2.voip.ms''') 208.100.60.30
*Denver 1, CO ('''denver.voip.ms''') 208.100.60.32
*Denver 2, CO ('''denver2.voip.ms''') 208.100.60.33
*Houston, TX ('''houston.voip.ms''') 208.100.60.15
*Houston 2, TX ('''houston2.voip.ms''') 208.100.60.16
*Los Angeles 1, CA ('''losangeles.voip.ms''') 208.100.60.35
*Los Angeles 2, CA ('''losangeles2.voip.ms''') 208.100.60.36
*Los Angeles 3, CA ('''losangeles3.voip.ms''') 208.100.60.37
*Los Angeles 4, CA ('''losangeles4.voip.ms''') 208.100.60.38
*New York 1, NY ('''newyork.voip.ms''') 208.100.60.66
*New York 2, NY ('''newyork2.voip.ms''') 208.100.60.67
*New York 3, NY ('''newyork3.voip.ms''') 208.100.60.68
*New York 4, NY ('''newyork4.voip.ms''') 208.100.60.69
*New York 5, NY ('''newyork5.voip.ms''') 208.100.60.11
*New York 6, NY ('''newyork6.voip.ms''') 208.100.60.12
*New York 7, NY ('''newyork7.voip.ms''') 208.100.60.13
*New York 8, NY ('''newyork8.voip.ms''') 208.100.60.14
*San Jose, CA ('''sanjose.voip.ms''') 208.100.60.40
*San Jose 2, CA ('''sanjose2.voip.ms''') 208.100.60.41
*Seattle 1, WA ('''seattle.voip.ms''') 208.100.60.42
*Seattle 2, WA ('''seattle2.voip.ms''') 208.100.60.43
*Seattle 3, WA ('''seattle3.voip.ms''') 208.100.60.44
*Tampa, FL ('''tampa.voip.ms''') 208.100.60.46
*Tampa 2, FL ('''tampa2.voip.ms''') 208.100.60.47
*Tampa 3, FL ('''tampa3.voip.ms''') 208.100.60.48
*Tampa 4, FL ('''tampa4.voip.ms''') 208.100.60.49
*Washington 1, DC ('''washington.voip.ms''') 208.100.60.63
*Washington 2, DC ('''washington2.voip.ms''') 208.100.60.64

</li>

<li style="display: inline-block; vertical-align:top;">
'''International'''
*Amsterdam, NL ('''amsterdam1.voip.ms''') 208.100.60.65
*London, UK ('''london1.voip.ms''') 208.100.60.34
*Paris, FR ('''paris1.voip.ms''') 208.100.60.39
*Sydney, AU ('''sydney1.voip.ms''') 208.100.60.45

</li>
</ul></div>

= Latency and its importance =

Latency is very important for Voip, this will determine the time that will take the data package transmission to reach the destination. A high latency will lead to a delay and echoes in the communication.

While on a VoIP call, our server will transmit you information and you will be transmiting information to our server, which is the middleman between the mobile provider, PSTN provider or VoIP provider and you. Basically, while on a call, you will be sending us your voice packets which we will relay accordingly and the incoming voice packets (the user speaking), we will be transmitting them to you. If the travel time is too long, that is when you will notice a long delay during your conversation, which is refered by latency. This of course can be due to your internet service provider among many other situations.

Latency is measured in milliseconds (ms) For example: a latency of 150ms is barely noticeable, thus acceptable. Higher than that, quality starts to suffer. When it gets higher than 300 ms, it becomes unacceptable.

[[category:guides]]

Panasonic KX-TGP 550

2020-05-14T12:31:16Z

John:

== Devices ==
* Panasonic KX-TGP-550 (cordless VoIP 'phone with one base handset, one remote handset)
* Panasonic KX-TGP-500 (cordless VoIP 'phone with one remote handset, no handset on base)
* Panasonic KX-TGP-551 (cordless VOIP 'phone with one base handset, one remote handset)
<center>
{|
||[[Image:Pana550b.jpg|300px|thumb|caption|Panasonic KX-TGP 550]]
||[[Image:Panasonic KX-TGP-500.jpg|175px|thumb|caption|Panasonic KX-TGP 500]]
|}</center>
<div style="width:100%;overflow:hidden;clear:both"></div>

== Register Handset(s). ==

*Remember to install batteries in your cordless handsets before you start the registration process.

