Category:Tutorials - Revision history

Mcefan at 00:06, 6 October 2012

2012-10-06T00:06:50Z

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	[[Category: Documentation]]		[[Category: Documentation]]
	Please add <nowiki>[[Category: Tutorials]]</nowiki> to the top of a page you wish to have listed here.		Please add <nowiki>[[Category: Tutorials]]</nowiki> to the top of a page you wish to have listed here.

			If you need help creating or editing pages on the wiki, please read [[Contributing to LMCE]].

Twodogs: Replaced content with "Category: Documentation Please add Category: Tutorials to the top of a page you wish to have listed here."

2011-08-26T20:56:46Z

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Twodogs at 20:53, 26 August 2011

2011-08-26T20:53:15Z

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	=Cisco/Linksys/Sipura Analog Telephone Adapter=		=Cisco/Linksys/Sipura Analog Telephone Adapter=

	~~by Twodogs, 25 Aug 2011~~

	Want to get rid of that expensive telephone service? Want to connect your phone to your LinuxMCE home automation system? Want to spend hours of frustrating software configuration? Of course!		Want to get rid of that expensive telephone service? Want to connect your phone to your LinuxMCE home automation system? Want to spend hours of frustrating software configuration? Of course!

Twodogs at 20:52, 26 August 2011

2011-08-26T20:52:46Z

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	~~[[Category: Documentation]]~~
	~~[[Category: Hardware]]~~
	~~[[Category:Telecom]]~~
	~~[[Category:VoIP]]~~
	~~{\| align="right"~~
	~~\| __TOC__~~
	\|}

	=Cisco/Linksys/Sipura Analog Telephone Adapter=		=Cisco/Linksys/Sipura Analog Telephone Adapter=

Twodogs at 20:52, 26 August 2011

2011-08-26T20:52:01Z

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	~~[[Category: Tutorials]]~~
	[[Category: Hardware]]		[[Category: Hardware]]
	[[Category:Telecom]]		[[Category:Telecom]]

Twodogs at 20:51, 26 August 2011

2011-08-26T20:51:01Z

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	===Step 4. Configure Asterisk Using FreePBX===		===Step 4. Configure Asterisk Using FreePBX===
	In LinuxMCE admin, select Advanced -> Configuration -> Phones Setup. This brings up the FreePBX window that is basically a front end used to configure Asterisk. If you chose a VOIP provider that LinuxMCE recognizes, (from the drop down list in Step 3.) you will be happy to see that Asterisk has been magically configured, but there are a few things to check and change. Only 4 tabs are of interest.		In LinuxMCE admin, select Advanced -> Configuration -> Phones Setup. This brings up the FreePBX window that is basically a front end used to configure Asterisk. If you chose a VOIP provider that LinuxMCE recognizes, (from the drop down list in Step 3.) you will be happy to see that Asterisk has been magically configured, but there are a few things to check and change. Only 4 tabs are of interest.
	Extensions Tab. Click the 3-digit extension number that matches the phone device you set up in step 2. Write down the important information to help you configure the ATA later. For me this was:
			'''Extensions Tab.''' Click the 3-digit extension number that matches the phone device you set up in step 2. Write down the important information to help you configure the ATA later. For me this was:

	Name – SPA3102		Name – SPA3102
Line 87:		Line 88:
	Port – 5060		Port – 5060

	Trunk Tab. Look on the right and click the sip trunk corresponding to the phone line you set up in step 3. (mine is SIP/broadvoice). Put a “1” in the maximum channels box, then delete everything from the “Dial Rules” box. I'll explain what this is all about shortly. If you hover your mouse over “Dial Rules” you'll get helpful information. When the changes are made, hit “submit” at the bottom, “apply” at the top, and “OK” in the popup. Or something like that – it takes three button pushes.		'''Trunk Tab.''' Look on the right and click the sip trunk corresponding to the phone line you set up in step 3. (mine is SIP/broadvoice). Put a “1” in the maximum channels box, then delete everything from the “Dial Rules” box. I'll explain what this is all about shortly. If you hover your mouse over “Dial Rules” you'll get helpful information. When the changes are made, hit “submit” at the bottom, “apply” at the top, and “OK” in the popup. Or something like that – it takes three button pushes.

