VoIP.ms Wiki - User contributions [en]

Yealink SIP-T28P

2019-09-04T15:39:25Z

Jannatul:

[[File:Yealink-sip-t28p.jpg|300px|thumb|left|Yealink SIP-T28P]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Configuration Details==

'''Get the IP Address'''

* Press the '''OK''' button
* The IP Address should appear on screen

'''Web Management'''

* Open your web browser and go to <nowiki>http://<ipaddress></nowiki>
* Login to web management (Default User : admin, default Password : admin)

'''Account Settings'''

* Go to '''Account''' tab
* Chose one account
* Fill in the following values:
** '''Account Active:''' On
** '''Label:''' 100000 (your VoIP.ms username)
** '''Display Name:''' John Smith (your name)
** '''Register Name:''' 100000 (your VoIP.ms username)
** '''User Name:''' 100000 (your VoIP.ms username)
** '''Password:''' ********* (account password)
** '''SIP Server:''' atlanta.voip.ms (one of our multiple servers)

'''Advanced'''
* '''Subscribe Period:''' 300
''300 is also to be used for Login Expire(seconds), to be changed from 3600''

[[File:Yealink_wib.png|594px|Yealink account settings]]

* Click on '''Confirm''' button to save your settings.

'''Preventing Direct IP Scanner Calls from 100'''

[[File:YealinkSipTrust.png|594px|Yealink account settings]]

Please make sure you have the latest firmware upgrade and look for and Enable the setting "Accept SIP Trust Server Only" which is at the bottom of the Advanced Tab under Account.

The newer firmware also has an option in the General Settings called "Allow IP Call", please set this to "Disabled".

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

User Manual : [http://support.yealink.com/attachmentDownload/download?path=upload%2Fattachment%2F2015-3-12%2F2%2Fe4dcfce0-a574-41a0-95bf-08370826f22d%2FYealink+SIP-T28P+Datasheet.pdf Download PDF]

[[category:SIP phones]]

Snom C520

2019-09-04T15:17:51Z

Jannatul:

[[File:snom_c520.png|250px|thumb|left]]
<br clear="all" />

The [https://www.snom.com/en/ip-phones/conferencing/c520 Snom C520 WiMi] 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 '''C520''' 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:snom_c520_main.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:snom_c520_selectAccount1.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:snom_c520_GralAccSett.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:snom_c520_Settings2.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:snom_c520_registered.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:snom_c520_SettingsTLS.png|thumb|none|400px]] </li>
<li style="display: inline-block;"> [[File:snom_c520_Signaling_TLS.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.

[https://helpdesk.snom.com/support/solutions/articles/6000191184-how-to-factory-reset-c520-conference-phone Source]

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

User Manual : [https://snom-website-data2.s3.amazonaws.com/filer_public/54/c4/54c43ec8-280d-4bea-bacc-37a3e42aba92/um_c520_en.pdf Download PDF]

[[category:SIP phones]]

Siemens Gigaset C450-Ip

2019-09-04T14:08:10Z

Jannatul:

[[File:Siemens_gigaset_c450_IP.jpg|300px|thumb|center|Siemens Gigaset C450-Ip]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Getting Started==
*Launch the web browser on your computer
*Enter the phone's IP address in the address field of the web browser, e.g. http://192.168.1.10
*Press the return key

A connection is established to the phone's web configuration interface

You can check the phone's current IP address on the handset following:
Menu -> Settings -> VoIP configuration -> IP Configuration -> IP Address

==Configuration Details==

'''Web configuration'''

* Select '''Telephony''' In the Web Configuration Page
* Select '''VoIP'''
* Enter the following values:
** '''Authentication Name:''' 100000 (your VoIP.ms username)
** '''Authentication password:''' *********** (account password)
** '''Confirm your password:''' ***********
** '''Username:''' 100000
** '''Domain:''' houston.voip.ms (one of our multiple servers)
** '''Proxy Server Address:''' houston.voip.ms
** '''Proxy Server Port:''' 5060
** '''Registrar Server:''' houston.voip.ms
** '''Registrar Server Port:''' 5060
** '''STUN enabled:''' NO
** '''Outbound proxy mode:''' Never

[[File:Siemens c450ip.jpg|Siemens Gigaset C450-IP Configuration]]

* Click on '''Set''' button to save your settings

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

User Manual : [http://gse.gigaset.com/fileadmin/legacy-assets/A31008-M1713-F101-3X-7219_it_CHE.pdf Download PDF]

[[Category:SIP phones]]

Polycom VVX 300, 400, etc

2019-09-04T11:43:41Z

Jannatul:

[[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).'''

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

==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 (Supposedly your Call ID if you are not overriding it on VoIP's side)<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>

<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.

==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]

Polycom SoundPoint IP 601

2019-09-04T11:40:01Z

Jannatul:

Polycom SoundPoint IP 601 [[File:Voipms-polycom601.jpg|200px]]

GENERAL INFORMATION
We will configure the SoundPoint IP 601 IP Phone to register to the Voip.ms servers and allow you, the end user, to place and receive calls normally.

You can also use this guide to configure similar Polycom products.

[http://www.polycom.com/common/documents/support/user/products/voice/soundpoint_ip600_601_user_guide_sip2.0.pdf SoundPoint 601 User Guide]

Configuring the Polycom SoundPoint IP 601
In this guide we will go through configuring the SoundPoint IP in order to register it to the Voip.ms servers so that you will be able to place and receive calls using your Voip.ms account. Before moving forward please ensure you have properly added the device to your network and that you have acquired the proper IP address. The phone should be turned on and running and you should have access to the device's configuration menus.

NOTE: As the SoundPoint IP is a standard IP Phone and not an IP PBX it will not be able to do standard call transfers. You will be able to join and conference your calls, however transferring a call may not be possible without the SoundPoint IP being involved, simply due to the way the device performs call transfers.

*Additionally please keep in mind that call transfer is a feature which would work best with an IP PBX like solution.

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

'''STEP 1''' Logging into your device
First we will acquire the IP address of your device, if you haven't already done so. You can acquire the IP of your device by:

Hitting the Menu button on the physical phone
Selecting Status
Selecting Network
Selecting TCP/IP Parameters

Your IP should be shown to you, along with other network information. If your IP is 0.0.0.0 please make sure that your device is properly connected to your network.

Once you have acquired your IP login to your device by using http://IP Address/. You should see a page similar to the following:

[[File:Voipms-Polycom-config.jpg]]

'''STEP 2''' Configuring your Voip.ms account.
We will first configure your Voip.ms account on your desired Line number. Please select the line by clicking the Line link and then entering the necessary credentials in your desired line number. Please use the information below to assist you:

'''Identification'''

Display Name: Your Desired Name, or VoipMs

Address: 100000 (Replace with your Main SIP account or Subaccount UserID, e.g. 198765 or 198765_sub)

Auth User ID: 100000 (Replace with your Main SIP account or Subaccount UserID, e.g. 198765 or 198765_sub)

Auth Password: Enter your SIP Password here. Your SIP password is the same password you created when you signed up for your Voip.ms account.
Optionally you may change it in your Account Settings/Security Tab.

Label: The name you would like this account to have on your phone's display screen

'''Server 1'''

Address: atlanta.voip.ms

Port: 5060

Transport: UDPonly

Expires: 60

Register: 1

And please leave the Server 2 information blank

Make sure to click Submit to save your changes. Wait for your device to reboot and then wait up to 60 seconds before attempting to access the web interface. Please NOTE your device may take upwards of 5 minutes to become fully accessible through the browser again. Please allow this much time to pass before determining that an error has occurred.

You can view an example configuration in the image below:

[[File:Soundpoint_601_config02.jpg]]

'''STEP 3''' Codec and DTMF
Now we will configure the codecs you will use with the Voip.ms services.
From the top of the page select General.
Then select Audio Processing.
Now set the priorities according to the information below:

'''Codec Preferences'''

G711Mu: 1

G711A: 2

G729AB: 3

'''NOTE:''' Out of these codes, only G711Mu is supported by Voip.ms. Therefore, you must make sure that your account has this specific codec enabled. This can be checked at Main Menu > Account Settings > Advanced for the Main Account and at Sub Accounts > Manage Sub Accounts > Edit > Advanced Options for the Sub Account.

You will then want to make sure the Payload Size is set to 20 for each codec's profile.
Here is an Example you can go by.

[[File:Polycom-601-payload.jpg]]

Now click on Submit on the bottom of the page and once your phone reboots it should register properly to Voip.ms and you can then make test calls.

'''STEP 4''' You`re Done! You can now make a phone call.
You can make a test call to 4443 (Echo Test), or if you have credit you can place a call to a traditional land line or mobile phone by dialing either:
the area code and number for calls to the US or Canada
Or
011 + the country code, area code, and number for calls worldwide (you may also use 00 instead of 011).

''NOTE'': If you have issues reaching an internal extension, make sure to add the following string in your dial plan: [1-9]xx

==Configuring Voicemail Messages==

Please go to the Message Center Section and put the following values.

Subscriber: [blank]

Callback Mode: Contact

Callback Contact: *97

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

Admin Manual : [https://support.polycom.com/content/dam/polycom-support/products/voice/soundstation-ip-series/user/en/soundpoint-ip-soundstation-ip-administrators-guide-sip-3-0-2-english-rev-a3.pdf Download PDF]

Polycom SoundPoint IP 501

2019-09-04T10:15:44Z

Jannatul:

[[File:Polycom501.jpg|258px|thumb|left|Polycom SoundPoint IP 501]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Configuration Details==

'''Get the IP'''

Start the Phone and it will get an IP from the DHCP Server, then it will display the IP on the Screen.

'''Configure Settings'''

* Enter the IP address on your web browser (PC and Device should be on the same network)
* The browser should load the configuration page for the IP 501
* Click on Sip Link
* Enter User and Password (default values are user: Polycom, pass: 456)
* Enter the Following values:

[[File:Poly501.gif|Settings]]

* Click on Submit. This will save settings and reboot unit.
* After the Device has rebooted, click now on Line Link and set your Identification values:

[[File:Poly501b.gif|Settings]]

* Click on Submit. This will save settings and reboot unit.
* After the Device has rebooted, your Device will be ready to be used with VoIP.ms

''NOTE'': If you have issues reaching an internal extension, make sure to add the following string in your dial plan: [1-9]xx

==Configuring Voicemail Messages==

Please go to the Message Center Section and put the following values.

Subscriber: [blank]

Callback Mode: Contact

Callback Contact: *97

[[category:SIP phones]]

Please click [http://www.voip-info.org/wiki/view/Polycom+SoundPoint+IP+MWI+audio here] to find out how to get rid of the MWI Audible Sound

==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/soundpointip/user/en/soundpoint-ip500-501-quick-user-guide.pdf Download PDF]

Admin Manual : [https://support.polycom.com/content/dam/polycom-support/products/voice/soundstation-ip-series/user/en/spip-ssip-vvx-admin-guide-sip-3-2.pdf Download PDF]

Grandstream GXP2200

2019-09-04T09:15:09Z

Jannatul:

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

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

==Enter the Grandstream GXP2200 Configuration Panel==

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

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

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

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

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

*'''Account Active:''' Checked
*'''Account Name''' Enter a name of your choice (it can be your name or company name)
*'''SIP Server:''' atlanta.voip.ms (Pick any one of VoIP.ms multiple [[Choosing Server|servers]])
*'''SIP User ID:''' 100000 (Replace with your Main SIP account or Subaccount UserID, e.g. 198765 or 198765_sub)
*'''SIP Authenticate ID:''' 100000 (Replace with your Main SIP account or Subaccount UserID, e.g. 198765 or 198765_sub)
*'''SIP Authenticate Password:''' ********* (Use the SIP account password-By default the same as the Customer Portal)
*'''Voice Mail UserID:''' *97
*'''Name:''' Enter a name of your choice (it can be your name or company name)
: If you want to register your Main account, you can find the Username and Password information by logging into your [https://www.voip.ms/m/samples/voxalot.php Customer Portal].
: If you want to register a sub-account you will have to use the credentials that you set when you created that sub-account.
[[File:GXP2200.png|800px]]
<div style="width:100%;overflow:hidden;clear:both"></div>

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

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

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

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

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

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

[[category:SIP phones]]

Grandstream GXP2135

2019-09-04T09:11:15Z

Jannatul:

[[File:GXP2135-device.jpg|300px|thumb|left|Grandstream GXP2135]]
<div style="width:100%;overflow:hidden;clear:both"></div>

A high-profile desktop phone built for the busy user. The GXP2135 is a sleek High-End IP phone that will bring higher levels of communication productivity to a network. Its modern design, high call capacity, and rich functionality make it the ideal choice for the call-intensive worker.

__TOC__

----

'''SIP ACCOUNT'''

Once you have connected your device to your local connection, it will be necessary to obtain the local IP address that was designated for it.

-From your IP phone navigate through the menus to obtain its IP address:

'''Menu >> Status >> Network Status >> IPv4 Address'''

-Now that you have the Local IP address, open it in your web browser of preference.

[[File:Pasteipchromegxp2135.PNG|thumb|none|600px]]

-Once you have opened the login page, use the following credentials to access to the configuration:

'''Username: admin'''

'''Password: admin'''

[[File:Loginadmingxp2135.PNG|thumb|none|600px]]

=== General Settings ===

-Once you see the settings page, go to the following:

'''Account >> Account # >> General Settings'''

<div><ul>
<li style="display: inline-block;">[[File:Accountsgxp2135.PNG|thumb|none|600px]]</li>
<li style="display: inline-block;">[[File:Account1gxp2135.PNG|thumb|none|600px]]</li>
<li style="display: inline-block;">[[File:Generalgxp2135.PNG|thumb|none|600px]]</li>
</ul></div>

-Change the following settings according to the information of your account:

'''A. Account Name''': The name for this account that will be displayed on the LCD screen of your phone.

'''B. SIP Server''': The server that you will use for the registration of this device. (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'''

'''C. SIP User ID''': The account or sub-account number that you will be using for this line.

'''D. Authenticate Password''': The password for the account or sub-account you will be using for this line.

'''E. Name''': The name that will be used as Caller-ID Name. ('''Notice that this only applies for Canada''')

'''F. Save and Apply'''

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

=== Basic SIP settings ===

Once this is done, we will need to set the time that the device will send the information for its registration status.

-Please go to the following:

'''Account # >> SIP Settings >> Basic Settings'''

<div><ul>
<li style="display: inline-block;">[[File:Account1gxp2135.PNG|thumb|none|600px]] </li>
<li style="display: inline-block;">[[File:Sipsettingsgxp1.PNG|thumb|none|600px]] </li>
<li style="display: inline-block;">[[File:Basicsettingsgxp01.PNG|thumb|none|600px]] </li>
</ul></div>

- Change the following:

'''SIP Registration:''' Yes

'''Register Expiration:''' 5 (this is in minutes)

'''Enable OPTIONS Keep Alive:''' Yes

Finally, click '''Save and Apply'''

===Common errors===

====Outgoing calls issue====
If you are able to receive incoming calls but outgoing calls are failing with error "No response", make the following changes to fix it:

1. Login into your device's settings and head to '''Accounts > Account X > SIP > Custom SIP Header''' and disable:

*'''Use X-Grandstream-PBX Header'''
*'''Use P-Access-Network-Info Header'''
*'''Use P-Emergency-Info Header'''

[[File:SIP Headers GXP.png|thumb|none|600px]]

2. Go to '''Accounts > Account X > SIP > Audio Settings''', and choose codec '''G729A/B''' as preferred Vocoder and the rest with '''PCMU'''.

