VoIP.ms Wiki - User contributions [en]

Polycom VVX 300, 400, etc

2024-08-26T10:20:34Z

Mauro: TLS addendum

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

'''This guide covers the Polycom VVX 300-series phones, tho it is also applicable to other VVX-series phones (such as the 400-series).'''

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==CONFIGURATION==

'''1. Access the Web Configuration Utility'''
----
<br>Connect the phone to a network with a DHCP server and wait for it to boot to the main interface. This typically takes about 1-2 minutes. Depending on the firmware version the phone has your IP may be displayed once it boots.<br>You can always access the current IP information by going to the following location in the menu:

&emsp;Home (Button) -> Settings -> Status -> Network -> TCP/IP Parameters

<br>Once you've obtained the phone's IP address, use your favorite web browser (Chrome/Firefox recommended) and visit the following address:<br>

&emsp;https://<phone IP Address>/

<br>The default admin login password is '456'. If everything goes as planned, you should be looking at a page that looks similar to below.<br><br>

&emsp;[[File:VVX311-Web-Conf-Util.png|400px]]

<br><br>'''2. Configure NTP Settings'''

<br>While not strictly required, it's a good idea to setup your backup NTP settings in case your DHCP server does not provide time service information. You can set this by going to the following location in the menu (red highlighted sections):

&emsp;Simple Setup -> Time Synchronization<br><br>

&emsp;[[File:VVX311-NTP-Settings.png|400px]]

<br>When finished, click "Save" at the bottom of the page.

'''<br><br>'''3. Configure SIP Settings''''''

<br>Begin the SIP configuration by going to the following menu location (red highlighted sections):<br>

&emsp;Settings -> Lines<br><br>

&emsp;[[File:VVX311-Lines-Menu.png|150px]]

<br>You should now be on the "Line 1" configuration screen. Each line button you want to use will need to be configured individually but all should generally require the same settings. This example will show you what was required for this specific setup but your requirements may vary.<br><br>

For each section listed below, you will need to enter this information (red highlighted sections):<br>

'''<br>"Identification" section:<br><br>'''

&emsp;• Display Name (This is your Outbound Caller ID name.) '''(See the requirements below)'''<br>
&emsp;• Address (Your main/sub account name)<br>
&emsp;• Label (The name listed next to the Line button on the phone)<br>
&emsp;• SRTP settings should all be set to "No"<br>
&emsp;• Server Auto Discovery should be "Disabled" since this is not used<br>

'''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.

'''<br>"Authentication" section:<br><br>'''

&emsp;• Use Login Credentials should be "Disabled"<br>
&emsp;• Domain (Leave blank)<br>
&emsp;• User ID (Your main/sub account name)<br>
&emsp;• Password (Your main/sub account password)<br>

'''<br>"Server 1" section:<br><br>'''

&emsp;• Address (Your selected VoIP.ms SIP server FQDN or IP address)<br>
&emsp;• Port (5060)<br>
&emsp;• Transport (UDPOnly)<br>
&emsp;• Expires (300, your value may depend on your gateway/router timeout values but this is typically fine)<br>
&emsp;• Subscription Expire(s) (300, same as Expires)<br>
'''
<br>"Message Center" section:<br><br>'''

&emsp;• Callback Mode (Contact)<br>
&emsp;• Callback Contact (*97 to dial the assigned extension's VM box)<br><br>

[[File:VVX311-Lines-Settings.png|500px]]

<br>When finished, click "Save" at the bottom of the page.

<br><br>'''4. Restart and Verify Operation'''

<br>To make sure that the settings take effect, I recommend you "restart" the phone by going to:

&emsp;Utilities -> Restart Phone<br>

When prompted, click "Yes".

Once your phone reboots, your display should show the phone online, the registration status section on VoIP's Control Panel should show registered and you should be able to make/receive calls.

===Configuring a Voice line using TLS===

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

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

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

=== Settings to change ===

In order to use TLS/call encryption with VVX 300, 400 IP phones, you'll need to modify the following parameters accordingly after following the configuration guidelines provided above. Please note that this is only available for some devices.

Under Settings > Lines use the following values:
:'''Enable SRTP''': Yes
:'''Offer SRTP''': Yes
:'''Require SRTP''': Yes

:'''Server 1'''
:'''Port''': 5061
:'''Transport''': TLS

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

User Manual : [https://support.polycom.com/content/dam/polycom-support/products/voice/business-media-phones/user/en/vvx300-quickuserguide-ucs-4-1-4.pdf Download PDF]

Admin Manual : [https://support.polycom.com/content/dam/polycom-support/products/voice/business-media-phones/user/en/vvx-300-310-400-410-admin-addendum-4-1-4.pdf Download PDF]

Softphones

2023-03-28T07:08:52Z

Mauro: Addition of SessionTalk to the list

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:AcrobitsLogo.png|left|Acrobits]]

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

'''Product:''' Acrobits

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

'''OS:''' IOS, Android

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

[[Acrobits|See Configuration Details]]
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==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]]
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==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]]
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==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]]
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==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]]
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==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]]
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==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]]
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==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]]
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==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]]
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==Grandstream Wave==

[[File:Grandstream Wave.png|200px|Thumb|left|Grandstream Wave]]

'''Product:''' Grandstream Wave

'''Company:''' Grandstream

'''OS:''' Android & iOS

'''Overview:'''
Grandstream Wave is a FREE softphone application that allows users to make and receive voice calls through their business or residential SIP accounts on any Android™ device (version 4.0+) from anywhere in the world.

This application supports integration of up to 6 SIP accounts, 6-way voice conferencing, and allows users to monitor their PBX (such as Grandstream's UCM6100 series IP PBX & UCM6510 IP PBX) while utilizing speed dial with up to 24 virtual BLF keys.

Grandstream Wave also supports advanced SIP telephony features including call transfer, LDAP phonebook integration and more.

[[Grandstream Wave|See Configuration Details]]
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==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]]
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==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]]
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==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]]
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==NetDial Sip Phone==

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

'''Product''': NetDial Sip Phone

'''Company''': NeoMecca

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

[[NetDial_Sip_Phone|See Configuration Details]]
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==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]]
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==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]]
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==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]]
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==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]]
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==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]]
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==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]]
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==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]]
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==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]]
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==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]]
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==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]]
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==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]]
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==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]]

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==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_for_Android|See Configuration Details for android]]<br/>
*[[ZoIPer_for_iOS|See Configuration Details for iOS]]<br/>
<div style="width:100%;overflow:hidden;clear:both"></div>
[[category:softphones]]

==ZoiPer Communicator==

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

'''Product:''' Zoiper Communicator

'''Company:''' Zoiper

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

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

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

File:SessionTalk logo.png

2023-03-28T07:03:10Z

Mauro: SessionTalk logo

SessionTalk logo

SessionTalk

2023-03-28T06:52:33Z

Mauro:

Session Talk is a free softphone available for Android and iOS users, it's a great free choice for iOS users. Their free version runs in background and works fine for Android and iOS, the free version does not include SMS SIP so you can only make and receive calls.

==Basic set up==
1. Download <b>"Session Talk"</b> from your phone store.

[[File:Sessiontalk01.png|300px|border]]
<br>
<br>

2. Allow access, notifications and permissions to your app.

[[File:sessiontalk02.png|300px|border]]
<br>
<br>

3. Generic SIP, tap the <b>'i'</b> icon

[[File:Sessiontalk03.png|300px|border]]
<br>
<br>

4. 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>
'''Display name:''' <b style="color:#b30000;">Type your name or company name (NOT guaranteed to be displayed)</b><br>
'''Username:''' <b style="color:#b30000;">Type your account number or subaccount name</b><br>
'''Password:''' <b style="color:#b30000;">Type your password</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>
<br>
<br>
From the '''advanced''' options find the audio codecs settings. In both the Wifi and Mobile plan settings use the '''Move''' option in the top right corner to adjust the settings. Set G.711u as the top option and slide the rest to the disabled section.

[[File:sessiontalk04.png|300px|border]] [[File:Popserver.png|600px|border]]

<br>
<br>

5. Tap the <b>'Enabled'</b> slider. Your account will be ready.

[[File:Sessiontalk06.png|300px|border]] [[File:Sessiontalk05.png|300px|border]]

<br>
<br>

6. Go to the dialpad and test the audio dialing 4443

[[File:Sessiontalk07.png|300px|border]] [[File:Sessiontalk08.png|300px|border]]

<br>
<br>

7. 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:CallerID.png|500px|border]]

Outgoing Calls not Working

2023-01-12T04:18:27Z

Mauro:

__TOC__

{| class="wikitable"
|+
|-
! Article en Français !! Artículo en Español
|-
| [[Les_appels_sortants_ne_fonctionnent_pas | Français ]] || [[Las_llamadas_salientes_no_funcionan | Español]]
|}

== Is your Account in Positive Stand? ==

By looking into your customer portal home page, you will see your current balance. If it is at $0 or in negative, this will not allow you to receive or make calls. Make sure that your balance is always in positive

[[File:Inbound1new.png|500px|border]]

== Are you Passing a Valid CallerID? ==

You can see this by heading into Main Menu, Account Settings, General tab. If you did not configure this yet, it is primordial that you configure your callerID to match one of your DID number or set a valid 10 digit callerID that you have but with another provider (your mobile phone for example).

[[File:Outbound2.png|500px|border]]

‘’’Note’’’: If you use a PBX server, you will need to send it from your PBX and not from the portal (some exceptions may apply).

== Are you Trying to Call an International Destination? ==

If you are trying to call internationally, make sure to verify the following 3 points:

===Is International Calls Enabled?===

By heading into Main Menu, Account Settings, Account Restrictions tab, make sure that ‘’’Allow International Calls’’ is set to ‘’’Yes’’’.

[[File:Outbound3.png|500px|border]]

=== Is the International Destination Allowed? ===

Still under Main Menu, Account Settings, Restrictions tab you will find at the bottom ‘’’Allow Calls to Countries’’’ setting which allows you to enable or disable international countries. Make sure that the country you wish to reach is enabled. If it is greyed out, is due some destinations with a higher per minute rate are locked by default, you can use this link https://www.voip.ms/m/high_cost_destinations.php or the "Allowed International Destination" link in your main page to unblock them.

[[File:Outbound4.png|500px|border]]

=== Additional Point to Verify for International Calls ===

: * Still under Account Restrictions, make sure that the setting ‘’’ International Amount Restriction’’’ is set above the actual cost of the destination you will reach. You can confirm the rate per minute of your destination under Rates, Check Rates Online.

: * If you have the country enabled but trying to reach a number still fails, by checking under Rates, Check Rates Online, type the number and see if a yellow lock appears. This means that there is currently no routes available to the destination you attempt to reach. Although please contact our support at support@voip.ms or via a ticket so that it can be looked into by our routing team.

== Is your Account Blocked by One of our Security Feature? ==

For your main account, this can easily be seen by heading into Main Menu, Account Settings, Security tab and seeing if any configuration has been done for the option ‘’’IP Restriction’’’ or ‘’’POP Restriction’’’. Same can be verified for your sub account by modifying your sub account.

[[File:Outbound5.png|500px|border]]

== Issue is Persisting ==

Please do contact our technical support:
: * By creating a [https://www.voip.ms/m/hp/tickets.php ticket],
: * Emailing us at '''Support@VoIP.ms'''
: * via [https://voip.ms/?startchat=1 Live Chat].

ZoIPer for Android

2022-10-26T02:52:51Z

Mauro:

__TOC__

== Tutorial Video ==

[[Image:Youtube_thumbnails_zoiper.jpg|200px|link=https://www.youtube.com/watch?v=-FB_1GwhGaA]]

==Basic Setup==
This is the very basic set up for '''Zoiper softphone''', there are only '''6 steps''' needed to have your device up and running. Refer to the advanced section if you want to improve your softphone performance and use some of the softphone features.

===Setting up Zoiper with your SIP/IAX credentials===

'''Step 1 -''' Download and install '''Zoiper IAX SIP VOIP Softphone''' from '''Play Store'''.

[[File:Zoiper PlayStore (2).png|thumb|none|300px|Click to enlarge]]

''' Step 2 -''' Start Zoiper and on the welcome screen, use your '''SIP/IAX credentials''' to fill the '''Username''' and '''Password''' then click on '''Create an account'''.

* '''Username:''' Your Main account or sub account username (six digit number) E.G 123456 / 123456_XX (the underscore has to be used for sub-accounts)
* '''Password:''' The password you set for the account / sub account

'''*Note 1:''' New accounts have received these '''main SIP/IAX credentials''' by e-mail when the account was created. By default, your
'''main SIP/IAX password''' is the same you use to log into your customer portal, however, for old accounts they could be
different if the portal password has been changed in the past. If you're not sure about your '''Main SIP/IAX password''',
update it from your customer portal at '''Main Menu>> Account Settings>> Security tab>> Main SIP/IAX password'''.

[[File:Zoiper Welcome.png|thumb|none|300px|Click to enlarge]]

'''Step 3 -''' On the host screen, fill the '''Hostname or Provider''' with one of VoIP.ms multiple [[Choosing Server#Choosing_a_Server | servers]], you can choose the one closest to your location, then click on '''Next'''.

'''Note 2:''' Bear in mind to use the same VoIP server your VoIP number is using. You can check what VoIP server
your VoIP number is using, from your VoIP.ms customer portal at '''DID Numbers>> Manage DIDs''', under '''POP''' column.
You can choose any server you want, as long as the one in your portal and the one in this field matches, otherwise,
incoming calls won't ring.

[[File:ZoipperApp_Host.png|thumb|none|300px|Click to enlarge]]

'''Step 4 -''' '''Skip''' the '''Authentication''' step, AuthUserID and Outbound Proxy are not needed.

[[File:ZoiperApp_authskip.png|thumb|none|300px|Click to enlarge]]

'''Step 5 -''' In this screen, you can select the protocol your account will use:
* SIP TLS
* SIP TCP
* SIP UDP
* IAX UDP

Select the one you prefer according to your network's settings, i.e. '''SIP UDP''' then click on the '''Finish''' button.

'''Note 3:''' If all these options show "Not found" but the Hostname is fine, just click '''Finish'''.

If you chose '''TLS''' please refer to [https://wiki.voip.ms/w/index.php?title=ZoIPer_for_Android&stable=0&shownotice=1&fromsection=Network_settings#Call_Encryption_TLS.2FSRTP section 2.3] for some extra adjustments, only if they are needed. When choosing TLS, Zoiper should adjust these parameters automatically.

[[File:ZoiperApp_acctype.png|thumb|none|300px|Click to enlarge]]

'''Step 6 -''' At this point your VoIP.ms account should be shown under Accounts section and marked on <font color="green">Green</font> with a tick arrow next to it. This means your softphone is ready to start placing calls. '''In case of failure''', your account will be marked on <font color="red">Red</font> and will display an error code. You can have more information about this error and how to fix it from [https://wiki.voip.ms/w/index.php?title=ZoIPer_for_Android&stable=0&shownotice=1&fromsection=Configuration_Detail#SIP_error_codes section 3.2].

'''Note 4:''' Bear in mind to set the '''Caller ID Number''' for your outgoing calls from your customer portal. If you're using the
main account at '''Main Menu>> Account Settings>> General tab>> Caller ID Number'''. If you're using a sub account at
'''Sub accounts>> Manage Sub Accounts>> Edit Sub-account''' (orange icon with a pen)'''>> Caller ID Number'''.

[[File:Zoiper acc ready.png|300px|Click to enlarge]]

'''Finally,''' you can check if your account is registered from your customer portal, at the main page. If your account is now registered and you have set your caller ID number for outgoing calls, you will be able to start placing outgoing calls.

[[File:Acc registered portal ATL1.png|600px|Click to enlarge]]

==Advanced setup and features==

=== Audio codecs ===

In order to use only supported codecs by VoIP.ms, please go to '''Settings>> Accounts>> Your account'''.

[[File:ZoiperApp_accsettings.png|thumb|none|300px|Click to enlarge]]

Scroll down to '''Audio Codec Settings''', and just enable '''"uLaw"''' and '''"GSM"'''. You can touch over any codec and move it to the top of the list in order to change priority order. We suggest keeping "uLaw" at the top. In case, you have bought '''"g729"''' make this your first option and move it right to the top.

[[File:Zoiper codecs.png|thumb|none|300px|Click to enlarge]]

'''IMPORTANT for Caller ID''':
- 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.

For best results, it is recommended to use WI-FI. But if using 4G/LTE and you experience poor sound quality, it has been reported that turning off '''ECHO CANCELLATION''' helps solve the issue.

=== Network settings ===
In order to have your softphone running all the time, proceed with the following steps:

'''1.''' Go to '''Settings>> Connectivity''' and enable '''"Keep Alive WIFI"''' and '''"Run in Background"'''.

[[File:Zoiper conectivity.png|thumb|none|300px|Click to enlarge]]

'''2.''' Go to '''Settings>> Accounts>> Your Account>> Network Settings''' and change '''"Registration Expiry Time"''' to 180.

[[File:Zoiper exp time.png|thumb|none|300px|Click to enlarge]]

'''Done''', at this point you have configured your softphone to work within our service using the best settings. These settings will help you to place and don't miss any incoming call.

===Call Encryption TLS/SRTP===
If you decided to use ''' SIP TLS - call encryption''' along with '''Zoiper''' please follow these steps:

'''1.''' Make sure your Main account or sub-account has '''"Encrypted SIP Traffic"''' enabled.

Bear in mind, if this setting is enabled and your device sends '''UDP/TCP''', '''RTP''' or '''video codec information'''
you will be rejected with error code 488.

Enable this setting for the Main Account at '''Main Menu>> Account settings>> Advanced tab'''.

[[File:Mainacc encryp.png|thumb|none|600px|Click to enlarge]]

For a sub-account enable it at '''Sub accounts>> Manage sub-accounts''' by clicking on the orange icon with a pen and finally click at '''"Advanced Options (Click here to display)"'''.

[[File:Subacc encryp.png|thumb|none|600px|Click to enlarge]]

'''2.''' Now that your account/sub-account has this setting enabled, your device only needs to send '''TLS''' and '''SRTP'''.

To send '''TLS''', go to '''Settings>> Accounts>> Your account''' and type:

*'''Host''': atlanta1.voip.ms:5061 (Use the same [http://wiki.voip.ms/article/Choosing_Server server] your phone number is at, you can check it out from your customer portal at '''Manage DIDs''' section, under '''POP''' column)
*'''Username''': Your account's number or sub-account name
*'''Password''': Your SIP/IAX password

'''Note:''' When using TLS is very important to specify the number of the server, in case the server's name doesn't have the number "1" included, you need to add it. Adding any of the SIP ports '''5061/5081/42873''' at the end of the '''Hostname''' is also required.

[[File:Tls zoiper.png|thumb|none|300px|Click to enlarge]]

Now scroll down and look for '''Network settings''' and change the "Transport mode" to '''TLS'''.

[[File:Zoiper tls.png|thumb|none|300px|Click to enlarge]]

Finally to send '''SRTP''', go back to your account settings and at '''"Encryption settings"''' enable '''SRTP'''.

[[File:Zoiper srtp.png|thumb|none|300px|Click to enlarge]]

Your account should be shown as registered and when you place a call you will see a closed padlock, which means the call is now secure.

[[File:Call tls.png|thumb|none|300px|Click to enlarge]]

You can also check if '''Zoiper''' is fully registered and using '''SIP-TLS''' protocol from your customer portal at Home page, '''Main menu>> Portal Home''' for each account/sub account registered or at '''Sub Accounts>> Manage Sub accounts''' tab to see all of your Sub Accounts registration status.