	Outbound Routes Tab. Look on the right and you'll see a route for each trunk. We only set up one trunk, so click on it (mine was “broadvoice”). You'll have to change some things in the “Dial Patterns” box. Make it look like this:		'''Outbound Routes Tab.''' Look on the right and you'll see a route for each trunk. We only set up one trunk, so click on it (mine was “broadvoice”). You'll have to change some things in the “Dial Patterns” box. Make it look like this:

	112		112
Line 109:		Line 110:
	So whenever I dialed 9+the number, Asterisk would know to route that call to the PSTN trunk. Of course we would have to remove the “9” prefix before actually dialing the number. To do that, we would open up the trunk tab and put in a “Dial Rule” to subtract the leading “9” whenever we dial an 8 or 11 digit number.		So whenever I dialed 9+the number, Asterisk would know to route that call to the PSTN trunk. Of course we would have to remove the “9” prefix before actually dialing the number. To do that, we would open up the trunk tab and put in a “Dial Rule” to subtract the leading “9” whenever we dial an 8 or 11 digit number.

	Inbound Routes Tab. Nothing to change here. At the bottom you'll see “Set Destination” will have a check next to “Custom App – custom-linuxmce,102,1”. This allows LinuxMCE to determine what is to be done with incoming calls. For instance, when you set your home's alarm, it can automatically redirect incoming calls to your cell phone.		'''Inbound Routes Tab.''' Nothing to change here. At the bottom you'll see “Set Destination” will have a check next to “Custom App – custom-linuxmce,102,1”. This allows LinuxMCE to determine what is to be done with incoming calls. For instance, when you set your home's alarm, it can automatically redirect incoming calls to your cell phone.

	===Step 5. Configure The ATA===		===Step 5. Configure The ATA===
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	We won't mess with any of the “Router” tabs. It defaults to DHCP and that's how we'll leave it. We only have to modify two “Voice” tabs.		We won't mess with any of the “Router” tabs. It defaults to DHCP and that's how we'll leave it. We only have to modify two “Voice” tabs.

	SIP Tab. Change “RTP Packet Size” to 0.020		'''SIP Tab.''' Change “RTP Packet Size” to 0.020

	Line 1 Tab. Get the info you wrote in step 4, and change the following:		'''Line 1 Tab.''' Get the info you wrote in step 4, and change the following:

	Line Enable: Yes		Line Enable: Yes

Twodogs at 20:48, 26 August 2011

2011-08-26T20:48:42Z

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	Inbound Routes Tab. Nothing to change here. At the bottom you'll see “Set Destination” will have a check next to “Custom App – custom-linuxmce,102,1”. This allows LinuxMCE to determine what is to be done with incoming calls. For instance, when you set your home's alarm, it can automatically redirect incoming calls to your cell phone.		Inbound Routes Tab. Nothing to change here. At the bottom you'll see “Set Destination” will have a check next to “Custom App – custom-linuxmce,102,1”. This allows LinuxMCE to determine what is to be done with incoming calls. For instance, when you set your home's alarm, it can automatically redirect incoming calls to your cell phone.

	Step 5. Configure The ATA		===Step 5. Configure The ATA===
	If the ATA is not new, you should reset the factory default. Connect an analog phone to the FXS port, and dial ****73738# from the telephone keypad (check your manual). It asks you to enter 1 to confirm.		If the ATA is not new, you should reset the factory default. Connect an analog phone to the FXS port, and dial ****73738# from the telephone keypad (check your manual). It asks you to enter 1 to confirm.