[[File:Audio codecs GXP.png|thumb|none|600px]]

==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/gxp2130_gxp2140_gxp2160_gxp2135_gxp2170_quick_user_guide_english..pdf Download PDF]

Admin Manual : [http://www.grandstream.com/sites/default/files/Resources/gxp21xx_administration_guide.pdf Download PDF]

Grandstream GXP2120 IP Phone

2019-09-04T09:06:20Z

Jannatul:

[[File:Gxp2110.png|300px|thumb|left|Grandstream GXP2120 IP Phone]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Configuration Detail==

'''Step 1:'''

You must first determine what IP address the IP Phone was assigned. To do this, press the round menu button, then the down arrow next to it until the display shows:

[2] IP Addr

Now, press the round menu button again and you will be shown the IP address of your device (e.g. 192.168.0.100).

'''Step 2:'''

Open your web browser and go to the IP you just found: https://[DeviceIP]

'''Step 3:'''

You will be prompted to enter a password, the default is "admin", then click the "Login" button.

'''Step 4:'''

Click on '''ACCOUNT''', followed by the '''ACCOUNT 1''' (to configure your line 1). You need to modify only a few settings from the factory default.

Fill in the followings fields.

* '''SIP Server:''' atlanta.voip.ms (You can enter the address of any of our multiple servers, this is just an sample with Atlanta server)
* '''SIP User ID:''' 100000 (Replace with your VoIP.ms username, it is a six digit number)
* '''Authenticate Password:''' ********* (Type your VoIP.ms account Password)
* '''SIP Registration:''' Yes
* '''Register Expiration:''' 5

'''Step 5:'''

Finally, please click the '''Update''' button and then '''Reboot''' at the bottom of the configuration screen.

[[File:Grandstream2021final1.jpg|800px|]]

==Ghost Calls==

If you receive Weird Calls such as from CallerID 100 or in the middle of the Night not showing in your CDR, please note that these calls are not going through our Network but rather through the internet directly to your IP Phone without authentication.

Users have the ability to deny calls that are not authenticated. You can find the option by navigating the phone web interface, clicking on Account X-->SIP Settings-->Security Settings and enabling "Authenticate Incoming invite"
You can also go to Account X -> SIP Settings -> Security Settings -> and enable "Accept Incoming SIP from Proxy Only"

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

User Manual : [http://www.grandstream.com/sites/default/files/Resources/gxp21xx_14xx_usermanual_english.pdf Download PDF]

Grandstream DP750/DP720

2019-09-02T17:14:51Z

Jannatul:

== Grandstream DP750 Configuration Detail ==

[[File:DP750-720.png|thumb|none|600px]]

1. Type the DP715 Base Station IP address in your PC browser e.g. ''http://192.168.2.1''

2. Log in using username and password of "admin" to configure the DP750 Base Station.

3. Navigate to the Profiles > Profiles 1, and select ' '''General Settings''' '.

[[File:DP750-Profile1-General.png|thumb|none|600px]]

* Select a name. This is only for your own use.
* One of VoIP.ms multiple [[Choosing Server | servers]], you can choose the one closest to your location
* Click on the '''Save and Apply''' button.

4. Navigate to the SIP Settings > Basic Settings menu on the left.

[[File:Sip_-_Basic_Settings_-_1.png|thumb|none|600px]]

* Select either '''UDP''', or if your account is configured to use [[Call_Encryption_-_TLS/SRTP | SRTP encryption]], select '''TLS/TCP'''.
* Set the Local SIP Port to 5061.
* Click on the '''Save and Apply''' button.

[[File:Profile_1_sip_basic_2.png|thumb|none|600px]]

5. Navigate to the Profiles > Profile 1 > '''Audio Settings'''.

[[File:Profile_1_Audio_Settings.png|thumb|none|600px]]

* Select both '''In-Audio''' and '''via RTP''' for the send DTMF option.

[[File:Profile_1_Audio_2.png|thumb|none|600px]]

* If you are using '''SRTP / TLS/TCP''', ensure the '''SRTP Mode''' option is set to '''Enabled and Forced'''.
* Click on the '''Save and Apply''' button.

6. Navigate to '''DECT''' > '''Sip Account Settings'''.

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

* In the Account 1 line, enter your SIP User ID into both the '''SIP User ID''' and '''Authenticate ID''' fields.
* Enter the password for the SIP account in the password field.
* Enter a name for the SIP account in the Name field. This name will be displayed on the handsets.
* Click on the '''Save and Apply''' button.

7. Navigate to '''DECT''' > '''Handset Line Settings'''.

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

* Ensure the SIP account that you created in step 6 is select in the '''Line 1''' column for each handset.
* Click on the '''Save and Apply''' button.

==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/DP750_DP720_User_Guide.pdf Download PDF]

Admin Manual : [http://www.grandstream.com/sites/default/files/Resources/DP750_DP720_Administration_Guide.pdf Download PDF]

Grandstream DP715/DP710

2019-09-02T17:10:44Z

Jannatul:

[[File:Grandstream 715-710.png|400px|thumb|left|Grandstream DP715/DP710]]
<div style="width:100%;overflow:hidden;clear:both"></div>

== Grandstream DP715 Configuration Detail ==

Configure using Interactive Voice Prompt:

1. From the registered Handset, press *** to get into the IVR
menu. Enter option 02 to obtain the DP715 IP address.

2. Type the DP715 Base Station IP address in your PC browser e.g. http://192.168.2.1

3. Log in using password “admin” to configure the DP715 Base Station.

[[File:Grandstream7xxlogin.png|400px]]

Once that you login you will see the following window:

[[File:Granstream7xxstatus.png|400px]]

4. Click on '''PROFILE 1''' to configure your first line you will see the following window:

[[File:DP750correction.png|600px]]

Complete the followings fields.

* '''Primary SIP Server:''' One of our multiple servers (e.g. atlanta.voip.ms )
* '''Register Expiration:''' Set to 5 minutes (300 seconds)

Then go to the bottom and press "Update"

5. Click on '''HANDSETS''' to configure your Handsets, you will see the following window:

[[File:Granstream7xxHandsets.png|600px]]

Complete the followings fields.

* '''Enable Handset''' Set to Yes
* '''SIP User ID:''' Your VoIP.ms SIP username (e.g. 100000)
* '''Authenticate ID:''' Your VoIP.ms SIP username (e.g. 100000)
* '''Authenticate Password:''' ********* (account password)
* '''Name:''' Your Name

Then go bottom and press "Update"

===Setting Your Time Zone===

Click on the " Basic Settings " tab link at the top of the Grandstream Device Configuration window.

Scroll down to the bottom of the Basic Settings page to the Time Zone drop-down list.

From the drop-down menu list, select your local Time Zone.

Click on the Update button at the bottom of the configuration window to save your changes.

===Voice Mail Waiting Indicator Or MWI===

By default, the MWI indicator is disabled. In which case, the only way you know you have a voice mail is by picking up the handset and pressing the off-hook button. Normally you will just hear a dial tone. But, when you have voice mail waiting, you will hear a stutter tone.

To enable MWI indicator on the DP715 base unit:

*Login to the phone via web browser.
*Navigate to the Advanced Settings page.
*Scroll down to the very bottom of the Advanced Settings page.
*Change the MWI Blinking setting from Disable to Enable.
*Click the Update button. Then Reboot.

Once done, you need to associate a mailbox to the registered account from the VoIP.ms customer portal. For the main account it is done from Main Menu>>Account Settings>>General Tab and for a sub account it is chosen as the internal extension voicemail from Sub Accounts>>Manage Sub Accounts>>Edit. Next time you receive a voice mail, you will see the base unit's green LED (handset off-hook indicator) will blink. When the voicemail is deleted, the handset off-hook indicator will no longer blink.

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

User Manual : [http://www.grandstream.com/sites/default/files/Resources/dp71x_usermanual_english.pdf Download PDF]

Gigaset A510 IP

2019-09-02T17:03:58Z

Jannatul:

[[File:Gigaset_a510_IP.jpg|400px|thumb|left|Gigaset a510 IP]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Configuration Detail==

'''Connect the Gigaset a510 IP'''

1. Connect your router with the supplied Ethernet network cable to the ethernet port found on the Gigaset's base station. Finally plug the supplied power cable into the base station. Wait 60 seconds after plugging the base station's power adapter.

2. Press the Paging button located on the face of the base station, as indicated by the screen shot below:

[[File:gigaset_a510_paging.png|300px]]

3. Once pressed, the IP address of the base station should be displayed on the handset, as indicated below:

[[File:gigaset_a510_IPshow.png|100px]]

4. Open a web browser on your computer such as Internet Explorer, or Firefox and enter the IP address you saw on the phone's interface. We will now login to your IP Phone. You should now see a page that looks like this:

[[File:gigaset_a510_web1.png|600px]]

5. Enter the system PIN for the A510 IP on the given field. The default system PIN for the A510 IP is 0000. After entering the system PIN and clicking the OK button, you should see a screen that looks similar to the one below:

[[File:gigaset_a510_web2.png|600px]]

6. We will now check your IP configuration. On the Home page, click on the Settings tab on the top menu bar. Now under the Settings configuration page, click on the Telephony link on the side menu bar as shown screenshot below:

[[File:gigaset_a510_web3.png|600px]]

7. To configure your VoIP.MS account settings, please select the first available line, indicated by the Not configured status:

[[File:gigaset_a510_web4.png|600px]]

8. Once you have selected your desired line click on the Show Advanced Settings button, and use the following settings.

'''Connection Name or Number:''' VoIP.MS

'''Authentication name:''' 100000 (your VoIP.ms username)

'''Authentication password:''' ********* (account password)

'''Username:''' 100000 (your VoIP.ms username)

'''Display name:''' 100000 (your VoIP.ms username)

'''Domain:''' atlanta.voip.ms (one of our multiple servers)

'''Proxy server address:''' atlanta.voip.ms (one of our multiple servers)

'''Proxy server port:''' 5060

'''Registration server:''' atlanta.voip.ms (one of our multiple servers)

'''Registration server port:''' 5060

'''Registration refresh time:''' 300

'''STUN enabled:''' No

'''STUN server address:''' (Leave Empty)

'''Outbound proxy mode:''' Always

'''Outbound server address:''' atlanta.voip.ms (one of our multiple servers)

'''Outbound proxy port:''' 5060

'''Select Network Protocol:''' UDP Only

9. After configuring your settings click the '''Set''' button to save your changes. Your configuration should look similar to the one shown in the picture below:

[[File:gigaset_a510_web5.png|600px]]

10. To receive calls properly you need to configure your Gigaset IP Phone. Click on '''Number Assignment''' link, and ensure both the radio button and the check box have been marked for the settings for outgoing calls and for incoming calls next to VoIP.MS, as shown below:

[[File:gigaset_a510_web7.png|600px]]

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

User Manual : [https://gse.gigaset.com/fileadmin/legacy-assets/A31008-M2230-R301-2-6019_en_US_CA.pdf Download PDF]

[[category: Analog Telephone Adapters]]

Cisco SPA504G Phone

2019-09-02T16:23:05Z

Jannatul:

[[File:Cisco_spa504g.jpg|300px|thumb|left|Cisco SPA504G Phone]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Documentation==
[https://www.cisco.com/c/dam/en/us/td/docs/voice_ip_comm/csbpipp/ip_phones/user/guide/50X_sip_user_guide_source/500_SIP_user.pdf User's Guide]

==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 versions of Cisco IP 'phone (typically multi-line devices from the SPA30x and SPA50x series) 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 dialing any codes.

'''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:''' 300<br>
'''Proxy Fallback Intvl:''' 300<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

(On multi-line 'phones like the SPA30x/50x this tab is labelled 'Ext1', 'Ext2', 'Ext3'. The initial defaults assign all individual line keys to 'Ext1' settings. Labels are specified on the 'Phone' tab)

'''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 (or Ext1 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. The IP 'phone may take up to 35 seconds to re-initialise.

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

Admin Manual : [https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/csbpipp/ip_phones/administration/guide/frame_source/500-series-7-5-7/ip_phone_admin.pdf Download PDF]

== 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. Multiline IP 'phones are easier to configure as the dialplan-translated number is displayed on the LCD when completing the call but the syntax is identical.

* You can check the most commonly used Star Codes from [[Cisco/Linksys Star Codes]]

[[Category: SIP phones]]

Cisco IP Phone 7940/7960

2019-09-02T16:10:52Z

Jannatul:

[[File:Cisco_7960g.png|300px|thumb|left|Cisco IP Phone 7940/7960]]
<div style="width:100%;overflow:hidden;clear:both"></div>

En Español: [[Cisco IP Phone 7940/7960 (Español)]]

'''Note:''' These devices need to have the latest firmware in order to have the SIP Configuration options mentioned on this article. For more details please contact your Cisco Service or review the following links:

* Cisco Download Software [http://software.cisco.com/download/navigator.html?mdfid=269065653]
* Converting a Cisco 7940/7960 CallManager Phone to a SIP Phone[http://www.cisco.com/en/US/products/hw/phones/ps379/products_tech_note09186a0080094584.shtml]

==Configuration from the phone's settings menu==

#Press '''**#''' to unlock the '''Configuration Protection'''
#*Some firmware versions must be unlocked by pressing the '''"Settings"''' button, scrolling down to the '''"Unlock Phone"''' option and pressing the '''"Select"''' softkey
#Press '''Settings''' button
#Go to '''"Network Configuration"''' to set the IP connectivity
#Go to SIP Configuration
#Select '''"Line 1"'''
##In '''Name''' and '''Authentication Name''' enter your VoIP.ms username(e.g. 100000)
##Enter your VoIP.ms password
##Enter any of the VoIP.ms servers (e.g. atlanta.voip.ms) as '''Proxy Address'''
##Configure '''Preferred Codec''' to g711u
##In Settings > SIP Configuration > Register Expires (You need to set a timeout < 3600s, please put 180)
##In Settings > SIP Configuration > Register with Proxy set this to YES
##Press '''Back'''
##Press '''Save''' to apply changes.

'''As an additional recommendation, Set "NAT enabled" to Yes, and on the "NAT address" parameter, enter the External (publicly addressable) IP address of Router or firewall, if any. '''

'''Important: Please make sure your timer_register_expires is set to 180 seconds. Lowering the setting to 180 seconds refreshes the SIP invite and creates a new dynamic route every 180 secs. The option is also available directly on the phone menu through: Settings -> SIP Configuration -> Register Expires'''

== External Links ==

Using the Cisco SIP IP Phone Menu Interface[http://www.cisco.com/en/US/docs/voice_ip_comm/cuipph/7960g_7940g/sip/1_0/english/administration/guide/install.html#wpxref97227]

Cisco SIP IP Phone 7960 Administrator Guide
[http://www.cisco.com/en/US/docs/voice_ip_comm/cuipph/7960g_7940g/sip/1_0/english/administration/guide/admingd.pdf]

* You can check the most commonly used Star Codes from [[Cisco/Linksys Star Codes]]

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

User Manual : [https://www.cisco.com/c/dam/en/us/td/docs/voice_ip_comm/cuipph/7960g_7940g/7_0/sccp/english/user/guide/6040sccp.pdf Download PDF]

[[Category: SIP phones]]

Cisco SPA525G

2019-09-02T16:06:26Z

Jannatul:

[[File:525.jpg]]

The Cisco SPA525G 5-line IP Phone with Color Display is a full-featured VoIP (Voice
over Internet Protocol) phone that provide voice communication over an IP
network. It provides traditional features, such as call forwarding, redialing, speed
dialing, transferring calls, conference calling, and accessing voice mail. Calls can
be made or received with a handset, headset or speaker.

