VoIP.ms Wiki - User contributions [en]

Acrobits Groundwire

2024-02-11T02:50:28Z

Aleclerc: Added instructions to make secure calls (TLS) working

GroundWire is an Android and iOS business Softphone application built by Acrobits, s.r.o. Fully certified, you will be able to register your VoIP.ms account quickly and easily.

__TOC__

== Application Features ==
[[File:Acrobitscert.png|200px|right|none|Acrobits|link=https://www.acrobits.net//?utm_campaign=itsp-partners&utm_medium=cp-certified-logo&utm_source=voipms]]

{| class="wikitable mw-collapsed"
! <strong>Features</strong>
|-
|Push Notifications for incoming calls for improved battery life
|-
|Multiline
|-
|Call Conferencing
|-
|Transfer and Attended Transfer
|-
|Voicemail
|-
|Call Forwarding
|-
|Bluetooth headset support
|-
|Secure calling through SRTP (SDES)
|-
|ZRTP support (through in app purchase), the latest in secure calling for SIP
|-
|Support for use on wifi and 3G networks
|-
|backgrounding support with status bar notification; check your email or other tasks while still available for calls on the Softphone
|-
|quick import of accounts from major VoIP Providers
|-
|call recorder and player, seamlessly integrated into the call history
|-
|Bluetooth headset support
|-
|HD Wideband audio through G.722 and Opus codecs
|-
|excellent sound quality, includes the G.711 (ulaw and alaw) and GSM audio codecs. Make an in app purchase to add G.729 Annex A for great quality over 3G networks
|-
|TLS support for encrypted SIP
|-
|audio codec manipulation, enabling you to prioritize the codecs used and disable ones you don’t want to use
|-
|comfortable and super-smooth user interface, fine-tuned specially for Android
|-
|easy, intuitive configuration
|-
|simultaneous registration of multiple SIP accounts, have multiple accounts registered to receive incoming calls and switch the account used for outgoing calls without leaving the keypad
|-
|Android contacts integration. Easy to call anyone in your contacts via SIP
|-
|add new contacts directly from Groundwire
|-
|ability to generate DTMF tones while in call, to control various PBX features or automated systems (use audio, rfc 2833 or SIP INFO)
|-
|speakerphone support
|-
|detailed call history
|-
|configurable RTP port range
|-
|SIP Proxy support
|-
|STUN server support, automatic service discovery using DNS SRV queries
|-
|phone number resolution. We present the phone numbers in a convenient format with grouped digits, display the flag and country name and even region/city name for some countries, including USA
|}

== Configuration Details ==

The configuration will be done in only 3 steps. Once you have downloaded GroundWire from Google Play/Apps Store, you will need to start the application.

On the initial start up for Android, the Acrobits Groundwire App will require some rights such as access to your microphone and call information to be able to fully use this application.

[[File:Acrobits_Groundwire_01.png|200px]] [[File:Acrobits_Groundwire_02.png|200px]] [[File:Acrobits_Groundwire_03.png|200px]] [[File:Acrobits_Groundwire_04.png|200px]] [[File:Acrobits_Groundwire_05.png|200px]] [[File:Acrobits_Groundwire_06.png|200px]]

Once this is done, you have to create your first SIP account. In this section, VoIP.ms has been certified by Acrobits and we are in the provider list. Simply type "VoIP.ms" into the upper search box and click on it to start the configuration.

[[File:Acrobits_Groundwire_Search.png|300px]]

In this section, you will need to fill the fields with your main account/sub account information. You will also need to specify the server that you want to be registered on.

[[File:Acrobits_Groundwire_AccountConfig.png|300px]]

*'''Title''': You will be able to rename this field if you plan to use more than one account on this apps.
*'''Username''': You main SIP account or sub account username
*'''Password''': The password you set for the account / sub account
*'''Domain''': One of VoIP.ms multiple [[Choosing Server#Choosing_a_Server | servers]], you can choose the one closest to your location.
*'''Advanced Settings: Secure Calls''': Set to "Disabled" unless you already have "Encrypted SIP Traffic" set to yes in your main Voip.ms account, and the sub account advanced settings also set to "Encrypted SIP Traffic" - Yes.
*'''Advanced Settings: Transport Protocol": Set to UDP, or if using Secure Calls, then set to "tls (sips)"

Finally, click on the upper right '''"Save"''' button and you will be able to start using your VoIP.ms account with Acrobits GroundWire softphone apps.

