Grandstream UCM6200

From VoIP.ms Wiki

Revision as of 21:48, 15 August 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

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

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Creating a trunk

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

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SIP Trunk

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

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

Note 1: Bear in mind to use the same VoIP server your VoIP number is using. 
You can check what VoIP server is your VoIP number using, from your VoIP.ms customer portal 
at DID Numbers>> Manage DIDs, under POP column. 
You can choose any server you want, as long as the one in your portal and the one in this field matches, 
otherwise, incoming calls won't ring.

Click Save button, do not click Apply Changes yet.

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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 fill:

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

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Audio codecs

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

Trunk Caller ID

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

"CalerIDName"<CallerIDNumer>

Note 2: No Caller ID Number must be set at your VoIP.ms portal or it will override the one sent by your trunk

Keep Alive Event

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

"Enable Heartbeat Detection Heartbeat Frequency: 50

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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.

Call Encryption TLS/SRTP

In order to use TLS along with your UCM please follow these steps:

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

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

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

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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)".

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

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

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

*Transport: TLS

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

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

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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.)

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

• 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

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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.

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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"

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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.

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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".

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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.

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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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