A <font color="green">green padlock</font> [[File:green_padlock.png]] will appears on the right of <font color="green">"Registered"</font> in green. If you don’t see the padlock, you need to revalidate some configuration.

[[File:TLS-SRTP-Steps0006.png|thumb|none|600px|Click to enlarge]]

[[File:TLS-SRTP-Steps0007.png|thumb|none|600px|Click to enlarge]]

===SMS SIP Account===
Zoiper is able to send and receive '''SMS''' using the '''SIP MESSAGE''' header. In order to use this feature along with Zoiper, there are some settings you need to adjust from your customer portal and your VoIP.ms number must support '''SMS''' service.

Bear in mind this feature is in '''BETA''' phase and we can't guarantee will work all the time.

Note: If your SIP device is offline, undelivered SMS will be trying to reach it during the next 24 hours after
they were received. After this time, they won't be delivered in your SIP device BUT they won't be lost.
You can check all your SMS from your customer portal at '''DID Numbers>> SMS Message Center''' at anytime.

====Receiving SMS====

You can use your '''Main Account''' or a '''Sub-account'''; from your customer portal enable the following:

'''1.''' Go to '''DID Numbers>> Manage DIDs''' section and click at '''Edit DID''' (orange icon with a pen), at the bottom of this section you will find '''Short Message Service (SMS)''' section. Make sure the following are enabled and ticked:

*'''Short Message Service (SMS):''' Enabled.<br>
*'''SMS SIP Account BETA:''' Choose the Main Account or Sub-account you are currently using in Zoiper. <br>

[[File:Zoiper SMSSIPAcc.png|thumb|none|600px|Click to enlarge]]

From Zoiper zoiper ->settings->accounts screen select your account and enable the following enable publish presence & enable subscribe presence.
At this point, you should be able to receive incoming '''SMS''' in Zoiper, they will show up under the '''Message''' icon in Zoiper. You can also reply to this message using the same thread.

[[File:Zoiper SMS1.png|thumb|none|300px|Click to enlarge]]

====Sending SMS====
You can use your '''Main Account''' or a '''Sub-account'''; from your customer portal enable the following:

'''1.''' Your Main Account or Sub-account is using the phone number with '''SMS''' service enabled as its own '''caller ID number'''. You can check this for your Main Account at '''Main Menu>> Account settings>> General tab''' and for a Sub-account at '''Sub accounts>> Manage Sub accounts''' by clicking at '''Edit Sub-Account''' (orange icon with a pen).

'''2.''' If the phone number is properly set in your customer portal, you will be able to send SMS using Zoiper from the '''Message''' tab and by typing the destination number using 10 digits. You can also type in the '''TO''' field the name of any of your contacts to send the SMS directly.

[[File:Zoiper SMS2.png|thumb|none|300px|Click to enlarge]]

=== Zoiper PUSH service ===

Zoiper PUSH service is a paid feature that allows you to receive incoming calls in a more reliable way than decreasing the registration expiry time to save battery life. Bear in mind that when using this feature there's an extra device in the path between your endpoint and our servers, so in some cases, you might need to contact Zoiper support for troubleshooting.

Bear in mind this is an extra and paid feature offered by Zoiper, this is an optional feature you can use to
improve the performance of your incoming calls if your network or phone conditions don't work with the suggested
settings in previous section 2.2. We do not force our customers to use this feature, but if they decide to use it, this
section will help you with some of its settings.

The default configuration provided by Zoiper is good enough to work properly within our service but the following settings are suggested:

'''1.''' At '''Settings >> Connectivity:'''
* Check/Enable '''"Enable PUSH Proxy"''' option.
* Set '''"Proxy protocols"''' to '''SIP+RTP'''.
* Set '''"Push transport"''' to '''TCP''' (Default).

[[File:Zoiper-PUSH.png|thumb|none|300px|Click to enlarge]]

'''2.''' At '''Settings >> Accounts >> Your account''':
* Set '''"Authentication user"''' with your Main Account number or Sub account name (It's the same value you have for '''"Username"''').

[[File:Zoiper-AuthID.png|thumb|none|300px|Click to enlarge]]

'''3.''' At '''Settings>> Accounts>> Your account >> Network settings''':
* Set '''"Transport Type"''' to '''TCP'''

[[File:Zoiper-transport.png|thumb|none|300px|Click to enlarge]]

==Common Errors==

=== No transport mode found ===

If there was a typo in the hostname or if there's a network DNS issue, Zoiper might show the account type options in red. You can select the protocol and hit '''Finish''' button even though, they're all red.

'''Note:''' Sometimes even when the username, password and server's name were typed fine, Zoiper will show all
those options as "Not found" in red. However, you can choose your desired selection and hit '''Finish'''.

[[File:ZoiperApp_accerror.png|thumb|none|300px|Click to enlarge]]

=== SIP error codes ===

In case your account has been marked on red, an error code number will be shown at '''Settings>> Accounts''' section. You can check the error response code and correct it by checking the list of common [https://wiki.voip.ms/article/SIP_Responses SIP Response codes].

==== 403 error ====
This is the most common error you can get and it's related to wrong information in your '''username''' and/or '''password'''. Please make sure you have the right credentials and no blank spaces are in your username or password.

[[File:ZoiperApp_SIPcode.PNG|thumb|none|300px|Click to enlarge]]

==== 408 error ====
On some networks, you might get this error code that could disappear when you change to a different network. This could be due to some limitation on the local SIP ports for this network. As a workaround, you can try the following modifications to your settings.

'''1.''' Go to '''Settings>> Conectivity>> Listening Ports''' and untick '''"Random Port"''' for Local SIP port only and type '''5080''' or '''42872''' in the '''"Port"''' field.

[[File:Zoiper ports.png|thumb|none|300px|Click to enlarge]]

'''2.''' Go to '''Settings>> Accounts>> Your account''' and at '''"Host"''' add ''':5080''' or ''':42872''' to the server's name.

[[File:Zoiper SIP acc port.png|thumb|none|300px|Click to enlarge]]

FreePBX / PBX in a Flash

2021-04-06T14:45:59Z

Mauro:

{{DISPLAYTITLE: FreePBX / PBX in a Flash}}

[https://www.freepbx.org/ '''FreePBX'''] is a web-based open source GUI (graphical user interface) that controls and manages Asterisk (PBX), an open source communication server. FreePBX is licensed under the GNU General Public License (GPL), an open source license. FreePBX can be installed manually or as part of the pre-configured FreePBX Distro that includes the system OS, Asterisk, FreePBX GUI and assorted dependencies.

==Important Security Information==

A '''critical vulnerability''' has been discovered that can affect FreePBX versions between 13.0.12 and 13.0.26. An unauthenticated remote attacker can run shell commands as the Asterisk user of any FreePBX machine with ‘Recordings’ This has been fixed in Recordings 13.0.27.

You can read more about this vulnerability including how to fix this here: http://wiki.freepbx.org/display/FOP/2016-08-09+CVE+Remote+Command+Execution+with+Privileged+Escalation

------------

We are also aware of an '''important''' and '''critical exploit''' related to all FreePBX versions prior to 12. This Zero-Day Remote Code Execution and Privilege Escalation exploit allows users to bypass authentication and gain ‘Full Administrator’ access to the FreePBX server when the ‘FreePBX ARI Framework module/Asterisk Recording Interface (ARI)’ is present on the system. This vulnerability may offer to any non authorized user full remote code execution access as the user running the Apache process. This exploit can be present also for users who have updated to version 12 from a prior version and did not remove the legacy FreePBX ARI Framework module.

Here are some recommendations for their product from the freepbx.org website for protection against this issue: http://www.freepbx.org/critical-freepbx-rce-vulnerability-all-versions/

== Creating a Trunk ==

To connect your FPBX server with ours, you need to create a '''trunk'''. To achieve this, once you are into your FBPX's GUI, follow this path: Connectivity >> Trunks >> Add SIP (chan_sip) Trunk.

[[File:FPBX_00.png|550px|thumb|left|Creating a trunk- click to enlarge]]
<br clear="all" />

Once you are there, you will see a list of options, create a '''SIP''' or '''IAX''' trunk (depending on your needs)

[[File:FPBX_01.png|550px|thumb|left|Creating a trunk- click to enlarge]]
<br clear="all" />

=== SIP Trunk ===

From here, use the following example to configure your SIP trunk:

==== General Settings ====

* Trunk name: Set your trunk name, a recommended one could be '''voipms''', remember that you can manage more than 1 DID number with the same trunk (using your inbound routes).

* Outbound CallerID: The 10 digit valid caller ID number that you will pass with this trunk for Outbound calls. This can be override from your extension's settings.

==== Dialed number Manipulation Rules ====

You can set here your outbound rules. These rules can manipulate the dialed number before sending it out this trunk. If no rule applies, the number is not changed. (Optional)

==== Outgoing Settings ====

This section is very important, below you'll find a sample, please replace the dummy information with yours and delete the comments. '''The information after the semicolon (;) is considered a comment and must be deleted for the trunk to work properly on some PBX versions.'''

canreinvite=nonat
nat=yes
context=from-trunk
host=atlanta.voip.ms ; (one of our multiple [http://wiki.voip.ms/article/Choosing_Server servers], you can choose the one closer to your location)
username=100000 ; (Replace with your 6 digit Main SIP Account User ID or Sub Account username, i.e. 123456 or 123456_sub)
fromuser=100000 ; (Replace with your 6 digit Main SIP Account User ID or Sub Account username, i.e. 123456 or 123456_sub)
secret=***** ; (password associated with the Main or Sub-account. '''Please avoid using the '#' character in the password as it will cause authentication issues''')
type=peer
disallow=all
allow=ulaw
; allow=g729 ; uncomment if you purchased g.729 from Digium
trustrpid=yes
sendrpid=yes
insecure=invite
qualify=yes

==== Incoming Settings ====

Please '''delete''' the default settings you'll find here, this section must be blank.

==== Registration ====

At this section you'll set your '''register string''', this is needed when you use "registration" as authentication method (If you use '''IP Authentication''' leave this in blank)

It is formed with your SIP username, password, server and registration port as below:

YourAccountNumber:Password@server.voip.ms:5060, for example:

100000:YourPassword@atlanta.voip.ms:5060

Finally, click on '''Submit changes''', after that you'll see a Red button in the top "'''Apply config'''", do not forget to click it to apply the changes.

[[File:FPBX_SIP_01.png|1000px|thumb|left|SIP trunk configuration- click to enlarge]]
<br clear="all" />

====TLS====
In order to use TLS along with FreePBX please follow these steps:

1. Make sure your Main account or sub-account has "Encrypted SIP Traffic" enabled. Bear in mind, if this setting is enabled and you use UDP/TCP you will be rejected with error code 488. Enable this for the Main Account at '''Main Menu>> Account settings>> Advanced tab''' and for a sub-account at '''Sub accounts>> Manage sub-accounts''' and by clicking on the orange icon with a pen and click at "Advanced Options Click here to display"

[[File:Mainacc encryp.png|thumb|none|300px|Click to enlarge]]

[[File:Subacc encryp.png|thumb|none|300px|Click to enlarge]]

2. Now that your account/sub-account has this setting enabled, your device only needs to send TLS and SRTP.

In freepbx make sure your '''peer details''' are:

host=atlanta1.voip.ms
username=your account/sub account
fromuser=your account/sub account
secret=your password
transport=tls
encryption=yes
qualify=yes
qualifyfreq=50
nat=yes
type=peer
directmedia=no
context=from-trunk
insecure=invite
sendrpid=yes
trustrpid=yes
disallow=all
allow=g729&ulaw&gsm

'''Register String'''
tls://Username:Password@atlanta1.voip.ms:5061~300

'''Note:''' When using TLS is very important to specify the number of the server, in case the name you have chosen doesn't use the number 1 you need to add it, at least when using TLS

Finally, in your freepbx go to '''Settings>> Asterisk SIP settings>> Chan SIP settings''' and at "TLS/SSL/SRTP Settings"

'''*Enable TLS''': Yes

'''*Don't verify server''': Yes

[[File:TLS freepbx.png|thumb|none|300px|Click to enlarge]]

=== IAX2 Trunk ===

From here, use the following example to configure your IAX2 trunk:

==== General Settings ====

* Trunk name: Set your trunk name, this is different from SIP trunks and it '''must''' be '''voipms''', otherwise you could experience issues with your registration and calls.

[[File:IAX_TrunkName.png|500px|thumb|left|IAX2 trunk configuration- click to enlarge]]
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* Outbound CallerID: The 10 digit valid caller ID number that you will pass with this trunk for Outbound calls. This can be override from your extension's settings.

==== Dialed number Manipulation Rules ====

You can set here your outbound rules. These rules can manipulate the dialed number before sending it out this trunk. If no rule applies, the number is not changed. (Optional)

==== Outgoing Settings ====

This section is very important, below you'll find a sample, please replace the dummy information with yours and delete the comments. '''The information after the semicolon (;) is considered a comment and must be deleted for the trunk to work properly on some PBX versions.'''

type=friend
username=100000 ; (Replace with your 6 digit Main SIP Account User ID or Sub Account username, i.e. 123456 or 123456_sub)
secret=***** ; (password associated with the Main or Sub-account. '''Please avoid using the '#' character in the password as it will cause authentication issues''')
context=from-trunk
host=atlanta.voip.ms ; (one of our multiple [http://wiki.voip.ms/article/Choosing_Server servers], you can choose the one closer to your location)
disallow=all
allow=ulaw
insecure=port,invite
requirecalltoken=no
qualify=yes

==== Incoming Settings ====

Please '''delete''' the default settings you'll find here, this section must be in blank.

==== Registration ====

At this section you'll set your '''register string''', this is needed when you use "registration" as authentication method (If you use '''IP Authentication''' leave this in blank)

It is formed with your SIP username, password, server and registration port as below:

YourAccountNumber:Password@server.voip.ms:4569, for this example:

100000:YourPassword@atlanta.voip.ms:4569

Finally, click on '''Submit changes''', after that you'll see a Red button in the top "'''Apply config'''", do not forget to click it to apply the changes.

[[File:FPBX_IAX_01.png|1000px|thumb|left|IAX2 trunk configuration- click to enlarge]]
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== Outbound routes ==

Once you have your trunk configured, you will need an outbound route to make calls.

To create an outbound route go to "Connectivity" menu and then select "Outbound routes".

=== Route Settings ===

* Route Name: Name of this route. Should be used to describe what type of calls this route matches (for example, 'local' or 'longdistance').

* Route CID: If set, this will override all CIDS specified except:
**Extension/device EMERGENCY CIDs if this route is checked as an EMERGENCY Route
**Trunk CID if trunk is set to force it's CID
**Forwarded call CIDs (CF, Follow Me, Ring Groups, etc)
**Extension/User CIDs if checked

=== Dial Patterns ===

A Dial Pattern is a unique set of digits that will select this route and send the call to the designated trunks. If a dialed pattern matches this route, no subsequent routes will be tried. If Time Groups are enabled, subsequent routes will be checked for matches outside of the designated time(s).

''' Rules: '''
* X: matches any digit from 0-9
* Z: matches any digit from 2-9
* [1237-9]: matches any digit in the brackets (example: 1,2,3,7,8,9)
* '''.''' : wildcard, matches one or more dialed digits
* Prepend: Digits to prepend to a successful match. If the dialed number matches the patterns specified by the subsequent columns, then this will be prepended before sending to the trunks.
* Prefix: Prefix to remove on a successful match. The dialed number is compared to this and the subsequent columns for a match. Upon a match, this prefix is removed from the dialed number before sending it to the trunks.
* Match patterns: The dialed number will be compared against the prefix + this match pattern. Upon a match, the match pattern portion of the dialed number will be sent to the trunks.
* Caller ID: If CallerID is supplied, the dialed number will only match the prefix + match pattern if the CallerID being transmitted matches this. When extensions make outbound calls, the CallerID will be their extension number and NOT their Outbound CID. The above special matching sequences can be used for CallerID matching similar to other number matches.

Recommended Dial patterns are:
* 1NXXNXXXXXX
* NXXNXXXXXX
* 4XXX (This one to be able to test our '''echo test''' and '''DTMF test''')

=== Trunk Sequence for Matched routes ===

The Trunk Sequence controls the order of trunks that will be used when the above Dial Patterns are matched.

Select there your voip.ms' trunk.

[[File:FPBX_Outbound.png|1000px|thumb|left|Outbound route configuration - click to enlarge]]
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== Inbound routes ==

If you have DID numbers with us and route calls to your trunks, you need inbound routes to manage them. To create an inbound route, go to "Connectivity" menu, option "Inbound routes"

=== Add Incoming Route ===

*Description: Provide a meaningful description of what this incoming route is.

*DID number: Define the expected DID Number if your trunk passes DID on incoming calls. Set your '''voip.ms DID number''' with only 10 digits (Without dots, commas, spaces or the 1 in front of the number).

=== Set Destination ===

Set here the destination for your incoming calls received at the DID you configured as '''DID number''' (an extension, IVR, recording, voice mail, etc).

[[File:FPBX_Inbound.png|1000px|thumb|left|Inbound route configuration - click to enlarge]]
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Once you have finished the basic configuration of your PBX server, do not forget to click on the red button "Apply Config"

[[File:After_Changes.png|550px|thumb|left|Apply Setting's button - click to enlarge]]
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= FreePBX using a PJSIP Trunk =

Please see our wiki article for the configuration [[FreePBX (PJSIP)| here]].

File:IAX TrunkName.png

2021-04-06T14:44:37Z

Mauro:

Primeros Pasos

2021-03-23T15:42:05Z

Mauro:

<br clear="all" />
[[File:Logo-VoIPms-light.png|left|400px]]
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{| class="wikitable"
|+
|-
! Article in English !! Article en Français
|-
| [https://wiki.voip.ms/article/Getting_Started English] || [https://wiki.voip.ms/article/Mise_en_Marche Français]
|}

Esta guía tiene como objetivo ayudarle a empezar a utilizar su cuenta [http://voip.ms Voip.ms] y sus funciones, de acuerdo a las necesidades de su negocio o de su uso personal. Una cuenta creada recientemente puede ocasionar muchas preguntas, la mayoría de estas preguntas pueden ser respondidas por el personal de soporte a través del chat en vivo o por el sistema de tickets. Sin embargo la información de este articulo pretende hacer más fácil su primer contacto con VoIP.

== Video Tutorial ==
[[Image:GetStartedThumbnail.png|200px|link=https://www.youtube.com/watch?v=XjD81OkSOGU]]

== Cree su cuenta ==

Usted puede crear de manera gratuita su cuenta, solo ingrese a esta liga: [https://www.voip.ms/signup.php Signup], a, ahi tendrá acceso a la página de ingreso, por favor ingrese la información solicitada y envíe su solicitud. Una vez completado este proceso, usted recibirá la confirmación mediante correo electrónico( esto valida la cuenta y la dirección de correo que usted proporcionó en la página de ingreso ).

Recuerde revisar todas las carpetas de su correo electrónico, para confirmar que usted recibió la confirmación, sino la recibiera, por favor
contacte a VoIP.ms en la dirección : support@voip.ms

[[File:Sign_Up.jpg|thumb|none|784px|Sing Up]]

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'''Una vez completado este proceso, tenga en mente lo que exactamente usted quiere hacer con su cuenta. Se recomienda que la primera vez que entre al portal del cliente revise la sección Account Settings, de esta manera usted podrá conocer un poco mejor las diferentes opciones que el sistema ofrece para configurar su línea teléfonica.'''