	The ATA has it's own admin screen – it is not configured through LinumMCE admin. The ATA manual tells how to get access the admin setup. I plugged a laptop directly into the “ethernet” port of the ATA, then opened a web browser and pointed it to IP address 192.168.0.1 (because the ATA manual told me).		The ATA has it's own admin screen – it is not configured through LinumMCE admin. The ATA manual tells how to get access the admin setup. I plugged a laptop directly into the “ethernet” port of the ATA, then opened a web browser and pointed it to IP address 192.168.0.1 (because the ATA manual told me).

	Look on the upper right side of the page and make sure you are logged in as “admin” and “advanced”.		Look on the upper right side of the page and make sure you are logged in as “admin” and “advanced”.
	We won't mess with any of the “Router” tabs. It defaults to DHCP and that's how we'll leave it. We only have to modify two “Voice” tabs.		We won't mess with any of the “Router” tabs. It defaults to DHCP and that's how we'll leave it. We only have to modify two “Voice” tabs.

	SIP Tab. Change “RTP Packet Size” to 0.020		SIP Tab. Change “RTP Packet Size” to 0.020

	Line 1 Tab. Get the info you wrote in step 4, and change the following:		Line 1 Tab. Get the info you wrote in step 4, and change the following:

	Line Enable: Yes		Line Enable: Yes

	SIP Port: 5060 (or whatever you phone device uses)		SIP Port: 5060 (or whatever you phone device uses)

	Proxy: 192.168.80.1 (fixed IP address of your LinuxMCE core – should be same for everyone)		Proxy: 192.168.80.1 (fixed IP address of your LinuxMCE core – should be same for everyone)

	Register: Yes		Register: Yes

	Register Expires: 3600		Register Expires: 3600

	Make Call Without Reg: yes		Make Call Without Reg: yes

	Ans Call Without Reg: yes		Ans Call Without Reg: yes

	Display Name: 200 (or your phone device's extension)		Display Name: 200 (or your phone device's extension)

	UserID: 200 (or your phone device's extension)		UserID: 200 (or your phone device's extension)

	Password: 200 (or your phone device's pasword)		Password: 200 (or your phone device's pasword)

	Use AuthID: no		Use AuthID: no

	Preferred Codec: G711u		Preferred Codec: G711u

	Use Pref Codec Only: no		Use Pref Codec Only: no

	Dial Plan: (xxx\|xxxx\|xxxxxxx\|xxxxxxxxxx\|1xxxxxxxxxx)		Dial Plan: (xxx\|xxxx\|xxxxxxx\|xxxxxxxxxx\|1xxxxxxxxxx)

	Submit the changes and cross your fingers while it thinks for a few seconds. When the admin screen returns, click on the “Router Status” tab and you should see that your ATA has registered and picked up a unique IP address from the core.		Submit the changes and cross your fingers while it thinks for a few seconds. When the admin screen returns, click on the “Router Status” tab and you should see that your ATA has registered and picked up a unique IP address from the core.

	Test It		===Test It===
	Grab your cell phone and dial your new VOIP number.		Grab your cell phone and dial your new VOIP number.

Twodogs: Cisco/Linksys/Sipura Analog Telephone Adapter (25 Aug 2011)

2011-08-26T20:46:19Z

Cisco/Linksys/Sipura Analog Telephone Adapter (25 Aug 2011)

←Older revision		Revision as of 20:46, 26 August 2011
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	=Cisco/Linksys/Sipura Analog Telephone Adapter (25 Aug 2011)=		=Cisco/Linksys/Sipura Analog Telephone Adapter=

			by Twodogs, 25 Aug 2011

	Want to get rid of that expensive telephone service? Want to connect your phone to your LinuxMCE home automation system? Want to spend hours of frustrating software configuration? Of course!		Want to get rid of that expensive telephone service? Want to connect your phone to your LinuxMCE home automation system? Want to spend hours of frustrating software configuration? Of course!