Your Cisco IP Phone provides a web interface for the phone user that allows you to
configure some features of your phone by using a web browser.

This article will guide you through the steps for basic configuration to make it work with VoIP.ms

----
<br>
<font size="4">'''Step 1'''</font>
<br>
'''''Get the IP address of your phone'''''

a. Press Setup.<br>
b. Select to Status > Network Status.<br>
c. Scroll to view IP Address. This is the IP address of your phone.<br>

You should now have a number which is similar to 192.168.xxx.xxx

<font size="4">'''Step 2'''</font><br>
'''''Logging in to the Phone Web User Interface'''''

*On your PC, open a Web browser window. Your PC must be on the same subnetwork as the phone.
*Enter the IP address in the browser address bar.

You will now see this screen:<br>

[[File:525 1.gif|center]]
<br>
*Click on the '''"Admin Login"''' button near the top right side of the screen, then click on the '''"Ext 1"''' tab.

[[File:525 2.gif|center]]

<font size="4">'''Step 3'''</font>
<br>
'''''Configure with your VoIP.ms account'''''

Find the following fields on the '''"Ext"''' tab and configure accordingly.

'''Nat Keep Alive:''' Yes<br>
'''Nat Mapping/Traversal:''' Yes<br>
'''Proxy:''' atlanta.voip.ms (one of the multiple VoIP.ms servers, you can choose the one closer to your location.)<br>
'''Register Expires:''' 300<br>
'''Display Name:''' John Smith (Replace with your name or company name)<br>
'''User ID:''' 100000 (Replace with your 6 digits Main SIP account UserID or Sub Account user name, e.g. 123456 or 123456_sub)<br>
'''Password:''' ******** (SIP Account Password)<br>
'''Use DNS SRV:''' NO<br>
'''DNS SRV Auto Prefix:''' NO<br>

[[File:cisco-spa525g.gif|center]]

*If a second extension is needed, click on '''"Ext 2"''' and repeat Step 3. Please make sure to increment the SIP port by one. For example, Ext 1 SIP port: 5060;<br> Ext 2 SIP port: 5061. Make sure you also click on Phone Tab to route the Extensions to the proper lines.

<font size="4">'''Step 4'''</font>
<br>

'''''Configure your dial plan'''''

This step can be considered optional however this is a dial plan that is optimized to work with VoIP.ms service.

Find the dial plan section of your line and enter the following string:

'''(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.)'''

<font size="2" color="blue">'''If done properly, after completing all these steps your phone will now be ready to place and receive calls!'''</font>

== '''Documentation:''' ==

* You can check the most commonly used Star Codes from [[Cisco/Linksys Star Codes]]

[http://www.cisco.com/en/US/docs/voice_ip_comm/csbpipp/ip_phones/user/guide/525g_sip_user_guide_source/spa525_sip_user.pdf User´s Manual]<br>

keywords: SPA525

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

Admin Manual : [https://www.cisco.com/c/dam/en/us/td/docs/voice_ip_comm/csbpipp/ip_phones/administration/guide/spa_wip_admin.pdf Download PDF]

Audiocodes 400HD

2019-09-02T05:59:29Z

Jannatul:

[[File:Audiocodes 420HD.jpg|300px|thumb|left]]
<br clear="all" />

Audiocodes 400HD series

The [https://www.audiocodes.com/solutions-products/products/ip-phones AudioCodes 400HD series] of IP phones is a range of easy-to-use, feature-rich desktop devices for the service provider hosted services, enterprise IP telephony and contact center markets. Based on the same advanced, field-proven underlying technology as our other VoIP products, AudioCodes high quality IP phones enable systems integrators and end customers to build end-to-end VoIP solutions.

We have tested and confirmed that the Audiophone's [https://www.audiocodes.com/solutions-products/products/ip-phones/405hd-ip-phone '''''405HD'''''] and [https://www.audiocodes.com/solutions-products/products/ip-phones/420hd-ip-phone '''''420HD'''''] are fully compatible with our services and offers you great flexibility and features

'''405HD''' running firmware version ''2.2.16.251''
'''420HD''' running firmware version ''2.2.16.251''

__TOC__

==Configuration Details==

In order to set up your new '''Audiocodes 420 HD IP phone''' you only need to get its IP address number and access to your device's settings from your web browser.

===Getting the IP address number===

As soon as you have plugged your phone into your local router, your phone will receive its own local IP address number. You can get this number by following these steps:

'''1.''' Press the '''MENU''' key located on the device.<br/>
'''2.''' Go to option '''5 "Device Status"''' and press '''OK''' button.<br/>
'''3.''' Choose option '''1 "Network Settings"''' and press '''OK''' button.<br/>
'''4.''' Scroll down to '''"IP Address"''' and take note of this number.

===Login into your device via Web Interface===
From your PC, open your favorite web browser and at the URL bar, type the IP address number you got from your phone. A pop-up window will show up asking for your phone's login credentials.

[[File:Sign in pop-up window.png|thumb|none|300px]]

*Username: '''admin'''<br/>
*Password: '''1234'''<br/>

You will reach the main page of your '''Audiocodes 420 HD IP phone'''

[[File:AudioCode420MainPage.png|thumb|none|700px]]

===Setting up your device with your SIP credentials===
From the left panel click on "Quick setup" folder and then at "Quick Setup" icon. Three sections will show up on the right panel, '''"LAN Setup"''', '''"SIP Proxy and Registrar"''' and '''"Line Settings"'''.

'''1.''' At '''"SIP Proxy and Registrar"''' we can set:

*'''Use SIP Proxy:''' Enable<br/>
*'''Proxy IP Address or Host Name:''' Type any of our [[Choosing Server | servers]]. i.e. denver.voip.ms<br/>
*'''Proxy Port:''' 5060<br/>
*'''Use SIP Proxy IP and Port for Registration:''' Enable<br/>
*'''Use SIP Registrar:''' Disable<br/>

'''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.

'''2.''' At '''"Line Settings"''' we can set: <br/>

*'''Line Number:''' 1<br/>
*'''Line 1 Activate:''' Enable<br/>
*'''Line 1 Display Name:''' Caller ID name field (Not guarantee so be sent)<br/>
*'''Line 1 User ID:''' 100000 (replace with your main VoIP.ms account number or sub-account name)<br/>
*'''Line 1 Authentication User Name:''' 100000 (replace with your main VoIP.ms account number or sub-account name)<br/>
*'''Line 1 Authentication Password:''' ********* (replace with your main SIP or sub-account password)<br/>

[[File:AudioCode420 SIPcredentials.png|thumb|none|700px]]

==Extra settings==
The following settings will improve your device's performance.

===Registration time and NAT Keep alive settings===
These settings will keep your device awake and ready to receive incoming calls all the time. From the left panel, go to '''Voice Over IP >> Signaling Protocols''' and from the right panel modify:

*'''Enable Registrar Keep Alive:''' Enable<br/>
*'''Registrar Keep Alive Period:''' 50 Seconds<br/>
*'''Registration Expires:''' 300 Seconds<br/>

[[File:AudioCode420 RegisterTime.png|thumb|none|700px]]

===Audio Codecs===
Make sure to enable only supported codecs by VoIP.ms. Go to '''Voice Over IP >> Media Streaming''', from the left panel. You can enable audio codecs '''G729''' and '''G711u''' as first and second option and set all the other options to '''none'''.

[[File:AudioCode420 AudioCodecs.png|thumb|none|700px]]

===Network Settings===

====Encryption TLS====
In order to use TLS along with your '''Audiocodes 420 HD IP phone''' 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 or 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|700px]]

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|700px]]

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

To enable '''TLS''' go to '''Voice Over IP >> Signaling Protocols''' and set these settings as follows:

*'''SIP Transport Protocol:''' TLS<br/>
*'''TLS Port:''' 5061<br/>
*'''SIP Local Port:''' 5081<br/>
*'''Proxy IP Address or Host Name:''' denver1.voip.ms:5061 (Use the same [[Choosing Server | server]] your phone number (DID) is at, you can check it out from your customer portal at Manage DIDs section, under '''POP''' column)<br/>
*'''Proxy Port:''' 5061<br/>
*'''Use SIP Proxy IP and Port for Registration:''' Disable<br/>
*'''Use SIP Outbound Proxy:''' Disable<br/>

'''Note:''' When using TLS is very important to specify the number of the server, in case the server you have chosen, doesn't use the number "1", you need to add it. Adding any of the SIP ports '''5061 | 5081 | 42873''' is also required.

[[File:AudioCode420 TLS.png|thumb|none|700px]]

To enable '''SRTP''' go to '''Voice Over IP >> Media Streaming''' and at '''SRTP''' section set the following settings:

*'''SRTP Encryption and Authentication:''' REQUIRE ENCRYPTION<br/>
*'''Method:''' AES_CM_128_ALL_METHODS<br/>
*'''Negotiation mode:''' Basic<br/>
*'''ARIA:''' Disable<br/>

[[File:AudioCode420 SRTP.png|thumb|none|700px]]

When your device is fully registered by using '''SIP-TLS''' protocol, you will be able to see the registration status in the portal Home page, at '''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|700px]]

[[File:TLS-SRTP-Steps0007.png|thumb|none|700px]]

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

User Manual : [https://www.audiocodes.com/media/9627/ltrt-11939-430hd-and-440hd-ip-phone-for-microsoft-skype-for-business-users-manual-ver-301.pdf Download PDF]

Admin Manual : [https://www.audiocodes.com/media/13525/400hd-series-ip-phone-for-microsoft-skype-for-business-administrators-manual-ver-312.pdf Download PDF]

Aastra 6730i VoIP Phone

2019-09-02T05:55:57Z

Jannatul:

[[File:Aastra6730i.jpg|300px|frame|left|Aastra 6730i VoIP Phone]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Documentation==

[https://www.mitel.com/en-ca/open-solutions/6730 Aastra 6730i User/Quick Reference/Installation Guides - Label Templates - Encryption Tools - Language Packs - Admin Guide - XML Api - Software Versions]

[https://www.mitel.com/fr-ca/open-solutions/6731 Aastra 6731i User/Quick Reference/Installation Guides - Label Templates - Encryption Tools - Language Packs - Admin Guide - XML Api - Software Versions]

==Configuration Details==

'''Get the IP Address'''

* Press the Spanner key
* Press the scroll down button to get to <b>Phone Status</b> and then press [ http://voip.ms/images/puce_gray.gif ] to confirm selection
* Select IP&MAC Addresses and press [ http://voip.ms/images/puce_gray.gif ]
* The IP Address will appear on the screen.

'''Web Management'''

* Open your web browser and go to the IP you just found: <nowiki>https://[DeviceIP]</nowiki>
* You will be presented with a warning, you need to confirm that you want to continue.
* You will be asked for user and password. Provide the user and password for the phone
** Username: Admin<br>
** Password: xxxxx
*On the left menu select <b>Line 1</b> under <b>Advanced Settings</b>
*Fill in the values as shown in the image.

[[File:Aastra6731i_det1b.png|Aastra 6730i VoIP Phone]]

'''Screen Name:''' John Smith (value of your choice)<br>
'''Phone Number:''' 100000 (Your VoIP.ms username)<br>
'''Caller ID:''' 100000 (Your VoIP.ms username)<br>
'''Authentication Name:''' 100000 (Your VoIP.ms username)<br>
'''Password:''' xxxxxxxxxxx (the account password)<br>
'''Proxy Server:''' atlanta.voip.ms (one of our multiple servers)*<br>
'''Proxy Port:''' 5060 (default sip port)

'''Backup Proxy Port:''' 0

'''Outbound Proxy Port:''' 0

'''Registrar Server:''' atlanta.voip.ms (one of our multiple servers)*<br>
'''Registrar Port:''' 5060 (default sip port)

'''Backup Registrar Port:''' 0
*According to some customers, using the server and the port number under '''Registrar Server''' help them obtaining registration with our server.
For example, atlanta.voip.ms:5060 .

* Click <b>Save Settings</b>.

'''Restart the Phone.'''
* Click on <b>Reset</b> under <b>Operations</b> on the left menu.
* Now click on <b>Restart</b>
* Phone will restart automatically
==Guide Links==
In the event where you need the guide directly from Aastra, you may find the admin manual guide below:

Admin Manual : [https://www.voipsupply.com/downloads/dl/file/id/38/aastra_6730i_install_guide.pdf Download PDF]

[[category:SIP phones]]

Auerswald COMpact 5010

2019-09-02T05:03:15Z

Jannatul:

[[File:Auerswald_5010.jpg|300px]]

After you have logged in to your device configuration screen, please enter the following information.

==Documentation==
[https://www.auerswald.de/en/pbx-systems/60-service/tk-anlagen/compacterserie/handbuch-compact-serie/548-handbuch-compact-5010voip-en.html Operator´s Manual / Installation / Configuration ]

==Configuration Details==

'''Step 1'''

Under '''''Voice Over IP (VoIP) >> Providers''''', enter the following:

'''Domain:''' london.voip.ms (Replace with the address of one of the multiple servers from VoIP.ms)

'''Registrar:''' london.voip.ms (Replace with the address of one of the multiple servers from VoIP.ms)

'''Port:''' 5060

'''Note:'''

* The (internal) SIP port set by the box, each configuration for a provider has it's own port
* The shown codec setting works only if G.711 and G.729 are selected in the portal, otherwise you have to select only these two codecs in the PBX configuration
* Format der eigenen Rufnummer (own number): With this setting, the PBX will send the correct caller ID.

[[File:Auerswald 5010 1.jpg|1000px]]

'''Step 2'''

Under the next tab '''''Voice Over IP (VoIP) >> Accounts''''', enter the following:

Select the provider that you have entered on Step 1 (Voipms), then under Kind of Connection select PTMP Connection, finally click the button Configure.

[[File:Aureswald 5010 2.png|1000px]]

Now on the following screen, set the following.

'''Account Name:''' The name that was chosen in Setup Account 1

'''Country Prefix:''' E.g. 001 for NAMP or 01141 for Switzerland

'''Area code:''' 201 for New York

'''Country:''' Technically the country that correspondence with the country prefix, however there is a back in this version of the firmware and only "Germany" can be chosen. It does not have any effect on the functionality

'''Exchange line access number (account number)''': The internal number in the PBX for this account. Can be used to dial-out directly and override the set up for external line preference

'''User name:''' 100000 (your VoIP.ms username)

'''Password:''' ********* (account password)

'''Authorization ID:''' Can be left blank

'''Multiple subscriber numbers (MSNs):''' The phone number(s) chosen/assigned by VoIP.ms

'''Display name:''' Assign a unique display name (16 characters max.) for the MSN

'''Ringer rhythm:''' Optionally, you can select the ringer rhythm that should be used for the MS

[[File:Aureswald 5010 3.png|1000px]]

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

Admin Manual : [http://www.auerswald-root.de/download/datei/handbuch/eng/COMpact50-VoIP/Konfigurationshandbuch/ver060811/KVoIPGB.pdf Download PDF]

Atcom AG188N

2019-09-02T04:56:11Z

Jannatul:

__TOC__

[[File:Atcom-ag188n.jpg|300px|thumb|left|Atcom AG188N]]
<div style="width:100%;overflow:hidden;clear:both"></div>

== '''Documentation:''' ==

[http://www.senatelecom.com/downloads/atcom/AG188N-USER%20MANUAL-W-V080408-En.pdf User´s Manual]<br>

=='''Accessing Your Device´s Configuration Page:'''==

Get the IP Address to access your AG-188N from the handset:

First please pick up the handset and dial #*111#

Then you will hear the AG-188N’s IP address, please write it down.

Enter the AG-188N’s IP address into a web browser and press enter, then you can access AG-188N’s web manage interface.