== Agregando Fondos==

Para realizar o recibir llamadas mediante el servicio de VoIP.ms, usted necesitará agregar fondos suficientes en su cuenta, recuerde que éste es un servicio de pre-pago, el monto agregado ( mínimo $15 USD ) se reflejará en el balance de la cuenta, usted puede encontrar esta opción en el menu "Finances" del portal del cliente.

[[File:AddFunds.png|thumb|none|784px|Add Funds]]

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Agregar fondos es un proceso sencillo, solo necesita dar un click en -> menú Finances -> opción Add funds desde su portal del cliente. Puede agregar fondos utilizando alguna de las diferentes opciones que se ofrecen: PayPal, tarjeta de credito ( cuyo monto mínimo es $15 USD ), o Bank Wire ( contacte al personal de VoIP.ms para obtener más detalles acerca de este método de pago).
Para monitorear sus gastos y checar balance de su cuenta visite esta [[Calculando mis gastos|liga]].
.
Recuerde que con VoIP.ms, las llamadas entrantes y salientes se cobran por separado. Por ejemplo, si usted compra un numero DID de USA( para recibir llamadas ), el costo puede ser de $0.85 USD mensuales y una tarifa al por minuto de $0.009 USD. Todas las llamadas salientes se cobrarán de acuerdo a nuestras tarifas ([http://voip.ms/rates.php Termination Rates]) bajo un esquema al por minuto, por ejemplo: la tarifa por minuto para llamar a USA48 es de $0.0100 USD.

== Llamadas Salientes ==

Si usted quiere usar los servicios de VoIP.ms para realizar llamadas salientes, necesitará primero tener un "dispositivo" o un "softphone". Es posible usar los servicios de VoIP.ms con cualquier dispositivo( no bloqueado ), servidor o aplicación que soporte los protocolos "SIP" o "IAX2", incluidos softphones gratuitos como X-lite, PBX asterisk, Voxalot, Trixbox, adaptadores ATA, teléfonos VoIP, hardware CISCO, Switches VOIP, etc. En resumidas cuentas, cualquier hardware o software que soporte alguno de los protocolos que ofrece VoIP.ms.

Una vez que decida que dispositivo utilizará con el servicio, será necesario configurar dicho dispositivo con las credenciales de la cuenta que usted recibió al momento de crearla. Su número de cuenta principal SIP consiste en un número de 6 dígitos, por ejemplo: 100000, junto con una contraseña SIP que usted eligió cuando creó la cuenta ( puede cambiar esta contraseña más tarde en la [[Configuraciones de la Cuenta (Account Settings)| Configuración de su cuenta]]. Ademas de eso, usted necesita el nombre de alguno de los servidores de VoIP.ms para obtener registro, para decidir cuál de ellos utilizar es recomendable checar esta [[Elegir servidor|liga]]. Incluso podría enviar algunos "pings" para determinar cuál de ellos le proporciona una menor latencia, o si usted lo decide puede usar el servidor más cercano a su ubicación.

[[File:Info11.jpg|thumb|none|784px|Account Information]]

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Para mas información acerca de los ejemplos de configuración de diferentes dispositivos, puede visitar esta [[Devices|liga]].

Una vez que su cuenta esta registrada, marque "4443" para alcanzar "echo test", esta herramienta es útil para probar la calidad de su línea, ya que es el único número al que puede llamar sin agregar fondos. Una vez que usted decida agregar fondos a su cuenta, podrá llamar a cualquier destino, no olvide revisar los costos al por minuto de acuerdo al destino en el portal del cliente, además de obtener tener una mejor idea del [[Costo del servicio|Costo del servicio]].

== Llamadas entrantes ==

Para recibir llamadas entrantes mediante el sistema de VoIP.ms, necesita tener un número DID en su cuenta, el cual funciona como cualquier número regular para recibir llamadas entrantes, además, es posible recibir llamadas entrantes mediante [[Dirección URI (SIP URI)|SIP URI]] usando una [[Sub Cuentas (Sub Accounts)|Sub Cuenta]], tome en cuenta que para usar la mayoría de las opciones incluidas en el portal será necesario tener un número DID.

Puede transferir a su cuenta voip.ms un número existente que actualmente tenga con otro poveedor o comprar un nuevo número de nuestro stock. Usted dispone de diferentes opciones para tener un número DID, puede tener desde un número local de USA o Canadá hasta un número Internacional o un número de lada sin costo. Además, se encuentran otras alternativas como los números Virtuales y iNums. Para ordenar el número que mejor se adapte a sus necesidades, lo puede hacer desde el portal del cliente -> menú DID numbers -> opción order DID.

[[File:Order.JPG|thumb|none|500px|Order DID]]
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Una vez que ordene un número DID, lo puede ver en su cuenta desde el portal del cliente -> menú DID numbers -> opción Manage DID, en esta sección tendrá las diferentes opciones para configurar su número. Para asegurarse de que recibió un número que funciona correctamente, se sugiere rutearlo a la opción System -> echo test

Aqui puede encontrar algunas de las opciones de ruteo para un número DID:

* Main or [[Sub Cuentas (Sub Accounts)|Sub Cuenta]] : Esto significa que usted puede rutear su número a un dispositivo o equipo previamente registrado o configurado.

* [[Recepcionista Digital|Recepcionista Digital]] : Esta opción es útil para administrar las llamadas entrantes de su negocio y dirigirlas hacia determinada extensión, persona o departamento.

* [[Llamadas en Cola (Calling Queues)|Llamadas en Cola (Calling Queues)]] : Los miembros de la cola, al estar logueados, pueden contestar las llamadas entrantes

* [[Condiciones de Tiempo (Time Conditions)|Condiciones de Tiempo (Time Conditions)]] : Es posible establecer diferentes destinos para las llamadas entrantes de sus números de acuerdo a la hora del día en el que se recibe.

* [[Desvío de Llamadas (Call Forwarding)|Desvío de Llamadas (Call Forwarding)]] : De esta forma, usted puede re-dirigir sus llamadas a un número externo como una linea local o un celular

* [[Dirección URI (SIP URI)]Dirección URI (SIP URI)] : Para llamar a otra persona o compañía mediante una dirección SIP URI.

* [[Timbre En Grupo (Ring Groups)|Timbre En Grupo (Ring Groups)]] : Al tener múltiples dispositivos registrados con diferentes subcuentas, es posible dirigir las llamadas de alguno de sus números a estos dispositivos mediante esta función

* [[Grabaciones (Recordings)|Grabaciones (Recordings)]] : Reproduce una grabación para las personas que llamen a su número DID. Tenga en cuenta que la llamada se desconectará después de que la grabación haya terminado.

* [[Regreso de LLamada (Callback)|Regreso de LLamada (Callback)]] : Para realizar llamadas desde un teléfono externo mediante el sistema de VoIP.ms.

* [[Acceso directo al sistema interno (DISA)|Acceso directo al sistema interno (DISA)]] : Parecido a la opción anterior, pero aquí deberá ingresar un PIN para realizar llamadas

* [[Buzón de voz (Voicemail)|Buzón de voz (Voicemail)]] : Las personas que le llamen, tendrán la opción de dejar un mensaje en su buzon de voz, para que posteriormente usted los contacte, si así lo desea.

* '''System''' : Principalmente utilizado para echo test y otras respuestas del sistema, como el tono de ocupado o un mensaje para las llamadas no deseadas.

== SubCuentas ==

La mayoría de los usuarios de VoIP.ms que han creado una cuenta recientemente, probablemente tengan preguntas acerca de [[Sub Cuentas (Sub Accounts)|Sub Cuentas (Sub Accounts)]], ¿Para que son?, ¿cual es el beneficio de crearlas?

[[File:SubAccounts.png|thumb|none|784px|Sub Accounts]]
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Con VoIP.ms usted tiene la posibilidad de registrar más de un dispositivo ( softphone o sistema ) de manera simultanea, ésto se puede lograr usando las subcuentas. Talvez, usted desee crear subcuentas a sus familiares o amigos para que ellos también puedan usar el servicio de VoIP.ms y así realizar y recibir llamadas desde su cuenta.

Puede crear tantas subcuentas como sean necesarias, de acuerdo al número de dispositivos que quiera registrar al mismo tiempo. Por favor, tenga en cuenta, que las subcuentas no son necesarias si sus dispositivos no están registrados simultáneamente. Por ejemplo, usted no necesitaría sub cuentas si usa su cuenta principal en su PC en casa, luego apaga la PC e inicia sesión desde el trabajo con algún otro software o dispositivo, ya que las cuentas no estarían registradas en el servidor al mismo tiempo.

Registrar más de un dispositivo de manera simultanea no es recomendable por que pueden causar algunos problemas de conectividad.

Cuando crea subcuentas, es como tener cuentas adicionales pero administradas todas juntas. Por cada subcuenta, obtendrá una ID de usuario y una contraseña diferente para registrar un dispositivo.

'''Por ejemplo :'''
'''Subcuenta : 100000_101'''
'''Password : la contraseña que estableció al momento de crear la subcuenta'''
'''Sip Proxy : atlanta.voip.ms'''

Las subcuentas son útiles para establecer comunicación entre 2 dispositivos, digamos que usted tiene 2 subcuentas registradas en el servidor: Atlanta.voip.ms, usted podrá realizar llamadas directamente entre una subcuenta y otra mediante las extensiones "internas", estas extensiones pueden ser configuradas durante el proceso de creación de las subcuentas, tome en cuenta que dichas extensiones también podrán ser utilizadas como direcciones SIP URI.

Usted siempre puede revisar el estado de registro de su dispositivo( ATA o softphone ) desde el portal del cliente -> menú principal -> opción portal home, ahi en la sección de registration status usted podrá confirmar cuales de sus sub cuentas están registradas correctamente.

== Llamadas Internacionales ==

Cualquier llamada fuera de los 48 estados oficiales de Estados Unidos o Canadá se considera como una llamada internacional, por esta razón, debe marcar utilizando el siguiente código:
00 ó 011 + código del país + número de teléfono

Para obtener códigos de marcación adicionales, consulte el artículo [[Códigos de Marcado|Códigos de Marcado]]

Si usted pretende utilizar los servicios de VoIP.ms para realizar llamadas "internacionales", tome en cuenta que por medidas de seguridad este tipo de llamadas están inhabilitadas por defecto en las configuraciones de su cuenta. Usted puede habilitar esta opción desde el portal del cliente -> menú principal ( main menu ) -> opción account settings -> pestaña Account Restrictions. Si usted decidiera habilitar este tipo de llamadas, le recomendamos utilizar las siguientes restricciones que se encuentran en la misma pestaña Account Settings.

* '''Allow International calls''' : Permitir llamadas Internacionales

* '''Max. Call Time for US48/Canadian Calls''' : Máximo tiempo permitido para una llamada hacia un numero que este incluido en nuestras rutas US48/CAN, una vez excedido este tiempo la llamada será terminada.

* '''Max. Call Time for International Calls''' : Máximo tiempo permitido para una llamada internacional, una vez excedido el tiempo la llamada será terminada.

* '''International Amount Restriction''' : Máximo monto permitido por minuto para llamadas internacionales, si el costo por minuto del destino al que usted desea llamar excede este valor, la llamada no se realizará.

VoIP.ms ofrece actualmente nuevas opciones que pueden ayudar con la seguridad, puede consultar desde el portal del cliente -> main menu -> account settings -> account restrictions. Estos ajustes definen las restricciones que el sitema utilizará cuando realice llamadas a números de USA48, Canadá o números internacionales.

[[File:rest11.jpg|thumb|none|784px|Account Restrictions]]
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NOTA: Por medidas de seguridad, algunos destinos están bloqueados por defecto, revise esta liga: Desbloqueo de todas las llamadas internacionales , Diríjase a main menu -> portal home -> Allowed International destinations, haga clic en el enlace [[Desbloqueo de todas las llamadas internacionales]], y siga las instrucciones.

== Registro de los detalles de las llamadas ==

En su cuenta VoIP.ms, usted encontrará una sección llamada [[Registro de Llamadas (CDR)|CDR and Reports]], ahí usted podrá ver los registros de todas las llamadas realizadas y recibidas en su cuenta, los diferentes filtros disponibles le permiten una manera más sencilla para encontrar una llamada en especifico.

[[File:cdr11.jpg|thumb|none|784px|Call Detail Records (CDR)]]
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* Búsqueda por un rango especifico de "Fechas"

* Filtrado de llamadas de acuerdo al estado de las mismas: Contestadas, no contestadas, ocupado, falladas.

* Filtrado de llamadas por tipo: salientes, entrantes, hacia un número DID en especifico.

* Filtrado de llamadas por tipo cobro: llamadas cobradas, llamadas gratuitas.

* Filtrado de llamadas por cuenta: cuenta principal o subcuentas.

* Zona Horaria (De esta manera puede ver la hora de las llamadas de acuerdo a la zona horaria que aplica en su localidad )

== Personal de Soporte ==

Si en algún momento usted necesitará contactar al personal de soporte de VoIP.ms, lo puede hacer desde el portal del cliente -> menu support, desde ahí podrá crear un ticket e inclusive revisar los tickets previamente creados desde su cuenta. Además podrá encontrar los siguientes links:

[[Welcome|Bienvenido]]( página de bienvenida del wiki de VoIP.ms )

[[Devices| Dispositivos]]( página de los ejemplos de configuración en la sección de wiki )

[[File:chat17.jpg|thumb|none|784px|Live Chat]]
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El personal de soporte le puede ayudar a solucionar diferentes situaciones, pero además puede encontrar información útil aqui:

* [[Problemas de audio|Problemas de audio]]( En caso de que tenga problemas con la calidad de las llamadas )

* [[Solucion de Problemas con su DID|Solucion de Problemas con su DID]] ( Para diagnosticar algún problema con su número DID )

* [[Problemas Para Iniciar Sesión|Problemas Para Iniciar Sesión]] ( para solucionar problemas de acceso al portal )

* [[Seguridad en PBX|Seguridad en PBX]] ( sugerencias para mejorar la seguridad en su sistema PBX )

* [[Solución de problemas de llamadas salientes|Solución de problemas de llamadas salientes]] ( Para problemas con llamadas salientes )

Desde la sección llamada "Feature Requests" usted puede realizar sus recomendaciones y sugerencias para generar mejores opciones en el sistema de VoIP.ms.

== Sección de las Tarifas ==

Si usted quiere saber la tarifa por minuto que aplicará para un determinado destino, puede obtener esta información desde el portal del cliente -> menú rates -> opción check rates online, ahí también encontrará algunos campos que le permiten agilizar esta búsqueda:

[[File:rates11.jpg|thumb|none|784px|International Rates]]
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* '''Búsqueda por país''' (revise la lista de precios por país).

* '''Prefijo de Número telefónico''' ( revise la tarifa de un número en particular).

* [[Value vs. Premium|Value vs. Premium]] (Puede ver las tarifas por destino de acuerdo a la ruta: value o premium)

== Notas ==

'''Las llamadas ENTRANTES y SALIENTES''' se cobran y manejan '''POR SEPARADO''' con nuestro servicio. Las llamadas entrantes se cobran según el plan del número DID y las llamadas salientes se cobrarán siempre al por minuto según el destino de las llamadas.

También tenga en cuenta que no es necesario tener un número DID para comenzar a realizar llamadas ya que un número DID está destinado a recibir llamadas únicamente.

'''Para recibir las llamadas correctamente''', el servidor de registro y el POP del número DID deben coincidir, es decir, si su dispositivo / softphone está registrado en dallas.voip.ms el POP del DID también debe tener el Punto de Presencia de Dallas.

[[category:Guías]]

Getting Started

2021-03-23T15:40:45Z

Mauro:

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[[File:Logo-VoIPms-light.png|left|400px]]
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{| class="wikitable"
|+
|-
! Article en Français !! Artículo en Español
|-
| [https://wiki.voip.ms/article/Mise_en_Marche Français] || [https://wiki.voip.ms/article/Primeros_Pasos Español]
|}

This guide is intended to help you start using your [http://voip.ms Voip.ms] account and its features. A recently created account can come with a lot of questions. Most of those questions can be answered by the Support staff via Live chat or the Ticket system. However, the information here is intended to make your first contact with VoIP easier and clearer.

== Tutorial Video ==

[[Image:GetStartedThumbnail.png|200px|link=https://www.youtube.com/watch?v=XjD81OkSOGU]]

== Create your account ==

You can create an account for free by clicking [https://www.voip.ms/signup.php Here]. You will access the sign-up page, please enter the information required and click ''submit''. After signing up you will receive an email to confirm your email address and validate the account.

Remember to check in your spam/junk folder to ensure you received the email. If you did not, please contact VoIP.ms by email at: support@voip.ms

[[File:Sign_Up.jpg|thumb|none|784px|Add Funds]]

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'''Once signed up, determine what you would like to accomplish with your new account. When you log in for the first time, we recommend to check the [http://wiki.voip.ms/article/Account_Settings Account Settings] so you can get familiar with the options offered to configure your line.'''

== Add Funds ==

In order to use VoIP.ms services to send and receive calls, you will need to add funds to your account. This is a prepaid service and the amount added will be reflected in the '''Account Balance''' found in the [https://www.voip.ms/m/balance.php Finances] section in the customer portal.

[[File:AddFunds.png|thumb|none|784px|Add Funds]]
<br clear="all" />

Adding funds is a simple procedure, you just need to click [https://www.voip.ms/m/payment.php Finances --> Add Funds] in the customer portal. Make a payment with any of the available payment options such as Credit Card or PayPal, the minimum payment is '''$15 USD'''. Contact VoIP.ms staff for the information needed to send a payment using Bank Wire if required.

To monitor your expenses and account balance, please visit the following article [http://wiki.voip.ms/article/Calculating_my_expenses calculate my expenses]

Remember that with VoIP.ms, the inbound and outbound calls are charged separately. For example, if you take a US DID number (for receiving calls), the cost can be $0.85 a month and $0.009 per minute. And all your outbound will still be charged as per the [http://voip.ms/rates.php Termination Rates]. '''For example, calling a US48 number is $0.009 per minute.'''

== Outgoing calls ==

If you intend to use VoIP.ms services for the purpose of making calls, you will first need a '''device or softphone'''. You can use any type of software or device (unlocked) which support '''SIP or IAX2''' protocols. Including free softphones such as [http://wiki.voip.ms/article/Softphones#Zoiper_Classic Zoiper], free open source [[PBXs]], VoIP [[Devices|ATAs and VoIP phones]], Hardware CISCO VoIP switches, etc. Basically, any hardware or software that supports one of the two protocols offered.

Once you have defined what equipment you will use with the service, you need to configure it using the credentials provided when creating the account. You will receive a '''SIP/IAX Main Username''' that consists of six digits (e.g. 100000), along with a password you chose when you created the account (you can change this password later in your [https://www.voip.ms/m/settings.php Account Settings]). The additional information missing now is the VoIP.ms server you will use to register. We recommend sending a ping to [http://wiki.voip.ms/article/Choosing_Server choose a server] so you can have an idea which server to use. Try to use the one with the less latency (lower ping), or the one closest to your geographical location.

[[File:Info11.jpg|thumb|none|784px|Account Information]]

<br clear="all" />

Refer also to the [http://wiki.voip.ms/article/Devices configurations samples] here on the Wiki to find specific instructions for your equipment.

Once registered, first dial '''4443''' to reach the "echo test". This tool is useful to test the quality of your line since it is the only number you can call without adding funds. After doing this, you are ready to add funds and call the destinations you want, remember to check the Termination rates inside the customer portal so you can get a better idea about the [http://wiki.voip.ms/article/Service_Cost service cost].

