Cisco/Linksys/Sipura Analog Telephone Adapters - Revision history

Mcefan at 21:13, 19 October 2012

2012-10-19T21:13:33Z

←Older revision		Revision as of 21:13, 19 October 2012
Line 1:		Line 1:
	{{versioninfo\|810Status=applies\|810UpdatedDate=25 Aug 2011\|810UpdatedBy=twodogs}}		{{versioninfo\|810Status=applies\|810UpdatedDate=25 Aug 2011\|810UpdatedBy=twodogs}}
	~~[[Category:Tutorials]]~~
	[[Category:ATA]]		[[Category:ATA]]
	[[Category:Telecom]]		[[Category:Telecom Tutorials]]
	~~[[Category:VoIP~~]]
	{\| align="right"		{\| align="right"
	\| __TOC__		\| __TOC__

Twodogs: /* Step 4. Configure Asterisk Using FreePBX */

2012-04-18T00:17:29Z

Step 4. Configure Asterisk Using FreePBX

←Older revision		Revision as of 00:17, 18 April 2012
Line 93:		Line 93:
	===Step 4. Configure Asterisk Using FreePBX===		===Step 4. Configure Asterisk Using FreePBX===

	Note - In LMCE 1004 Freepbx is not used. Use Wizard -> devices -> phone lines to set up trunks.		'''Note''' - In LMCE 1004 Freepbx is not used. Use Wizard -> devices -> phone lines to set up trunks.

	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. The big picture here is that Asterisk is powerful software that can manage telecom for a business. There might be many extensions in different offices, and several different phone lines (trunks) coming in. So Asterisk needs to know about the extensions and trunks it has to manage. It needs rules to route incoming calls to the correct extension. Finally, it needs rules to route outbound calls to the correct trunk. We are only setting up one simple VOIP line, but we still need to visit the 4 tabs below.		For LMCE 0810, 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. The big picture here is that Asterisk is powerful software that can manage telecom for a business. There might be many extensions in different offices, and several different phone lines (trunks) coming in. So Asterisk needs to know about the extensions and trunks it has to manage. It needs rules to route incoming calls to the correct extension. Finally, it needs rules to route outbound calls to the correct trunk. We are only setting up one simple VOIP line, but we still need to visit the 4 tabs below.

Twodogs: /* Step 4. Configure Asterisk Using FreePBX */

2012-04-18T00:16:42Z

Step 4. Configure Asterisk Using FreePBX

←Older revision		Revision as of 00:16, 18 April 2012
Line 92:		Line 92:

	===Step 4. Configure Asterisk Using FreePBX===		===Step 4. Configure Asterisk Using FreePBX===

			Note - In LMCE 1004 Freepbx is not used. Use Wizard -> devices -> phone lines to set up trunks.

	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. The big picture here is that Asterisk is powerful software that can manage telecom for a business. There might be many extensions in different offices, and several different phone lines (trunks) coming in. So Asterisk needs to know about the extensions and trunks it has to manage. It needs rules to route incoming calls to the correct extension. Finally, it needs rules to route outbound calls to the correct trunk. We are only setting up one simple VOIP line, but we still need to visit the 4 tabs below.		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. The big picture here is that Asterisk is powerful software that can manage telecom for a business. There might be many extensions in different offices, and several different phone lines (trunks) coming in. So Asterisk needs to know about the extensions and trunks it has to manage. It needs rules to route incoming calls to the correct extension. Finally, it needs rules to route outbound calls to the correct trunk. We are only setting up one simple VOIP line, but we still need to visit the 4 tabs below.

Twodogs at 00:14, 18 April 2012

2012-04-18T00:14:23Z

←Older revision		Revision as of 00:14, 18 April 2012
Line 9:		Line 9:


	~~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!~~		This tutorial shows how to get your existing analog phone to work with VOIP using the SPA3102. However, any Sipura/Linksys/Cisco SPA-XXXX is configured almost identically. This family of devices was originally designed by Sipura, then bought by Linksys (owned by Cisco).

	This tutorial shows how to get your existing analog phone to work with VOIP using the Broadvoice service provider and an SPA3102. However, any Sipura/Linksys/Cisco SPA-XXXX is configured almost identically. This family of devices was originally designed by Sipura, then bought by Linksys (owned by Cisco). Also, if you chose another VOIP provider, it should not cause too many problems as long as it is recognized by linuxMCE.

	~~http://wiki.linuxmce.org/index.php/VOIP_Service_Providers~~


Line 30:		Line 28:
	==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~~. I chose Broadvoice for $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. When choosing a VOIP provider, you may want to consider whether they offer 911 service (some don't) and whether there is an existing template in LinuxMCE.