Remember the account login is admin/admin for administrator and guest/guest for user.

=='''SIP Configuration Section:'''==

*'''Fill the following fields according to your account:'''
**'''Register Server Addr:''' atlanta.voip.ms (one of our multiple servers)
**'''Register Server Port:''' 5060
**'''Register Username:''' 100000 (your VoIP.ms username)
**'''Register Password:''' ******** (the account password)
**'''Detect Interval Time:''' 60
**'''DTMF Mode:''' DTMF_RFC2833
**'''Enable Via rport:''' Enabled
**'''Local SIP Port:''' 5060
**'''Register Expire Time:''' 180
**'''RFC Protocol Edition:''' RFC3261
**'''Server Type:''' common
**'''SIP Default Protocol:''' Enabled

*'''Click on the "Apply" button at the bottom of the form.'''

'''Configuration Screen'''

[[File:Atcom.JPG|Atcom AG188N Configuration Screen]]

==Known Issues with this Device:==

'''Problems when dialing *97 for Voicemail:'''

With one phone hooked to the ‘phone’ jack, you can dial a star code to access the PSTN line instead of the SIP/IAX configured on the ATA. So, one analog phone enables you to use either SIP/IAX or the PSTN. However, people using the ATA with Asterisk have a problem when they want to dial Asterisk * codes like *97 for voicemail (as the * takes you to PSTN). The solution is to delete the “lifeline *T” option from “Dial-Peer” menu.

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

User Manual : [http://www.atcom.cn/uploadfile/2014/1219/ata/ag198/AG198-Quick%20Start%20Guide-W-D110728-EN.pdf Download PDF]

[[category:Analog Telephone Adapters]]

OBi 100/110 & OBi 200

2019-09-02T04:43:52Z

Jannatul:

[[File:OBi110-ATA.jpg|none|200px|center|link=http://www.polycom.com/voice-conferencing-solutions/voip-adapters.html|Obi]]

<br/>''The OBi100 & OBi200 are single phone port ATA adapters that support SIP VoIP services. The OBi100 & OBi200 are perfect for customers who do not have a traditional phone service, yet need a similar solution and want the savings and simplicity of using a VoIP service for all their calls. To start configuring your OBi100 or OBi200 device you will need to plug it in to your router/modem via its Internet port with an Ethernet cable and connect a regular handset phone to it's Phone port, then follow the next steps.''<br/>
<br/>
== Manual Configuration Details ==
<br/>
Start by dialing ''' * * * ''' from the connected phone, then press '''1''' to confirm your choice, this will return the IP address of your device being a number similar to '''192.168.xxx.xxx'''.<br/>
Once you get the IP address, enter it in the URL address bar '''"http://"''' of your Internet Browser to get access to the Graphic User Interface of the OBi100.

For an OBi202 please do the following to enable the GUI Web Interface:

Dial *** from the phone connected to the OBi202
Enter 0 For Advanced
Enter 30# Check Mark from
Press 1 to Enter a New Value
Press 1# to Enable
Press 1 to Save
Hang up

If done properly, the following window should appear on your screen:
[[File:ObiLogin.png|300px|thumb|left|Authentication Window - click to enlarge]]
<br clear="all" />

Once you get the Authentication Required window you will be prompted for a User Name and Password, the default credentials are:

'''User Name:''' admin

'''Password:''' admin

After this, you should now be able to see the OBi Web interface.

Now on the left side of the screen please find the next options and follow the next steps:

===Disabling auto-provisioning===

'''**NOTE :''' You may use this guide to configure an OBi110 as well. This is the VoIP.ms recommended configuration versus using the Obihai configuration dashboard (more on this later on this page) and you may also not find all new VoIP.ms servers on the Obihai Dahsboard. In order to make sure there will be no conflicts between this Manual configuration and the Obihai dashboard, please perform the following steps to disable auto-provisioning:

*System Management -> Auto Provisioning -> Auto Firmware Update -> Method : Disabled
*System Management -> Auto Provisioning -> ITSP Provisioning -> Method : Disabled
*System Management -> Auto Provisioning -> OBiTALK Provisioning -> Method : Disabled
*Voice Services -> OBiTALK Service -> Enable : Unchecked

Please note you must remove the check mark from the "default" column, then under "Method" please use the ''''Drop Down Selection'''' and choose '''Disabled'''.

[[File:Step1.png|450px|thumb|left|Disabling Auto Provision - click to enlarge]]
<br clear="all" />
After this, save all changes and you are ready to move on to the actual configuration.
<br />
===Configuring the ITSP Profile===

====General Section====
In this section you will set the name and the DigiMap you will use in the profile you configure. By default you will configure the profile A, unless you use the same device with another provider.

:'''Name''': 100000 (Replace with your 6 digit Main SIP account UserID or Sub Account user name, i.e. 123456 or 123456_sub)<br/>
:'''DigitMap''': Copy the line, including parenthesis, in the Digitmap field in the ITSP Profile and replace the "555" digits in the following lines by the area code of your choice:

::Dial Plan (recommended):
(1xxxxxxxxxx|<1555>[2-9]xxxxxx|<1>[2-9]xxxxxxxxx|911|011xx.|xx.|*xx.|***xxx|4xxx|(Mipd)|[^*]@@.)

Also, some clients have been successful by dialing '''**1 and *97''' for line 1 or '''**2 and *97''' for line 2.

:*If you need to set the dial plan back to Default, you can use this:

(1xxxxxxxxxx|<1>[2-9]xxxxxx|<1>[2-9]xxxxxxxxx|011xx.|***xxx|xx.|(Mipd)|[^*]@@.)

[[File:Step2.png|550px|thumb|left|ITSP profile, General - click to enlarge]]
<br clear="all" />

====SIP Section====
In this section you can set the server and the port that you wish to register to.

Please note that in order to change the settings, you need to uncheck the Default box on the right hand side.

*ProxyServer: denver.voip.ms (one of VoIP.ms multiple [http://wiki.voip.ms/article/Choosing_Server servers], you can choose the one closest to your location.)
*ProxyServerPort: 5060
*RegistrarServer: denver.voip.ms (one of VoIP.ms multiple servers [http://wiki.voip.ms/article/Choosing_Server servers], you can choose the one closest to your location.)
*RegistrarServerPort: 5060

[[File:Step3.png|550px|thumb|left|ITSP profile, SIP - click to enlarge]]
<br clear="all" />

Additionally, you may want to change the RegisterExpires value to 300, scroll down, deselect the default box and set the value there from 3600 to 300.

[[File:Step4.png|550px|thumb|left|ITSP profile, SIP (Register Expires)- click to enlarge]]
<br clear="all" />

===Configuring Voice Services===
<br/>
In this section you can set your Main account/sub_account credentials like User name and Password. The Main account password by default is the same password as the Customer Portal.

*AuthUserName: 100000 (Replace with your 6 digits Main SIP account UserID or Sub Account user name, i.e. 123456 or 123456_sub)
*AuthPassword: ****** (''Your SIP Account Password'')

[[File:Step5.png|550px|thumb|left|Voice Services (SIP Credentials) - click to enlarge]]
<br clear="all" />

Once you have finished changing all those settings, click on the button ''Submit'' to save the changes and ''reboot your OBi device'', your device should now be registered.

== Configuration Using OBi Dashboard ==

Besides the Manual Configuration previously explained, Obihai also provides us with their own API dashboard where you can add your device, to complete the configuration in easy steps.
Add your device to the OBiTALK service in the OBi Dashboard [http://www.obitalk.com/obinet/]. Instructions for this are included with the OBi110 and are not discussed here.

After the OBi110 is added, edit the device. You can select '''Service Provider 1''' or '''Service Provider 2''' under the '''Configure Voice Services''' heading. This will take you to a page where you can select ''voip.ms''. Follow the instructions and once you are done the configuration will be downloaded to your Obi110.

==Known Issues and Resolutions==
=== Ghost Calls ===
<br/>
Some customers have reported receiving constant calls from "100" or "101" as callerID. These calls are going directly to your device and do not pass through our servers, so we cannot filter them. You can read more about these calls, including how to prevent them, here: [[Sip Scanner Ghost Calls]]

You can also follow these suggestions, specific for the OBi brand devices:
*You can just disable (by unchecking Enable) for SP2 and OBiTALK under your Voice Tab (If you are using our service as SP1).

*You can restrict which IP addresses that can connect to your OBi. Going to "Service Providers -> ITSP Profile A -> SIP -> X_AccessList" : voip.ms_ip_address. You can see the IP address of the server you are currently using from this link: [http://wiki.voip.ms/article/Choosing_Server#IPs Server's IPs]

*You can also change your Obi Firewall Setting X_InboundCallRoute to : {(?|x|xx|xxx|xxxx|xxxxx|xxxxxx|un@@.|anon@@.):}, ph
This will only allow 7 digit or greater numbers through.

*Another alternative: OBi Interface>> Voice Services>> SP1 Service>> X_InboundCallRoute: {>('Insert your AuthUserName here'):ph}, example:

{>('100000'):ph} where 100000 is replaced with your own six digit SIP account UserID or the sub-account registered with your device.

By default, OBi devices accept calls destined for any username. The above syntax rejects calls that are not intended for whatever you have configured as AuthUserName.

<br/>

===10 Second Delay Reaching voip.ms Voicemail Attendant when dialing *97 or *98===

The Obi 100, 110 and 202 devices have non-configurable 'short' and 'long' delays if a dialed sequence does not match a digitmap. So you may have a 10 second delay when you dial into your voip.ms voicemail because of the built-in 'long' delay. This can be resolved in a couple of ways. Simply dial a # sign after you dial *97 or *98. Or include literals in your digitmap under the Service Provider / ITSP profile A or B / General / digitmap. Here is an example digitmap with a *97 literal included:

(1xxxxxxxxxx|<1>[2-9]xxxxxxxxx|011xx.|xx.|*xx.|'*97'|(Mipd)|[^*#]@@.)

The literal in the example is '*97'. You could also add a literal for '*98'.

Then when you dial *97, the device immediately sends it instead of waiting 10 seconds.

Read more on digitmaps under the topic Digit Map Configuration in the Obi Device Admin Guide.

=== Call Drops ===

If you experience random call drops while in the middle of a call or if the person you talk to remains silent for over a minute (60 seconds by default), OBi will hang up the call. Please go here and check and increase the following setting (Physical interface -> LINE port -> DetectFarEndLongSilence / SilenceTimeThreshold)

=== Enable Message Waiting Indicator MWI ===

To enable MWI please refer to the following section of the OBI web page:

Voice Services -> SP1 Service -> Calling Features ->

'''MWIEnable''' - Uncheck the box at the far right, to be able to check the box at the left, this enables the option.

'''X_VMWIEnable''' - Uncheck the box at the far right, to be able to check the box at the left, this enables the option.

'''MessageWaiting''' - Mark the checkbox

After these steps, the MWI should be active and working.

'''If you are trying to place an outbound call and get a recorded message ¨There is no service to complete your call¨ Please do the following to resolve this.'''
In Your OBi Device please go to Physical Interfaces >> PHONE Port which by default it is PSTN and it needs to be changed to Trunk Group 1

=== Using the OBi Network ===

You can use your OBi device to make calls directly to other OBi devices "''The OBi comes out of the box ready to make FREE calls to other OBi endpoints using the OBiTALK network. Dialing **9 + obi account number will use the OBiTALK feature and does not place calls to regular numbers nor use our network. ''" (you can get more information about [http://www.obihai.com/features-and-set-up here]), be aware that those calls will not pass through our network. If you need assistance with that feature, please contact [http://www.obihai.com/request-support OBI's support].
<br/>
=== Can not dial *98 even if is on your DigiMap ===
By default your OBi device uses *98 as Blind transfer code. If you want to be able to dial *98 from your device, you should change this code. You can achieve this in the settings of your device at: ''Star Codes >> Star code profile (A/B)'', unmark the "default" box and change *98 for something else (like *99)

[[File:Step6.png|550px|thumb|left|Changing *98 default code - click to enlarge]]
<br clear="all" />

=== Using the Phone book of your customer portal ===

If you plan on using the Phone Book in your Customer Portal and Speed Dial *75. Please log into your OBi and change the built-in speed dial code from *75 in the device to something else.

=== An additional note regarding outgoing calls===

In at least one instance it was necessary to specify a non-default outbound calling route in the OBi110 to be able to place calls using the voip.ms service. The default setting had the OBi110 attempting to place calls using the PSTN port on the device. The relevant setting is:

'''Physical Interfaces >> PHONE Port '''
*PrimaryLine: (Select from drop-down)

[[File:ObiPhoneport.JPG|550px|thumb|left|Changing Phone Port - click to enlarge]]
<br clear="all" />

The default is PSTN. Select SP1 Service if you only have one SIP account configured on the device. Select Trunk Group 1 to have it attempt to place calls using SP1 first, then SP2. Additional Trunk groups can be configured under Voice Services >> Gateways and Trunk Groups.

Portions of this article have been taken from [http://www.toao.net/500-mangos-guide-to-configuring-an-obi100-obi110-and-obi202-ata Mango's Guide to Configuring an OBi ATA]. Used with permission.

==Guide Links==
In the event where you need the guide directly from OBIHAI Technology Inc, you may find the admin manual guide below:

Admin Manual : [https://www.obitalk.com/info/documents/admin_guide/OBiDeviceAdminGuide.pdf Download PDF]

[[Category:Analog Telephone Adapters]]

== OBI 100/110 OBI 200/202 star codes ==

Here's a list of the most common used Star codes available for the OBI. <br>
You may also consult the following link in order to see the full list of star codes available for the OBI devices:
<br>
[http://www.obihai.com/docs/OBiFeatureStarCodes.pdf OBI feature star codes]
<br>

* *81 Block Caller ID (Persistent Mode)
* *82 Unblock Caller ID (Persistent Mode)
* *67 Block Caller ID (One Time)
* *68 Unblock Caller ID (One Time)
* *72 Call Forward All (Enter Number + #)
* *73 Disable Call Forward All
* *60 Call Forward on Busy (Enter Number + #)
* *61 Disable Call Forward in Busy
* *62 Call Forward on No Answer (Enter Number + #)
* *63 Disable Call Forward No Answer
* *56 Enable Call Waiting
* *57 Disable Call Waiting
* *78 Do Not Disturb - Turn On
* *79 Do Not Disturb – Disable

Netgear WGR615V

2019-09-02T04:38:41Z

Jannatul:

[[File:NetgearWGR615V.jpg|300px|thumb|left|Netgear WGR615V]]
<div style="width:100%;overflow:hidden;clear:both"></div>

== '''Documentation:''' ==
[ftp://downloads.netgear.com/files/wgr614v5_ref_manual.pdf User´s Manual]<br>

==Configuration Detail==
Go to http://192.168.61.1/set_voip.htm

Complete the following fields:

'''SIP Proxy''': Your voip.ms server

'''SIP Control Port''': 5060 or 5080

...

'''Line Enable:''' Yes

'''Display Name:''' whatever you want call display to show

'''Telephone Number:''' voip.ms username

'''User Name:''' voip.ms username

'''Password:''' voip.ms password

'''Register Expire Time (sec.):''' 30

'''Registration Head Start Time (sec.):''' 3

'''Anonymous Incoming Call:''' Allow

'''Distinctive Ring Valid:'''

'''Prefer Ring ID:'''

'''Prefer Fax Codec:'''

'''Prefer Codec:'''

'''Support Codec:'''

'''Note:''' special characters such as "_", "@", "#" and possibly others cannot be included in the phone number, username or password fields. Where sub-accounts contain an underscore, you can only use this device with your main voip.ms account.