Line 42:		Line 45:
	I chose the Linksys SPA3102 because there seemed to be a lot of information on how to configure it for LinuxMCE. There is a line of very similar ATA devices originally designed by Sipura, then bought by Linksys (owned by Cisco). So just about any ATA called a Sipura/Linksys/Cisco SPA-XXXX will probably work with this tutorial. The difference between these devices is mainly the number of different ports. For instance, the SPA2102 has two FXS ports for two home phone lines. The SPA3102 trades one FXS port for an FXO port (in case I ever wanted to get back with my phone company). I could configure the ATA so that I could access either my VOIP line or my PSTN line (by dialing “9” just like a corporate office). I am not using this feature, so this tutorial will not cover setting up a PSTN line.		I chose the Linksys SPA3102 because there seemed to be a lot of information on how to configure it for LinuxMCE. There is a line of very similar ATA devices originally designed by Sipura, then bought by Linksys (owned by Cisco). So just about any ATA called a Sipura/Linksys/Cisco SPA-XXXX will probably work with this tutorial. The difference between these devices is mainly the number of different ports. For instance, the SPA2102 has two FXS ports for two home phone lines. The SPA3102 trades one FXS port for an FXO port (in case I ever wanted to get back with my phone company). I could configure the ATA so that I could access either my VOIP line or my PSTN line (by dialing “9” just like a corporate office). I am not using this feature, so this tutorial will not cover setting up a PSTN line.

	=Make It Work=		==Make It Work==
	We'll cover this in five steps. I suggest following the order below in order to prevent having to revisit screens.		We'll cover this in five steps. I suggest following the order below in order to prevent having to revisit screens.
	1. Install hardware
	2. Configure LMCE phone device		1. Install hardware.
	3. Configure LMCE phone line device
	4. Configure Asterisk using FreePBX		2. Configure LMCE phone device.
	5. Configure The ATA
			3. Configure LMCE phone line device.

			4. Configure Asterisk using FreePBX.

			5. Configure The ATA.

	There are a zillion different screens, tabs, options, etc. to fiddle with. The configuration process is like opening a safe: if you know the combination it only takes a few seconds. If you don't know, it could take forever. To make the process easier, we'll talk about what not to touch. It also helps to remember that LinuxMCE is basically a router. We're not really trying to get all these devices to talk with each other so much as we are trying to get them each to talk with LinuxMCE. This is helpful because it eliminates variables during troubleshooting. For instance, when you get your phone line device to register, that means LMCE is properly connected to your VOIP provider. If your ATA fails to register, the problem is your ATA configuration, not your phone line.		There are a zillion different screens, tabs, options, etc. to fiddle with. The configuration process is like opening a safe: if you know the combination it only takes a few seconds. If you don't know, it could take forever. To make the process easier, we'll talk about what not to touch. It also helps to remember that LinuxMCE is basically a router. We're not really trying to get all these devices to talk with each other so much as we are trying to get them each to talk with LinuxMCE. This is helpful because it eliminates variables during troubleshooting. For instance, when you get your phone line device to register, that means LMCE is properly connected to your VOIP provider. If your ATA fails to register, the problem is your ATA configuration, not your phone line.

	Step 1. Network		===Step 1. Network===
	I thought I might have to fix some IP addresses or fool around with the modem or router, but no, I just plugged everything in as follows:		I thought I might have to fix some IP addresses or fool around with the modem or router, but no, I just plugged everything in as follows:

Line 59:		Line 67:
	The SPA3102 has 4 ports: internet, ethernet, FXS, and FXO. I use only two: the internet port connects it to the core through the switch, and the FXS port connect to my analog phone. During setup, I connected a laptop to the ethernet port to access the SPA3102 admin screens. I do not have a PSTN line so the FXO port goes unused. Your ATA quick setup manual is your friend.		The SPA3102 has 4 ports: internet, ethernet, FXS, and FXO. I use only two: the internet port connects it to the core through the switch, and the FXS port connect to my analog phone. During setup, I connected a laptop to the ethernet port to access the SPA3102 admin screens. I do not have a PSTN line so the FXO port goes unused. Your ATA quick setup manual is your friend.