	===Get An ATA===		===Get An ATA===
Line 42:		Line 40:


	I chose the Linksys SPA3102 because it has an FXO port that allows you to set up PSTN as well as VOIP, then choose between them (by dialing 9, for instance). I thought it would be useful to keep my PSTN line until I got VOIP to work properly. As it turned out, PSTN is just as difficult to configure as VOIP. I actually got VOIP to work first, so I fired my ~~Qwest~~ telephone service and never looked back. We won't talk about setting up PSTN, though there are other tutorials that cover this. So I could have saved a few bucks by choosing an ATA without an FXO port like an SPA2100 (eBay for $15-20).		I chose the Linksys SPA3102 because it has an FXO port that allows you to set up PSTN as well as VOIP, then choose between them (by dialing 9, for instance). I thought it would be useful to keep my PSTN line until I got VOIP to work properly. As it turned out, PSTN is just as difficult to configure as VOIP. I actually got VOIP to work first, so I fired my land telephone service and never looked back. We won't talk about setting up PSTN, though there are other tutorials that cover this. So I could have saved a few bucks by choosing an ATA without an FXO port like an SPA2100 (eBay for $15-20).

	==Make It Work==		==Make It Work==

Twodogs: /* Step 2. Phone Device */

2011-12-20T23:35:45Z

Step 2. Phone Device

←Older revision		Revision as of 23:35, 20 December 2011
Line 78:		Line 78:
	Don't touch the orbiter phone. At the upper left, it says "controlled by:Onscreen Orbiter" and it is phone type "SIP6051".		Don't touch the orbiter phone. At the upper left, it says "controlled by:Onscreen Orbiter" and it is phone type "SIP6051".

	The phone we want says "controlled by: Asterisk" and is phone type "SIP". 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, but you must change it to something more secure later). 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 "~~add device~~".		The phone we want says "controlled by: Asterisk" and is phone type "SIP". 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, but you must change it to something more secure later). 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 "update".

	'''Success:''' Generating an extension number, password, and port number		'''Success:''' Generating an extension number, password, and port number

Twodogs at 13:57, 8 November 2011

2011-11-08T13:57:17Z

←Older revision		Revision as of 13:57, 8 November 2011
Line 22:		Line 22:
	* PSTN – Public Switched Telephone Network.		* PSTN – Public Switched Telephone Network.
	* FXS – Foreign Exchange Station.		* FXS – Foreign Exchange Station.
	*`FXO – Foreign Exchange Office.		* FXO – Foreign Exchange Office.
	* RJ45 – Ethernet jack.		* RJ45 – Ethernet jack.
	* RJ11 – phone jack.		* RJ11 – phone jack.

Twodogs at 13:39, 8 November 2011

2011-11-08T13:39:39Z

←Older revision		Revision as of 13:39, 8 November 2011
Line 1:		Line 1:
	{{versioninfo\|810Status=applies\|810UpdatedDate=25 Aug 2011\|810UpdatedBy=twodogs}}		{{versioninfo\|810Status=applies\|810UpdatedDate=25 Aug 2011\|810UpdatedBy=twodogs}}
	[[Category: Tutorials]]		[[Category:Tutorials]]
	[[Category: ~~Hardware~~]]		[[Category:ATA]]
	[[Category:Telecom]]		[[Category:Telecom]]
	[[Category:VoIP]]		[[Category:VoIP]]

Marie.o at 13:19, 23 October 2011

2011-10-23T13:19:28Z

←Older revision		Revision as of 13:19, 23 October 2011
Line 17:		Line 17:

	==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.
			* PSTN – Public Switched Telephone Network.
	POTS – Plain Old Telephone Service.		* FXS – Foreign Exchange Station.
			*`FXO – Foreign Exchange Office.
	PSTN – Public Switched Telephone Network.		* RJ45 – Ethernet jack.
			* RJ11 – phone jack.
	FXS – Foreign Exchange Station.

	FXO – Foreign Exchange Office.

	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 station (telephone) plugged into it. An FXO port wants to connect to the office downtown. 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 station (telephone) plugged into it. An FXO port wants to connect to the office downtown. These are both RJ11 jacks.


	==Preliminary Steps==		==Preliminary Steps==

Twodogs: /* Step 4. Configure Asterisk Using FreePBX */

2011-09-28T16:57:07Z

Step 4. Configure Asterisk Using FreePBX

←Older revision		Revision as of 16:57, 28 September 2011
Line 130:		Line 130:


	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 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
Line 147:		Line 147:

	*Getting the pattens and rules correct. You won't know for sure until you test the phone.		*Getting the pattens and rules correct. You won't know for sure until you test the phone.


	===Step 5. Configure The ATA===		===Step 5. Configure The ATA===

Twodogs: /* Important */

2011-09-28T14:00:04Z

Important

←Older revision		Revision as of 14:00, 28 September 2011
Line 236:		Line 236:

	===Important===		===Important===
	Once the phone is working, go back and change your phone's password under LMCE admin "phone device". The password should automatically be wizard'd to asterisk (check this in the FreePBX extensions tab). Then manually enter the same password in your ATA admin configuration. If you use phone extension 200 and password 200, your asterisk line can be easily hacked!		Once the phone is working, go back and change your phone's password under LMCE admin "phone device". The password should automatically be wizard'd to asterisk (check this in the FreePBX extensions tab). Then manually enter the same password in your ATA admin configuration. If you use phone extension 200 and password 200, your asterisk line can be easily hacked! Google "secure password generator" to get a strong password.