* Plug the cordless handset in the base unit.
* Press the center button to select the '''Menu''' option, and once you enter the '''Menu''' select the '''Initial Settings''' option (the icon is a wrench).
* Once you enter the '''Initial Settings''' menu select the '''Registration''' option (it will be on the 2nd page), and after you enter the sub-menu select '''Register handset''' (should be highlighted)
* On the base unit, press the '''ALL''' button for 4 seconds, the unit may start beeping.
*On the TGP500 (black box), it will be located on the right side of the unit.
*On the TGP550 (phone base), it will be the fourth button on the top row.
Then press the OK soft button on the handset, it may take a few seconds for the unit to register. Repeat for every wireless handset you have.

== Enter the Panasonic KX-TGP 550 Configuration Panel. ==

* On the handset, press the center button to select '''Menu''' and once you enter the '''Menu''' select '''IP Service''' (looks like a toolbox).
* On the '''IP Service''' menu the '''Network Setting''' option should be highlighted, select it and on the sub-menu select '''IP Settings''' and the IP address of the base unit should be displayed, '''record this IP address''', as you will need it momentarily.
*If the IP address doesn't display you may need to repeat the handset registration procedure or check that it has a connection to the network and a DHCP server is available.
* Press the Back soft button to be returned to the menu, and select '''Embedded web'''. By default, '''Off''' is highlighted, press the middle button up once to highlight '''On''' and press it once to save the setting. The handset will beep if successful.
* Enter the IP address of the base unit into your web browser, the default username is '''admin''' and password is '''adminpass'''.

== Register the Panasonic KX-TGP 550 with Voip.ms ==

Complete the following fields in the VOIP Tab:

* '''Phone number:''' 100000 (Replace with your Main SIP account or Subaccount UserID, e.g. 198765 or 198765_sub)
* '''Line ID:''' 100000 (Replace with your Main SIP account or Subaccount UserID, e.g. 198765 or 198765_sub)
* '''Registrar Server Address:''' servername.voip.ms (Pick one of VoIP.ms multiple [http://wiki.voip.ms/article/Choosing_Server servers])
* '''Registrar Server port:''' 5060 (Default SIP Port)
* '''Proxy server Address:''' servername.voip.ms (Pick one of VoIP.ms multiple [http://wiki.voip.ms/article/Choosing_Server servers])
* '''Proxy server port:''' 5060
* '''Presence server port:''' 5060
* '''Service domain:''' servername.voip.ms (Pick one of VoIP.ms multiple [http://wiki.voip.ms/article/Choosing_Server servers])
Remember to match the '''Registrar Server Address''', '''Proxy server Address''' and the '''Service domain''' options with each other.
Also remember to use the same server that you set as '''Point of Presence''' (POP) to your DID.
* '''Source port:''' 5060 (You may choose from our supported ports: 5060, 5080 or 42872)
* '''**Authentication ID:''' 100000 (Replace with your Main SIP account or Subaccount UserID, e.g. 198765 or 198765_sub)
* '''**Authentication password:''' ********* (Use the SIP account password-By default the same as the Customer portal)
* '''Keep Alive Interval:''' Change from 0 to around 15

**You can also find this information by logging to your [https://www.voip.ms/m/accountinfo.php Customer portal].

[[File:Panasonic.png]]

==Additional Details==

On the TGP500 model currently it isn't possible to use sub accounts, as the device wont accept the underscore (_) for certain fields on which the SIP account userID needs to be entered.