	Step 2. Phone Device		===Step 2. Phone Device===
	Log in to the LinuxMCE web admin. Go to Wizard -> Devices -> Phones -> Add Device (bottom of the page). For device template select “Generic SIP softphone” which is template 1734. When the device is added, it will show an extension number – don't change it. It also shows a password – change this to match the extension number (just to make life easier). On the left of the screen, rename the phone to whatever you like – I chose SPA3102. Under that field, use the drop down to assign the phone to a room. Write down the extension and password because we'll use it later. The new extension also has been assigned to a port (probably 5060 but we'll check that later). Now hit the update button.		Log in to the LinuxMCE web admin. Go to Wizard -> Devices -> Phones -> Add Device (bottom of the page). For device template select “Generic SIP softphone” which is template 1734. When the device is added, it will show an extension number – don't change it. It also shows a password – change this to match the extension number (just to make life easier). On the left of the screen, rename the phone to whatever you like – I chose SPA3102. Under that field, use the drop down to assign the phone to a room. Write down the extension and password because we'll use it later. The new extension also has been assigned to a port (probably 5060 but we'll check that later). Now hit the update button.

	Step 3. Phone Line Device		===Step 3. Phone Line Device===
	Sarah can create a phone line during the installation wizard, but I thought it was easier to do through the LinuxMCE admin page. Besides, you'll probably screw up the configuration and have to delete and reinstall everything anyway. Your VOIP provider's website has the information to set up the phone line (Broadvoice has an Account->Show Settings tab). To get everything ready, log into LinuxMCE admin, then open another tab and log into your VOIP provider. Now tab back to LinuxMCE admin and click Phone Line-> Add. A screen will pop up allowing you to enter your VOIP provider information. Start up at the top. Leave the “prepend” boxes blank. For most people the local number length is 7, but here in Denver we have several area codes and I had to enter 10. The stuff below is copied directly from your web provider. Broadvoice is one of the VOIP providers appearing in the drop down window, so I just selected it. If you chose an unrecognized provider, you'll have to look for a tutorial telling you what to do. A gotcha here is that the username and password are the ones for your phone line, not the ones to log into the website. For Broadvoice, the username is your 10 digit phone number, and the password is a bunch of random letters. Hit submit when done. The phone line should almost immediately show “registered”. If not, you entered something wrong. Trying to correct it doesn't work too well. Best to delete the phone device and start over – it only takes a minute. Go no further until you get a “registered” indication.		Sarah can create a phone line during the installation wizard, but I thought it was easier to do through the LinuxMCE admin page. Besides, you'll probably screw up the configuration and have to delete and reinstall everything anyway. Your VOIP provider's website has the information to set up the phone line (Broadvoice has an Account->Show Settings tab). To get everything ready, log into LinuxMCE admin, then open another tab and log into your VOIP provider. Now tab back to LinuxMCE admin and click Phone Line-> Add. A screen will pop up allowing you to enter your VOIP provider information. Start up at the top. Leave the “prepend” boxes blank. For most people the local number length is 7, but here in Denver we have several area codes and I had to enter 10. The stuff below is copied directly from your web provider. Broadvoice is one of the VOIP providers appearing in the drop down window, so I just selected it. If you chose an unrecognized provider, you'll have to look for a tutorial telling you what to do. A gotcha here is that the username and password are the ones for your phone line, not the ones to log into the website. For Broadvoice, the username is your 10 digit phone number, and the password is a bunch of random letters. Hit submit when done. The phone line should almost immediately show “registered”. If not, you entered something wrong. Trying to correct it doesn't work too well. Best to delete the phone device and start over – it only takes a minute. Go no further until you get a “registered” indication.

	Finally click “settings” to ensure that all drop downs say “ring extensions” and put a checkmark next to the phone you setup in step 2. Otherwise when we're all done anyone who tries to call will get a “party not available” message.		Finally click “settings” to ensure that all drop downs say “ring extensions” and put a checkmark next to the phone you setup in step 2. Otherwise when we're all done anyone who tries to call will get a “party not available” message.