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

User Manual : [http://www.downloads.netgear.com/files/wgr614v5_ref_manual.pdf Download PDF]

[[Category:Networking Devices]]

Grandstream HandyTone 486

2019-08-31T04:25:45Z

Jannatul:

[[File:HT486.jpg|300px|thumb|left|Grandstream HandyTone 486]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Configuration Detail==

'''STEP 0'''

Before powering the device, plug in an Ethernet cable from your Cable or DSL modem into the WAN port of the Handytone 486.

Doing this before powering the device facilitates the device getting an IP address via DHCP.

'''STEP 1'''

Connect an Ethernet cable from your PC to the Handytone 486 LAN port. Make sure that your PC Ethernet Internet Protocol (TCP/IP) is set to “Obtain an IP address automatically” .

'''STEP 2'''

Go to any browser equipped computer on your network and enter the address: http://192.168.2.1

'''STEP 3'''

You will be prompted to enter a password, the default is "admin", then click the "Login" button.

'''STEP 4'''

Go to BASIC SETTINGS tab.

If you wish to fall back on PPPOE if DHCP does not work and if your ISP assigns you IP number by PPPOE, you can enter the information assigned by your ISP into the circled areas.

[[File:Grandstream-install8 (1).gif|400px]]

'''STEP 5'''

Go to ADVANCED SETTINGS tab.

'''SIP Server''': atlanta.voip.ms (one of our multiple servers, you can choose the one closer to your location.)

'''SIP User ID''': 100000 (Replace with your 6 digits Main SIP account UserID or Sub Account user name, i.e. 123456 or 123456_sub)

'''Authenticate ID:''' 100000 (Replace with your 6 digits Main SIP account UserID or Sub Account user name, i.e. 123456 or 123456_sub)

'''Authenticate Password''': ********* (SIP Account Password)

'''Name:''' Enter your full name, this will show up as part of your caller ID.

'''User ID is phone number''': No

'''SIP Registration:''' Yes

'''NAT transversal''' : 10

[[File:Step5.jpeg]]

'''STEP 6'''

Click on the '''"Update"''' button at the bottom of the form.
[[File:Example.jpg]]

'''STEP 7'''

You will be brought to a confirmation screen on which you will click on the only option available: '''"Click to view your changes"'''.

'''STEP 8'''

Click on the '''"Reboot"''' button at the bottom of the form.
[[File:Example2.jpg]]

== See also ==
* [[Grandstream HandyTone 286]]
[[category: Analog Telephone Adapters]]

Cisco SPA2102 Phone Adapter with Router

2019-08-31T04:17:20Z

Jannatul:

[[File:Spa2102b.jpg|300px|thumb|left|Cisco SPA2102 Phone Adapter with Router]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Configuration Details==

''This is a guide for the Initial Configuration of the SPA2102/3102.''

'''Step 1'''

*Power off your network devices, including your modem and PC.
*Connect an Ethernet Cable from the Ethernet port of the SPA to the Ethernet port of the PC.
*Connect an Ethernet Cable from the Internet port of the SPA to the LAN/Ethernet port of the Modem.
*Connect your regular handset phone, to the Line port of the SPA2102.
*Then power up modem, then the SPA2102 and then the PC.
*Launch a web browser from the PC and enter "'''http://192.168.0.1/advanced'''" in the URL address bar field.

After these steps you should now have access to the Web Interface page of the SPA2102 to start with the initial configuration.

'''NOTE: If this is not working or the Browser can't find the page, you may also need to enable the Administration web service.<br>'''
'''Dial 7932#, then when prompted press 1 to enable and 1 to confirm'''<br>'''

You can also check this information from the Linksys Quick Guide: http://wiki.voip.ms/files/linksys-spa-2102-user-guide.pdf

'''Step 2'''

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 3'''

'''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 3.5'''

'''On the SIP tab, Under NAT Support Parameters, set the following.'''

'''Handle VIA received:''' Yes<br>
'''Substitute VIA Addr:''' Yes<br>

'''Step 4 (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:

<br>
(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.) <br>

[[File:SPA2102Line1.png]]

'''Step 5'''

Click on the "Save Settings " button at the bottom of the form.

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

== How to avoid the long delay to hear the ringtone ==

If you ever experience some delay to hear the ringtone when you make outgoing calls with your SPA. Changing the SPA's Interdigit Long Timer value can help resolve the issue. Follow the next steps in order to change that setting:

'''Note''': However before changing that option, test if calling the number with an # at the end of the number works(e.g. 5554441234#).
If that doesn't work you need to contact the support staff in voip.ms.

*1- First access the SPA's web interface.
*2- Click on the '''Admin Login''' and then click on the '''Voice''' tab.
*3- Click on the '''Regional''' tab and look for the '''Control Timer Values (sec)''' section.
*4- Enter the desire value in the '''Interdigit Long Timer''' field (for example lower this value to 4).

[[File:Ctrl timer values.jpg|800px]]

'''Note''': The image correspond to the PAP2 device. If your device doesn't have this setting, contact voip.ms support.

'''Additionally, whenever the registration is dropped and its having problems to gain it back, you can try this simple trick:'''
'''On Advanced view, go to the Line tab, look for the SIP port setting, and change it to another, i.e. if you have port 5060 change to 5080, if it is 5061 you can change to 5081.'''

== Known Issues ==

If you are receiving phone calls at strange times and with a strange CallerID and they do not show up in your CDR then these calls could be Direct IP Device Calls trying to connect to you. To remove these types of calls please follow these directions.

Please Login to your Cisco Adapter and from Admin Login > Advanced > Voice > System > System Configuration > Restricted Access Domain

In this field put the servers you connect to: I.E. montreal.voip.ms, toronto.voip.ms
Please remember you did this in case you change VoIP.ms servers.

Once the Restricted Access Domain field is populated with Point-of-Presence (POP) servers, any server specified in Proxy field (Voice > LINE1> Proxy & Registration) will not be able to establish registration UNLESS the same server is included in the Restricted Access Domain field.

== Factory Reset Procedure ==

Step 1: Connect a phone into TEL1 port

Step 2: On the handset dial digits **** (4 Stars)

Step 3: Dial 73738# then press 1 to confirm

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

Admin Manual : [https://www.cisco.com/c/dam/en/us/td/docs/voice_ip_comm/csbpvga/ata/administration/guide/ATA_AG_v3_NC-WEB.pdf Download PDF]

== See also ==
* [[Dial Plan for Linksys ATAs]]
* You can check the most commonly used Star Codes from [[Cisco/Linksys Star Codes]]

[[Category:Analog Telephone Adapters]]

[http://tinyurl.com/lf7f3bo PDF Owner`s Manual]

Cisco WRP400

2019-08-31T04:06:48Z

Jannatul:

[[File:WRP400.jpg|300px|thumb|left|Cisco WRP400]]
<div style="width:100%;overflow:hidden;clear:both"></div>

The '''Cisco WRP400'''[http://www.cisco.com/en/US/products/ps10028/] and '''Cisco WRP500'''[http://www.cisco.com/c/en/us/products/unified-communications/wrp500-wireless-ac-broadband-router-2-phone-ports/index.html] are wireless broadband routers with two phone ports, one wired Internet uplink port and four wired LAN ports. They provide both the networking capability of a standard wi-fi router and the voice capability of common Cisco/Sipura two-line adapters (such as the [[Cisco SPA112|SPA112/122]] series) all in one box.

== Configuration Details ==
This section describes configuration of the voice portion of the WRP400/WRP500 routers only. It presumes that your PC is connected to one of the router's local area network ports (as wi-fi is disabled until configured by the user). The router is manually configurable at its default LAN address of 192.168.15.1 unless this address has been modified by a previous user configuration change.

The telephony capabilities of the WRP500 ([http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/csbpvga/wrp500/admin/wrp500-ata-admin-guide.pdf admin manual]) are very similar to its predecessor; the WRP500 was introduced in 2015 to provide faster wi-fi (802.11ac + 802.11bgn dual band on WRP500, versus 802.11bg on WRP400) and the older model discontinued in 2017.

'''Step 1'''

Power up the router and allow it to initialise. Connect a PC to one of the router's local area network ports.

'''Step 2'''

Open a web browser to http://192.168.15.1/admin/voice/ and log in with username ''admin'' and default password ''admin''.

[[Image:Cisco-wrp400-line1tabheader.png|412px]]

If the web interface does not appear at the default address (192.168.15.1) try dialling '''* * * * 110 #''' or '''* * * * 210 #''' from one of the telephone handsets. As with other Cisco/Sipura ATA's, the device will announce its local and external Internet addresses by voice.

On the WRP500, 110# returns the router's external address, not its LAN address. If 192.168.15.1 is in use by another network device, the WRP500 will use 192.168.16.1

'''Step 3'''

Click on Voice, then configure Line 1 as follows:

'''Proxy:''' atlanta.voip.ms (You can choose any of our multiple VoIP.ms servers)

'''Display Name:''' Your name (Example: John Smith)

'''User ID:''' Your (6 digits) VoIP.MS SIP Account number (or subaccount)

'''Password:''' Your VoIP.MS SIP Password

'''Dial Plan:''' (911S0|<:1555>[2-9]xxxxxx|1[2-9]xx[2-9]xxxxxxS0|<:1>[2-9]xx[2-9]xxxxxxS0|*xx|*xx.|***xxx|[3468]11|0|00|4xxx|**275x.|xxxxxxxxxxxx.)

(note: Replace 555 in the dial plan with your area code. See [[Dial Plan for Linksys ATAs]] for details.)

Click Submit to save settings.

'''Step 4'''

Click on Voice, then Line 1

If the WRP400/WRP500 is behind another router (which will rarely be the case) set '''NAT Mapping Enable''' to Yes, then set '''NAT Keep Alive Enable''' to Yes. If this device is connected directly to your modem, in most cases you can leave these settings disabled.

Under '''Proxy and Registration''' set '''Register Expires''' to 180, '''Proxy Fallback Intvl''' to 180

[[File:WRP400_Proxy1.png]]

Also confirm the following settings:<br>
'''Use DNS SRV:''' NO<br>
'''DNS SRV Auto Prefix:''' NO

Click Submit to submit these changes

'''Step 5'''

Cisco hardware arrives preconfigured for California (Pacific, GMT-8). If you are in this time zone, most of the steps to set local time may be safely skipped.

Setup the time zone and configure the system time by synchronizing with time server (NTP) or set time manually (Manual Setting).

; WRP500: Click '''Administration''' > '''Time Setup''' and set the '''time zone''' field to match your local time. Select DST if applicable. NTP settings may be left as-is.

; WRP400:

: Click '''Voice''', Admin Login, then go to '''Regional''', then scroll down to '''Miscellaneous'''

: Set your '''time zone''' to the local time for your location.

: Set your "Daylight Saving Time Rule" (the box directly under "Time Zone") to match your location.

: If you are in Canada/USA and still see ''start=4/1/7;end=10/-1/7;save=1'' this is incorrect. That rule set the clock forward on the first Sunday in April and back on the last Sunday in October. As of 2007, the time change for Canada/USA takes place in March and November.

: '''Daylight Saving Time Rule:''' ''start=3/8/7;end=11/1/7;save=1''

: (If you're in a location with no DST, such as Arizona or Saskatchewan, you may want to leave this box empty.)

Click Submit to save the changes

'''Step 6'''

;WRP400 only: The WRP400 has separate time setting pages for the telephone adapter and the wi-fi router. For this model, set the router time to GMT with no daylight savings.

: Click '''Setup''', then go to '''Basic Setup''', then scroll down to '''Time Setting'''

: Set the timezone to '''GMT''' with no daylight saving time (ie: Liberia) as the time zone is already specified on the Voice - Regional tab.

: Optionally, select your NTP settings (by default, an NTP server will be automatically selected - setting time manually doesn't work with this router). A good time server choice is 0.pool.ntp.org.

: (As of version 2.00.32, a bug in Cisco's firmware continues to cause the WRP400 to send incorrect time to call display on the handsets if you attempt to use the Basic Setup tab to specify your local time zone. It is unclear why Cisco provided two different - and buggy - configuration pages to configure local time on the WRP400. WRP500 appears not to have these issues.)

Click Submit to save the changes

'''Step 7'''

Click '''Voice''', then go to '''SIP'''.

Set SIP Timer Values (sec)
:SIP T1: '''1'''

Set RTP Parameters
:RTP Packet Size: '''0.02'''

Cisco's defaults (SIP T1 = 0.5 sec, RTP packet size 0.030 on most Sipura adapters) respectively may cause unnecessary retransmission of commands over connections with high latency and create issues with outbound audio "breaking up".

Click Submit to save the changes

== WRP400 Firmware Upgrade ==
This requirement applies to the WRP400 only; as of 2018, the original WRP500 firmware (v1.01) with which the newer model was released in 2015 remains current and valid.

;WRP400: As of 2013, Cisco was continuing to ship the WRP400 and SPA112/122 with outdated (ver 1.01.00) firmware. This firmware is buggy, causing the web menus to display with errors for Mozilla Firefox users. Some Cisco gear also will have caller ID problems until firmware is upgraded to a current version (using, presumably, some other web browser).

: Updated WRP400 firmware (version 2.00.32, released 13 March 2013) is available for download at https://software.cisco.com/download/release.html?mdfid=282414113&softwareid=282463656&release=2.00.32 - this is likely to be the last update as this model was discontinued in 2017.

[[Image:Cisco-wrp400-fwupgrade.png|thumb|414px]]
: The downloaded file is a .ZIP archive; unzip it. This yields three files; a .pdf of documentation and two .bin files with similar names (but one with "FCC" and the other with "ETSI" in the name). The "FCC" version is for North American users.

: Go to http://192.168.15.1/Upgrade.asp (the default local address of the router if you have not changed it) and enter the username/password (by default, ''admin'' for both). You will be prompted to "select a file to upgrade the firmware" and press "start to upgrade". Select the .bin file for your region and allow the upgrade to complete.

: Once the upgrade is complete, see http://www.cisco.com/en/US/docs/voice_ip_comm/csbpvga/wrp400/administration/guide/WRP400_UG_v2-0_OL-18475-01.pdf for instructions on how to configure the networking features of the WRP400, such as the wireless local area network.

: The new firmware will replace the "Linksys" logo and branding with "Cisco" on all configuration screens.

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

Admin Manual : [https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/csbpvga/wrp500/admin/wrp500-ata-admin-guide.pdf Download PDF]

== See also ==
* [[Dial Plan for Linksys ATAs]]
* Commonly used [[Cisco/Linksys Star Codes]]

== External links ==
* Cisco WRP500 Administration Guide [https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/csbpvga/wrp500/admin/wrp500-ata-admin-guide.pdf (PDF)] [https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/csbpvga/wrp500/admin/wrp500-ata-admin-guide.html (HTML)]

[[category:Networking Devices]]
[[category:routers]]

Cisco WRP400

2019-08-31T04:06:06Z

Jannatul:

[[File:WRP400.jpg|300px|thumb|left|Cisco WRP400]]
<div style="width:100%;overflow:hidden;clear:both"></div>

The '''Cisco WRP400'''[http://www.cisco.com/en/US/products/ps10028/] and '''Cisco WRP500'''[http://www.cisco.com/c/en/us/products/unified-communications/wrp500-wireless-ac-broadband-router-2-phone-ports/index.html] are wireless broadband routers with two phone ports, one wired Internet uplink port and four wired LAN ports. They provide both the networking capability of a standard wi-fi router and the voice capability of common Cisco/Sipura two-line adapters (such as the [[Cisco SPA112|SPA112/122]] series) all in one box.