The default registration expiry time on this device is 3600 seconds. You may have to reduce that if you lose registration. Unfortunately, that can not be set on the web interface. You need to load a provisioning configuration file from a web or FTP server (details [http://www.voicesonic.com/panasonic/manuals/panasonic-others/TGP500-550-Admin-Guide.pdf here], page 132 for file format). The parm is REG_EXPIRE_TIME_[n]="mmm", where n is line number, and mmm is the registration interval. e.g. REG_EXPIRE_TIME_1="180" to set line 1 registration time to 180 seconds. A symptom of this problem is that the telephone web interface shows that the line is registered, but the voip.ms control panel indicates that it is not registered. This device supports multiple provisioning files in a hierarchy.

The "Phone Number" field may contain only digits, and *must* contain at least one digit, or the base unit won't try to register it. Within those restrictions it can be any number you want. When using the handset (or base unit) to select a line from which to dial, this number will be displayed next to the entry for the associated line. I think Panasonic's idea is that this is supposed to be a DID associated with the SIP address, but it doesn't have to be.

Enter your SIP username in the "Line ID" field and again in the "Authentication ID" field. If you are registering a voip.ms main account your SIP username will be your six-digit account number. If you are registering a sub-account it will be your main account number followed by an underscore followed by the sub-account number.

You can register at any voip.ms server (e.g. "atlanta.voip.ms") but you must enter your chosen server name in the "Registrar Server Address" and "Proxy Server Address" and "Service Domain" fields.

After saving these settings, check the "VoIP Status" page in the "Status" section to see if the status of this line is "registered". This product can register up to eight SIP addresses at once. If any of the SIP addresses you configured into the device do not show their status on this page as being "registered", then the status light on the base unit will flash yellow. (This is not mentioned in either the user guide nor the admin guide for this product.)

The above information applies to both the TGP-550 (where the base unit has a handset, dialpad, LCD display, etc.) and the TGP-500 (where the base unit is a faceless black box that sits in your network closet).

==Dial Plans==

If you wish to have 7 digit dialing within a USA area code, go to "Telephone" "Call Control" "Line #" and in the "Dial Plan" box, enter: <:585>[23456789]xxxxxx
With the addition of immediate *11 calls (such as 411 or 911): [23456789]11|<:585>[23456789]xxxxxx

[[Category:SIP phones]]

OBi 100/110 & OBi 200

2019-10-10T17:32:02Z

John:

[[File:OBi110-ATA.jpg|none|200px|center|link=http://www.polycom.com/voice-conferencing-solutions/voip-adapters.html|Obi]]

<br/>''The OBi100 & OBi200 are single phone port ATA adapters that support SIP VoIP services. The OBi100 & OBi200 are perfect for customers who do not have a traditional phone service, yet need a similar solution and want the savings and simplicity of using a VoIP service for all their calls. To start configuring your OBi100 or OBi200 device you will need to plug it in to your router/modem via its Internet port with an Ethernet cable and connect a regular handset phone to it's Phone port, then follow the next steps.''<br/>
<br/>
== Manual Configuration Details ==
<br/>
Start by dialing ''' * * * ''' from the connected phone, then press '''1''' to confirm your choice, this will return the IP address of your device being a number similar to '''192.168.xxx.xxx'''.<br/>
Once you get the IP address, enter it in the URL address bar '''"http://"''' of your Internet Browser to get access to the Graphic User Interface of the OBi100.

For an OBi202 please do the following to enable the GUI Web Interface:

Dial *** from the phone connected to the OBi202
Enter 0 For Advanced
Enter 30# Check Mark from
Press 1 to Enter a New Value
Press 1# to Enable
Press 1 to Save
Hang up

If done properly, the following window should appear on your screen:
[[File:ObiLogin.png|300px|thumb|left|Authentication Window - click to enlarge]]
<br clear="all" />

Once you get the Authentication Required window you will be prompted for a User Name and Password, the default credentials are:

'''User Name:''' admin

'''Password:''' admin

After this, you should now be able to see the OBi Web interface.