	Step 4. Configure Asterisk Using FreePBX		===Step 4. Configure Asterisk Using FreePBX===
	In LinuxMCE admin, select Advanced -> Configuration -> Phones Setup. This brings up the FreePBX window that is basically a front end used to configure Asterisk. If you chose a VOIP provider that LinuxMCE recognizes, (from the drop down list in Step 3.) you will be happy to see that Asterisk has been magically configured, but there are a few things to check and change. Only 4 tabs are of interest.		In LinuxMCE admin, select Advanced -> Configuration -> Phones Setup. This brings up the FreePBX window that is basically a front end used to configure Asterisk. If you chose a VOIP provider that LinuxMCE recognizes, (from the drop down list in Step 3.) you will be happy to see that Asterisk has been magically configured, but there are a few things to check and change. Only 4 tabs are of interest.
	Extensions Tab. Click the 3-digit extension number that matches the phone device you set up in step 2. Write down the important information to help you configure the ATA later. For me this was:		Extensions Tab. Click the 3-digit extension number that matches the phone device you set up in step 2. Write down the important information to help you configure the ATA later. For me this was:

	Name – SPA3102		Name – SPA3102

	Extension – 200		Extension – 200

	Secret – 200 (this is the password you supplied)		Secret – 200 (this is the password you supplied)

	Port – 5060		Port – 5060

	Trunk Tab. Look on the right and click the sip trunk corresponding to the phone line you set up in step 3. (mine is SIP/broadvoice). Put a “1” in the maximum channels box, then delete everything from the “Dial Rules” box. I'll explain what this is all about shortly. If you hover your mouse over “Dial Rules” you'll get helpful information. When the changes are made, hit “submit” at the bottom, “apply” at the top, and “OK” in the popup. Or something like that – it takes three button pushes.		Trunk Tab. Look on the right and click the sip trunk corresponding to the phone line you set up in step 3. (mine is SIP/broadvoice). Put a “1” in the maximum channels box, then delete everything from the “Dial Rules” box. I'll explain what this is all about shortly. If you hover your mouse over “Dial Rules” you'll get helpful information. When the changes are made, hit “submit” at the bottom, “apply” at the top, and “OK” in the popup. Or something like that – it takes three button pushes.

	Outbound Routes Tab. Look on the right and you'll see a route for each trunk. We only set up one trunk, so click on it (mine was “broadvoice”). You'll have to change some things in the “Dial Patterns” box. Make it look like this:		Outbound Routes Tab. Look on the right and you'll see a route for each trunk. We only set up one trunk, so click on it (mine was “broadvoice”). You'll have to change some things in the “Dial Patterns” box. Make it look like this:

	112		112

	411		411

	911		911

	NXXXXXX		NXXXXXX

	NXXNXXXXXX		NXXNXXXXXX

	What are patterns and rules about? When you dial a number on your phone, Asterisk checks the dial pattern. So if it sees 112, 411, 911, any 7-digit number, or any 10-digit number, then it will route those calls to my Broadvoice sip trunk. If I had kept my PSTN line and setup another trunk, then I could make a dial pattern for that trunk like:		What are patterns and rules about? When you dial a number on your phone, Asterisk checks the dial pattern. So if it sees 112, 411, 911, any 7-digit number, or any 10-digit number, then it will route those calls to my Broadvoice sip trunk. If I had kept my PSTN line and setup another trunk, then I could make a dial pattern for that trunk like:

	9NXXXXXX		9NXXXXXX

	9NXXNXXXXXX		9NXXNXXXXXX

	So whenever I dialed 9+the number, Asterisk would know to route that call to the PSTN trunk. Of course we would have to remove the “9” prefix before actually dialing the number. To do that, we would open up the trunk tab and put in a “Dial Rule” to subtract the leading “9” whenever we dial an 8 or 11 digit number.		So whenever I dialed 9+the number, Asterisk would know to route that call to the PSTN trunk. Of course we would have to remove the “9” prefix before actually dialing the number. To do that, we would open up the trunk tab and put in a “Dial Rule” to subtract the leading “9” whenever we dial an 8 or 11 digit number.