== Incoming calls ==

To receive calls with VoIP.ms, you will need a '''DID number''' which works just as any regular telephone number used to receive incoming calls. You can still receive calls via [http://wiki.voip.ms/article/SIP_URI Sip Uri] using a [http://wiki.voip.ms/article/Sub_Accounts sub account], but to use most of the features included in the portal it will be necessary to get a DID number for the incoming calls and different options that are offered.

You can either [[Porting a Number|port]] an existing number that you own over to VoIP.ms or purchase a new number from our stock. There are several DID options available, from local US & Canadian numbers to International or Toll free numbers. You will also find some interesting alternatives such as Virtual numbers or iNUMs. Refer to [https://www.voip.ms/m/dids.php DID Numbers >> Order DID(s)] in the customer portal and pick the number that suits best your needs.

[[File:Order.JPG|thumb|none|500px|Order DID]]
<br clear="all" />

Once you have a DID number you will be able to see it in your portal listed under DID Numbers >>[https://www.voip.ms/m/managedid.php Manage DID(s)]. In [[Manage DID|this section]] you will have different options to configure your number. The first recommended test is to route it to '''System >> Echo test''' to make sure you received a number that is working properly.

You can find different options for the Routing of your number:

* Main or [http://wiki.voip.ms/article/Sub_Accounts sub account] (this basically means you can route your number to a device or equipment previously configured and registered).

* [http://wiki.voip.ms/article/Digital_Receptionist_(IVR) IVR (digital receptionist)] this is useful to handle your calls at your business and direct the callers to a specific department.

* [http://wiki.voip.ms/article/Calling_Queues Calling queue] (members that can log in and answer your calls).

* [http://wiki.voip.ms/article/Time_Conditions Time conditions] (specify different routes for your numbers depending on the time of the day).

* [http://wiki.voip.ms/article/Call_Forwarding Call forwarding] (route your calls to an external number such a landline or cellphone number).

* [http://wiki.voip.ms/article/SIP_URI Sip Uri], to call another person or company via SIP.

* [http://wiki.voip.ms/article/Ring_Groups Ring group] (you have different devices registered, using different sub accounts, you can route one number to all of them with this feature).

* [http://wiki.voip.ms/article/Recordings Recordings], play a simple message for people to hear when calling your number. Please note the call will disconnect after the recording has finished playing.

* [http://wiki.voip.ms/article/Callback Call back], (you can use for example your cellphone to make calls via VoIP.ms service).

* [http://wiki.voip.ms/article/DISA DISA] (works similarly to call back but instead you have to enter a PIN number first).

* [http://wiki.voip.ms/article/Voicemail Voicemail] (give people calling you an option to leave a message, so you can contact them later).

* '''System (Used for echo test and other responses, such a busy tone or message for unwanted calls).'''

== Sub Accounts ==

Most users new to VoIP.ms will probably have questions regarding [http://wiki.voip.ms/article/Sub_Accounts sub accounts], please continue reading to learn what they are and the benefit to using them.

[[File:SubAccounts.png|thumb|none|784px|Sub Accounts]]
<br clear="all" />

With VoIP.ms you have the possibility to register more than one device (e.g., softphone or system) '''simultaneously''' and this can be achieved by using the '''sub accounts''' feature. Or maybe you wish to create sub accounts for your relatives/friends so they can use the VoIP.ms service to place and receive calls under your account.

You can create as many sub accounts as you need, depending on how many devices you want to register at the same time. Please keep in mind that you do not need sub accounts if your devices are not logged in simultaneously. For instance, you wouldn't need sub accounts if you use your main account at your PC at home, then turn the PC off, and log in from work and use a software phone, since the accounts are not registered to the server at the same time.

Registering multiple devices only with the main account is not recommended since this can cause some connectivity issues.

When you create sub accounts, it is like having additional accounts but managed all together. You will get a different User ID and password to register a device:

'''For example:'''
'''Sub Account : 100000_101'''
'''Password: You set it on the sub account'''
'''Sip Proxy : atlanta.voip.ms'''

Sub accounts are also useful to communicate between devices, let say you have 2 sub accounts registered to the Atlanta server, you can make calls directly between the sub accounts using an '''internal extension''' that can be configured in the sub account settings. This same internal extension can also be used as a [[SIP URI]].

You can always check the registration status of your ATA / softphones on the '''Portal Home Page ([https://www.voip.ms/m/ Main Menu >> Portal Home])''' found near the bottom of the page.

== International calls ==

Any call outside of USA48 or Canada is treated as an '''International call''', for that reason you need to dial using the following code:

00 or 011 + country code + phone number

For additional dialing codes check the [http://wiki.voip.ms/article/Dialing_Codes Dialing Codes] article.

If you intend to use VoIP.ms services for '''International calls''', note these are disabled by default. This is done to enforce the security on your account and if you decide to enable them, VoIP.ms recommends highly to use the restrictions in the [http://wiki.voip.ms/article/Account_Settings Account Settings]:

* Allow International calls.

* Max. Call Time for US48/Canadian Calls (exceeding the time will cause the call to hang up automatically).

* Max. Call Time for International Calls (Max. Call Time for International Calls).

* International Amount Restriction (If exceeds the maximum amount, the call will not be connected).

VoIP.ms currently offers new options that can help for the security, you can refer to '''[https://www.voip.ms/m/settings.php Main Menu >> Account Settings >> Account Restrictions].''' These settings define the restrictions the system will use when you place calls to either USA48, Canada or International Numbers.

[[File:rest11.jpg|thumb|none|784px|Account Restrictions]]
<br clear="all" />

Note also that some destinations are locked by default for all accounts. In order to have [http://wiki.voip.ms/article/Unlocking_All_International_Destinations full access to all international destinations], refer to Main Menu >> Portal Home >> Allowed International destinations, click on the link [https://www.voip.ms/m/verification.php "Some Areas Locked (Click for details)"] and follow the instructions.

== Call detail records ==

In your VoIP.ms account you will notice a section called "CDR and Reports", here you can find your Call Detail Records which basically represent all the calls made and received in your account. You can search for a specific call by using the different filters that are available.

[[File:cdr11.jpg|thumb|none|784px|Call Detail Records (CDR)]]
<br clear="all" />

* Search for a specific range of '''dates'''

* Filter calls by '''disposition''' in the CDR (answered, no answered, busy, failed).

* Filter calls by '''type''' (Outgoing, Incoming, calls to specific DID number).

* Filter calls by '''call billing''' (billed calls, free calls).

* Filter calls by '''account''' (main or sub account calls).

* '''Time Zone''' (Set the time zone to adjust your CDR to your local time).

== Support staff ==

If need to [http://voip.ms/contactus.php contact] the VoIP.ms staff, there is a section named 'Support' in the customer portal. You can open tickets from there and also see all the tickets you have previously created. There is also a link for the [http://wiki.voip.ms/article/Devices configuration samples] and a direct link to the [http://wiki.voip.ms/article/Welcome Wiki page].

[[File:chat17.jpg|thumb|none|784px|Live Chat]]
<br clear="all" />

Support staff can help you with the trouble shooting in different scenarios, but you can also find useful information here:

* [[How_To_Guides#How_To_Add_Funds| Some Getting Started 'How To' Guides]]

* [[How_it_works| An overview on how the service works]]

* [http://wiki.voip.ms/article/Call_quality_issues Call quality issues]

* [http://wiki.voip.ms/article/DID_Troubleshooting DID trouble shooting]

* [http://wiki.voip.ms/article/Problems_logging_in Problems logging in]

* Tips for your [http://wiki.voip.ms/article/PBX_Security PBX security]

* [http://wiki.voip.ms/article/Troubleshooting_Outgoing_Calls Trouble shooting outgoing calls]

The section called '''"Feature Request"''' is the one you can use to report things that are not currently available with VoIP.ms service but you think will be good to have them in the future, like specific tools in the portal.

== Rates section ==

You want to know how much you will pay when calling a specific number? You need to visit the [https://www.voip.ms/m/rates/rates.php Rates section] in the customer portal. There are fields that will make your search quicker:

[[File:rates11.jpg|thumb|none|784px|International Rates]]
<br clear="all" />

* '''Search country''' (make your search by looking at the different rates for a specific country in the termination list).

* '''Phone number prefix''' (look for a specific number, you will get the rate for the prefix that matches the number).

* [http://wiki.voip.ms/article/Value_vs_Premium Value & Premium] (you can choose between Value and Premium route for the Canada, International & Toll free calls). The US route is only available on the Premium route with a rate of $0.01 (one cent) per minute.

== Notes ==

'''INCOMING and OUTGOING calls are charged and handled SEPARATELY with our service. The incoming calls are charged depending on the plan of the DID
'''and the outgoing calls will be charged always per minute and according to the destination of the calls.

Also note that it is not necessary to have a DID number to start placing outgoing calls, a DID number is meant to receive calls only.

'''In order to receive calls properly the registration server and the POP of the DID number must match, i.e., if your device/softphone is registered to dallas.voip.ms
'''the POP of the DID must have the Point of Presence of Dallas as well.

Grandstream UCM6200

2020-12-24T16:42:56Z

Mauro:

[[File:Ucm6202.jpg|300px|thumb|left]]

Designed to provide a centralized solution for the communication needs of businesses, the UCM6200 series IP PBX appliance combines enterprise-grade voice, video, data, and mobility features in an easy-to-manage solution. This IP PBX series allows businesses to unify multiple communication technologies, such as voice, video calling, video conferencing, video surveillance, data tools, mobility options, and facility access management onto one common network that can be managed and/or accessed remotely. The secure and reliable UCM6200 series delivers enterprise-grade features without any licensing fees, costs-per-feature or recurring fees.

'''Website''': [http://www.grandstream.com/products/ip-pbxs/ucm-series-ip-pbxs/product/ucm6200-series UCM 6200 Series]

<br clear="all" />

== Login into your device ==

#Connect a computer to the same network as the UCM6202
#Ensure the UCM 6202 is properly powered on and displays the IP address on the LCD screen
#Open a web browser on the computer and enter the displayed IP address into the search bar in the following format '''''http(s)://ipaddress:portnumber'''''. The default protocol is HTTPS and the default port number is 8089.
#The Web portal should be shown (see figure below). Enter the default administrator credentials:

'''Username''': admin
'''Password''': admin

[[File:UCM main screnn.png|thumb|none|550px|Click to enlarge]]

== Creating a trunk ==
To create a new VoIP.ms trunk head to '''Extension/Trunk>> VoIP trunks''', from the left panel. In this section, you can choose to create a SIP or an IAX trunk.

[[File:UCM-trunk.png|thumb|none|550px|Click to enlarge]]

===SIP Trunk===

To create a SIP trunk you only need to fill some basic information.

* '''''Type''''': Register SIP trunk
* '''''Provider Name''''': VoIP.ms (any name can be used)
* '''''Host Name''''': Type any of our [http://wiki.voip.ms/article/Choosing_Server servers], i.e. toronto5.voip.ms
* '''''Keep trunk CID''''': Enable it, if you want the trunk to send its own CID number or disable it if your extensions are going to send their own CID number
* '''''Username''''': 100000 (replace with your main VoIP.ms account number or sub-account name)
* '''''Password''''': ********* (replace with your main SIP/IAX password or sub-account password)

'''Note:''' Bear in mind to use the same VoIP server your VoIP number is using.
You can check what VoIP server is your VoIP number using, from your VoIP.ms customer portal
at '''DID Numbers>> Manage DIDs''', under '''POP''' column.
You can choose any server you want, as long as the one in your portal and the one in this field matches,
otherwise, incoming calls won't ring.

Click '''Save''' button, do not click '''Apply Changes''' yet.

[[File:New SIP trunk.png|thumb|none|800px|Click to enlarge]]

====Extra SIP settings====

Once your basic SIP trunk has been created we will proceed to improve some settings on it, click on the edit icon for your trunk.

'''SIP Headers'''

To send the SIP '''"FROM"''' header as we require, make:

'''"fromuser:"''' 100000 (replace with your main VoIP.ms account number or sub-account name)

[[File:SIP-basic settings.png|thumb|none|800px|Click to enlarge]]

'''Audio codecs'''

Click at "Advanced Settings" and at "Codec Preference" use only the supported codecs by VoIP.ms: '''G.729''', '''PCMU''' & '''GSM''', in this order.

'''Trunk Caller ID'''

You can use '''PAI (P-Asserted-Identity)''' header if you want to send the CID name & number from your trunk, you only need to use this format:

'''"CallerIDName"<CallerIDNumber>'''

'''Note:''' Enabling the PAI header but leaving the field blank (null) will force the pbx to use the CID priority rules from Grandstream thus enabling you to passthorugh the CID from the extensions

'''Note:''' No Caller ID Number must be set at your VoIP.ms portal or it will override the one sent by your trunk

'''NAT Keep Alive'''

In order to avoid your modem closing your local SIP ports, enable:

'''*Enable Heartbeat Detection:''' Enabled<br>
'''*Heartbeat Frequency:''' 50

[[File:SIP advanced settings.png|thumb|none|550px|Click to enlarge]]

'''Finally''', click '''Save''' button and at this stage, you can also click on '''Apply Changes'''. Your trunk should be shown as '''Registered''' from your VoIP.ms dashboard, however, no calls will work until you set up your outbound and inbound routes.

====Call Encryption TLS/SRTP====
In order to use TLS along with your UCM please follow these steps:

'''1.''' Make sure your Main account or sub-account has '''"Encrypted SIP Traffic"''' enabled.

Bear in mind, if this setting is enabled and your device sends '''UDP/TCP''', '''RTP''' you will be rejected
with error code 488.

Enable this setting for the Main Account at '''Main Menu>> Account settings>> Advanced tab'''.

[[File:Mainacc encryp.png|thumb|none|600px|Click to enlarge]]

For a sub-account enable it at '''Sub accounts>> Manage sub-accounts''' by clicking on the orange icon with a pen and finally click at '''"Advanced Options (Click here to display)"'''.

[[File:Subacc encryp.png|thumb|none|600px|Click to enlarge]]

'''2.''' Now that your account/sub-account has this setting enabled, your device only needs to send '''TLS''' and '''SRTP'''.

Go to '''Extension/Trunk>> VoIP Trunks''' and click on "Edit trunk". In this section make sure you have the following settings:

'''*Host Name:''' toronto5.voip.ms:5061 (Use the same server your phone number is at, you can check it out from your customer portal at Manage DIDs section).

'''*Transport:''' TLS

'''Note:''' When using TLS is very important to specify the number of the server, in case the server's name doesn't have the number "1" included, you need to add it. Adding any of the SIP ports '''5061/5081/42873''' at the end of the '''Hostname''' is also required.

[[File:Ucm trunk tls.png|thumb|none|800px|Click to enlarge]]

Go to '''Advanced Settings''' and set '''SRTP''' to "Enabled and forced"

[[File:Ucm srtp.png|thumb|none|800px|Click to enlarge]]

===IAX2 Trunk===
To create an IAX2 trunk just fill the following information:

* '''''Type''''': Register IAX trunk
* '''''Provider Name''''': VoIP.ms (any name can be used)
* '''''Host Name''''': Type any of our [http://wiki.voip.ms/article/Choosing_Server servers], i.e. toronto5.voip.ms
* '''''Keep trunk CID''''': Enable it, if you want the trunk to send its own CID number or disable it if your extensions are going to send their own CID number
* '''''Caller ID''''': Type the CID number your trunk will be sending.
* '''''Caller ID Name''''': Type the name your trunk will be sending.
* '''''Username''''': 100000 (replace with your main VoIP.ms account number or sub-account name)
* '''''Password''''': ********* (replace with your main SIP/IAX password or sub-account password)

'''Note:''' Bear in mind to use the same VoIP server your VoIP number is using.
You can check what VoIP server is your VoIP number using, from your VoIP.ms customer portal
at '''DID Numbers>> Manage DIDs''', under '''POP''' column.
You can choose any server you want, as long as the one in your portal and the one in this field matches,
otherwise, incoming calls won't ring.

Click '''Save''' button, do not click '''Apply Changes''' yet.

[[File:IAX2 trunk.png|thumb|none|550px|Click to enlarge]]

====Extra IAX settings====
Once your basic IAX trunk has been created we will proceed to improve some settings on it, click on the edit icon for your trunk.

'''Audio codecs'''

Click at "Advanced Settings" and at "Codec Preference" use only the supported codecs by VoIP.ms: '''G.729''', '''PCMU''' & '''GSM''', in this order.

'''NAT Keep Alive'''

In order to avoid your modem closing your local ports, enable:

'''*Enable Heartbeat Detection:''' Enabled<br>
'''*Heartbeat Frequency:''' 50

[[File:IAX advanced settings.png|thumb|none|550px|Click to enlarge]]

'''Finally''', click '''Save''' button and at this stage, you can also click on '''Apply Changes'''. Your trunk should be shown as '''Registered''' from your VoIP.ms dashboard, however, no calls will work until you set up your outbound and inbound routes.

== Creating your outbound route ==
Outbound routes are the ones in charge of making match your dialing pattern and send your call through the proper trunk

'''VoIP.ms suggest to include the following patterns into your outbound route: '''

_1NXXXXXXXXX
_NXXXXXXXXX
_4XXX
_00.
_011.
_033.
_044.

All your different dial patterns must be prefixed by the character "_"

To create your outbound routes click on '''Extension/Trunk>> Outbound routes''', from the left panel and click on "Add"

[[File:Outbound routes general.png|thumb|none|550px|Click to enlarge]]

In this section you only need to fill the following fields:

* '''''Calling Rule Name''''': Any name you want for this route.
* '''''Pattern''''': The desired patter your callers need to dial. You can use the suggested dial pattern by VoIP.ms shown above.
* '''''Trunk''''': Choose your VoIP.ms trunk on where your call will be sent through.
* '''''Privilege Level''''': Choose the desired privilege your extensions must have to be able to use this route. If you use our dial pattern we suggest to set it to "International" since you will be able to place international calls. Bear in mind that an extension with an inferior privilege won't be able to use this route. You can hover over the field's name to gather more information.

[[File:Outbound routes setup.png|thumb|none|800px|Click to enlarge]]

Note: If you want to include a dial-out prefix, you can type it after the "_" character in your dial patterns. This number will need to be stripped off, you can do this by using the "Strip" field, you can choose how many digits you can strip after the "_" character.

<pre>
For example: If you want to use "9" to dial out, then your pattern will need to be _9NXXXXXXXXX
To strip off this number "9" when dialing out, you will need to set the "Strip" field to "1", this way only
one character (in this case number "9") will be stripped off.
</pre>

== Creating your inbound route ==
Thanks to your inbound routes you can use only one single trunk to receive all the incoming calls from all your phone numbers. This way you don't need to use more than one trunk for your phone numbers, inbound routes will receive all of them and route them into the proper destination in your UCM.

'''Note:''' We do not suggest using more than one VoIP.ms trunk on the same device.

From the left panel, head to '''Extension/Trunk>> Inbound Routes''' and click on "Add".

[[File:Inbound routes main.png|thumb|none|550px|Click to enlarge]]

In this section, fill the following fields:

*'''Trunks:''' Choose your VoIP.ms trunk, on where the incoming calls should be sent by us.
*'''Pattern:''' Type your VoIP.ms DID phone number exactly as it is shown under "Manage DIDs" section from your customer portal. This number should be prefixed by the character "_"
*'''Default Destination:''' Choose the default destination on where your PBX should send your incoming calls to. This could be an internal extension, an IVR, a ring group and so on; managed locally by your PBX.

