Grandstream UCM6200

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[quality revision]

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Revision as of 18:20, 30 April 2019 (view source) Joseph (Talk | contribs) (Created page with "left Designed to provide a centralized solution for the communication needs of businesses, the UCM6200 series IP PBX appliance combines enterpri...") ← Older edit		Revision as of 20:52, 2 May 2019 (view source) Roy (Talk | contribs) (→Creating your outbound route) Newer edit →
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	'''VoIP.ms suggest to include the following patterns into your outbound route: '''		'''VoIP.ms suggest to include the following patterns into your outbound route: '''
-	_1NXXXXXXXX	+	_1NXXXXXXXXX
	_NXXXXXXXXX		_NXXXXXXXXX
	_4XXX		_4XXX
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	== Creating your inbound route ==		== 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.	+	Thanks to your inbound routes you can use only one single trunk to receive all the incoming calls from all your phone numbers. This way you don't need to use more than one trunk for your phone numbers, inbound routes will receive all of them and route them into the proper destination in your UCM.
		+
		+	'''Note:''' We do not suggest using more than one VoIP.ms trunk on the same device.
	From the left panel, head to Extension/Trunk>> Inbound Routes and click on "Add".		From the left panel, head to Extension/Trunk>> Inbound Routes and click on "Add".
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	[[File:Inbound routes setup.png|thumb|none|550px|Click to enlarge]]		[[File:Inbound routes setup.png|thumb|none|550px|Click to enlarge]]
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		+	'''Note''': Remember to send your phone number in the '''TO''' SIP header, this way your PBx will match it with your inbound routes. You can do this very easily by setting '''Device Type''' from "ATA adapter, IP phone or Softphone" to "IP PBX Server, Asterisk or Softswitch". You will find this setting from your VoIP.ms customer portal at Main Menu>> Account settings>> Inbound settings, if you're using the main account or at Sub accounts>> Manage Sub accounts and by clicking on the orange icon with a pen, if you're using a sub-account
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		+	[[File:PBx setting VoIP panel.png|thumb|left|550px|Click to enlarge]]
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		+	[[File:PBx setting VoIP panel2.png|thumb|left|550px|Click to enlarge]]

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Revision as of 20:52, 2 May 2019

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

Website: UCM 6200 Series

Contents 1 Login into your device 2 Creating a trunk 2.1 SIP Trunk 2.1.1 SIP settings enhanced 2.1.1.1 SIP headers 2.1.1.2 Audio codecs 2.1.1.3 Trunk Caller ID 2.1.1.4 Keep Alive Event 2.2 IAX2 Trunk 2.2.1 IAX settings enhanced 3 Creating your outbound route 4 Creating your inbound route

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

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.

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 (one of VoIP.ms multiple 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 

• 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 the "Save" button. Your new trunk will be created but changes won't take effect until you click on the "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"

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.

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 the "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.

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 (one of VoIP.ms multiple servers, you can choose the one closest to your location.)
• 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

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.

Creating your outbound route

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

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

_1NXXXXXXXXX
_NXXXXXXXXX
_4XXX
_00.
_011.
_033.
_044.

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

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

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.

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

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

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

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

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.

Note: Remember to send your phone number in the TO SIP header, this way your PBx will match it with your inbound routes. You can do this very easily by setting Device Type from "ATA adapter, IP phone or Softphone" to "IP PBX Server, Asterisk or Softswitch". You will find this setting from your VoIP.ms customer portal at Main Menu>> Account settings>> Inbound settings, if you're using the main account or at Sub accounts>> Manage Sub accounts and by clicking on the orange icon with a pen, if you're using a sub-account