Now on the left side of the screen please find the next options and follow the next steps:

===Disabling auto-provisioning===

'''**NOTE :''' You may use this guide to configure an OBi110 as well. This is the VoIP.ms recommended configuration versus using the Obihai configuration dashboard (more on this later on this page) and you may also not find all new VoIP.ms servers on the Obihai Dahsboard. In order to make sure there will be no conflicts between this Manual configuration and the Obihai dashboard, please perform the following steps to disable auto-provisioning:

*System Management -> Auto Provisioning -> Auto Firmware Update -> Method : Disabled
*System Management -> Auto Provisioning -> ITSP Provisioning -> Method : Disabled
*System Management -> Auto Provisioning -> OBiTALK Provisioning -> Method : Disabled
*Voice Services -> OBiTALK Service -> Enable : Unchecked

Please note you must remove the check mark from the "default" column, then under "Method" please use the ''''Drop Down Selection'''' and choose '''Disabled'''.

[[File:Step1.png|450px|thumb|left|Disabling Auto Provision - click to enlarge]]
<br clear="all" />
After this, save all changes and you are ready to move on to the actual configuration.
<br />
===Configuring the ITSP Profile===

====General Section====
In this section you will set the name and the DigiMap you will use in the profile you configure. By default you will configure the profile A, unless you use the same device with another provider.

:'''Name''': 100000 (Replace with your 6 digit Main SIP account UserID or Sub Account user name, i.e. 123456 or 123456_sub)<br/>
:'''DigitMap''': Copy the line, including parenthesis, in the Digitmap field in the ITSP Profile and replace the "555" digits in the following lines by the area code of your choice:

::Dial Plan (recommended):
(1xxxxxxxxxx|<1555>[2-9]xxxxxx|<1>[2-9]xxxxxxxxx|911|011xx.|xx.|*xx.|***xxx|4xxx|(Mipd)|[^*]@@.)

Also, some clients have been successful by dialing '''**1 and *97''' for line 1 or '''**2 and *97''' for line 2.

:*If you need to set the dial plan back to Default, you can use this:

(1xxxxxxxxxx|<1>[2-9]xxxxxx|<1>[2-9]xxxxxxxxx|011xx.|***xxx|xx.|(Mipd)|[^*]@@.)

[[File:Step2.png|550px|thumb|left|ITSP profile, General - click to enlarge]]
<br clear="all" />

====SIP Section====
In this section you can set the server and the port that you wish to register to.

Please note that in order to change the settings, you need to uncheck the Default box on the right hand side.

*ProxyServer: denver.voip.ms (one of VoIP.ms multiple [http://wiki.voip.ms/article/Choosing_Server servers], you can choose the one closest to your location.)
*ProxyServerPort: 5060
*RegistrarServer: denver.voip.ms (one of VoIP.ms multiple servers [http://wiki.voip.ms/article/Choosing_Server servers], you can choose the one closest to your location.)
*RegistrarServerPort: 5060

[[File:Step3.png|550px|thumb|left|ITSP profile, SIP - click to enlarge]]
<br clear="all" />

Additionally, you may want to change the RegisterExpires value to 300, scroll down, deselect the default box and set the value there from 3600 to 300.

[[File:Step4.png|550px|thumb|left|ITSP profile, SIP (Register Expires)- click to enlarge]]
<br clear="all" />

===Configuring Voice Services===
<br/>
In this section you can set your Main account/sub_account credentials like User name and Password. The Main account password by default is the same password as the Customer Portal.

*AuthUserName: 100000 (Replace with your 6 digits Main SIP account UserID or Sub Account user name, i.e. 123456 or 123456_sub)
*AuthPassword: ****** (''Your SIP Account Password'')

[[File:Step5.png|550px|thumb|left|Voice Services (SIP Credentials) - click to enlarge]]
<br clear="all" />

Once you have finished changing all those settings, click on the button ''Submit'' to save the changes and ''reboot your OBi device'', your device should now be registered.

== Configuration for usage with call encryption TLS/SRTP ==

In order to use TLS/call encryption with OBi devices, you'll need to modify the following parameters accordingly. Please note that this is only available for some OBi devices, and the screenshots are from the most recent firmware version.