[[File:Inbound routes setup.png|thumb|none|800px|Click to enlarge]]

'''Note''': Remember to send your phone number in the '''"TO"''' SIP header, this way your PBx will match it with your inbound routes. You can do this very easily by setting '''Device Type''' setting from '''"ATA adapter, IP phone or Softphone"''' to '''"IP PBX Server, Asterisk or Softswitch"'''.

You will find this setting from your VoIP.ms customer portal at '''Main Menu>> Account settings>> Inbound settings''', if you're using the main account or at '''Sub accounts>> Manage Sub accounts''' and by clicking on the orange icon with a pen, if you're using a sub-account

[[File:PBx setting VoIP panel.png|thumb|left|750px|Click to enlarge]]

[[File:PBx setting VoIP panel2.png|thumb|left|750px|Click to enlarge]]

SMS

2020-08-02T17:24:13Z

Mauro:

'''SMS Service'''

Short Message Service (SMS). This feature will allow you to send and receive messages with your DID Number (US and Canada DID Numbers Only).

Please note that this feature is for regular customer usage. '''No automation, telemarketing, bulk sending or receiving will be allowed.'''

'''Our SMS service is no longer in BETA. Starting August 1st 2020, SMS messages will be charged at a rate of $0.0075 per messages.'''

__TOC__

== Important information to know about the SMS Service ==

* The SMS Service is only available for US and Canadian local DID Numbers marked with the distinctive SMS Icon.

* 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.

* The SMS Service can only be used to send SMS Messages to Canadian and US 10 digit numbers at this time. We also cannot guarantee that International SMS will be properly received.

* VoIP.ms reserves the right, at its sole discretion, to limit or disable the SMS service from any account that may present marketing patterns / automation patterns / bulk sending or receiving.

* At this time we cannot guarantee 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.

* For '''Ported In Numbers''' compatible with the feature: The SMS Functionality will be available up to 48 hours after the porting process is marked as Completed.

If you have further questions don't hesitate to contact the Support Staff on the Live Chat or Ticket System.

== Identifying a SMS DID Number ==

The first thing you need to know is that this feature is only available for '''local US and Canadian numbers''' at the moment. You will note that some numbers have a little icon of a cellphone device, this indicates that the number supports SMS. Please note that not all locations support SMS at this time.

You can start ordering a DID Number from your portal >> DID Numbers >> [https://www.voip.ms/m/dids.php Order DID] >> Local Numbers >> (US or Canada) Numbers and you will be able to purchase your desired number, just make sure it supports the SMS feature.

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

== Configuring the SMS service ==

Once purchased the SMS Service needs to be activated on the DID settings, from your portal >> DID Numbers >> Manage DID >> Edit Selection >> SMS.

On that section you can activate and configure some forward options for the SMS service.

[[File:SMSSIPAccount.png]]

The first action to take is to mark the '''Short Message Service (SMS) service''', by enabling this field the SMS Service will be active and free until further notice.

* '''VoIP.ms SMS 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 Email Address''': By activating this option all the SMS Messages will be sent to the email address you configure in this field. 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.

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 [#USXXX] Message sent to 5555555555.

*'''SMS 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.

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.

*'''SMS 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.

The destination ("To" header field) of the SIP MESSAGE will be your account name, instead of the destination number of the original SMS.

*'''SMS 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:

'''{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
'''Example''': http: //mysite.com/sms.php?to={TO}&from={FROM}&message={MESSAGE}&id={ID}&date={TIMESTAMP}

*'''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.

== Send and Receive Messages (Web Interface)==

===SMS Message Center===

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===

[[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 (SIP Protocol) ==

'''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'''

=== 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 ===

*DID must have the '''"Short Message Service (SMS)"''' 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 ===

It is possible to configure a FreePBX system to send SMS, please refer [[SIP/SMS with FreePBX | here]] for detailed instructions.

Grandstream GXP2170

2020-06-23T14:14:03Z

Mauro:

[[File:Grandstream_GXP2170.png|300px|thumb|left|Grandstream GXP2170]]
<div style="width:100%;overflow:hidden;clear:both"></div>

The GXP2170 is a powerful High-End IP phone that is ideal for busy users who handle high call volumes. Receptionists, administrators, sales staff and other call-intensive rolls can enjoy efficiency by utilizing the GXP2170’s 12 line keys, 4.3 inch color display LCD and 48 digital, on-screen speed dial/BLF keys.

__TOC__

==Setting up a Grandstream GXP2170 with VOIP.ms==

These are the settings that I used to get the Grandstream GXP2170 working on voip.ms. I used a combination of guides here on the wiki and other resources. I am not sure that all these settins are needed, but this combination of settings got the phone working for me.

Note:
* If you want to use the BLF feature, you will need to contact voip.ms support and request that they activate it for the server that you are using.
* If you are going to be using an incoming DID number, you must ensure it's setup in the DID's settings to forward to the server you are going to be connecting to.
* These instructions setup encrypted SIP traffic. You will want to ensure that this is activated in your sub-account advanced settings.
* I am starting out with a phone running firmware 1.0.9.135 and starting from a clean factory default.

===What you need to get started===
To get started, you will need the following:
* A Grandstream GXP2170 IP Phone
* Valid login credentials for a voip.ms account or sub account
* The server you want to use (I'll be using denver1.voip.ms here)

You will also need to know the IP address if your phone. The phone by default will pull an IP address via DHCP. To check the IP address of your phone you:
# Press the center hat button
# Select "Status" (or press 2)
# Select "Network Status"
# Find the IP address the phone us using (I'll be using 10.1.10.137 in my example here)

=== Connecting to the phone===
Connect to the phone via browser of your choice. If this is the first time you have logged into the phone, you will be prompted to update the password. The default username/password are both admin.

[[File:GXP2170_Login.png|thumb|none|600px]]

[[File:GXP2170_PasswordUpdate.png|thumb|none|600px]]

==Updating Phone Settings==

The following are the settings that need to be updated.

===General Settings===

Go to Accounts -> Account 1 -> General Settings

[[File:GXP2170_GeneralSettingsGetTo.png|thumb|none|600px]]
[[File:GXP2170_GeneralSettings.png|thumb|none|600px]]

These are the settings changed from their default values:
*'''Account Name''' - Name that will be displayed for the account on the phone's screen
*'''SIP Server''' - Network address of the voip.ms SIP server you will connect to
*'''BLF Server''' - Network address of the BLF server you will connect to (This is optional, but if used should be the same as your SIP Server)
*'''SIP User ID''' - Username from your voip.ms Sub Accounts page
*'''Authenticate ID''' - Username from your voip.ms Sub Accounts page (same as SIP User ID)
*'''Authenticate Password''' - Password from your voip.ms Sub Accounts page
*'''Name''' - Caller ID Display Name (Enter it here, but it will likely be overwritten by settings in your account)

===SIP Basic Settings===

Go to Accounts -> Account 1 -> SIP Settings -> Basic Settings

[[File:GXP2170_SipBasicSettingsGetTo.png|thumb|none|600px]]
[[File:GXP2170_SipBasicSettings.png|thumb|none|600px]]

These are the settings changed from their default values:
*'''Register Expiration''' - Set to "5" *minutes)
*'''Enable OPTIONS Keep Alive''' - Select "Yes"
*'''SIP Transport''' - Select "TLS/TCP" (This is important for Secure SIP to work)
*'''SIP Listening Mode''' - Select "Dual(Secured)" (This is important for Secure SIP to work)
*'''Use Actual Ephemeral Port in Contact with TCP/TLS''' - Select "Yes" (This is important for Secure SIP to work)
*'''SUBSCRIBE for MWI''' - Select "Yes"
*'''SUBSCRIBE for Registration''' - Select "Yes"

===SIP Custom Headers===

Go to Accounts -> Account 1 -> SIP Settings -> Custom SIP Headers

[[File:GXP2170_SipCustomHeadersGetTo.png|thumb|none|600px]]
[[File:GXP2170_SipCustomHeaders.png|thumb|none|600px]]

These are the settings changed from their default values:
*'''Use Privacy Header''' - Select "Yes"
*'''Use P-Preferred-Identity Header''' - Select "Yes"
*'''Use X-Grandstream-PBX Header''' - Select "No"
*'''Use P-Access-Network-Info Header''' - Select "No"
*'''Use P-Emergency-Info Header''' - Select "No"

===SIP Audio Settings===

Go to Accounts -> Account 1 -> Audio Settings

[[File:GXP2170_SipAudioSettingsGetTo.png|thumb|none|600px]]

These are the settings changed from their default values:
*'''Preferred Vocoder - choice 1''' - Select "PCMU"
*'''Preferred Vocoder - choices 2 to 8''' - are mostly irrelevant however you can choose g.729a, GSM or G.722. Make sure you only select those enabled on your account or subaccount settings to avoid issues like packet fragmentation.

'''Additional Note'''

*'''SRTP mode''' - should be disabled if encryption isn't enabled on your account advanced settings and you're not using TLS.

[[File:GXP2170_SipAudioSettings.png|thumb|none|600px]]

Messagerie vocale

2019-12-12T14:58:32Z

Mauro:

VoIP.ms dispose d'une fonction de messagerie vocale de haut niveau, optionnelle, vous offrant la possibilité de transférer vos messages vers votre adresse de courriel sous forme de pièce jointe.

Afin d'utiliser la fonction de messagerie vocale avec ([https://www.voip.ms/ voip.ms]) vous devez d'abord créer une entrée vocale, puis, ''associer à l'un de vos DIDs ou de vos comptes.

== Création de la Messagerie Vocale ==

Dans votre portail client, allez à Numéros DID > Messagerie Vocale

Vous allez voir le tableau qui suit:

[[File:MessagerieVocaleCreate.png|thumb|none|900px]]

Détails des différentes options pour créer votre messagerie vocale:
*'''Numéro de messagerie vocale''': Vous pouvez définir jusqu'à 99999 comptes de messagerie vocale. Par exemple: 1xxxx-1 ou 1xxxx-54321.

*'''Nom''': Ceci peut être utilisé comme simple note ou description afin de facilement identifier vos comptes de messagerie vocale.

*'''Mot de Passe''': Le mot de passe est utilisé pour accéder aux options de votre messagerie vocale, telles l'écoute de vos messages et l'enregistrement votre message d'accueil.

*'''Ignorer l'invite de mot de passe''': Si la valeur est à OUI et que vous composez *97 à partir d'un compte associé à cette messagerie vocale, le système ne vous demandera pas votre mot de passe.

*'''Enregistrement de message non disponible''': Vous pouvez choisir un enregistrement que vous avez téléchargé en avant ou les valeurs par défaut du système.

*'''Courriel''': Si ce champ contient une valeur, la boîte aux lettres enverra donc un avis au courriel inscrit dans ce champ, chaque fois que vous recevrez un nouveau message.

*'''Joindre message au courriel''': Si la valeur est à OUI, la messagerie vocale joindra un fichier .WAV contenant le nouveau message à chaque fois qu'il envoie un avis par courriel.

*'''Supprimer message''': Si la valeur est à OUI, la messagerie vocale supprimera automatiquement le nouveau message, suite à l'envoi du courriel d'avis avec la pièce jointe.

*'''Format de la pièce jointe''': Format du fichier qui sera joint dans le message courriel.

*'''Langue de messagerie vocale''': Cette option définit la langue que l'appelant et vous entendez lorsque les instructions ou les menus sont joués par la messagerie vocale.

*'''Dicter les instructions''': Si la valeur est à OUI, l'appelant entendra les instructions sur la façon de laisser un message dans votre messagerie vocale, avant que le bip sonore se fasse entendre.

*'''Dicter la date ou l'heure''': Si la valeur est à OUI, lors de la vérification de votre message, vous entendrez la date et l'heure auxquelles le message a été reçu.

*'''Fuseau horaire''': L'option "Fuseau horaire" utilise la zone de temps configurée ici pour fournir la date et l'heure exactes de la réception des messages.

*'''Dicter identification de l'appelant''': Si la valeur est à OUI, quand vous consulterez vos messages, vous entendrez le Caller ID de l'expéditeur du message.

== Assigner votre messagerie vocale à votre DID ==

Après avoir créé votre messagerie vocale, vous pouvez l'assigner à l'un de vos DIDs à partir de votre portail principal. Veuillez vous référer à Numéros DID > Gestion des DID > Modifier DID > '''Messagerie Vocale associée avec le DID'''.

Toujours sous la même section vous pouvez définir la durée de la sonnerie (Le temps maximum où un appel vers votre DID peut rester en "état de sonnerie" avant d'annuler l'appel. (Veuillez noter que 30 secondes sont équivalentes à 6 sonneries).

[[File:MessagerieVocaleDID.png|thumb|none|900px]]

== Assigner votre messagerie vocale à votre Compte ==

Si vous souhaitez assigner une messagerie vocale à votre compte principal, rendez-vous dans votre portail, Menu Principal > Paramètres du compte > Généralités > '''Messagerie Vocale associée au compte principal'''.

[[File:MessagerieVocaleMain.png|thumb|none|900px]]

Maintenant, si vous avez besoin d'attribuer une messagerie vocale à un sous-compte spécifique, vous devez aller dans votre compte, et suivre les options: Sous-Comptes > Gestion des Sous-comptes > Modifier Sous-comptes

Vous verrez au bas de cette page, l'option "'''Messagerie Vocale de poste interne'''" , où vous pourrez la configurer.

[[File:MessagerieVocaleSousCompte.png|thumb|none|900px]]

== Codes d'accès de la Messagerie Vocale ==

*'''Astérisque(*)97''': est utilisé pour accéder directement à la messagerie vocale associée au compte que vous avez choisi. Si vous souhaitez vérifier si la messagerie vocale est associée à votre compte, alors consultez l'option '''Assigner votre messagerie vocale à votre Compte''', expliquée ci-dessus. (Le système vous demandera votre mot de passe, uniquement si l'option « Ignorer l'invite de mot de passe » est désactivée)

*'''Astérisque(*)98''': est utilisé pour accéder à votre messagerie vocale et choisir un de vos comptes de messagerie vocale. (Le système vous demandera votre ID et votre mot de passe de messagerie vocale).

Si pour une raison quelconque vous n'avez pas accès à votre service VoIP, sachez que vous pouvez consulter votre messagerie vocale de n'importe où, en composant votre numéro. Une fois que le système de messagerie vocale répondra à votre appel, appuyez sur la touche astérisque (*). Lorsque vous serez connecté à votre messagerie vocale, appuyez sur 0 (zéro) pour les options. Vous pouvez aussi enregistrer vos messages d'accueil et temporaires à partir de là.

'''Note:''' Le message d'accueil temporaire a priorité sur tous les autres messages d'accueil standards enregistrés (non disponible et occupé).

[[category:Guides en français]]

Grandstream HandyTone 802 - HT802

2019-11-17T15:40:07Z

Mauro:

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

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

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

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

<br clear="all" />

__TOC__

----

===Configuring the HandyTone 802===

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

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

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

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

====Plugging the HT802====

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.

====Getting IP address for the GUI====

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

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

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

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

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

====Loging into the device====

You should now see a page that looks like this:

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

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

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

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

====Configuring device's port FXS====

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.

Use the following settings to configure your VoIP.ms account: Configuration Page Settings:

* '''''Primary SIP Server''''': servername.voip.ms (one of VoIP.ms multiple [[Choosing_Server#Choosing_a_Server | '''''servers''''']], you can choose the one closest to your location.)

'''Notice that it is necessary to use the same server for both the device and the DID number in order to get incoming calls correctly'''

You can also find this information by logging into your Customer portal.

* '''''Failover SIP Server''''': (Please leave this Blank)

* '''''Outbound Proxy''''': servername.voip.ms (Use the same server you used as '''''Primary SIP Server'''''.
''For firmware 1.0.15.4 and higher we recommend leaving blank the outbound proxy field''

* '''''NAT Traversal''''': Keep-Alive

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

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

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

* '''''DNS Mode''''': A Record

* '''''SIP Registration''''': Yes

* '''''Unregister On Reboot''''': No

* '''''Outgoing Call Without Registration''''': Yes

* '''''Register Expiration''''': 5

* '''''Allow Incoming SIP Messages from SIP Proxy Only''''': Yes

* '''''Preferred DTMF method''''': In-audio, RFC2833

* '''''Use P-Access-Network-Info Header''''': No

* '''''Use P-Emergency-info Header''''': No

* '''''Enable Call Features''''': No

* '''''Dial Plan''''': {[x*]+}

* '''''Preferred Vocoder''''': PCMU, PCMA, G729

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

====Saving the changes====

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

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

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

Or

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

== 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 HT802 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.

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

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

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

Grandstream HandyTone 802 - HT802

2019-10-21T18:02:07Z

Mauro: /* Configuring device's port FXS */

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

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

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

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

<br clear="all" />

__TOC__

----

===Configuring the HandyTone 802===

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

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

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

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

====Plugging the HT802====

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.

====Getting IP address for the GUI====

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

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

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

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

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

====Loging into the device====

You should now see a page that looks like this:

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

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

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

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

====Configuring device's port FXS====

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.

Use the following settings to configure your VoIP.ms account: Configuration Page Settings:

* '''''Primary SIP Server''''': servername.voip.ms (one of VoIP.ms multiple [[Choosing_Server#Choosing_a_Server | '''''servers''''']], you can choose the one closest to your location.)

'''Notice that it is necessary to use the same server for both the device and the DID number in order to get incoming calls correctly'''

You can also find this information by logging into your Customer portal.

* '''''Failover SIP Server''''': (Please leave this Blank)

* '''''Outbound Proxy''''': servername.voip.ms (Use the same server you used as '''''Primary SIP Server'''''

* '''''NAT Traversal''''': Keep-Alive

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

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

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

* '''''DNS Mode''''': A Record

* '''''SIP Registration''''': Yes

* '''''Unregister On Reboot''''': No

* '''''Outgoing Call Without Registration''''': Yes

* '''''Register Expiration''''': 5

* '''''Allow Incoming SIP Messages from SIP Proxy Only''''': Yes

* '''''Preferred DTMF method''''': In-audio, RFC2833

* '''''Use P-Access-Network-Info Header''''': No

* '''''Use P-Emergency-info Header''''': No

* '''''Enable Call Features''''': No

* '''''Dial Plan''''': {[x*]+}

* '''''Preferred Vocoder''''': PCMU, PCMA, G729

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

====Saving the changes====

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

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

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

Or

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

== 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 HT802 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.

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

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

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

File:TOTP code enteredFr.PNG

2019-09-05T20:57:49Z

Mauro: uploaded a new version of "File:TOTP code enteredFr.PNG"

File:TOTP codeFr.PNG

2019-09-05T20:55:01Z

Mauro: uploaded a new version of "File:TOTP codeFr.PNG"

User:Mauro/Sandbox

2019-09-04T17:30:49Z

Mauro:

Two-factor authentication is a method for securing the account, there are several methods available however in VoIP.ms we offer 2 options.
<div><ul> <li style="display: inline-block; vertical-align:top;"> </li> </ul></div>
*2 step authentication using a one time code sent via email (OTP) You can find the guide for this feature in our wiki page [[Two-step Verification | Two Step Verification]]

*And authentication based on a time-based one-time Password (TOTP) using an authenticator app like FreeOTP, Google Autheticator, Microsoft Authenticator, Authy, etc.

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

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

__TOC__

== Activate This Option for your Account ==

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

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

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

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

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

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

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

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

== Pairing your authenticator app ==

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

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

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

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

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

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

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

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

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

User:Mauro/Sandbox

2019-09-04T17:14:08Z

Mauro:

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

2 step authentication using a one time code sent via email (OTP) and authentication based on a time-based one-time Password (TOTP) using an authenticator app like FreeOTP, Google Autheticator, Microsoft Authenticator, Authy, etc.