	Inbound Routes Tab. Nothing to change here. At the bottom you'll see “Set Destination” will have a check next to “Custom App – custom-linuxmce,102,1”. This allows LinuxMCE to determine what is to be done with incoming calls. For instance, when you set your home's alarm, it can automatically redirect incoming calls to your cell phone.		Inbound Routes Tab. Nothing to change here. At the bottom you'll see “Set Destination” will have a check next to “Custom App – custom-linuxmce,102,1”. This allows LinuxMCE to determine what is to be done with incoming calls. For instance, when you set your home's alarm, it can automatically redirect incoming calls to your cell phone.

Twodogs: Cisco/Linksys/Sipura Analog Telephone Adapter (25 Aug 2011)

2011-08-26T20:40:59Z

Cisco/Linksys/Sipura Analog Telephone Adapter (25 Aug 2011)

←Older revision		Revision as of 20:40, 26 August 2011
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	=~~Subject~~=		=Cisco/Linksys/Sipura Analog Telephone Adapter (25 Aug 2011)=
	Want to get rid of that expensive telephone service? Want to connect your phone to your LinuxMCE home automation system? Want to spend hours of frustrating software configuration? Of course ~~we do~~!		Want to get rid of that expensive telephone service? Want to connect your phone to your LinuxMCE home automation system? Want to spend hours of frustrating software configuration? Of course!

	=Glossary=		==Glossary==
	VOIP – Voice Over Internet Protocol.		VOIP – Voice Over Internet Protocol.
	ATA – Analog Telephone Adapter
	POTS – Plain Old Telephone Service		ATA – Analog Telephone Adapter.
	PSTN – Public Switched Telephone Network
			POTS – Plain Old Telephone Service.

			PSTN – Public Switched Telephone Network.

	FXS – Foreign Exchange Station.		FXS – Foreign Exchange Station.

	FXO – Foreign Exchange Office.		FXO – Foreign Exchange Office.
	RJ45 – Ethernet jack
	RJ11 – phone jack		RJ45 – Ethernet jack.

			RJ11 – phone jack.

	Its easy to confuse FXS and FXO. If you can remember that telecom engineers call telephones “stations”, then you can remember that an FXS port wants to have a telephone plugged into it. An FXO port wants to connect to the telephone office. These are both RJ11 jacks.		Its easy to confuse FXS and FXO. If you can remember that telecom engineers call telephones “stations”, then you can remember that an FXS port wants to have a telephone plugged into it. An FXO port wants to connect to the telephone office. These are both RJ11 jacks.

	=Preliminary Steps=		==Preliminary Steps==
	Get a VOIP provider		===Get a VOIP provider===
	You will need a VOIP provider to give you a phone number and provide access to the PSTN. This allows you to call your mom and vice versa. Without a VOIP provider you could only make computer to computer calls to your geek friends. I wanted a provider that offered 911 service (some don't) and that had an existing template in LinuxMCE. That's why I chose Broadvoice. It costs me $11 per month instead of the $70 I paid to the phone company.		You will need a VOIP provider to give you a phone number and provide access to the PSTN. This allows you to call your mom and vice versa. Without a VOIP provider you could only make computer to computer calls to your geek friends. I wanted a provider that offered 911 service (some don't) and that had an existing template in LinuxMCE. That's why I chose Broadvoice. It costs me $11 per month instead of the $70 I paid to the phone company.