Under Service Providers > ITSP Profile > SIP use the following values:
:'''ProxySeverPort''': 5061
:'''ProxyServerTransport''': TLS
:'''RegistrarServerPort''': 5061
:'''OutboundProxyPort''': 5061
:'''X_OutboundProxyTransport''': TLS

[[File:TLSOBi1.png|550px|thumb|left|ITSP profile, SIP (TLS)- click to enlarge]]
<br clear="all" />

Under Voice Services > SP Service use the following values:
:'''X_KeepAliveServerPort''': 5061
:'''X_SRTP''': Use SRTP Only

[[File:TLSOBi2.png|550px|thumb|left|Voice Services, SP Service (SRTP)- click to enlarge]]
<br clear="all" />

== Configuration Using OBi Dashboard ==

Besides the Manual Configuration previously explained, Obihai also provides us with their own API dashboard where you can add your device, to complete the configuration in easy steps.
Add your device to the OBiTALK service in the OBi Dashboard [http://www.obitalk.com/obinet/]. Instructions for this are included with the OBi110 and are not discussed here.

After the OBi110 is added, edit the device. You can select '''Service Provider 1''' or '''Service Provider 2''' under the '''Configure Voice Services''' heading. This will take you to a page where you can select ''voip.ms''. Follow the instructions and once you are done the configuration will be downloaded to your Obi110.

==Known Issues and Resolutions==
=== Ghost Calls ===
<br/>
Some customers have reported receiving constant calls from "100" or "101" as callerID. These calls are going directly to your device and do not pass through our servers, so we cannot filter them. You can read more about these calls, including how to prevent them, here: [[Sip Scanner Ghost Calls]]

You can also follow these suggestions, specific for the OBi brand devices:
*You can just disable (by unchecking Enable) for SP2 and OBiTALK under your Voice Tab (If you are using our service as SP1).

*You can restrict which IP addresses that can connect to your OBi. Going to "Service Providers -> ITSP Profile A -> SIP -> X_AccessList" : voip.ms_ip_address. You can see the IP address of the server you are currently using from this link: [http://wiki.voip.ms/article/Choosing_Server#IPs Server's IPs]

*You can also change your Obi Firewall Setting X_InboundCallRoute to : {(?|x|xx|xxx|xxxx|xxxxx|xxxxxx|un@@.|anon@@.):}, ph
This will only allow 7 digit or greater numbers through.

*Another alternative: OBi Interface>> Voice Services>> SP1 Service>> X_InboundCallRoute: {>('Insert your AuthUserName here'):ph}, example:

{>('100000'):ph} where 100000 is replaced with your own six digit SIP account UserID or the sub-account registered with your device.

By default, OBi devices accept calls destined for any username. The above syntax rejects calls that are not intended for whatever you have configured as AuthUserName.

<br/>

===10 Second Delay Reaching voip.ms Voicemail Attendant when dialing *97 or *98===

The Obi 100, 110 and 202 devices have non-configurable 'short' and 'long' delays if a dialed sequence does not match a digitmap. So you may have a 10 second delay when you dial into your voip.ms voicemail because of the built-in 'long' delay. This can be resolved in a couple of ways. Simply dial a # sign after you dial *97 or *98. Or include literals in your digitmap under the Service Provider / ITSP profile A or B / General / digitmap. Here is an example digitmap with a *97 literal included:

(1xxxxxxxxxx|<1>[2-9]xxxxxxxxx|011xx.|xx.|*xx.|'*97'|(Mipd)|[^*#]@@.)

The literal in the example is '*97'. You could also add a literal for '*98'.

Then when you dial *97, the device immediately sends it instead of waiting 10 seconds.

Read more on digitmaps under the topic Digit Map Configuration in the Obi Device Admin Guide.

=== Call Drops ===

If you experience random call drops while in the middle of a call or if the person you talk to remains silent for over a minute (60 seconds by default), OBi will hang up the call. Please go here and check and increase the following setting (Physical interface -> LINE port -> DetectFarEndLongSilence / SilenceTimeThreshold)

=== Enable Message Waiting Indicator MWI ===

To enable MWI please refer to the following section of the OBI web page:

Voice Services -> SP1 Service -> Calling Features ->

'''MWIEnable''' - Uncheck the box at the far right, to be able to check the box at the left, this enables the option.