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

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

__TOC__

== Activate This Option for your Account ==

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

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

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

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

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

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

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

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

== Pairing your authenticator app ==

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

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

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

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

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

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

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

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

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

User:Mauro/Sandbox

2019-09-04T17:12:51Z

Mauro:

User:Mauro/Sandbox

2019-09-04T17:12:01Z

Mauro:

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

2 step authentication using a one time code sent via email (OTP) and authentication based on a time-based one-time Password (TOTP) using an authenticator app like FreeOTP, Google Autheticator, Microsoft Authenticator, Authy, etc.

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

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

__TOC__

== Activate This Option for your Account ==

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

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

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

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

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

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

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

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

== Pairing your authenticator app ==

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

:[[File:TOTP_GAUTH.png|border|thumb|none|600px]]

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

:[[File:TOTP_GAUTH2.png|border|thumb|none|600px]]

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

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

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

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

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

User:Mauro/Sandbox

2019-09-04T17:10:00Z

Mauro:

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

2 step authentication using a one time code sent via email (OTP) and authentication based on a time-based one-time Password (TOTP) using an authenticator app like FreeOTP, Google Autheticator, Microsoft Authenticator, Authy, etc.

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

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

__TOC__

== Activate This Option for your Account ==

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

:[[File:TLS-SRTP-Steps0001.png|border]]

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

:[[File:TOTP_enable.PNG|border]]

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

:[[File:TOTP_OK.PNG|border]]

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

:[[File:TOTP_code.PNG|border]]

== Pairing your authenticator app ==

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

:[[File:TOTP_GAUTH.png|border]]

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

:[[File:TOTP_GAUTH2.png|border]]

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

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

:[[File:TOTP_code.PNG|border]]

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

:[[File:TOTP_warning.PNG|border]]

File:TOTP warning.PNG

2019-09-04T17:09:36Z

Mauro:

File:TOTP GAUTH2.png

2019-09-04T17:08:45Z

Mauro:

File:TOTP GAUTH.png

2019-09-04T17:04:26Z

Mauro:

File:TOTP code.PNG

2019-09-04T16:57:16Z

Mauro:

File:TOTP OK.PNG

2019-09-04T16:51:39Z

Mauro:

File:TOTP enable.PNG

2019-09-04T16:50:57Z

Mauro:

User:Mauro/Sandbox

2019-09-04T16:47:08Z

Mauro:

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

2 step authentication using a one time code sent via email (OTP) and authentication based on a time-based one-time Password (TOTP) using an authenticator app like FreeOTP, Google Autheticator, Microsoft Authenticator, Authy, etc.

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

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

__TOC__

== Activate This Option for your Account ==

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

:[[File:TLS-SRTP-Steps0001.png|border]]

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

:[[File:TLS-SRTP-Steps0001.png|border]]

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

:[[File:TLS-SRTP-Steps0001.png|border]]

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

:[[File:TLS-SRTP-Steps0001.png|border]]

5) After scanning the code or adding the string on the app you'll receive a timed code of 6 digits that you can input in the window where the code was shown. Click on the "Validate" button to complete the validation.

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

:[[File:TLS-SRTP-Steps0001.png|border]]

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

User:Mauro/Sandbox

2019-09-04T16:36:44Z

Mauro:

User:Mauro/Sandbox

2019-09-04T16:32:45Z

Mauro:

The Account Settings page is one of the most important pages in the Customer Portal, it will help you out as its name points out, to set all the proper ways to use your account, and these settings will affect the global settings on your account, from Device type to routing and even security of your account.

__TOC__

== Account Routing ==

These settings define the routes the system will use when you place calls to either Canada, Toll Free or International Numbers.

[[File:Accsettings1.png|thumb|none|600px|Account Routing]]

'''Canada Routing''': You can choose between the [[Value vs Premium|'''Value''' or '''Premium''']] Route. If you are looking for the best wholesale price without any volume commitment you can use the '''Value''' route, however the CallerID number is not always guaranteed to work in this route.
The '''Premium''' route has a better level of quality, at a price that is a little higher than the value option. The CallerID Number is guaranteed to work for this route. Please note that the '''Canada Route''' can also be set independently per sub account, under the [[Sub Accounts]] settings.

Please note the US route is only available on the Premium route with a rate of $0.009 per minute.

'''International Routing''': You can choose also the [[Value vs Premium|'''Value''' or '''Premium''']] route. Both routes are reliable, however you're going to find better quality if you're using the '''Premium''' route. The CallerID Number is not guaranteed to work for international calls even if you use the '''Premium''' route.

'''Toll Free Routing''': This define the route the system will use when you place a Toll-Free call to Canada or USA. You can choose between the '''Value''' route (free to use) or the '''Premium''' (at 1.06 cent per minute). As the other settings, the '''Premium''' route give you better quality and the CallerID Number is guaranteed to work.

'''Toll Free Carrier''': This setting defines the type of carrier that will be used when you dial a Toll-Free number. You can choose between '''Server Default''', '''American Carrier''' or '''Canadian Carrier'''.

'''Server Default:''' The Carrier for the termination of your toll-free call will be chosen depending on the geographical location of the server
you are registering or connecting to. American Servers will dial through an American Carrier and Canadian servers will use a Canadian Toll-Free carrier.
'''American Carrier:''' Your Toll-Free calls will always go through an American Carrier, even if you use a Canadian server.
'''Canadian Carrier:''' Your Toll-Free calls will always go through a Canadian Carrier, even if you use an American server.

For example, this feature can be useful for west based Canadian customers who wish to use the Seattle server but have their toll-free termination calls go through a Canadian carrier.

'''Note''': Please refer to [[Service Cost]] for information about the rates of the outgoing calls.

== Account Restrictions ==

These settings define the restrictions the system will use when you place calls to either USA, Canada or International Numbers.

[[File:Account Restrictions.jpg|thumb|none|600px|Account Restrictions]]

'''Allow 411 dialing''': When enabled, you can call 411 directory assistance at the cost of $0.99 per call.

'''Note''': XXX-555-1212 is disabled within VoIP.ms. The only way to reach directory assistance is by dialing 411.

'''Allow International Calls''': When Disabled, calls placed to destinations outside USA/Canada will be automatically rejected. If you are using [[Sub Accounts]] you need to setup this directly in the sub account.

'''Note''': This option is set to "No" by default for New accounts.

'''Max. Call Time for US48/Canadian Calls''': With these setting you can choose how long the call will last when you call to US or Canada Numbers. If a call to a US or Canada Number exceeds the maximum time you have set in this option, it will be automatically hung up.

'''Max. Call Time for International Calls''': Works exactly as the '''Max. Call Time for US48/Canadian Calls''' setting. If a call to an International Number exceeds the maximum time set, it will be automatically hung up.

'''International Amount Restriction''': With this option you can set the maximum amount per minute for calls to international destinations. If the Per minute rate for an International destination exceeds the value you have set in this option, the call will not be connected.

For example, let say you can set this option to $0.150 and you want to call a UK mobile number with area code 4470, the call will not go through because the per minute rate for this area code is $0.3048.

'''Note:''' This settings depends on the International Routing you have chosen between Value or Premium.

'''Allow Calls to Countries''': Here you can select Regions or Specific Countries to allow outgoing calls from your account.

In order to allow calls to a specific country you can click the name of the region to expand the countries within it and select the desired countries only. You can select '''Select All''' so all the countries within the continent will be selected. You will also be able to search for a specific country using the '''Search Country''' field. If you call to a country not allowed in this section, the call will not be connected.

Please make sure to only allow those destinations that you are planning to dial and avoid having unplanned destinations allowed.

[[File:Allow Inter Count.jpg|thumb|none|600px|Allow Calls to Countries]]

'''Note''': If you have international calls disabled, it will prevent calling to the international countries you have enabled in this list.

== General ==

These are the general settings used by the system when you make or receive calls.

[[File:AcctSet Gen.jpg|thumb|none|600px|General]]

'''e911 Default CallerID''': When you dial 911 from any of your SIP accounts, and the caller ID doesn't match a 911-enabled DID, the "e911 Default CallerID" will be used as the caller ID allowing you to connect your call instead of a busy tone.

'''Dialing Mode''': This setting allows you to set the way you're going to dial other numbers.

*North America (Recommended): You can dial to countries part of the North American Numbering Plan Administration by dialing 10 or 11 digits. (with or without the 1 prefix). You'll need to use the 00 or 011 prefix to call international numbers.

*E164: You dial numbers by using Country Code first. You don't need to send the 00 or 011 prefix when using this mode, but it's still supported.

'''CallerID Number''': You can set the [[Caller ID|callerID Number]] you want to pass if you are using an [[Devices|ATA]], IP Phone or [[Softphones|Softphone]]. It is important to pass a valid [[Caller ID|caller id]] to ensure proper termination. If you have a a device capable of passing its own [[Caller ID|CallerID number]] such as a soft switch or [[PBXs|PBX]], you can leave this blank to set the [[Caller ID|CallerID Number]] on your side.

'''Voicemail Associated to the Main Account''': You can select the mailbox associated to the main account.

*Accessing Voice Mail: When you dial *97 from your main account, it will not prompt for the [[voicemail]] ID. It will also not prompt for the password if you have selected "Skip Password" option in the [[voicemail]] configuration.

*Message Waiting Indicator: When there are new messages, notice of new message(s) will be sent. This will lead to different results depending on your type of adapter, soft phone or IP phone. For example, when using a Linksys [[Devices|ATA]] adapter, it's usually a stutter tone with periodic ring, IP phones usually use a stutter dial tone when you pick up the line and are equipped with a blinking light, soft phones usually show up a Voicemail Icon in the dial display.

'''Note:'''This setting only affects the device registered with the main account.
If you're using a [[Sub Accounts|sub account]] you need to set the '''Internal Extension Voicemail''' for that subaccount.

'''Music On Hold''': Most IP Phones and [[Softphones|Softphones]] inform the VoIP server they are connected to when the HOLD Button is pressed. If you would like to present music to the person you place on hold, you can select it here.<br>
The following options for music are available:<br>

*'''No Music, Intermittent Bleep:''' Silent, but with a subtle intermittent bleep sound to let your callers know that they are still on the line while they hold.
*'''Away in the Tropics:''' From Hawaii to the Caribbean, these tracks deliver sounds of ukulele, steel drums, and steel guitars.
*'''Coffee and Sunrise:''' Uplifting without being perky, and positive without being too smiley.
*'''Coffee Shop Acoustic:''' Soothing, acoustic guitar tracks makes for a relaxed feel and sets a great atmosphere.
*'''Easy Listening:''' Smooth, casual tunes.
*'''Guitar Alchemy:''' Clever harmonics and progressive chord sequences to create a joyful and warming musical experience.
*'''Happy Endings:''' Uplifting, commercial style. Guitar, drums, ukulele, harmonica and bells.
*'''Light and Casual:''' Soothing and peaceful songs with a light, positive feeling.
*'''Orchestral Moods:''' Emotional and dramatic tales spun by violins, pianos and full orchestras.
*'''Piano Mix:''' Smooth Piano
*'''Rock Me Easy:''' Feel-good music to create a relaxing atmosphere.
*'''Spa Sounds:''' Soft, slow and serene instrumentals.

You can test the categories by dialing the following codes: <br>

'''*** 89''' Away in the Tropics <br>
'''*** 90''' Coffee and Sunrise <br>
'''*** 91''' Coffee Shop Acoustic <br>
'''*** 92''' Easy Listening <br>
'''*** 93''' Guitar Alchemy <br>
'''*** 94''' Happy Endings <br>
'''*** 95''' Light and Casual <br>
'''*** 96''' Orchestral Moods <br>
'''*** 97''' Piano Mix <br>
'''*** 98''' Rock Me Easy <br>
'''*** 99''' Spa Sounds<br>
'''***100''' No Music, Intermittent Bleep<br>

== Security ==

Here you can change the password to access your Customer Portal as well your SIP/IAX passwords for your main account.

[[File:Securityen.png|thumb|none|600px|Security]]

'''Customer Portal Password''': Here you can change the password used to log in your VoIP.ms Customer Portal.

'''Main SIP/IAX Password''': Here is where you can change the password used to register your Main Account to one of the VoIP.ms servers.

'''Note:''' For default, the Main SIP/IAX Password is the same you use to login in your Customer Portal.

== Inbound Settings==

In this tab, you can set the protocol used for inbound calls and the device you're using for your main account.

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

'''Protocol for inbound DIDs''': Here you can select either SIP or IAX2 protocol. This will depend on the device you want to use with your Main account. SIP is the recommended protocol.

'''Device type''': Make sure to select the correct type of device you're going to use with the main account to properly receive incoming calls in your device.

== Notifications ==

The notifications tab holds a complete list of alerts based upon most of the features that you will be able to see in the customer portal, in regards of changes over your configuration system structure, e911 dialing and API configuration changes.

You will be able to select rather to receive an email or not when these events occur within your configuration, added to the always useful Balance Threshold.

[[File:Accsettings2.png|thumb|none|600px|Notifications]]

'''Balance Threshold''': Here you can set an amount and when your balance goes below this amount you will receive an email to the address set in the option below. You can select a value between $1 up to $200, or if you like you can disable the notifications (this is not recommended).

'''Note''': Set this according to the monthly use you have in your account.

'''Email''': Enter the email address in which you want to receive the notifications. Make sure to enter a proper email address. If you want to receive notifications in more than one email address, you can list the emails separated with commas (,). This will only work for the Balance Threshold

'''Note''': While the balance is below your threshold, you're going to keep receiving emails until you add new funds in your account or you change the balance threshold.

== Default DID Routing ==

These settings let you define the preconfigured Routing options that are going to be applied to each new DIDs that you order. However, you are still able to change the settings for the DID in the order page.

[[File:DefaultDIDRouting.png|thumb|none|600px|Default DID routing]]

'''Select Plan''': Incoming Calls and Monthly Fee for the DID will be charged according to the Plan selected here. Plan prices change depending on the rate center. These prices will be shown when purchasing numbers.

'''CallerID Name Lookup''': When activated, the system will perform a lookup in the LIBD/CNAM database to find the name matching the number of callers with a US/Canadian CallerID Number. The result of this query will be displayed with the following format "Caller ID name portion" <Caller ID number>.

'''DID POP''': Here you can set the server in which your software/device is registered to or receiving the call from.
'''Note''': Always make sure to register your software/device to the same server you set in this option to properly receive incoming calls.

[[File:DefaultRouting.png|thumb|none|600px|Default DID POP]]

'''Routing''': Here you can select the destination of the incoming calls to your DID number. You can routed directly to one account/sub account, an [[Digital Receptionist (IVR)]], [[Calling Queues]], [[Time Conditions]], [[Ring Groups]], etc.

'''Failover Options''': This setting let you define where to redirect the call when the destination is '''Busy''', '''Unreachable''' or '''No Answer'''. These options can be displayed by clicking the '''Show Failover Options''' button

Note: You can change these settings later or enable additional settings in your Customer Portal>>DID Numbers>>Manage DID(s)

== Newsletter ==

'''Newsletter Subscription''': You can set if you want to receive news from voip.ms

[[File:Accsettings3.png|thumb|none|600px|Newsletter]]

== Advanced ==

These settings are recommended for advanced users only. If you are not familiar with them, you can leave them with the default values.

[[File:Maxexpiry.png|thumb|none|600px|Advanced]]

'''NAT (Network Address Translation)''': This setting should be set to Yes if you're behind a NAT, if not set to No. If you're unsure what setting means, is highly recommended that you leave it to Yes.

'''DTMF Mode''': This allows you to select the DTMF mode that is going to be used for your account. If you set this to AUTO the RFC2833 (AVT) is going to be used and automatically switch to INBAND if the other end doesn't support RFC2833.
Note: Its recommended that you select the same DTMF mode in your device.

'''Allowed Codecs''': This setting allows you to select the codecs that you're going to use, you can choose to allow one or more codecs at the time.
Note: It's recommended that you select Allow All and only change it if you have an specific reason to do so.

'''Encrypted SIP Traffic''': This setting allows you to encrypt the communication between your device and our server, by using the SIP-TLS ''(Transport Layer Security)'' and SRTP ''(Secure Real-Time Transport Protocol)'' protocol.

Before using this feature, please consult the article about the [[Call Encryption - TLS/SRTP]]

'''Note:''' If enabled, all the SIP traffic will be encrypted for the main account.
''' ''' Please note that if encrypted calls are enabled then you need to configure your device to make and receive encrypted calls.

'''Max Expiry:''' Sets the '''maximum''' amount of time (in seconds) until a device or phone system registration expires. The default value is 3600 but the range can go from 60 to 3600 seconds. If your device or phone system requests a lower registration expiry time to the server, this will be respected. MaxExpiry also applies to subscriptions, Message-Waiting Indicators (MWI) as well as Presence (BLF).

'''RTP Time Out:''' Sets the amount of time (in seconds) of '''no RTP activity (silence)''' before the call is terminated when the call is '''NOT''' on Hold. If the field is left empty the default value of 60 seconds will be used. RTP time out can range from 1 to 3600 seconds but it must always be equal or lower than RTP Hold Time Out.

'''RTP Hold Time Out:''' Sets the amount of time (in seconds) of '''no RTP activity (silence)''' before terminating a call that is '''On-Hold'''. If the field is left empty the default value of 600 seconds will be used. RTP Hold Time Out can range from 1 to 3600 seconds but it must always be equal or greater than RTP Time Out.

Note: '''Max Expiry, RTP Time Out, and RTP Hold Time Out''' settings are available for the main account and subaccounts independently.
[[category:guides]]

User:Mauro/Sandbox

2019-08-29T16:50:04Z

Mauro:

[[File:VCS754-thumb.PNG|250px|thumb|left]]
<br clear="all" />

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

All Eristation conference phones come with with portable DECT 6.0 microphones with Orbitlink Wireless Technology, making conference calls more flexible and engaging than ever before.

<br clear="all" />

__TOC__

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

In order to configure the '''VCS754''' to be used along with our service, it is required to access it's web interface settings. For this, the IP address of the device must be acquired.

To perform this click on the "'''''Menu'''''" button of the phone. Once on the settings menu, scroll down to: " '''''Status''''' " , then select: " '''''Network''''' ". On this section, you'll need to take note of the IP address shown beside: " '''''IP''''' ". It should read back something as: " '''''192.168.0.1''''' ".