	Get An ATA		===Get An ATA===
	The ATA is a device the size of a router. It has one or more RJ45 jacks that connect it to your LinuxMCE network, and it has one or more RJ11 jacks that provide an FXS port for your existing analog telephone. An ATA is only one of three ways to use VOIP with LinuxMCE. You could...		The ATA is a device the size of a router. It has one or more RJ45 jacks that connect it to your LinuxMCE network, and it has one or more RJ11 jacks that provide an FXS port for your existing analog telephone. An ATA is only one of three ways to use VOIP with LinuxMCE. You could...
	1. Plug a USB headset into your orbiter and use the softphone. (cheap, but inconvenient)		1. Plug a USB headset into your orbiter and use the softphone. (cheap, but inconvenient)
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	=Make It Work=		=Make It Work=
	We'll cover this in five steps. I suggest following the order below in order to ~~avoid~~ having to revisit screens.		We'll cover this in five steps. I suggest following the order below in order to prevent having to revisit screens.
	1. Install hardware		1. Install hardware
	2. Configure LMCE phone device		2. Configure LMCE phone device

Twodogs: /* Glossary */

2011-08-26T20:31:38Z

Glossary

←Older revision		Revision as of 20:31, 26 August 2011
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	=Glossary=		=Glossary=
	VOIP – Voice Over Internet Protocol		VOIP – Voice Over Internet Protocol.
	ATA – Analog Telephone Adapter		ATA – Analog Telephone Adapter
	POTS – Plain Old Telephone Service		POTS – Plain Old Telephone Service
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	Its easy to confuse FXS and FXO. If you can remember that telecom engineers call telephones “stations”, then you can remember that an FXS port wants to have a telephone plugged into it. An FXO port wants to connect to the telephone office. These are both RJ11 jacks.		Its easy to confuse FXS and FXO. If you can remember that telecom engineers call telephones “stations”, then you can remember that an FXS port wants to have a telephone plugged into it. An FXO port wants to connect to the telephone office. These are both RJ11 jacks.

			=Preliminary Steps=
	Get a VOIP provider		Get a VOIP provider
	You will need a VOIP provider to give you a phone number and provide access to the PSTN. This allows you to call your mom and vice versa. Without a VOIP provider you could only make computer to computer calls to your geek friends. I wanted a provider that offered 911 service (some don't) and that had an existing template in LinuxMCE. That's why I chose Broadvoice. It costs me $11 per month instead of the $70 I paid to the phone company.		You will need a VOIP provider to give you a phone number and provide access to the PSTN. This allows you to call your mom and vice versa. Without a VOIP provider you could only make computer to computer calls to your geek friends. I wanted a provider that offered 911 service (some don't) and that had an existing template in LinuxMCE. That's why I chose Broadvoice. It costs me $11 per month instead of the $70 I paid to the phone company.
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	I chose the Linksys SPA3102 because there seemed to be a lot of information on how to configure it for LinuxMCE. There is a line of very similar ATA devices originally designed by Sipura, then bought by Linksys (owned by Cisco). So just about any ATA called a Sipura/Linksys/Cisco SPA-XXXX will probably work with this tutorial. The difference between these devices is mainly the number of different ports. For instance, the SPA2102 has two FXS ports for two home phone lines. The SPA3102 trades one FXS port for an FXO port (in case I ever wanted to get back with my phone company). I could configure the ATA so that I could access either my VOIP line or my PSTN line (by dialing “9” just like a corporate office). I am not using this feature, so this tutorial will not cover setting up a PSTN line.		I chose the Linksys SPA3102 because there seemed to be a lot of information on how to configure it for LinuxMCE. There is a line of very similar ATA devices originally designed by Sipura, then bought by Linksys (owned by Cisco). So just about any ATA called a Sipura/Linksys/Cisco SPA-XXXX will probably work with this tutorial. The difference between these devices is mainly the number of different ports. For instance, the SPA2102 has two FXS ports for two home phone lines. The SPA3102 trades one FXS port for an FXO port (in case I ever wanted to get back with my phone company). I could configure the ATA so that I could access either my VOIP line or my PSTN line (by dialing “9” just like a corporate office). I am not using this feature, so this tutorial will not cover setting up a PSTN line.

	Make It Work		=Make It Work=
	We'll cover this in five steps. I suggest following the order below in order to ~~prevent~~ having to revisit screens.		We'll cover this in five steps. I suggest following the order below in order to avoid having to revisit screens.
	1. Install hardware		1. Install hardware
	2. Configure LMCE phone device		2. Configure LMCE phone device