User:Roy/Sandbox

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Revision as of 17:04, 20 April 2019 (view source) Roy (Talk | contribs) (→Creating your outbound route) ← Older edit		Latest revision as of 18:26, 4 August 2025 (view source) Roy (Talk | contribs)
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-	== Login into your device ==	+	==HOW DOES THIS WORK?==
		+	You can use our service to place and receive calls just using your internet connection, no matter where you are as long as you have internet. You can use an IP phone, ATA adapter (to use a traditional phone) or a softphone (an App installed on your cellphone or PC)
		+	==RATES==
		+	VoIP.ms is a prepaid service where you add funds to your account ($15 USD min) and the following rates are deducted from it:
-	1. Connect a computer to the same network as the UCM6202.	+	You only pay a monthly fee for each of your numbers + in/out minutes usage.
-	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	+	'''Monthly fee for the phone number'''
-	*Password: admin	+
-	[[File:UCM main screnn.png|thumb|none|550px|Click to enlarge]]	+	*Per minute plan:<br>
		+	$0.85 USD Monthly + $0.009 USD per minute inbound<br>
		+	$0.01 USD outbound per minute for US/CAN<br>
-	== Creating a trunk ==	+	*Flat rate plan:<br>
-	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.	+	$4.25 USD Monthly, 3500 free inbound minutes<br>
		+	$0.01 USD outbound per minute for US/CAN<br>
-	[[File:UCM-trunk.png|thumb|none|550px|Click to enlarge]]	+	SMS/MMS rates:<br>
		+	$0.0075 USD per SMS<br>
		+	$0.02 USD per MMS<br>
		+	==FIRST STEPS - ADD FUNDS==
		+	1. Go to Finances -> Add funds, and make your 1st payment. DO NOT use the autobilling for your 1st payment.<br>
		+	2. WAIT 5 minutes for balance to be updated and before you try to buy or port a number.<br>
		+	==PORTING FROM VOIPO==
		+	1. Log in to your VoIPo account at account.voipo.com.<br>
		+	2. Go to Menu -> Account -> Invoices<br>
		+	3. Print your invoice and save it as PDF/JPG<br>
		+	4. Now log into your voipms portal and go to DID Numbers -> DID portability<br>
		+	5. Look for "If you want to start a porting process from VOIPo, Click here" and click it<br>
		+	6. Fill in the following:<br>
		+	*VOIPo Account number: One of your voipo phone numbers
		+	*Proof of Ownership: upload your voipo invoice
		+	*List of Numbers: list your voipo number(s) you want to bring
		+	*Click "Submit"
-	===SIP Trunk===	+	'''Done! The port process takes 1-2 business days an email notification will be send once the port is completed.'''
-	To create a SIP trunk you only need to fill some basic information.	+	==SETTING UP YOUR HT==
		+	0. Factory reset your HT, press the RESET button located at the rear panel for 7-10 seconds
-	*Type: Choose "Register SIP trunk" from the dropdown menu	+	1. Pick up your phone and dial *** to access into the system configuration menu, once in there press 0 2 and your device will play back its IP address, a number like 192.168.xxx.xxx, write it down
-	*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.	+	2. Open a new tab on your web browser and type this number, you will be attempting to log into your device's settings. The password to access is admin
-	If you get a pop-up window letting you know about the SIP port number, just click "Ok"	+	3. Once logged in, head to FXS1 tab and fill the following:
-	[[File:New SIP trunk.png|thumb|none|550px|Click to enlarge]]	+	*'''Primary SIP Server:'''
		+	*'''SIP User ID:'''
		+	*'''Authenticate ID:'''
		+	*'''Authenticate Password:'''
		+	*Name:
-	====SIP settings enhanced====	+	4. Extra settings for better performance (might not be available for some models):
-	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 Transport: TCP
-		+	*Register Expiration: 5 minutes or 300 seconds
-	=====SIP headers=====	+	*Enable SIP OPTIONS keep alive: Yes
-		+	*Allow incoming SIP messages from SIP Proxy only: Yes
-	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.	+	*Use P-Access-Network-Info Header: No
-		+	*Use P-Emergency-info Header: No
-	[[File:SIP-basic settings.png|thumb|none|550px|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.	+
-		+
-	[[File:SIP advanced settings.png|thumb|none|550px|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	+
-		+
-	[[File:IAX2 trunk.png|thumb|none|550px|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.	+
-		+
-	[[File:IAX advanced settings.png|thumb|none|550px|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"	+
-		+
-	[[File:Outbound routes general.png|thumb|none|550px|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.	+
-		+
-	[[File:Outbound routes setup.png|thumb|none|550px|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".	+
-		+
-	[[File:Inbound routes main.png|thumb|none|550px|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.	+
-		+
-	[[File:Inbound routes setup.png|thumb|none|550px|Click to enlarge]]	+

---

Latest revision as of 18:26, 4 August 2025

Contents 1 HOW DOES THIS WORK? 2 RATES 3 FIRST STEPS - ADD FUNDS 4 PORTING FROM VOIPO 5 SETTING UP YOUR HT

HOW DOES THIS WORK?

You can use our service to place and receive calls just using your internet connection, no matter where you are as long as you have internet. You can use an IP phone, ATA adapter (to use a traditional phone) or a softphone (an App installed on your cellphone or PC)

RATES

VoIP.ms is a prepaid service where you add funds to your account ($15 USD min) and the following rates are deducted from it:

You only pay a monthly fee for each of your numbers + in/out minutes usage.

Monthly fee for the phone number

• Per minute plan:
  

$0.85 USD Monthly + $0.009 USD per minute inbound
$0.01 USD outbound per minute for US/CAN

• Flat rate plan:
  

$4.25 USD Monthly, 3500 free inbound minutes
$0.01 USD outbound per minute for US/CAN

SMS/MMS rates:
$0.0075 USD per SMS
$0.02 USD per MMS

FIRST STEPS - ADD FUNDS

1. Go to Finances -> Add funds, and make your 1st payment. DO NOT use the autobilling for your 1st payment.
2. WAIT 5 minutes for balance to be updated and before you try to buy or port a number.

PORTING FROM VOIPO

1. Log in to your VoIPo account at account.voipo.com.
2. Go to Menu -> Account -> Invoices
3. Print your invoice and save it as PDF/JPG
4. Now log into your voipms portal and go to DID Numbers -> DID portability
5. Look for "If you want to start a porting process from VOIPo, Click here" and click it
6. Fill in the following:

• VOIPo Account number: One of your voipo phone numbers
• Proof of Ownership: upload your voipo invoice
• List of Numbers: list your voipo number(s) you want to bring
• Click "Submit"

Done! The port process takes 1-2 business days an email notification will be send once the port is completed.

SETTING UP YOUR HT

0. Factory reset your HT, press the RESET button located at the rear panel for 7-10 seconds

1. Pick up your phone and dial *** to access into the system configuration menu, once in there press 0 2 and your device will play back its IP address, a number like 192.168.xxx.xxx, write it down

2. Open a new tab on your web browser and type this number, you will be attempting to log into your device's settings. The password to access is admin

3. Once logged in, head to FXS1 tab and fill the following:

• Primary SIP Server:
• SIP User ID:
• Authenticate ID:
• Authenticate Password:
• Name:

4. Extra settings for better performance (might not be available for some models):

• SIP Transport: TCP
• Register Expiration: 5 minutes or 300 seconds
• Enable SIP OPTIONS keep alive: Yes
• Allow incoming SIP messages from SIP Proxy only: Yes
• Use P-Access-Network-Info Header: No
• Use P-Emergency-info Header: No