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

Default's credentials are:
user: '''''admin'''''
password: '''''admin'''''

=='''System configuration'''==

Once logged in you'll be presented with this screen:

[[File:VCS754-1.PNG|thumb|none|600px]]

Here, please navigate to: ''System''. Once on this page, select '''''Account 1''''' under ''"SIP account Management"'' from the colum on the left (or the one you will configure):

[[File:VCS754-2.png|thumb|none|600px]]

===System Account Management Account X===

====General Account Settings====

:*'''''Account Label:''''': The name you wish to pass when calling out
:*'''''Display Name''''': The name you wish to pass when calling out
:*'''''User Identifer''''': 100000 (replace with your SIP main account or subaccount)
:*'''''Authentication Name''''': 100000 (replace with your SIP main account or subaccount)
:*'''''Authentication Password''''': Your SIP account's password
:*'''''Dial Plan''''': ''x+P'' (By default)

====SIP Server====

:*'''''Server Address''''': One of VoIP.ms multiple ''[[Choosing Server | SERVERS]]'', you can choose the one closest to your location
:*'''''Port''''': 5060 (Aditionally you could use 5080 or 42872)

[[File:VCS754-3.PNG|thumb|none|600px]]

====Registration====

:*'''''Server Address''''': Be sure that you use the same one you used at '''Server Address'''
:*'''''Port''''': 5060 (Aditionally you could use 5080 or 42872)
:*'''''Expiration (secs)''''': 300
:*'''''Registration Freq (secs)''''': 10

====Outbound Proxy====

:*'''''Server Address''''': Be sure that you use the same one you used at '''Server Address'''
:*'''''Port''''': 5060 (Aditionally you could use 5080 or 42872)

====Backup Outbound Proxy====

''Leave these settings as they are by default''

====Audio====

We strongly recommend to have set here only the audio codec(s) you have enabled in your SIP account's settings (preferrably only one). In case you need to have more than 1 Audio codec, it can be set as follows:

:*'''''Ringer Tone''''': 1 (or the one you prefer)
:*'''''Codec Priority 1''''': G.711u
:*'''''Codec Priority 2''''': G.729a/b
:*'''''Codec Priority 3''''': None
:*'''''Codec Priority 4''''': None
:*'''''Codec Priority 5''''': None

[[File:VCS754-4.PNG|thumb|none|600px]]

====Signaling Settings====

:*'''''Local SIP Port''''': 5060 (Aditionally you could use 5080 or 42872)
:*'''''Transport''''': UDP or TCP (Depending on your network's configuration)

You can leave all the other values as they are unless you actually need to specify them.

Scroll down and click on '''''"Save"''''' then you can go back to the '''''"Status"''''' tab, if all is correct, you will see your account as '''Registered'''

[[File:VCS754-5.PNG|thumb|none|600px]]

=== Using TLS/Encryption ===

The Vtech VCS754 is compatible with [[Call Encryption - TLS/SRTP | Call Encryption]]. However to use it, you need to change some settings in your device:

==== Ports for TLS ====

When you enable encryption calls from our system, the ports need to be changed to the encrypted ones. This is just adding " 1 " to the used port, leaving the available ones as 5061, 5081, 42873. Make the proper changes in your device's settings to have it working

==== Audio Section ====

Mark this "[]Enable Voice Encryption (SRTP)"

<div><ul>
<li style="display: inline-block;"> [[File:VCS754-6.PNG|thumb|none|400px]] </li>
<li style="display: inline-block;"> [[File:VCS754-7.PNG|thumb|none|600px]] </li>
</ul></div>

User:Mauro/Sandbox

2019-08-29T16:48:59Z

Mauro:

[[File:VCS754-thumb.PNG|250px|thumb|left]]
<br clear="all" />

The [https://businessphones.vtech.com/pd/3439/VCS754-ErisStation-SIP-Conference-Phone-with-Four-Wireless-Mics Vtech VCS754 ErisStation] conference phone features a compact, all-in-one design makes it easy to keep everything together—no clutter, no hassle.

All Eristation conference phones come with with portable DECT 6.0 microphones with Orbitlink Wireless Technology, making conference calls more flexible and engaging than ever before.

<br clear="all" />

__TOC__

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

In order to configure the '''VCS754''' to be used along with our service, it is required to access it's web interface settings. For this, the IP address of the device must be acquired.

To perform this click on the "'''''Menu'''''" button of the phone. Once on the settings menu, scroll down to: " '''''Status''''' " , then select: " '''''Network''''' ". On this section, you'll need to take note of the IP address shown beside: " '''''IP''''' ". It should read back something as: " '''''192.168.0.1''''' ".

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

Default's credentials are:
user: '''''admin'''''
password: '''''admin'''''

=='''System configuration'''==

Once logged in you'll be presented with this screen:

[[File:VCS754-1.PNG|thumb|none|600px]]

Here, please navigate to: ''System''. Once on this page, select '''''Account 1''''' under ''"SIP account Management"'' from the colum on the left (or the one you will configure):

[[File:VCS754-2.png|thumb|none|600px]]

===System Account Management Account X===

====General Account Settings====

:*'''''Account Label:''''': The name you wish to pass when calling out
:*'''''Display Name''''': The name you wish to pass when calling out
:*'''''User Identifer''''': 100000 (replace with your SIP main account or subaccount)
:*'''''Authentication Name''''': 100000 (replace with your SIP main account or subaccount)
:*'''''Authentication Password''''': Your SIP account's password
:*'''''Dial Plan''''': ''x+P'' (By default)

====SIP Server====

:*'''''Server Address''''': One of VoIP.ms multiple ''[[Choosing Server | SERVERS]]'', you can choose the one closest to your location
:*'''''Port''''': 5060 (Aditionally you could use 5080 or 42872)

[[File:VCS754-3.PNG|thumb|none|600px]]

====Registration====

:*'''''Server Address''''': Be sure that you use the same one you used at '''Server Address'''
:*'''''Port''''': 5060 (Aditionally you could use 5080 or 42872)
:*'''''Expiration (secs)''''': 300
:*'''''Registration Freq (secs)''''': 10

====Outbound Proxy====

:*'''''Server Address''''': Be sure that you use the same one you used at '''Server Address'''
:*'''''Port''''': 5060 (Aditionally you could use 5080 or 42872)

====Backup Outbound Proxy====

''Leave these settings as they are by default''

====Audio====

We strongly recommend to have set here only the audio codec(s) you have enabled in your SIP account's settings (preferrably only one). In case you need to have more than 1 Audio codec, it can be set as follows:

:*'''''Ringer Tone''''': 1 (or the one you prefer)
:*'''''Codec Priority 1''''': G.711u
:*'''''Codec Priority 2''''': G.729a/b
:*'''''Codec Priority 3''''': None
:*'''''Codec Priority 4''''': None
:*'''''Codec Priority 5''''': None

[[File:VCS754-4.PNG|thumb|none|600px]]

====Signaling Settings====

:*'''''Local SIP Port''''': 5060 (Aditionally you could use 5080 or 42872)
:*'''''Transport''''': UDP or TCP (Depending on your network's configuration)

You can leave all the other values as they are unless you actually need to specify them.

Scroll down and click on '''''"Save"''''' then you can go back to the '''''"Status"''''' tab, if all is correct, you will see your account as '''Registered'''

[[File:VCS754-5.PNG|thumb|none|600px]]

=== Using TLS/Encryption ===

The Vtech VCS754 is compatible with [[Call Encryption - TLS/SRTP | Call Encryption]]. However to use it, you need to change some settings in your device:

==== Ports for TLS ====

When you enable encryption calls from our system, the ports need to be changed to the encrypted ones. This is just adding " 1 " to the used port, leaving the available ones as 5061, 5081, 42873. Make the proper changes in your device's settings to have it working

==== Audio Section ====

Mark this "[]Enable Voice Encryption (SRTP)"

<div><ul>
<li style="display: inline-block;"> [[File:VCS754-6.PNG|thumb|none|400px]] </li>
<li style="display: inline-block;"> [[File:VCS754-7.PNG|thumb|none|600px]] </li>
</ul></div>

User:Mauro/Sandbox

2019-08-28T20:50:52Z

Mauro: /* Factory Reset */

[[File:VCS754-thumb.PNG|250px|thumb|left]]
<br clear="all" />

The [https://businessphones.vtech.com/pd/3439/VCS754-ErisStation-SIP-Conference-Phone-with-Four-Wireless-Mics Vtech VCS754 ErisStation] is a modern and cost-effective conferencing solution for enterprises of any scale.

With its modern and sleek design, the C520 fits seamlessly into your working day. Two detachable DECT microphones can be positioned freely or carried in the room as required to ensure the best sound and voice quality. Built-in charging stations with magnetic bays directly on the base station mean both microphones are always charged and ready for use in the next meeting. The conference phone also features automatic volume control and digital noise reduction so that all call participants can be understood in best sound quality.

<br clear="all" />

__TOC__

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

In order to configure the '''VCS754''' to be used along with our service, it is required to access it's web interface settings. For this, the IP address of the device must be acquired.

To perform this click on the "'''''Menu'''''" button of the phone. Once on the settings menu, scroll down to: " '''''Status''''' " , then select: " '''''Network''''' ". On this section, you'll need to take note of the IP address shown beside: " '''''IP''''' ". It should read back something as: " '''''192.168.0.1''''' ".

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

Default's credentials are:
user: '''''admin'''''
password: '''''admin'''''

=='''System configuration'''==

Once logged in you'll be presented with this screen:

[[File:VCS754-1.PNG|thumb|none|600px]]

Here, please navigate to: ''System''. Once on this page, select '''''Account 1''''' under ''"SIP account Management"'' from the colum on the left (or the one you will configure):

[[File:VCS754-2.png|thumb|none|600px]]

===System Account Management Account X===

====General Account Settings====

:*'''''Account Label:''''': The name you wish to pass when calling out
:*'''''Display Name''''': The name you wish to pass when calling out
:*'''''User Identifer''''': 100000 (replace with your SIP main account or subaccount)
:*'''''Authentication Name''''': 100000 (replace with your SIP main account or subaccount)
:*'''''Authentication Password''''': Your SIP account's password
:*'''''Dial Plan''''': ''x+P'' (By default)

====SIP Server====

:*'''''Server Address''''': One of VoIP.ms multiple ''[[Choosing Server | SERVERS]]'', you can choose the one closest to your location
:*'''''Port''''': 5060 (Aditionally you could use 5080 or 42872)

[[File:VCS754-3.PNG|thumb|none|600px]]

====Registration====

:*'''''Server Address''''': Be sure that you use the same one you used at '''Server Address'''
:*'''''Port''''': 5060 (Aditionally you could use 5080 or 42872)
:*'''''Expiration (secs)''''': 300
:*'''''Registration Freq (secs)''''': 10

====Outbound Proxy====

:*'''''Server Address''''': Be sure that you use the same one you used at '''Server Address'''
:*'''''Port''''': 5060 (Aditionally you could use 5080 or 42872)

====Backup Outbound Proxy====

''Leave these settings as they are by default''

====Audio====

We strongly recommend to have set here only the audio codec(s) you have enabled in your SIP account's settings (preferrably only one). In case you need to have more than 1 Audio codec, it can be set as follows:

:*'''''Ringer Tone''''': 1 (or the one you prefer)
:*'''''Codec Priority 1''''': G.711u
:*'''''Codec Priority 2''''': G.729a/b
:*'''''Codec Priority 3''''': None
:*'''''Codec Priority 4''''': None
:*'''''Codec Priority 5''''': None

[[File:VCS754-4.PNG|thumb|none|600px]]

====Signaling Settings====

:*'''''Local SIP Port''''': 5060 (Aditionally you could use 5080 or 42872)
:*'''''Transport''''': UDP or TCP (Depending on your network's configuration)

You can leave all the other values as they are unless you actually need to specify them.

Scroll down and click on '''''"Save"''''' then you can go back to the '''''"Status"''''' tab, if all is correct, you will see your account as '''Registered'''

[[File:VCS754-5.PNG|thumb|none|600px]]

=== Using TLS/Encryption ===

The Vtech VCS754 is compatible with [[Call Encryption - TLS/SRTP | Call Encryption]]. However to use it, you need to change some settings in your device:

==== Ports for TLS ====

When you enable encryption calls from our system, the ports need to be changed to the encrypted ones. This is just adding " 1 " to the used port, leaving the available ones as 5061, 5081, 42873. Make the proper changes in your device's settings to have it working

==== Audio Section ====

Mark this "[]Enable Voice Encryption (SRTP)"

<div><ul>
<li style="display: inline-block;"> [[File:VCS754-6.PNG|thumb|none|400px]] </li>
<li style="display: inline-block;"> [[File:VCS754-7.PNG|thumb|none|600px]] </li>
</ul></div>

User:Mauro/Sandbox

2019-08-28T20:50:16Z

Mauro: /* Using TLS/Encryption */

[[File:VCS754-thumb.PNG|250px|thumb|left]]
<br clear="all" />

The [https://businessphones.vtech.com/pd/3439/VCS754-ErisStation-SIP-Conference-Phone-with-Four-Wireless-Mics Vtech VCS754 ErisStation] is a modern and cost-effective conferencing solution for enterprises of any scale.

With its modern and sleek design, the C520 fits seamlessly into your working day. Two detachable DECT microphones can be positioned freely or carried in the room as required to ensure the best sound and voice quality. Built-in charging stations with magnetic bays directly on the base station mean both microphones are always charged and ready for use in the next meeting. The conference phone also features automatic volume control and digital noise reduction so that all call participants can be understood in best sound quality.

<br clear="all" />

__TOC__

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

In order to configure the '''VCS754''' to be used along with our service, it is required to access it's web interface settings. For this, the IP address of the device must be acquired.

To perform this click on the "'''''Menu'''''" button of the phone. Once on the settings menu, scroll down to: " '''''Status''''' " , then select: " '''''Network''''' ". On this section, you'll need to take note of the IP address shown beside: " '''''IP''''' ". It should read back something as: " '''''192.168.0.1''''' ".

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

Default's credentials are:
user: '''''admin'''''
password: '''''admin'''''

=='''System configuration'''==

Once logged in you'll be presented with this screen:

[[File:VCS754-1.PNG|thumb|none|600px]]

Here, please navigate to: ''System''. Once on this page, select '''''Account 1''''' under ''"SIP account Management"'' from the colum on the left (or the one you will configure):

[[File:VCS754-2.png|thumb|none|600px]]

===System Account Management Account X===

====General Account Settings====

:*'''''Account Label:''''': The name you wish to pass when calling out
:*'''''Display Name''''': The name you wish to pass when calling out
:*'''''User Identifer''''': 100000 (replace with your SIP main account or subaccount)
:*'''''Authentication Name''''': 100000 (replace with your SIP main account or subaccount)
:*'''''Authentication Password''''': Your SIP account's password
:*'''''Dial Plan''''': ''x+P'' (By default)

====SIP Server====

:*'''''Server Address''''': One of VoIP.ms multiple ''[[Choosing Server | SERVERS]]'', you can choose the one closest to your location
:*'''''Port''''': 5060 (Aditionally you could use 5080 or 42872)

[[File:VCS754-3.PNG|thumb|none|600px]]

====Registration====

:*'''''Server Address''''': Be sure that you use the same one you used at '''Server Address'''
:*'''''Port''''': 5060 (Aditionally you could use 5080 or 42872)
:*'''''Expiration (secs)''''': 300
:*'''''Registration Freq (secs)''''': 10

====Outbound Proxy====

:*'''''Server Address''''': Be sure that you use the same one you used at '''Server Address'''
:*'''''Port''''': 5060 (Aditionally you could use 5080 or 42872)

====Backup Outbound Proxy====

''Leave these settings as they are by default''

====Audio====

We strongly recommend to have set here only the audio codec(s) you have enabled in your SIP account's settings (preferrably only one). In case you need to have more than 1 Audio codec, it can be set as follows:

:*'''''Ringer Tone''''': 1 (or the one you prefer)
:*'''''Codec Priority 1''''': G.711u
:*'''''Codec Priority 2''''': G.729a/b
:*'''''Codec Priority 3''''': None
:*'''''Codec Priority 4''''': None
:*'''''Codec Priority 5''''': None

[[File:VCS754-4.PNG|thumb|none|600px]]

====Signaling Settings====

:*'''''Local SIP Port''''': 5060 (Aditionally you could use 5080 or 42872)
:*'''''Transport''''': UDP or TCP (Depending on your network's configuration)

You can leave all the other values as they are unless you actually need to specify them.

Scroll down and click on '''''"Save"''''' then you can go back to the '''''"Status"''''' tab, if all is correct, you will see your account as '''Registered'''

[[File:VCS754-5.PNG|thumb|none|600px]]

=== Using TLS/Encryption ===

The Vtech VCS754 is compatible with [[Call Encryption - TLS/SRTP | Call Encryption]]. However to use it, you need to change some settings in your device:

==== Ports for TLS ====

When you enable encryption calls from our system, the ports need to be changed to the encrypted ones. This is just adding " 1 " to the used port, leaving the available ones as 5061, 5081, 42873. Make the proper changes in your device's settings to have it working

==== Audio Section ====

Mark this "[]Enable Voice Encryption (SRTP)"

<div><ul>
<li style="display: inline-block;"> [[File:VCS754-6.PNG|thumb|none|400px]] </li>
<li style="display: inline-block;"> [[File:VCS754-7.PNG|thumb|none|600px]] </li>
</ul></div>

== Factory Reset ==

This is '''NOT''' always required and will wipe all your settings. Do not do it unless strictly necessary.

In case you need to factory C520 phone, you could to following this sequence :

Press '''[Menu] key''' + '''0 7 # * 1 8 5 0 #'''

This means start with the Arrow up key an then type: '''07#*1850#'''

If everything went right, it shows a Message. And than reboots.

[[category:SIP phones]]

User:Mauro/Sandbox

2019-08-28T20:45:16Z

Mauro: /* System configuration */

[[File:VCS754-thumb.PNG|250px|thumb|left]]
<br clear="all" />

The [https://businessphones.vtech.com/pd/3439/VCS754-ErisStation-SIP-Conference-Phone-with-Four-Wireless-Mics Vtech VCS754 ErisStation] is a modern and cost-effective conferencing solution for enterprises of any scale.

With its modern and sleek design, the C520 fits seamlessly into your working day. Two detachable DECT microphones can be positioned freely or carried in the room as required to ensure the best sound and voice quality. Built-in charging stations with magnetic bays directly on the base station mean both microphones are always charged and ready for use in the next meeting. The conference phone also features automatic volume control and digital noise reduction so that all call participants can be understood in best sound quality.

<br clear="all" />

__TOC__

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

In order to configure the '''VCS754''' to be used along with our service, it is required to access it's web interface settings. For this, the IP address of the device must be acquired.

To perform this click on the "'''''Menu'''''" button of the phone. Once on the settings menu, scroll down to: " '''''Status''''' " , then select: " '''''Network''''' ". On this section, you'll need to take note of the IP address shown beside: " '''''IP''''' ". It should read back something as: " '''''192.168.0.1''''' ".

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

Default's credentials are:
user: '''''admin'''''
password: '''''admin'''''

=='''System configuration'''==

Once logged in you'll be presented with this screen:

[[File:VCS754-1.PNG|thumb|none|600px]]

Here, please navigate to: ''System''. Once on this page, select '''''Account 1''''' under ''"SIP account Management"'' from the colum on the left (or the one you will configure):

[[File:VCS754-2.png|thumb|none|600px]]

===System Account Management Account X===

====General Account Settings====

:*'''''Account Label:''''': The name you wish to pass when calling out
:*'''''Display Name''''': The name you wish to pass when calling out
:*'''''User Identifer''''': 100000 (replace with your SIP main account or subaccount)
:*'''''Authentication Name''''': 100000 (replace with your SIP main account or subaccount)
:*'''''Authentication Password''''': Your SIP account's password
:*'''''Dial Plan''''': ''x+P'' (By default)

====SIP Server====

:*'''''Server Address''''': One of VoIP.ms multiple ''[[Choosing Server | SERVERS]]'', you can choose the one closest to your location
:*'''''Port''''': 5060 (Aditionally you could use 5080 or 42872)

[[File:VCS754-3.PNG|thumb|none|600px]]

====Registration====

:*'''''Server Address''''': Be sure that you use the same one you used at '''Server Address'''
:*'''''Port''''': 5060 (Aditionally you could use 5080 or 42872)
:*'''''Expiration (secs)''''': 300
:*'''''Registration Freq (secs)''''': 10

====Outbound Proxy====

:*'''''Server Address''''': Be sure that you use the same one you used at '''Server Address'''
:*'''''Port''''': 5060 (Aditionally you could use 5080 or 42872)

====Backup Outbound Proxy====

''Leave these settings as they are by default''

====Audio====

We strongly recommend to have set here only the audio codec(s) you have enabled in your SIP account's settings (preferrably only one). In case you need to have more than 1 Audio codec, it can be set as follows:

:*'''''Ringer Tone''''': 1 (or the one you prefer)
:*'''''Codec Priority 1''''': G.711u
:*'''''Codec Priority 2''''': G.729a/b
:*'''''Codec Priority 3''''': None
:*'''''Codec Priority 4''''': None
:*'''''Codec Priority 5''''': None

[[File:VCS754-4.PNG|thumb|none|600px]]

====Signaling Settings====

:*'''''Local SIP Port''''': 5060 (Aditionally you could use 5080 or 42872)
:*'''''Transport''''': UDP or TCP (Depending on your network's configuration)

You can leave all the other values as they are unless you actually need to specify them.