'''X_VMWIEnable''' - Uncheck the box at the far right, to be able to check the box at the left, this enables the option.

'''MessageWaiting''' - Mark the checkbox

After these steps, the MWI should be active and working.

'''If you are trying to place an outbound call and get a recorded message ¨There is no service to complete your call¨ Please do the following to resolve this.'''
In Your OBi Device please go to Physical Interfaces >> PHONE Port which by default it is PSTN and it needs to be changed to Trunk Group 1

=== Using the OBi Network ===

You can use your OBi device to make calls directly to other OBi devices "''The OBi comes out of the box ready to make FREE calls to other OBi endpoints using the OBiTALK network. Dialing **9 + obi account number will use the OBiTALK feature and does not place calls to regular numbers nor use our network. ''" (you can get more information about [http://www.obihai.com/features-and-set-up here]), be aware that those calls will not pass through our network. If you need assistance with that feature, please contact [http://www.obihai.com/request-support OBI's support].
<br/>
=== Can not dial *98 even if is on your DigiMap ===
By default your OBi device uses *98 as Blind transfer code. If you want to be able to dial *98 from your device, you should change this code. You can achieve this in the settings of your device at: ''Star Codes >> Star code profile (A/B)'', unmark the "default" box and change *98 for something else (like *99)

[[File:Step6.png|550px|thumb|left|Changing *98 default code - click to enlarge]]
<br clear="all" />

=== Using the Phone book of your customer portal ===

If you plan on using the Phone Book in your Customer Portal and Speed Dial *75. Please log into your OBi and change the built-in speed dial code from *75 in the device to something else.

=== An additional note regarding outgoing calls===

In at least one instance it was necessary to specify a non-default outbound calling route in the OBi110 to be able to place calls using the voip.ms service. The default setting had the OBi110 attempting to place calls using the PSTN port on the device. The relevant setting is:

'''Physical Interfaces >> PHONE Port '''
*PrimaryLine: (Select from drop-down)

[[File:ObiPhoneport.JPG|550px|thumb|left|Changing Phone Port - click to enlarge]]
<br clear="all" />

The default is PSTN. Select SP1 Service if you only have one SIP account configured on the device. Select Trunk Group 1 to have it attempt to place calls using SP1 first, then SP2. Additional Trunk groups can be configured under Voice Services >> Gateways and Trunk Groups.

Portions of this article have been taken from [http://www.toao.net/500-mangos-guide-to-configuring-an-obi100-obi110-and-obi202-ata Mango's Guide to Configuring an OBi ATA]. Used with permission.

==Guide Links==
In the event where you need the guide directly from OBIHAI Technology Inc, you may find the admin manual guide below:

Admin Manual : [https://www.obitalk.com/info/documents/admin_guide/OBiDeviceAdminGuide.pdf Download PDF]

[[Category:Analog Telephone Adapters]]

== OBI 100/110 OBI 200/202 star codes ==

Here's a list of the most common used Star codes available for the OBI. <br>
You may also consult the following link in order to see the full list of star codes available for the OBI devices:
<br>
[http://www.obihai.com/docs/OBiFeatureStarCodes.pdf OBI feature star codes]
<br>

* *81 Block Caller ID (Persistent Mode)
* *82 Unblock Caller ID (Persistent Mode)
* *67 Block Caller ID (One Time)
* *68 Unblock Caller ID (One Time)
* *72 Call Forward All (Enter Number + #)
* *73 Disable Call Forward All
* *60 Call Forward on Busy (Enter Number + #)
* *61 Disable Call Forward in Busy
* *62 Call Forward on No Answer (Enter Number + #)
* *63 Disable Call Forward No Answer
* *56 Enable Call Waiting
* *57 Disable Call Waiting
* *78 Do Not Disturb - Turn On
* *79 Do Not Disturb – Disable

File:TLSOBi2.png

2019-10-10T17:28:50Z

John:

File:TLSOBi1.png

2019-10-10T17:28:30Z

John:

File:VSP7XXRegistered.PNG

2019-08-28T22:33:44Z

John: uploaded a new version of "File:VSP7XXRegistered.PNG"