== Configuration Details ==
This section describes configuration of the voice portion of the WRP400/WRP500 routers only. It presumes that your PC is connected to one of the router's local area network ports (as wi-fi is disabled until configured by the user). The router is manually configurable at its default LAN address of 192.168.15.1 unless this address has been modified by a previous user configuration change.

The telephony capabilities of the WRP500 ([http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/csbpvga/wrp500/admin/wrp500-ata-admin-guide.pdf admin manual]) are very similar to its predecessor; the WRP500 was introduced in 2015 to provide faster wi-fi (802.11ac + 802.11bgn dual band on WRP500, versus 802.11bg on WRP400) and the older model discontinued in 2017.

'''Step 1'''

Power up the router and allow it to initialise. Connect a PC to one of the router's local area network ports.

'''Step 2'''

Open a web browser to http://192.168.15.1/admin/voice/ and log in with username ''admin'' and default password ''admin''.

[[Image:Cisco-wrp400-line1tabheader.png|412px]]

If the web interface does not appear at the default address (192.168.15.1) try dialling '''* * * * 110 #''' or '''* * * * 210 #''' from one of the telephone handsets. As with other Cisco/Sipura ATA's, the device will announce its local and external Internet addresses by voice.

On the WRP500, 110# returns the router's external address, not its LAN address. If 192.168.15.1 is in use by another network device, the WRP500 will use 192.168.16.1

'''Step 3'''

Click on Voice, then configure Line 1 as follows:

'''Proxy:''' atlanta.voip.ms (You can choose any of our multiple VoIP.ms servers)

'''Display Name:''' Your name (Example: John Smith)

'''User ID:''' Your (6 digits) VoIP.MS SIP Account number (or subaccount)

'''Password:''' Your VoIP.MS SIP Password

'''Dial Plan:''' (911S0|<:1555>[2-9]xxxxxx|1[2-9]xx[2-9]xxxxxxS0|<:1>[2-9]xx[2-9]xxxxxxS0|*xx|*xx.|***xxx|[3468]11|0|00|4xxx|**275x.|xxxxxxxxxxxx.)

(note: Replace 555 in the dial plan with your area code. See [[Dial Plan for Linksys ATAs]] for details.)

Click Submit to save settings.

'''Step 4'''

Click on Voice, then Line 1

If the WRP400/WRP500 is behind another router (which will rarely be the case) set '''NAT Mapping Enable''' to Yes, then set '''NAT Keep Alive Enable''' to Yes. If this device is connected directly to your modem, in most cases you can leave these settings disabled.

Under '''Proxy and Registration''' set '''Register Expires''' to 180, '''Proxy Fallback Intvl''' to 180

[[File:WRP400_Proxy1.png]]

Also confirm the following settings:<br>
'''Use DNS SRV:''' NO<br>
'''DNS SRV Auto Prefix:''' NO

Click Submit to submit these changes

'''Step 5'''

Cisco hardware arrives preconfigured for California (Pacific, GMT-8). If you are in this time zone, most of the steps to set local time may be safely skipped.

Setup the time zone and configure the system time by synchronizing with time server (NTP) or set time manually (Manual Setting).

; WRP500: Click '''Administration''' > '''Time Setup''' and set the '''time zone''' field to match your local time. Select DST if applicable. NTP settings may be left as-is.

; WRP400:

: Click '''Voice''', Admin Login, then go to '''Regional''', then scroll down to '''Miscellaneous'''

: Set your '''time zone''' to the local time for your location.

: Set your "Daylight Saving Time Rule" (the box directly under "Time Zone") to match your location.

: If you are in Canada/USA and still see ''start=4/1/7;end=10/-1/7;save=1'' this is incorrect. That rule set the clock forward on the first Sunday in April and back on the last Sunday in October. As of 2007, the time change for Canada/USA takes place in March and November.

: '''Daylight Saving Time Rule:''' ''start=3/8/7;end=11/1/7;save=1''

: (If you're in a location with no DST, such as Arizona or Saskatchewan, you may want to leave this box empty.)

Click Submit to save the changes

'''Step 6'''

;WRP400 only: The WRP400 has separate time setting pages for the telephone adapter and the wi-fi router. For this model, set the router time to GMT with no daylight savings.

: Click '''Setup''', then go to '''Basic Setup''', then scroll down to '''Time Setting'''

: Set the timezone to '''GMT''' with no daylight saving time (ie: Liberia) as the time zone is already specified on the Voice - Regional tab.

: Optionally, select your NTP settings (by default, an NTP server will be automatically selected - setting time manually doesn't work with this router). A good time server choice is 0.pool.ntp.org.

: (As of version 2.00.32, a bug in Cisco's firmware continues to cause the WRP400 to send incorrect time to call display on the handsets if you attempt to use the Basic Setup tab to specify your local time zone. It is unclear why Cisco provided two different - and buggy - configuration pages to configure local time on the WRP400. WRP500 appears not to have these issues.)

Click Submit to save the changes

'''Step 7'''

Click '''Voice''', then go to '''SIP'''.

Set SIP Timer Values (sec)
:SIP T1: '''1'''

Set RTP Parameters
:RTP Packet Size: '''0.02'''

Cisco's defaults (SIP T1 = 0.5 sec, RTP packet size 0.030 on most Sipura adapters) respectively may cause unnecessary retransmission of commands over connections with high latency and create issues with outbound audio "breaking up".

Click Submit to save the changes

== WRP400 Firmware Upgrade ==
This requirement applies to the WRP400 only; as of 2018, the original WRP500 firmware (v1.01) with which the newer model was released in 2015 remains current and valid.

;WRP400: As of 2013, Cisco was continuing to ship the WRP400 and SPA112/122 with outdated (ver 1.01.00) firmware. This firmware is buggy, causing the web menus to display with errors for Mozilla Firefox users. Some Cisco gear also will have caller ID problems until firmware is upgraded to a current version (using, presumably, some other web browser).

: Updated WRP400 firmware (version 2.00.32, released 13 March 2013) is available for download at https://software.cisco.com/download/release.html?mdfid=282414113&softwareid=282463656&release=2.00.32 - this is likely to be the last update as this model was discontinued in 2017.

[[Image:Cisco-wrp400-fwupgrade.png|thumb|414px]]
: The downloaded file is a .ZIP archive; unzip it. This yields three files; a .pdf of documentation and two .bin files with similar names (but one with "FCC" and the other with "ETSI" in the name). The "FCC" version is for North American users.

: Go to http://192.168.15.1/Upgrade.asp (the default local address of the router if you have not changed it) and enter the username/password (by default, ''admin'' for both). You will be prompted to "select a file to upgrade the firmware" and press "start to upgrade". Select the .bin file for your region and allow the upgrade to complete.

: Once the upgrade is complete, see http://www.cisco.com/en/US/docs/voice_ip_comm/csbpvga/wrp400/administration/guide/WRP400_UG_v2-0_OL-18475-01.pdf for instructions on how to configure the networking features of the WRP400, such as the wireless local area network.

: The new firmware will replace the "Linksys" logo and branding with "Cisco" on all configuration screens.

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

User Manual : [https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/csbpvga/wrp500/admin/wrp500-ata-admin-guide.pdf Download PDF]

== See also ==
* [[Dial Plan for Linksys ATAs]]
* Commonly used [[Cisco/Linksys Star Codes]]

== External links ==
* Cisco WRP500 Administration Guide [https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/csbpvga/wrp500/admin/wrp500-ata-admin-guide.pdf (PDF)] [https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/csbpvga/wrp500/admin/wrp500-ata-admin-guide.html (HTML)]

[[category:Networking Devices]]
[[category:routers]]

Cisco SPA112

2019-08-31T04:00:51Z

Jannatul:

[[File:SPA112.jpg|300px|thumb|left|Cisco SPA112]]
<div style="width:100%;overflow:hidden;clear:both"></div>

: '''There have been some reports of issues with this device from customers of both VoIP.ms and other providers.'''
: '''Make sure to install the latest firmware from [https://software.cisco.com/download/release.html?mdfid=283998771&softwareid=282463187&release=1.4.1%20SR3&relind=AVAILABLE&rellifecycle=&reltype=latest Cisco Software].'''
: '''Version 1.1 or later should be used for proper Caller ID support. '''
: '''Some People have reported issues using Firefox to Configure this device; please try Chrome or IE. '''

==Documentation==
[https://www.cisco.com/c/en/us/support/unified-communications/spa112-2-port-phone-adapter/model.html#~tab-documents Official Configuration guide / User guide / Data Sheets]

== Configuration Details ==

=== Getting the IP address of your device ===

There are two ways to retrieve the IP address of your Cisco SPA112: via analog phone menu, and via your internet router.

{| class="wikitable"
! style="text-align: center;" | Analog phone interface
! style="text-align: center;" | Internet Router
|-
|
# Attach the Cisco SPA112 to your network and attach an analog phone to one of the phone ports, then do the following:
## Dial **** from the phone, even though there is no dial tone.
## When you hear "System Configuration Menu," dial 1 1 0 # slowly. The current IP address will be read back. (e.g. 192.168.X.X)

'''If you hear 0.0.0.0, check your network connection and DHCP server. If necessary, a static IP address'''
'''can be assigned by using option 111# at the IVR, then entering the IP address with your phone's keypad'''
'''(for example, 10*1*27*2 for 10.1.27.2). The network mask can be set with option 121# and the default'''
'''gateway can be sent with option 131#'''
Learn more about the IVR menu options from the https://supportforums.cisco.com/docs/DOC-9900 document.

Be sure to allow at least a minute or two for the box to initialize; even a correctly configured and installed SPA112/122 will give no power or dialtone to the phone until initialization is complete.

Note that the SPA122 is basically a SPA112 with a second network port, intended for installation between a local network hub (LAN) and an upstream Internet (WAN) connection. The SPA122 may be configured as either a "NAT" or "bridge". Depending on configuration, this leaves the SPA122 with two addresses; a local area network address (such as 192.168.15.1) and an outside Internet address. Dialing ****110# will give one address, ****210# will give the other.
|
# Attach the Cisco SPA112 to your network
# Access your router's remote administration interface via your web browser (typical addresses may be <code>192.168.0.1</code> or <code>192.168.1.1</code>). Refer to your router instructions for more information.
# Enter your username/password if asked. If you have not set one, then it is likely the unchanged default password.
# In the router's menu, there should be a page showing a list of connected clients, with their internal IP address. Find the entry corresponding to the Cisco SPA112. It should identify itself in the list as "SPA112"
# Navigate to this IP address via your web browser
|}

=== Accessing to the device's settings page ===

Open your web browser and go to the IP address you obtained in step 1 (for example, http://192.168.2.1).
The default username is admin, and the default password is also admin.

For the SPA122, if one address does not return the web interface (or has some functions greyed/disabled), try the other.

=== Configuring the Quick Setup screen ===

Go to Quick Setup and configure Line 1 as follows:

'''Proxy:''' atlanta.voip.ms (You can choose any of our multiple VoIP.ms [http://wiki.voip.ms/article/Choosing_Server servers])

'''Display Name:''' Your name

'''User ID:''' 100000 (Replace with your 6 digit Main SIP account UserID or Sub Account user name, i.e. 123456 or 123456_sub)

'''Password:''' Your VoIP.MS SIP Password

'''Dial Plan:''' (911S0|310xxxx|<:1555>[2-9]xxxxxx|1[2-9]xx[2-9]xxxxxxS0|[2-9]xx[2-9]xxxxxxS0|*xx|***xxx|*xx.|[3468]11|822|0|00|4xxx|**275*x.|xxxxxxxxxxxx.)

('''''Note''''': Replace 555 in the dial plan with your area code, See [[Dial Plan for Linksys ATAs]] for details.)

Click Submit to save settings.

[[File:quick_setup_test.png|800px|thumb|left|Quick Setup Page - Click to enlarge]]
<br clear="all" />

=== Configuring the Voice Line ===
==== Nat Settings ====

Click on Voice, then Line 1

Set '''NAT Mapping Enable''' to Yes, then set '''NAT Keep Alive Enable''' to Yes. If your environment does not use NAT, you can leave these settings disabled. These features can usually be disabled on the SPA122 if it is connected directly to your modem since its traffic will not be subject to NAT in this configuration.

If using the second phone line on an SPA122 device, change the SIP Port for one of the lines to e.g. 5080.

[[File:VL_1_nat_settings.png|800px|thumb|left|NAT Settings - Click to enlarge]]
<br clear="all" />

==== Proxy and Registration ====

Under '''Proxy and Registration''', set the server you will use as registration server and the proper values for the Register Expires and Proxy Fallback Intvl:

'''Proxy''': atlanta.voip.ms (one of VoIP.ms multiple [http://wiki.voip.ms/article/Choosing_Server servers], you can choose the one closest to your location)
'''Register Expires''' to 300
'''Proxy Fallback Intvl''' to 300

Also confirm the following settings:<br />
'''Register:''' YES <br />
'''Use DNS SRV:''' NO<br />
'''DNS SRV Auto Prefix:''' NO

[[File:VL_2_proxyAndRegistration.png|800px|thumb|left|Proxy and Registration - Click to enlarge]]
<br clear="all" />

Click Submit to save these changes

==== Subscriber Information ====

In this section please confirm that you have the proper account information:

'''Display Name''': Your name (that will be shown as callerID name)
'''User ID''': 100000 (Replace with your 6 digit Main SIP account UserID or Sub Account user name, i.e. 123456 or 123456_sub)
'''Password''': Your VoIP.ms SIP Password

[[File:VL_3_subscriberInformation.png|800px|thumb|left|Subscriber Information - Click to enlarge]]
<br clear="all" />

==== Audio Configuration ====

You can verify or change the audio codec that will be used with the calls. Please verify that you have the same codec selected in your SIP account's settings.

Preferred codec: g711u (or G729)

[[File:VL_4_audioConfig.png|800px|thumb|left|Audio configuration - Click to enlarge]]
<br clear="all" />

==== Dial Plan ====

We recommend to use this dial plan.

(911S0|310xxxx|<:1555>[2-9]xxxxxx|1[2-9]xx[2-9]xxxxxxS0|[2-9]xx[2-9]xxxxxxS0|*xx|***xxx|*xx.|[3468]11|822|0|00|4xxx|**275*x.|xxxxxxxxxxxx.)

('''''Note''''': Replace 555 in the dial plan with your area code, See [[Dial Plan for Linksys ATAs]] for details.)

[[File:VL_5_dialPlan.png|800px|thumb|left|Dial Plan - Click to enlarge]]
<br clear="all" />

You can create your own dial plan if you need it. See [[Dial Plan for Linksys ATAs]]

=== Optional settings ===

==== Outbound audio "breaking up". ====

Cisco's defaults (SIP T1 = 0.5 sec, RTP packet size 0.030 on most Sipura adapters) respectively may cause unnecessary retransmission of commands over connections with high latency and create issues with outbound audio "breaking up".

Click '''Voice''', then go to '''SIP'''.

Set SIP Timer Values (sec)

SIP T1: 1

Set RTP Parameters

RTP Packet Size: 0.02
RTP Port Min: 10000
RTP Port Max: 20000

Click Submit to save the changes

[[File:VS_sipAndRTP.png|800px|thumb|left|SIP Values - Click to enlarge]]
<br clear="all" />

==== Caller ID display showing incorrect time ====

Sometimes the hour shown in your caller ID is incorrect. Following this suggestion usually solves the issue:

Enter your device's settings and click '''Network Setup''', then go to '''Basic Setup''', then click '''Time Settings'''

Set your time zone and (optional) NTP settings. A good time server choice is 0.pool.ntp.org. Setting the proper time zone will ensure that the time which appears on your Caller ID display is correct.

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

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

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

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

====Verifying the device's Firmware version====

First, check your firmware's version. This is, from your Device's Configuration Utility at ''Status >> System Information''

We '''strongly recommend''' to use the latest firmware version available, up today is the ''1.4.1 (SR3) Apr 3 2019''. If you do not have this version, you may consider its upgrade.

[[File:SPA_FW_Ver.png|800px|thumb|left|Click to enlarge]]
<br clear="all" />

==== Enabling TLS for the line ====

Go to the User's line you will use (If you use the Line 1 go to User 1) and navigate to the '''''Supplementary Service Settings''''', there set:
:''Secure Call Setting'': '''''yes'''''

[[File:SPA_User_SCS.png|800px|thumb|left|Click to enlarge]]
<br clear="all" />

==== Configuring the transport and port ====

Go to the line you will be using with TLS and navigate to the section ''SIP Settings'', then set:
:''SIP Transport'' : '''''TLS'''''
:''SIP Port'' : '''''5061'''''

[[File:SPA_Voice_Line.png|800px|thumb|left|Click to enlarge]]
<br clear="all" />

==== CA Certificate ====

As per CISCO's requirements, a CA certificate is needed to use Secure calls with the SPA112's device. To achieve this you will need to import the CA Cert.

Go to ''Voice >> Provisioning'' and once there navigate to '''''CA Settings''''', at ''Custom CA URL'' enter the following:

http://spa1xx.voip.ms/cca.pem

[[File:SPA_Prov_CA.png|800px|thumb|left|Click to enlarge]]
<br clear="all" />

Click ''Submit'', the device will reboot and after that will register and you will be ready to use it with TLS

==== Secure Call Indication Tone ====

Once the secure call feature is enabled, during all the duration of your calls you will hear a couple of tones (beeps), this is normal and beyond ''VoIP.ms'' control, however you can disable this notification going to: ''Voice >> Regional >> Secure Call Indication Tone''

'''''Note:''''' Please notice that this setting is not an on/off, you will need to remove all the line
In case you need to set it back, the default value is
397@-19,507@-19;15(0/2/0,.2/.1/1,.1/2.1/2)

[[File:SPA_Regional_Tone.png|800px|thumb|left|Click to enlarge]]
<br clear="all" />

Click Submit to save the changes

==Known Issues==

=== '''Phone will not ring on handset''' ===

Sometimes the Phone you are using is designed for a certain voltage and ring waveform. If someone tries to call you and the phone appears to be ringing for the caller but your phone never rings, please follow these steps to hopefully resolve this issue for you.

Step 1: First access the SPA web interface.

Step 2: Click on the '''Admin Login''' and then click on '''(switch to advanced view)'''

Step 3: Click on your Regional tab on the top menu.

Step 4: Go halfway down the page until you see the heading '''Ring and Call Waiting Tone Spec'''

[[File:Pap2Ring.jpg|800px|thumb|left| Ring and Call Waiting - Click to enlarge]]
<br clear="all" />

Step 5: Change the Ring Waveform setting to Sinusoid or Trapezoid, the opposite of what you have set. You can also change the Ring Voltage in increments of 5 to 90 or 95.

Step 6: Save settings and test an incoming call.

=== Receiving Unwanted Calls in the middle of the Night (i.e. CallerID 100) that do not appear in your CDR: ===

These calls are not going through our Network but rather through the internet directly to your ATA Device.

Please look under the Voice>> Line 1 page in your SPA device for the following setting: Restrict Source IP and make sure it's enabled.

This way the ATA device will block any traffic not coming from our servers.

[[File:VL_1_restrictSourceIP.png|800px|thumb|left|Restrict IP - Click to enlarge]]
<br clear="all" />

=== Firmware Upgrade ===

SPA112 and SPA122 adapters were distributed with outdated (1.0.x) firmware at least as late as 2012; affected boxes will not show Caller ID on any inbound call, even though the caller names and numbers are visible in the call detail record on VoIP.ms (or other provider's) web interface.

Updated firmware is available from the Cisco site [https://software.cisco.com/download/release.html?mdfid=283998771&softwareid=282463187&release=1.4.1%20SR3&relind=AVAILABLE&rellifecycle=&reltype=latest Cisco Firmware] as a .ZIP archive which contains two files (a .BIN with the actual firmware and a .PDF with documentation). Download and unzip this file. Go to the 'administration' tab on the web interface (on the SPA122, this needs to be done from the LAN side with SPA122's built-in networking set to NAT mode). On the left sidebar, click 'update firmware' (as most of the administration menu does not appear for Firefox users, downgrade to MS IE or another browser temporarily). Click the 'upload' button and indicate the location of the unzipped .BIN file. A box will appear with a progress indicator and a warning not to interrupt the upgrade. When the upgrade is completed, the SPA112/122 will reset and will likely take a minute or more to reinitialize, reconnect to the network and restore dial tone. SPA122 users who have installed the device in-line between the local PCs and the Internet will be disconnected from the Internet until reinitialization is complete.

Once the new firmware is deployed, call display will operate normally and the configuration web page will display in Firefox without missing options in the administration menu.

A manual for Cisco's SPA100 series adapters is online at http://www.cisco.com/en/US/docs/voice_ip_comm/csbpvga/spa100-200/admin_guide_SPA100/spa100_ag.html

* You can check the most commonly used [[Cisco/Linksys Star Codes]]

=== SPA Star Codes ===

{| class="wikitable" style="margin-left: center; margin-right: auto; border: none;"
! colspan="3" | SPA Star Codes
|-
! scope="col" | Star Code
! scope="col" | Name
! scope="col" | Result
|-
| *69 || Call Return Code || This code calls the last caller.
|-
| *07 || Call Redial Code || Redials the last number called. (Not in pap2t)
|-
| *98 || Blind Transfer Code || Begins a blind transfer of the current call to the extension specified after the activation code.
|-
| *66 || Call Back Act Code || Starts a callback when the last outbound call is not busy.
|-
| *86 || Call Back Dea t Code || Cancels a callback.
|-
| *05 || Call Back Busy Act Code || Starts a callback when the last outbound call is busy. (Not in pap2t)
|-
| *72 || Cfwd All Act Code || Forwards all calls to the extension specified after the activation code.
|-
| *73 || Cfwd All Deact Code || Cancels call forwarding of all calls.
|-
| *90 || Cfwd Busy Act Code || Forwards busy calls to the extension specified after the activation code.
|-
| *91 || Cfwd Busy Deact Code || Cancels call forwarding of busy calls.
|-
| *92 || Cfwd No Ans Act Code || Forwards no-answer calls to the extension specified after the activation code.
|-
| *93 || Cfwd No Ans Deact Code || Cancels call forwarding of no-answer calls.
|-
| *63 || Cfwd Last Act Code || Forwards the last inbound or outbound calls to the extension specified after the activation code.
|-
| *83 || Cfwd Last Deact Code || Cancels call forwarding of the last inbound or outbound calls.
|-
| *60 || Block Last Act Code || Blocks the last inbound call.
|-
| *80 || Block Last Deact Code || Cancels blocking of the last inbound call.
|-
| *64 || Accept Last Act Code || Accepts the last outbound call. It lets the call ring through when do not disturb or call forwarding of all calls are enabled.
|-
| *84 || Accept Last Deact Code || Cancels the code to accept the last outbound call.
|-
| *56 || CW Act Code || Enables call waiting on all calls.
|-
| *57 || CW Deact Code || Disables call waiting on all calls.
|-
| *71 || CW Per Call Act Code || Enables call waiting for the next call.
|-
| *70 || CW Per Call Deact Code || Disables call waiting for the next call.
|-
| *67 || Block CID Act Code || Blocks caller ID on all outbound calls.
|-
| *68 || Block CID Deact Code || Removes caller ID blocking on all outbound calls.
|-
| *81 || Block CID Per Call Act Code || Blocks caller ID on the next outbound call.
|-
| *82 || Block CID Per Call Deact Code || Removes caller ID blocking on the next inbound call.
|-
| *77 || Block ANC Act Code || Blocks all anonymous calls.
|-
| *87 || Block ANC Deact Code || Removes blocking of all anonymous calls.
|-
| *78 || DND Act Code || Enables the do not disturb feature.
|-
| *79 || DND Deact Code || Disables the do not disturb feature.
|-
| *65 || CID Act Code || Enables caller ID generation.
|-
| *85 || CID Deact Code || Disables caller ID generation.
|-
| *25 || CWCID Act Code || Enables call waiting, caller ID generation.
|-
| *45 || CWCID Deact Code || Disables call waiting, caller ID generation.
|-
| *26 || Dist Ring Act Code || Enables the distinctive ringing feature.
|-
| *46 || Dist Ring Deact Code || Enables the distinctive ringing feature. The default is *46 ||.
|-
| *74 || Speed Dial Act Code || Assigns a speed dial number.
|-
| *16 || Secure All Call Act Code || Makes all outbound calls secure.
|-
| *17 || Secure No Call Act Code || Makes all outbound calls not secure.
|-
| *18 || Secure One Call Act Code || Makes the next outbound call secure. (It is redundant if all outbound calls are secure by default.)
|-
| *19 || Secure One Call Deact Code || Secure One Call Deact Code Makes the next outbound call not secure. (It is redundant if all outbound calls are not secure by default.)
|}

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

Admin Manual : [https://www.cisco.com/c/dam/en/us/td/docs/voice_ip_comm/csbpvga/spa100-200/admin_guide_SPA100/spa100_ag_1_3_5.pdf Download PDF]

[[category:Analog Telephone Adapters]]

Cisco Linksys PAP2T

2019-08-31T03:45:14Z

Jannatul:

[[File:Pap2t.jpg|300px|thumb|left|Cisco Linksys PAP2T]]
<div style="width:100%;overflow:hidden;clear:both"></div>

Haga clic aquí para la versión en Español:: [[Cisco Linksys PAP2T (Español)]]
__TOC__

'''If you are running the PAP2t with a tomato firmware router, this may represent a persistent issue after'''
'''rebooting the device, generally it won't register back as usual, in order to fix this you only need to set'''
'''Tomato UDP Unreplied timeout down to 10 (from default of 30).'''

==Documentation==
[https://www.cisco.com/c/dam/en/us/td/docs/voice_ip_comm/csbpvga/pap2t/administration/guide/pap2t_user.pdf Official User Guide]

==Configuration Details==

'''Step 1'''

The first step is to find out what IP Address your adapter is currently using. To do this, pick a phone connected on Line 1 and do the following:

'''Dial: **** (That is 4 asterisks)'''<br>
'''Once this is done, dial: 110# (110 followed by a square)'''

The system should now playback the IP Address your device has been assigned.<br>
(Example: 192.168.1.2)

'''Step 2'''

'''Using your favorite web browser from a computer on the same network, point the address to the IP address of your adapter.'''<br>
<nowiki>(example http://192.168.1.2)</nowiki> Replace 192.168.1.2 by the IP Address your device is currently using.

'''Step 3'''

You should now see the web interface of your Linksys/Sipura.

'''At the top of the page below the tabs, click on the link "Admin", and once the page has reloaded, click again on the link "Advanced View".'''

'''Step 4'''

'''Under the LINE 1 Tab, Find the following fields and fill them with the following information'''

'''Nat Keep Alive:''' Yes<br>
'''Nat Mapping/Traversal:''' Yes<br>
'''Proxy:''' atlanta.voip.ms (You can choose any of our multiple VoIP.ms servers)<br>
'''Register Expires:''' 180<br>
'''Proxy Fallback Intvl:''' 180<br>
'''Display Name:''' John Smith (Replace with your name or company name)<br>
'''User ID:''' 100000 (Replace with your VoIP.ms account number or sub account ID)<br>
'''Password:''' ******** (Type Main account password, or sub account password)<br>
'''Use DNS SRV:''' NO<br>
'''DNS SRV Auto Prefix:''' NO

'''Step 5'''

You can configure your adapter with a better dial plan, allowing faster dialing of 10 digits number (Local US/Canada) and also enable 7 digits dialing in one area code of your choice.

'''At the bottom of Line 1 TAB, you will find a field called Dial Plan'''

Replace the 555 digits in the following line by the area code of your choice and copy the line, including parenthesis, in the Dialplan field in Line 1 Tab at the bottom of the page:

(911S0|310xxxx|<:1555>[2-9]xxxxxx|1[2-9]xx[2-9]xxxxxxS0|[2-9]xx[2-9]xxxxxxS0|*xx|***xxx|*xx.|[3468]11|822|0|00|4xxx|**275*x.|xxxxxxxxxxxx.)

[[Dial Plan for Linksys ATAs|Click here for more information on Linksys dial plans]]

'''Step 6'''

Click on the "Save Settings " button at the bottom of the form.

'''Configuration Screens'''

[[File:PAP2T001.JPG|PAP2T Configuration Screen|frame|left]]

[[File:PAP2T002.JPG|PAP2T Configuration Screen|frame|left]]

[[File:PAP2T003.JPG|PAP2T Configuration Screen|frame|left]]

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

'''Customer Submitted Information:''' <br>
For North America:<br>
Found this link on configuring the PAP2-NA hardware to work better in North America and specifically with VOIP.ms.<br>
Read the article called: [http://www.toao.net/25-linksys-ata-configuration Configure your Linksys VoIP ATA the right way!]

To upgrade a firmware version from a Windows system, the PAP2T-NA documentation nor Cisco's web site does not say how to do this.
Go to [http://www.cisco.com/en/US/products/ps10029/index.html Cisco PAP2T Internet Phone Adapter with 2 VoIP Ports]. Click on the link "Download Firmware", which downloads the .zip file. Run the .exe file, then enter the IP number of the ATA device (called “SPA” in the program). It then upgrades the device. This is provided for your information: the author is not saying you have to upgrade the firmware.

== How to avoid the long delay to hear the ringtone ==

If you ever experience some delay to hear the ringtone when you make outgoing calls with your PAP2T. Changing the PAP2T's Interdigit Long Timer value can help resolve the issue. Follow the next steps in order to change that setting:

'''Note''': However before changing that option, test if calling the number with an # at the end of the number works(e.g. 5554441234#).
If that doesn't work you need to contact the support staff in voip.ms.

*1- First access the PAP2T's web interface.
*2- Click on the '''Admin Login''' and then click on '''(switch to advanced view)'''

[[File:Pap2admlog.jpg|800px]]

*3- Click on the '''Regional''' tab and look for the '''Control Timer Values (sec)''' section.

[[File:Regional.jpg|800px]]

*4- Enter the desire value in the '''Interdigit Long Timer''' field (for example lower this value to 4).

[[File:Ctrl timer values.jpg|800px]]

== '''Phone will not ring on handset''' ==

Sometimes the Phone you are using is designed for a certain Voltage and Ring Waveform. If someone tries to call you and the phone appears to be ringing for the caller but your phone never rings please follow these steps to hopefully resolve this issue for you.

Step 1: First access the PAP2's web interface.

Step 2: Click on the '''Admin Login''' and then click on '''(switch to advanced view)'''

Step 3: Click on your Regional Tab on the Top Menu.

Step 4: Go Halfway Down the Page until you see the Heading '''Ring and Call Waiting Tone Spec'''

[[File:Pap2Ring.jpg|800px]]

Step 5: Change the Ring Waveform setting to Sinusoid or Trapezoid, the opposite of what you have set. You can also change the Ring Voltage in increments of 5 to 90 or 95.

Step 6: Save Settings and Test an Incoming Call

== Receiving Unwanted Calls in the middle of the Night ( i.e. CallerID 100) that do not appear in your CDR: ==

These calls are not going through our Network but rather through the internet directly to your ATA Device.

Please look under the Voice>> Line 1 page in your Linksys device for the following setting: Restrict Source IP and make sure it's enabled.

This way the ATA device will block any traffic not coming from our servers.

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

==Linksys PAP2T ATA Adapter Reset Procedure==

Sometimes it will be very helpful to reset your linksys ATA adapter to factory default settings.

*Connect a telephone to line 1 of the PAP2T unit and power it on.
*Disconnect your PAP2T adapter from the internet connection(unplug the Ethernet cable from the PAP2T hardware unit). Resetting with internet connection may mess up the unit making it completely useless.
*Dial ****, and wait for the Interactive Voice Menu (IVM) to get activated.
*Type in the following number including the # symbol.

73738#

(This number spells RESET.)

*Confirm this by pressing 1.

Your linksys ATA unit will now go back to it factory default settings.

'''Note:''' There are some PAP2/PAP2T devices in circulation which were originally 'locked' to one provider and subsequently unlocked by end users. Do *not* use the RESET$ command with these boxes.

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

User Manual : [https://www.cisco.com/c/dam/en/us/td/docs/voice_ip_comm/csbpvga/pap2t/administration/guide/pap2t_user.pdf Download PDF]

== See also ==
* [[Dial Plan for Linksys ATAs]]
* You can check the most commonly used Star Codes from [[Cisco/Linksys Star Codes]]

[[Category:Analog Telephone Adapters]]

=== SPA Star Codes ===

* *69 Call Return Code
This code calls the last caller.

* *07 Call Redial Code
Redials the last number called. (Not in pap2t)

* *98 Blind Transfer Code
Begins a blind transfer of the current call to the extension specified after the activation code.

* *66 Call Back Act Code
Starts a callback when the last outbound call is not busy.

* *86 Call Back Deact Code
Cancels a callback.

* *05 Call Back Busy Act Code
Starts a callback when the last outbound call is busy. (Not in pap2t)

* *72 Cfwd All Act Code
Forwards all calls to the extension specified after the activation code.

* *73 Cfwd All Deact Code
Cancels call forwarding of all calls.

* *90 Cfwd Busy Act Code
Forwards busy calls to the extension specified after the activation code.

* *91 Cfwd Busy Deact Code
Cancels call forwarding of busy calls.

* *92 Cfwd No Ans Act Code
Forwards no-answer calls to the extension specified after the activation code.

* *93 Cfwd No Ans Deact Code
Cancels call forwarding of no-answer calls.

* *63 Cfwd Last Act Code
Forwards the last inbound or outbound calls to the extension specified after the activation code.

* *83 Cfwd Last Deact Code
Cancels call forwarding of the last inbound or outbound calls.

* *60 Block Last Act Code
Blocks the last inbound call.

* *80 Block Last Deact Code
Cancels blocking of the last inbound call.

* *64 Accept Last Act Code
Accepts the last outbound call. It lets the call ring through when do not disturb or call forwarding of all calls are enabled.

* *84 Accept Last Deact Code
Cancels the code to accept the last outbound call.

* *56 CW Act Code
Enables call waiting on all calls.

* *57 CW Deact Code
Disables call waiting on all calls.

* *71 CW Per Call Act Code
Enables call waiting for the next call.

* *70 CW Per Call Deact Code
Disables call waiting for the next call.

* *67 Block CID Act Code
Blocks caller ID on all outbound calls.

* *68 Block CID Deact Code
Removes caller ID blocking on all outbound calls.

* *81 Block CID Per Call Act Code
Blocks caller ID on the next outbound call.

* *82 Block CID Per Call Deact Code
Removes caller ID blocking on the next inbound call.

* *77 Block ANC Act Code
Blocks all anonymous calls.

* *87 Block ANC Deact Code
Removes blocking of all anonymous calls.

* *78 DND Act Code
Enables the do not disturb feature.

* *79 DND Deact Code
Disables the do not disturb feature.

* *65 CID Act Code
Enables caller ID generation.

* *85 CID Deact Code
Disables caller ID generation.

* *25 CWCID Act Code
Enables call waiting, caller ID generation.

* *45 CWCID Deact Code
Disables call waiting, caller ID generation.

* *26 Dist Ring Act Code
Enables the distinctive ringing feature.

* *46 Dist Ring Deact Code
Disables the distinctive ringing feature. The default is *46.

* *74 Speed Dial Act Code
Assigns a speed dial number.

* *16 Secure All Call Act Code
Makes all outbound calls secure.

* *17 Secure No Call Act Code
Makes all outbound calls not secure.

* *18 Secure One Call Act Code
Makes the next outbound call secure. (It is redundant if all outbound calls are secure by default.)

* *19 Secure One Call Deact Code
Secure One Call Deact Code Makes the next outbound call not secure. (It is redundant if all outbound calls are not secure by default.)

Grandstream HandyTone 802 - HT802

2019-08-30T02:40:31Z

Jannatul:

[[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

* '''''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

2019-08-30T02:38:47Z

Jannatul:

[[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

* '''''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 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

2019-08-30T00:57:12Z

Jannatul:

[[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

* '''''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 Positron, you may find the user 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 502 - HT502

2019-08-30T00:56:00Z

Jannatul:

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

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

==Connecting the HandyTone==

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

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

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

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

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

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

[[File:502.jpg]]

==Accessing to the Web Configuration==

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

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

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

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

==Settings==

'''Step 1:'''

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

'''Step 2:'''

Fill the followings fields.

*'''Account Active:''' Yes
* '''Primary SIP Server:''' atlanta.voip.ms (Pick one of VoIP.ms multiple [http://wiki.voip.ms/article/Choosing_Server VoIP Servers])
*'''Outbound Proxy:''' Set the same server you've configured at the '''Primary SIP Server''' field
*'''NAT Traversal:''' No, But Keep-alive
* '''SIP User ID:''' 100000 (Replace with your Main SIP account or Subaccount UserID, e.g. 198765 or 198765_sub)
*'''Authenticate ID:''' 100000 (Replace with your Main SIP account or Subaccount UserID, e.g. 198765 or 198765_sub)
* '''Authenticate Password:''' ********* (account password)

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

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

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

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

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

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

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

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

===VoIP.ms===

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

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

Unfortunately, in the Grandstream dial plans, as soon as one rule is matched (except for x+ or x. which allow an arbitrary number of additional digits) the call is sent out immediately. (By contrast, the Linksys dial plan will wait for a short [typically five second] delay unless 'S0' is specified explicitly). It's therefore not possible to deal with multiple conflicting alternatives by waiting for the user to dial additional digits which could potentially match some other valid pattern.

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

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

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

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

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

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

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

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

== Known Issues and Resolutions ==

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

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

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

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

[[category: Analog Telephone Adapters]]

Grandstream HandyTone 802 - HT802

2019-08-30T00:54:10Z

Jannatul:

[[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

* '''''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 Positron, you may find the user manual guide 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 502 - HT502

2019-08-30T00:41:42Z

Jannatul:

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

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

==Connecting the HandyTone==

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

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

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

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

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

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

[[File:502.jpg]]

==Accessing to the Web Configuration==

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

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

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

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

==Settings==

'''Step 1:'''

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

'''Step 2:'''

Fill the followings fields.

*'''Account Active:''' Yes
* '''Primary SIP Server:''' atlanta.voip.ms (Pick one of VoIP.ms multiple [http://wiki.voip.ms/article/Choosing_Server VoIP Servers])
*'''Outbound Proxy:''' Set the same server you've configured at the '''Primary SIP Server''' field
*'''NAT Traversal:''' No, But Keep-alive
* '''SIP User ID:''' 100000 (Replace with your Main SIP account or Subaccount UserID, e.g. 198765 or 198765_sub)
*'''Authenticate ID:''' 100000 (Replace with your Main SIP account or Subaccount UserID, e.g. 198765 or 198765_sub)
* '''Authenticate Password:''' ********* (account password)

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

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

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

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

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

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

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

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

===VoIP.ms===

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

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

Unfortunately, in the Grandstream dial plans, as soon as one rule is matched (except for x+ or x. which allow an arbitrary number of additional digits) the call is sent out immediately. (By contrast, the Linksys dial plan will wait for a short [typically five second] delay unless 'S0' is specified explicitly). It's therefore not possible to deal with multiple conflicting alternatives by waiting for the user to dial additional digits which could potentially match some other valid pattern.

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

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

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

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

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

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

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

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

== Known Issues and Resolutions ==

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

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

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

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

[[category: Analog Telephone Adapters]]

Grandstream HandyTone 486

2019-08-30T00:39:26Z

Jannatul:

[[File:HT486.jpg|300px|thumb|left|Grandstream HandyTone 486]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Configuration Detail==

'''STEP 0'''

Before powering the device, plug in an Ethernet cable from your Cable or DSL modem into the WAN port of the Handytone 486.

Doing this before powering the device facilitates the device getting an IP address via DHCP.

'''STEP 1'''

Connect an Ethernet cable from your PC to the Handytone 486 LAN port. Make sure that your PC Ethernet Internet Protocol (TCP/IP) is set to “Obtain an IP address automatically” .

'''STEP 2'''

Go to any browser equipped computer on your network and enter the address: http://192.168.2.1

'''STEP 3'''

You will be prompted to enter a password, the default is "admin", then click the "Login" button.

'''STEP 4'''

Go to BASIC SETTINGS tab.

If you wish to fall back on PPPOE if DHCP does not work and if your ISP assigns you IP number by PPPOE, you can enter the information assigned by your ISP into the circled areas.

[[File:Grandstream-install8 (1).gif|400px]]

'''STEP 5'''

Go to ADVANCED SETTINGS tab.

'''SIP Server''': atlanta.voip.ms (one of our multiple servers, you can choose the one closer to your location.)

'''SIP User ID''': 100000 (Replace with your 6 digits Main SIP account UserID or Sub Account user name, i.e. 123456 or 123456_sub)

'''Authenticate ID:''' 100000 (Replace with your 6 digits Main SIP account UserID or Sub Account user name, i.e. 123456 or 123456_sub)

'''Authenticate Password''': ********* (SIP Account Password)

'''Name:''' Enter your full name, this will show up as part of your caller ID.

'''User ID is phone number''': No

'''SIP Registration:''' Yes

'''NAT transversal''' : 10

[[File:Step5.jpeg]]

'''STEP 6'''

Click on the '''"Update"''' button at the bottom of the form.
[[File:Example.jpg]]

'''STEP 7'''

You will be brought to a confirmation screen on which you will click on the only option available: '''"Click to view your changes"'''.

'''STEP 8'''

Click on the '''"Reboot"''' button at the bottom of the form.
[[File:Example2.jpg]]

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

[https://www.802.cz/manualy/manual-ht-486-en.pdf Download PDF]

== See also ==
* [[Grandstream HandyTone 286]]
[[category: Analog Telephone Adapters]]

Grandstream HandyTone 486

2019-08-28T18:30:42Z

Jannatul:

[[File:HT486.jpg|300px|thumb|left|Grandstream HandyTone 486]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Configuration Detail==

'''STEP 0'''

Before powering the device, plug in an Ethernet cable from your Cable or DSL modem into the WAN port of the Handytone 486.

Doing this before powering the device facilitates the device getting an IP address via DHCP.

'''STEP 1'''

Connect an Ethernet cable from your PC to the Handytone 486 LAN port. Make sure that your PC Ethernet Internet Protocol (TCP/IP) is set to “Obtain an IP address automatically” .

'''STEP 2'''

Go to any browser equipped computer on your network and enter the address: http://192.168.2.1

'''STEP 3'''

You will be prompted to enter a password, the default is "admin", then click the "Login" button.

'''STEP 4'''

Go to BASIC SETTINGS tab.

If you wish to fall back on PPPOE if DHCP does not work and if your ISP assigns you IP number by PPPOE, you can enter the information assigned by your ISP into the circled areas.

[[File:Grandstream-install8 (1).gif|400px]]

'''STEP 5'''

Go to ADVANCED SETTINGS tab.

'''SIP Server''': atlanta.voip.ms (one of our multiple servers, you can choose the one closer to your location.)

'''SIP User ID''': 100000 (Replace with your 6 digits Main SIP account UserID or Sub Account user name, i.e. 123456 or 123456_sub)

'''Authenticate ID:''' 100000 (Replace with your 6 digits Main SIP account UserID or Sub Account user name, i.e. 123456 or 123456_sub)

'''Authenticate Password''': ********* (SIP Account Password)

'''Name:''' Enter your full name, this will show up as part of your caller ID.

'''User ID is phone number''': No

'''SIP Registration:''' Yes

'''NAT transversal''' : 10

[[File:Step5.jpeg]]

'''STEP 6'''

Click on the '''"Update"''' button at the bottom of the form.
[[File:Example.jpg]]

'''STEP 7'''

You will be brought to a confirmation screen on which you will click on the only option available: '''"Click to view your changes"'''.

'''STEP 8'''

Click on the '''"Reboot"''' button at the bottom of the form.
[[File:Example2.jpg]]

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

[https://www.802.cz/manualy/manual-ht-486-en.pdf Download PDF]

== See also ==
* [[Grandstream HandyTone 286]]
[[category: Analog Telephone Adapters]]

Grandstream HandyTone 486

2019-08-28T18:16:31Z

Jannatul:

[[File:HT486.jpg|300px|thumb|left|Grandstream HandyTone 486]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Configuration Detail==

'''STEP 0'''

Before powering the device, plug in an Ethernet cable from your Cable or DSL modem into the WAN port of the Handytone 486.

Doing this before powering the device facilitates the device getting an IP address via DHCP.

'''STEP 1'''

Connect an Ethernet cable from your PC to the Handytone 486 LAN port. Make sure that your PC Ethernet Internet Protocol (TCP/IP) is set to “Obtain an IP address automatically” .

'''STEP 2'''

Go to any browser equipped computer on your network and enter the address: http://192.168.2.1

'''STEP 3'''

You will be prompted to enter a password, the default is "admin", then click the "Login" button.

'''STEP 4'''

Go to BASIC SETTINGS tab.

If you wish to fall back on PPPOE if DHCP does not work and if your ISP assigns you IP number by PPPOE, you can enter the information assigned by your ISP into the circled areas.

[[File:Grandstream-install8 (1).gif|400px]]

'''STEP 5'''

Go to ADVANCED SETTINGS tab.

'''SIP Server''': atlanta.voip.ms (one of our multiple servers, you can choose the one closer to your location.)

'''SIP User ID''': 100000 (Replace with your 6 digits Main SIP account UserID or Sub Account user name, i.e. 123456 or 123456_sub)

'''Authenticate ID:''' 100000 (Replace with your 6 digits Main SIP account UserID or Sub Account user name, i.e. 123456 or 123456_sub)

'''Authenticate Password''': ********* (SIP Account Password)

'''Name:''' Enter your full name, this will show up as part of your caller ID.

'''User ID is phone number''': No

'''SIP Registration:''' Yes

'''NAT transversal''' : 10

[[File:Step5.jpeg]]

'''STEP 6'''

Click on the '''"Update"''' button at the bottom of the form.
[[File:Example.jpg]]

'''STEP 7'''

You will be brought to a confirmation screen on which you will click on the only option available: '''"Click to view your changes"'''.

'''STEP 8'''

Click on the '''"Reboot"''' button at the bottom of the form.
[[File:Example2.jpg]]

==Guide Links==
In the event where you need the guides directly from (Company Name, in this case, Cisco), you may find the company guides below:

User Manual: [(https://www.802.cz/manualy/manual-ht-486-en.pdf) Download PDF

== See also ==
* [[Grandstream HandyTone 286]]
[[category: Analog Telephone Adapters]]