[[File:Acrobits_Groundwire_vms.png|300px]]

If you need assistance during the configuration, don't hesitate to contact VoIP.ms support.

==Call Encryption - TLS/SRTP==

To use [[Call_Encryption_-_TLS/SRTP#Configuration_on_SIP_Client | encrypted calls (TLS)]], once your main Voip.ms account, or any sub accounts have been properly configured, the following setting must be set in Groundwire:

* Open the advanced settings of the account to configure for TLS:
** Settings -> Accounts -> (select the account) -> Advanced Settings

* Once in Advanced Settings:
** Set '''Transport Protocol''' to “tls (sip)”
** Go in '''Secure calls''' setting.
*** In the '''SEDS (RFC 4568)''' section, set both '''''Incoming Calls''''' and '''''Outgoing Calls''''' to “Required”.
* Validate all these changes, and your secured call connection (TLS) should register as expected.

[[File:Acrobits_Groundwire_transport_protocol.jpg|200px]] [[File:Acrobits_Groundwire_SDES.png|200px]]

== Outbound CallerID Name ==
Under the apps settings, click on "Accounts" then on the SIP account you need to add an Outbound Caller ID to. Click on "Advanced settings". Scroll down to "Caller Identification".

[[File:Acrobits_Groundwire_3dot.png|200px]] [[File:Acrobits_Groundwire_setting.png|200px]] [[File:Acrobits_Groundwire_settingsVMS.png|200px]] [[File:Acrobits_Groundwire_settingsVMS_adv.png|200px]] [[File:Acrobits_Groundwire_menuCID.png|200px]]

== Voicemail settings ==

If you would prefer to have easy access to your Voicemail instead of dialing *97/*98, you can set your voicemail number directly in your settings.
Doing this, you will be able to click directly on a "voicemail" button directly from your dialpad when you have a voicemail.

Under the app's settings, click on "Accounts", then the SIP account you need to add a Outbound Caller ID. Click on "Advanced settings". Scroll down to "Voicemail Settings"

[[File:Acrobits_Groundwire_3dot.png|200px]] [[File:Acrobits_Groundwire_setting.png|200px]] [[File:Acrobits_Groundwire_settingsVMS.png|200px]] [[File:Acrobits_Groundwire_settingsVMS_adv.png|200px]] [[File:Acrobits_Groundwire_menuVM.png|200px]] [[File:Acrobits_Groundwire_menuVMon.png|200px]]

== Codec G.729a ==
The codec G.729a is an advanced biterate codec that requires less bandwidth, ideal if you have only access to a 3G network. This codec requires you to purchase the license to be able to used it. To do so, you need to go on your in-apps settings, above "Application settings" click on "Complement", then under Codecs "G.729 Codec of Annexe A", you will be able to purchase it.

[[category:guides]]

File:Acrobits Groundwire SDES.png

2024-02-11T02:46:05Z

Aleclerc:

File:Acrobits Groundwire transport protocol.jpg

2024-02-11T02:45:28Z

Aleclerc: uploaded a new version of "File:Acrobits Groundwire transport protocol.jpg"

File:Acrobits Groundwire transport protocol.jpg

2024-02-11T02:43:29Z

Aleclerc:

Grandstream HandyTone 802 - HT802

2024-02-11T02:16:31Z

Aleclerc: Added instructions to set call encryption (TLS)

[[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)

* '''''Name''''': Outbound callerID Name* '''See the requirements below.'''

* '''''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

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

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

====Call Encryption - TLS/SRTP====

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

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

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

====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]

[[Category:Analog Telephone Adapters]]