User:Roy/Sandbox - VoIP.ms Wiki

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User:Roy/Sandbox

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== Login into your device ==
== Login into your device ==
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1. Connect a computer to the same network as the UCM6202.
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1. Connect a computer to the same network as the UCM6202.<br>
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2. Ensure the UCM 6202 is properly powered on and displays the IP address on the LCD screen.  
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2. Ensure the UCM 6202 is properly powered on and displays the IP address on the LCD screen.<br>
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3. Open a web browser on the computer and enter the displayed IP address into the search bar in the following format http(s)://ipaddress:portnumber
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3. Open a web browser on the computer and enter the displayed IP address into the search bar in the following format http(s)://ipaddress:portnumber <br>
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The default protocol is HTTPS and the default port number is 8089.
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The default protocol is HTTPS and the default port number is 8089. <br>
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4. The Web portal should be shown (see figure below). Enter the default administrator credentials:
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4. The Web portal should be shown (see figure below). Enter the default administrator credentials: <br>
*Username: admin
*Username: admin

Revision as of 23:11, 22 April 2019

Contents

Login into your device

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

  • Username: admin
  • Password: admin
Click to enlarge

Creating a trunk

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

Click to enlarge

SIP Trunk

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

*Type: Choose "Register SIP trunk" from the dropdown menu
*Provider Name: Type VoIP.ms if not listed (any name can be used)
*Host Name: Type the domain name of the server your phone number(s) will be using. You can see our VoIP servers list from here: https://wiki.voip.ms/article/Choosing_Server
*Keep trunk CID: Enable it, if you want the trunk to send its own CID number or disable it if your extensions are going to send their own CID number
*Username: Type your account's number or if you're using a sub-account type its name. 
*Password: Type your SIP password for this account

When done, just click on "Save" button. Your new trunk will be created but changes won't take effect until you click on "Apply changes" button, we can do this later when the set up is complete.

If you get a pop-up window letting you know about the SIP port number, just click "Ok"

Click to enlarge

SIP settings enhanced

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

SIP headers

In order to send your SIP headers as we require, please make sure the "fromuser" field has your account or sub-account name in it. Bear in mind "username" and "fromuser" fields should contain the same information.

Click to enlarge
Audio codecs

Now click at "Advanced Settings" and at "Codec Preference" use only the supported codecs by VoIP.ms. At the right column choose only G.729, PCMU & GSM, in this order.

Trunk Caller ID

If your trunk is going to send its own CID number you can use PAI (P-Asserted-Identity) header. You only need to enable "Send PAI Header" and type the caller ID name and Caller ID number your trunk will send by using the following format: YourName<YourPhoneNumber>.

In order to send PAI header successfully, bear in mind that no CID number must be set from your VoIP.ms portal at your account or sub-accounts section, otherwise the CID number set in your portal will override the one sent by your trunk.

Keep Alive Event

We also suggest enabling "Enable Heartbeat Detection" in order to send SIP OPTIONS messages to our server and avoid your modem to close your local SIP ports. Make "Heartbeat Frequency" to 50 seconds, this should be enough time to renew your local SIP port connection.


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

Click to enlarge

IAX2 Trunk

To create an IAX2 trunk just fill the following information:

*Type: Choose "Register IAX trunk"
*Provider Name: Type VoIP.ms if not listed (any name can be used)
*Host Name: Type the domain name of the server your phone number(s) will be using. You can see our VoIP servers list from here: https://wiki.voip.ms/article/Choosing_Server
*Keep trunk CID: Enable it, if you want the trunk to send its own CID number or disable it if your extensions are going to send their own CID number
*Caller ID: Type the CID number your trunk will be sending.
*Caller ID Name: Type the name your trunk will be sending. 
*Username: Type your account's number or if you're using a sub-account type its name. 
*Password: Type your SIP password for this account
Click to enlarge

IAX settings enhanced

Click on the edit icon for your trunk and go to "Advanced settings". Make sure only supported codecs by VoIP.ms are listed in the right column.

Also, enable "Heartbeat Frequency" to send Keep Alive events to our server and avoid closing your local ports by your modem. "Heartbeat Frequency" should be set to 50 seconds.

Click to enlarge

Creating your outbound route

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

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

_1NXXXXXXXX
_NXXXXXXXXX
_4XXX
_00.
_011.
_033.
_044.

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

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

Click to enlarge

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

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

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

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

Creating your inbound route

Your inbound route will allow you to receive your incoming calls using your PBx and process it into the proper destination of your choice.

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

Click to enlarge

In this section, fill the following fields:

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