Scroll down and click on '''''"Save"''''' then you can go back to the '''''"Status"''''' tab, if all is correct, you will see your account as '''Registered'''

[[File:VCS754-5.PNG|thumb|none|600px]]

=== Using TLS/Encryption ===

The Snom C520 is compatible with [[Call Encryption - TLS/SRTP | Call Encryption]]. However to use it, you need to change some settings in your device:

==== Ports for TLS ====

When you enable encryption calls from our system, the ports need to be changed to the encrypted ones. This is just adding " 1 " to the used port, leaving the available ones as 5061, 5081, 42873. Make the proper changes in your device's settings to have it working

==== Audio Section ====

Mark this "[]Enable Voice Encryption (SRTP)"

<div><ul>
<li style="display: inline-block;"> [[File:VCS754-6.PNG|thumb|none|400px]] </li>
<li style="display: inline-block;"> [[File:VCS754-7.PNG|thumb|none|600px]] </li>
</ul></div>

== Factory Reset ==

This is '''NOT''' always required and will wipe all your settings. Do not do it unless strictly necessary.

In case you need to factory C520 phone, you could to following this sequence :

Press '''[Menu] key''' + '''0 7 # * 1 8 5 0 #'''

This means start with the Arrow up key an then type: '''07#*1850#'''

If everything went right, it shows a Message. And than reboots.

[[category:SIP phones]]

User:Mauro/Sandbox

2019-08-28T20:43:10Z

Mauro: Created page with "left <br clear="all" /> The [https://businessphones.vtech.com/pd/3439/VCS754-ErisStation-SIP-Conference-Phone-with-Four-Wireless-Mics Vtech..."

[[File:VCS754-thumb.PNG|250px|thumb|left]]
<br clear="all" />

The [https://businessphones.vtech.com/pd/3439/VCS754-ErisStation-SIP-Conference-Phone-with-Four-Wireless-Mics Vtech VCS754 ErisStation] is a modern and cost-effective conferencing solution for enterprises of any scale.

With its modern and sleek design, the C520 fits seamlessly into your working day. Two detachable DECT microphones can be positioned freely or carried in the room as required to ensure the best sound and voice quality. Built-in charging stations with magnetic bays directly on the base station mean both microphones are always charged and ready for use in the next meeting. The conference phone also features automatic volume control and digital noise reduction so that all call participants can be understood in best sound quality.

<br clear="all" />

__TOC__

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

In order to configure the '''VCS754''' to be used along with our service, it is required to access it's web interface settings. For this, the IP address of the device must be acquired.

To perform this click on the "'''''Menu'''''" button of the phone. Once on the settings menu, scroll down to: " '''''Status''''' " , then select: " '''''Network''''' ". On this section, you'll need to take note of the IP address shown beside: " '''''IP''''' ". It should read back something as: " '''''192.168.0.1''''' ".

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

Default's credentials are:
user: '''''admin'''''
password: '''''admin'''''

=='''System configuration'''==

Once logged in you'll be presented with this screen:

[[File:VCS754-1.PNG|thumb|none|600px]]

Here, please navigate to: ''System''. Once on this page, select '''''Account 1''''' under ''"SIP account Management"'' from the colum on the left (or the one you will configure):

[[File:VCS754-2.PNG|thumb|none|600px]]

===System Account Management Account X===

====General Account Settings====

:*'''''Account Label:''''': The name you wish to pass when calling out
:*'''''Display Name''''': The name you wish to pass when calling out
:*'''''User Identifer''''': 100000 (replace with your SIP main account or subaccount)
:*'''''Authentication Name''''': 100000 (replace with your SIP main account or subaccount)
:*'''''Authentication Password''''': Your SIP account's password
:*'''''Dial Plan''''': ''x+P'' (By default)

====SIP Server====

:*'''''Server Address''''': One of VoIP.ms multiple ''[[Choosing Server | SERVERS]]'', you can choose the one closest to your location
:*'''''Port''''': 5060 (Aditionally you could use 5080 or 42872)

[[File:VCS754-3.PNG|thumb|none|600px]]

====Registration====

:*'''''Server Address''''': Be sure that you use the same one you used at '''Server Address'''
:*'''''Port''''': 5060 (Aditionally you could use 5080 or 42872)
:*'''''Expiration (secs)''''': 300
:*'''''Registration Freq (secs)''''': 10

====Outbound Proxy====

:*'''''Server Address''''': Be sure that you use the same one you used at '''Server Address'''
:*'''''Port''''': 5060 (Aditionally you could use 5080 or 42872)

====Backup Outbound Proxy====

''Leave these settings as they are by default''

====Audio====

We strongly recommend to have set here only the audio codec(s) you have enabled in your SIP account's settings (preferrably only one). In case you need to have more than 1 Audio codec, it can be set as follows:

:*'''''Ringer Tone''''': 1 (or the one you prefer)
:*'''''Codec Priority 1''''': G.711u
:*'''''Codec Priority 2''''': G.729a/b
:*'''''Codec Priority 3''''': None
:*'''''Codec Priority 4''''': None
:*'''''Codec Priority 5''''': None

[[File:VCS754-4.PNG|thumb|none|600px]]

====Signaling Settings====

:*'''''Local SIP Port''''': 5060 (Aditionally you could use 5080 or 42872)
:*'''''Transport''''': UDP or TCP (Depending on your network's configuration)

You can leave all the other values as they are unless you actually need to specify them.

Scroll down and click on '''''"Save"''''' then you can go back to the '''''"Status"''''' tab, if all is correct, you will see your account as '''Registered'''

[[File:VCS754-5.PNG|thumb|none|600px]]

=== Using TLS/Encryption ===

The Snom C520 is compatible with [[Call Encryption - TLS/SRTP | Call Encryption]]. However to use it, you need to change some settings in your device:

==== Ports for TLS ====

When you enable encryption calls from our system, the ports need to be changed to the encrypted ones. This is just adding " 1 " to the used port, leaving the available ones as 5061, 5081, 42873. Make the proper changes in your device's settings to have it working

==== Audio Section ====

Mark this "[]Enable Voice Encryption (SRTP)"

<div><ul>
<li style="display: inline-block;"> [[VCS754-6.PNG|thumb|none|400px]] </li>
<li style="display: inline-block;"> [[File:VCS754-7.PNG|thumb|none|600px]] </li>
</ul></div>

== Factory Reset ==

This is '''NOT''' always required and will wipe all your settings. Do not do it unless strictly necessary.

In case you need to factory C520 phone, you could to following this sequence :

Press '''[Menu] key''' + '''0 7 # * 1 8 5 0 #'''

This means start with the Arrow up key an then type: '''07#*1850#'''

If everything went right, it shows a Message. And than reboots.

[[category:SIP phones]]

File:VCS754-thumb.PNG

2019-08-28T20:42:41Z

Mauro:

File:VCS754-7.PNG

2019-08-28T20:31:08Z

Mauro:

File:VCS754-6.PNG

2019-08-28T20:30:59Z

Mauro:

File:VCS754-5.PNG

2019-08-28T20:30:49Z

Mauro:

File:VCS754-4.PNG

2019-08-28T20:30:35Z

Mauro:

File:VCS754-3.PNG

2019-08-28T20:30:19Z

Mauro:

File:VCS754-2.png

2019-08-28T20:30:04Z

Mauro:

File:VCS754-1.PNG

2019-08-28T20:29:31Z

Mauro:

Cisco Linksys SPA942 NA

2019-05-13T16:07:59Z

Mauro:

[[File:SPA942-0.jpg|300px|thumb|left|Cisco Linksys SPA942 NA]]
<div style="width:100%;overflow:hidden;clear:both"></div>

Haga clic aquí para Español: [[Cisco Linksys SPA942 NA (Español)]]

==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)

Note: Some multi-line Cisco IP 'phones do provide a [setup] key (icon is one piece of paper with one corner folded). On these models, pressing [setup] then selecting 'Network' from the menu provides its IP without dialling any codes. The remainder of the configuration process remains the same.

'''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.

'''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)<br>
'''User ID:''' 100000 (Replace with your VoIP.ms username)<br>
'''Password:''' ******** (Type in the account password)<br>
'''Use DNS SRV:''' NO<br>
'''DNS SRV Auto Prefix:''' NO

'''Step 5 (Optional)'''

Optionally, 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.)

'''Step 6'''

Click on the "Save Settings " button at the bottom of the form.

==Encrypted Calls configuration changes==

For Encrypted calls to work a few changes in the configuration are required.

Under the '''LINE 1''' Tab make the following changes:

'''Proxy:''' atlanta1.voip.ms (It can be any of our servers just make sure you use the numeric value of the domain address,)<br>
'''Transport type''' TLS <br>
''' SIP port''' 5061 (5081, 42873 are accepted too)
<br>

After those changes go to the '''SIP''' Configuration tab and set:

'''"SRTP Method:"''' to "s-descriptor"
<br>

Lastly in the '''User''' Configuration page change:

'''"Secure Call Setting:"''' to "yes"

== See also ==
* [[Dial Plan for Linksys ATAs|Dial Plan for Cisco and Linksys devices]]; the same rules apply to dial plans for Cisco's IP 'phones as to Linksys ATA's.

* You can check the most commonly used Star Codes from [[Cisco/Linksys Star Codes]]
[[Category: SIP phones]]

Positron Telecom Systems

2019-01-03T16:25:32Z

Mauro: subaccount name from 123456 to 100000

Positron IP PBX solutions offer small and medium sized businesses powerful VoIP phone systems that combine voice and data into one easy-to-use device, also known as Unified Communications. These features, which were traditionally only available to large companies allow you to be more efficient and productive.

== Configuring your PBX ==

Assuming that your VoIP.ms account is active and your DID has been chosen, you may follow the steps below:

:1) Verify that the DTMF mode is set to "'''AUTO'''" in the VoIP.ms portal. To do so, log onto [https://voip.ms/m VoIP.ms portal], head over to Main Menu → Account Settings → Advanced and select "'''AUTO'''".

:[[File:Positron - DTMF.PNG|thumb|none|800px]]

:2) Make sure that you have your account username, password and server address. In case of doubt, you may find your username and server address in the Main Menu → Portal Home and under '''Main Account Registration Status'''.

:3) On the Positron PBX, go to PBX → Trunks/lines → Trunks/lines and click '''Add'''

:4) Give the trunk a name and the following information of your account:

::*'''IP Address / Domain''': One of VoIP.ms multiple [[Choosing Server | servers]], you can choose the one closest to your location
::*'''User name''': 100000 (Replace with your 6 digit Main SIP account UserID or Sub Account user name, i.e. 100000_sub)
::*'''Password''': Your VoIP.ms SIP Password

:After that, click '''Save'''.

:[[File:Positron_1.png|thumb|none|600px]]

:5) Once the trunk has been created, click '''Edit''' beside the trunk and in the larger amount of settings available, remove the username from the "'''From User'''" field and enter the DID under "'''Custom'''" in the P-Asserted Identity field, then click '''Save'''.

:[[File:Positron_2.PNG|thumb|none|600px]]

:6) You should now be able to apply configurations. Once applied, you will see the trunk in the VoIP section of the status screen.

== Set up Outgoing Calls ==

:1) On the Positron PBX, head over to PBX → Trunks/lines → Outgoing line groups and create a new group linked to the trunk. The trunk name created before should be available in the drop down menu.

:2) Once completed, go to PBX → Call Handling → Outgoing call rules and choose the ruleset that extensions will be using. We recommend that you create a few rules to test standard patterns e.g. 10-digit dialing and long-distance calling and make sure that they are using the Outgoing line group you have just created.

== Set up Incoming Calls ==

:1) On the Positron PBX, go to PBX → Call Handling → Incoming call rules and create a new group and make sure it is linked to the new trunk (by using the checkboxes).

:2) Edit the new group and enter the DID in the '''DID field''' and choose the extension (either IVR, ring group or simple extension) that the incoming call should go to.

:3) At this point you should be set up to for incoming and outgoing calls, but if you have another SIP trunk with another provider, you should also check that your '''SIP Registration Timer''' is set at a minimum of 600. To do so, on the Positron PBX, head over to PBX → PBX Settings → SIP → and set the '''SIP Registration Timer''' at a minimum of 600.

:[[File:Positron_3.png|thumb|none|600px]]

== Guide links ==
In the event where you need the guides directly from Positron, you may find the installation and the admin guides below:

Installation Guide : [http://www.positrontelecom.com/PDF_FILES/G124_1212_Installation_Guide_Apr_29-13.pdf Download PDF]

Admin Guide : [http://www.positrontelecom.com/PDF_FILES/124-1212_Admin_Guide%20-%20Feb_28_2013_FINAL.pdf Download PDF]

[[Category:PBXes]]

ZoIPer for Android

2018-04-23T16:40:24Z

Mauro:

==Configuration Detail==

* Start Zoiper IAX SIP VOIP Softphone App.

[[File:Welcome.png|thumb|none|300px|Click to enlarge]]

* On the welcome screen fill the '''Username''' and '''Password''' then click on '''Create an account'''.

** '''Username:''' 100000 (your VoIP.ms main or sub account) @server.voip.ms is optional.
** '''Password:''' ********* (account password)<br/>

[[File:ZoiperApp_accountsetup.png|thumb|none|300px|Click to enlarge]]

* On the host screen fill the domain/host with any of our [http://wiki.voip.ms/article/Choosing_Server servers]. I.E. toronto.voip.ms then click on '''Next'''

[[File:ZoipperApp_Host.png|thumb|none|300px|Click to enlarge]]

* '''Skip''' the '''Authentication''' step. AuthUserID and Outbound Proxy are not needed.

[[File:ZoiperApp_authskip.png|thumb|none|300px|Click to enlarge]]

* In this screen you can select the protocol that your SIP account will use:
** SIP TCP
** SIP UDP
** IAX UDP

Select the one you prefer according to your network's settings, I.E. '''SIP TCP''' then click on the '''Finish''' button.

Please notice that we do '''not support SIP TLS'''.

[[File:ZoiperApp_acctype.png|thumb|none|300px|Click to enlarge]]

* This the account configuration screen and will show a summary of your account information:
** '''User:''' 100000 (this is your SIP/IAX Main Username showing up in your [https://www.voip.ms/m/accountinfo.php customer portal])
** '''Password:''' ********* (this is your SIP password.)
** '''Host:''' atlanta.voip.ms (use one of the multiple VoIP.ms [http://wiki.voip.ms/article/Choosing_Server servers])

[[File:ZoiperApp_accsettings.png|thumb|none|300px|Click to enlarge]]

* Scroll down to '''Audio Codec Settings.''', the ones with a check mark are the codecs selected, leave only as selected codecs, uLaw or GSM. We also offer G.729 however it's a paid feature in Zoiper. Set the rest unchecked.

[[File:ZoiperApp_aucodec.png|thumb|none|300px|Click to enlarge]]

* Click on the '''<-''' in the left corner to leave the settings.

* Your Zoiper should be ready to make calls

==Common Errors==

*If there was a typo in your password or if there's a DNS issue Zoiper might show the account type options in red. You can select the protocol and hit the '''Finish''' button even though they're all red.

[[File:ZoiperApp_accerror.png|thumb|none|300px|Click to enlarge]]

*On the account settings screen you can check the error response code and correct it. You can check the list of common [https://wiki.voip.ms/article/SIP_Responses SIP Response codes] for this.

[[File:ZoiperApp_SIPcode.PNG|thumb|none|300px|Click to enlarge]]

**For best results, it is recommended to use wifi. But if using 4G/LTE and you experience poor sound quality, it has been reported that turning off ECHO CANCELLATION helps solve the issue.

[[category:softphones]]

ZoIPer for Android

2018-04-19T20:29:29Z

Mauro:

==Configuration Detail==

* Start Zoiper IAX SIP VOIP Softphone App.

[[File:Welcome.png|thumb|none|300px|Click to enlarge]]

* On the welcome screen fill the '''Username''' and '''Password''' then click on '''Create an account'''.

** '''Username:''' 100000 (your VoIP.ms main or sub account) @server.voip.ms is optional.
** '''Password:''' ********* (account password)<br/>

[[File:ZoiperApp_accountsetup.png|thumb|none|300px|Click to enlarge]]

* On the host screen fill the domain/host with any of our [http://wiki.voip.ms/article/Choosing_Server servers]. I.E. toronto.voip.ms then click on '''Next'''

[[File:ZoipperApp_Host.png|thumb|none|300px|Click to enlarge]]

* '''Skip''' the '''Authentication''' step. AuthUserID and Outbound Proxy are not needed.

[[File:ZoiperApp_authskip.png|thumb|none|300px|Click to enlarge]]

* In this screen you can select the protocol that your SIP account will use:
** SIP TCP
** SIP UDP
** IAX UDP

Select the one you prefer according to your network's settings, I.E. '''SIP TCP''' then click on the '''Finish''' button.

Please notice that we do '''not support SIP TLS'''.

[[File:ZoiperApp_acctype.png|thumb|none|300px|Click to enlarge]]

* This the account configuration screen and will show a summary of your account information:
** '''User:''' 100000 (this is your SIP/IAX Main Username showing up in your [https://www.voip.ms/m/accountinfo.php customer portal])
** '''Password:''' ********* (this is your SIP password.)
** '''Domain:''' toronto.voip.ms (use one of the multiple VoIP.ms [http://wiki.voip.ms/article/Choosing_Server servers])

[[File:ZoiperApp_accsettings.png|thumb|none|300px|Click to enlarge]]

* Scroll down to '''Audio Codec Settings.''', the ones with a check mark are the codecs selected, leave only as selected codecs, uLaw or GSM. We also offer G.729 however it's a paid feature in Zoiper. Set the rest unchecked.

[[File:ZoiperApp_aucodec.png|thumb|none|300px|Click to enlarge]]

* Click on the '''<-''' in the left corner to leave the settings.

* Your Zoiper should be ready to make calls

==Common Errors==

*If there was a typo in your password or if there's a DNS issue Zoiper might show the account type options in red. You can select the protocol and hit the '''Finish''' button even though they're all red.

[[File:ZoiperApp_accerror.png|thumb|none|300px|Click to enlarge]]

*On the account settings screen you can check the error response code and correct it. You can check the list of common [https://wiki.voip.ms/article/SIP_Responses SIP Response codes] for this.

[[File:ZoiperApp_SIPcode.PNG|thumb|none|300px|Click to enlarge]]

**For best results, it is recommended to use wifi. But if using 4G/LTE and you experience poor sound quality, it has been reported that turning off ECHO CANCELLATION helps solve the issue.

[[category:softphones]]

File:ZoiperApp SIPcode.PNG

2018-04-19T18:43:29Z

Mauro: