LinuxMCE - User contributions [en]

Linksys SPA3000

2012-01-16T15:47:35Z

Tred: /* General Information */

[[Category:Hardware]]
[[Category:Telecom]]
[[Category:VoIP]]
[[Category:Tutorials]]
[[Category:ATA]]
{{merge|Linksys SPA3102}}
== General Information ==

Note: '''these instructions are obsolete'''. Go to the Linksys SPA3102 page for newest.

The Sipura (now bought out by Linksys) spa 3000 is a pstn (analog phone line) to Asterisk gateway device that provides a SIP interface for 1 FXO port (analog phone line) and 1 FXS port (analog phone). This allows you to use the Telecom section of LMCE, and use Asterisk to send/recieve/route phone calls.

Most office environments have a phone in each room with their own extension number. Most homes, however, are set up with 1 common line (extension) that all phones are connected to - when a call comes in, all phones ring together. LMCE will allow you to have either setup (if you have enough FXS ports/SIP Phones to support the number of extensions you want). Since the spa 3000 has only 1 FXS port, I am going to show you how to set it up so that all phones on the main house phone line will be treated as one extension and ring together. The picture below illustrates the installation method that I'm going to outline here.
[[Image:spa3000_setup.png]]

[[Image:spa3000_setup_DSL.png]]

Another interesting feature of the spa3000, is that if the LMCE network goes down, or if there is a loss of power, the spa3000 will bridge the FXO and FXS ports, bypassing LMCE and Asterisk altogether, so that you will still have use of your phones in such emergencies.

Also, please realize that these instructions are for manually setting up the spa3000. In the near future, the process will be automated, and if this feature is available at the time you are reading this, then you should go the automatic route.

Please set aside some time to do this manual setup - as you will see, there are a lot of steps involved and it may take you quite some time!

:-:These Instructions are obsolete. Review the Linksys SPA3102 for current ones:-:

Note: these instructions are obsolete. Go to the Linksys SPA3102 page for newest.

== Setup ==
We are going to accomplish this setup in 3 steps:
1- Configuring the Phone Line in the web admin (FreePBX)
2- Adding the device for the internal phone line (FXS port)
3) Configuring the spa3000 (as we will need information from the previous 2 steps to complete)

'''Step 1 - Configuring the Phone Line'''

First, log into the web admin at 192.168.80.1, then navigate to Advanced->Configuration->Phones Setup. This will bring you to the FreePBX Admin page
On the left hand side, click on "Trunks", then click "Add SIP Trunk" on the page the follows. You will now have to fill out the folowing information:

Outbound Caller ID: House Line

Never Override Caller ID: leave unchecked

Maximum Channels: 1 (this must be set to 1 as a pstn line can only handle 1 call at a time)

Disable Trunk: leave unchecked

Monitor Trunk Failures: leave the Enable checkbox unchecked

Dial Rules: Leave blank

Dial rules wizards: leave this alone

Outbound Dial Prefix: leave blank

Trunk Name: House Line

Peer Details:
allow=ulaw
canreinvite=no
context=from-trunk
disallow=all
dtmfmode=rfc2833
host=dynamic
incominglimit=1
nat=never
port=5061
qualify=yes
secret=lmce
type=friend
username=House Line

USER Context: leave blank

USER Details: leave blank

Register String: Leave blank

go ahead and save the changes now!

Next, we are going to setup the Outbound Routes. In FreePBX, on the left, click on "Outbound Routes", and add the following for the new route:

Route Name: House Line Out

Route Password: leave blank

Emergency Dialing: leave unchecked

Intra Company Route: leave unchecked

Music On Hold?: leave at default

Dial Paterns:
112
411
911
9|.XXX

Dial Patterns Wizards: leave this alone

Trunk Sequence: select the SIP/House Line option from the drop down at the top position (this is the trunk we created earlier). If you already have VOIP setup, and you are adding the pstn as a second line, you may want to change the sequence order to suit your needs. All this sequence does, is if a call fails while trying the first sequence, then the call will be tried with the next one, and so on.

Submit your changes!

Next, we will setup the inbound routes. In FreePBX, click on "Inbound Routes", and add the following for the new route:

Description: Leave blank

DID Number : your line phone number, I.e. 800-555-1212

Caller ID Number: leave blank

Zaptel Channel: Leave blank

Fax Handling Section - leave these as-is

Privacy Manager : No

Options Settings: Leave as-is

lastly, in the Set Destination settings, select the last radio button, Custom App. Type the following in the text box for Custom App: custom-linuxmce,102,1

submit your changes.

One last thing to do while we are in FreePBX. You should see towards the top of the page an orange bar that should say "Apply Configuration Changes". Click this, then when prompted, select to continue with reload.

Next, go to Wizard->Devices->Phone Lines (on the left pane in the LMCE web admin, not in FreePBX). We are going to add a dummy line (NOTE: this is a temporary hack for now! I won't go into too many details other than saying that it will allow you to use the "Settings" link next to the listing to do some call routing on your pstn line!)
Use the dropdown to select broadvoice. Once you do this, you will see a form to fill in some data. Just put whatever you want in the fields, they won't be used with this hack! After you are done, you will see it listed as a phone line - use the "settings" link next to it to do call routing depending on security mode!
(Note: After creating this "dummy" phone line, go back to the FreePBX admin, and look at the Incoming Route for the broadvoice line. Look at the Custom App option at the bottom of the page. It should contain that same custom-linuxmce,102,1. If it does not, go back to the Incoming Route for the House Line, and change its custom app line to be the same as this one! From my tests, it should be the same (though the 3 digit number can change, so check this to be sure).

'''Step 2 - Adding the device template for the FXS port (your internal phone/phones)'''
In the web admin, on the left pane under devices, select Phones. On the resulting page, you should see all of the Orbiter Embedded Phones that your system already has. At the bottom of this page, click the "Add Device" button, then select "Generic Phone" from the Device Template picker. Click the "Pick device template" button.
Now notice that you have a new phone device in you list. Change the name to something like "House Line", set the PK_FloorplanObjectType to Pluto Telephone, set the Phone Type to SIP. Notice that the PhoneNumber is already filled out. LEAVE THIS ALONE AND REMEMBER IT! This will be the extension of your internal analog phones (or whatever single phone you have plugged into the FXS Port!) Also, you can assign this phone a room (it really doesn't matter which one you choose) and make sure the "Controlled By" column says Asterisk. If it does not say "Asterisk", click on it and choose Asterisk (Asterisk) - from the dropdown. When finished, Hit the Update button at the bottom of the screen to save these changes. You will have to reload your router, and do a quick regen on all orbiters manually (from the Devices->Orbiters section)

'''Step 3 - Configuring the spa3000 (almost done!)'''
This part assumes a factory default starting point. To do this, hook up an analog phone to the FXS (phone) port, and dial ****
When prompted, enter 73738# from the telephone keypad. This will reset the unit to factory defaults.

Now go ahead and connect the spa3000's LAN port to your internal network switch. We have to assign it a static IP address, so again dial **** from the connected handset. When prompted, enter 110# and the IP address will be spoken back to you. Write it down. Open up a web browser on the network, and enter the IP address to go to the spa3000 admin page. In the upper right hand corner, click on the links for both Admin Login and Advanced.

Go to the System tab. Under "Internet Connection Type" do the following:

DHCP: no
Static IP: enter a static IP address. I used 192.168.80.253
Netmask: 255.255.255.0
Gateway: 192.168.80.1

Update for SPA-3102 
Router -> WAN Setup 
set WAN Web Server to YES 
disconnect the LAN port from the network and connect the WAN port to the network. 

Next, go to the SIP tab. Change the following:

RTP Packet Size: 0.020

Next, go to the Line 1 tab. Change the following:

Line Enable: Yes

SIP Port: 5060

Proxy: 192.168.80.1

Register: Yes

Register Expires: 300

Make Call Without Reg: yes

Ans Call Without Reg: yes

Display Name: <extension number LMCE assigned your Generic Phone device> (mine was 206 for example)

UserID: <again, extension number LMCE assigned your Generic Phone device>

Password: <once again, extension number LMCE assigned your Generic Phone device>

Use AuthID: no

Preferred Codec: make sure its set to G711u

Lastly, go to the PSTN Line tab. Ensure the following settings:

Line Enable: Yes

SIP Port: 5061

Proxy: 192.168.80.1

Register: yes

Register Expires: 300

Make Calls Without Reg: Yes

Ans Calls Without Reg: Yes

Display Name: PSTN Call (this will be displayed in place of unknown caller ID's. You may want to name this "Unknown Call" or similar)

user ID: House Line (this must match the trunk name you did in the FreePBX setup!)

Password: lmce

Use Auth ID: no

Preferred Codec: make sure it is set to G711u

DTMF Process INFO: yes

DTMF Process AVT: yes

DTMF TxMethod: Auto

Dial Plan 2: (S0<:phone_number_here>) i.e. (S0<:800-555-1212>) (this is very important - it must match the DID number entered in FreePBX)

VoiP-To-PSTN Gateway Enable: yes

Voip Caller Auth Method: None

Voip PIN Max Retry: 3

One Stage Dialing: yes

Line1 VoIP Caller DP: None

ViOP Valler Default DP: None

Line1 Fallback DP: None (this may need changed to make sure incoming calls come in when the power goes out - will experiment)

PSTN-To-VoIP Gateway Enable: yes

PSTN Caller Auth Method: none

PSTN Right Thru Line 1: no

PSTN Pin Max Retry: 3

PSTN CID for VioP CID: yes

PSTN Caller Default DP: 2 (this is the dialplan 2 that we set up above!)

Off Hook While Calling VoIP: no

Line 1 Signal Hook Flash to PSTN: Disabled

VoIP Answer Delay: 0

PSTN Answer Delay: 3 (this allows enough time for the caller ID information to be passed along)

SPA to PSTN Gain and PSTN to SPA Gain - these can be adjusted to amplify the incoming/outgoing volume to a comfortable level. I like mine set at 6

That should be it! Enjoy LMCE Telecom on your old analog phone line!

== Disconnect Tone ==
If your line doesn't hang up after a call then you may need to change the disconnect tone. Different countries have different disconnect tones. Go to the SPA3000's configuration page, click on Admin Login and then Advanced. Now head to the PSTN Line tab. Scroll down to the PSTN Disconnect Detection. There is a setting called "Disconnect Tone" and that's what needs to change.

The format entered needs to be of the following: X@-30,X@-30;Y(A/B/1+2)

where:

X = The frequency in Hz.

Y = The cycles the pattern repeats

A = Time in seconds for the frequency

B = Time in seconds for the silence

The -30 is the level in dB.

The 1+2 is saying cycle the first and second tone. So for the example below you can just have 1 instead of 1+2.

The default is 480@-30,620@-30;4(.25/.25/1+2) however this didn't work for me. I live in Switzerland where the disconnect tone is at 425Hz for 200ms followed by a silence at 200ms.
So my settings is at 425@-30,425-30;4(.20/.20/1+2). Due to the simplicity of the Swiss disconnect tone I may not have needed to display the same Hz twice nor have a 4 repeat cycle pattern but it works this way and I haven't tested further.

== BUG REPORTS ==
Please note any bugs, improvements, fixes or Suggestions below so I can make sure it gets included in the final plug-and-play setup!

- Upon power outtage, or network failure/core crash, the FXO and FXS lines are bridged so you can still use the phones. Dialing out works fine (just unplug the power and/or network cable from your spa3000 to see!). However, incoming calls in this failsafe mode only ring once. (not yet fixed)

- Incoming calls don't directly ring the house phone extension. All orbiters alert of the call, and the call can be directed to go to the house line, but I would have expected that the houseline would ring and could be picked up on a call with no orbiter interaction at all. (not quite sure if this a bug or not, of if this is how LMCE should handle it. Can anyone with VoIP confirm the behavior?) (UPDATE: Found the problem. This will not work until do the web admin portion of this - it will essentiall fix its self when I implement that code)

Linksys SPA3000

2012-01-16T15:46:27Z

Tred: /* Setup */

[[Category:Hardware]]
[[Category:Telecom]]
[[Category:VoIP]]
[[Category:Tutorials]]
[[Category:ATA]]
{{merge|Linksys SPA3102}}
== General Information ==

The Sipura (now bought out by Linksys) spa 3000 is a pstn (analog phone line) to Asterisk gateway device that provides a SIP interface for 1 FXO port (analog phone line) and 1 FXS port (analog phone). This allows you to use the Telecom section of LMCE, and use Asterisk to send/recieve/route phone calls.

Most office environments have a phone in each room with their own extension number. Most homes, however, are set up with 1 common line (extension) that all phones are connected to - when a call comes in, all phones ring together. LMCE will allow you to have either setup (if you have enough FXS ports/SIP Phones to support the number of extensions you want). Since the spa 3000 has only 1 FXS port, I am going to show you how to set it up so that all phones on the main house phone line will be treated as one extension and ring together. The picture below illustrates the installation method that I'm going to outline here.
[[Image:spa3000_setup.png]]

[[Image:spa3000_setup_DSL.png]]

Another interesting feature of the spa3000, is that if the LMCE network goes down, or if there is a loss of power, the spa3000 will bridge the FXO and FXS ports, bypassing LMCE and Asterisk altogether, so that you will still have use of your phones in such emergencies.

Also, please realize that these instructions are for manually setting up the spa3000. In the near future, the process will be automated, and if this feature is available at the time you are reading this, then you should go the automatic route.

Please set aside some time to do this manual setup - as you will see, there are a lot of steps involved and it may take you quite some time!

:-:These Instructions are obsolete. Review the Linksys SPA3102 for current ones:-:

Note: these instructions are obsolete. Go to the Linksys SPA3102 page for newest.
== Setup ==
We are going to accomplish this setup in 3 steps:
1- Configuring the Phone Line in the web admin (FreePBX)
2- Adding the device for the internal phone line (FXS port)
3) Configuring the spa3000 (as we will need information from the previous 2 steps to complete)

'''Step 1 - Configuring the Phone Line'''

First, log into the web admin at 192.168.80.1, then navigate to Advanced->Configuration->Phones Setup. This will bring you to the FreePBX Admin page
On the left hand side, click on "Trunks", then click "Add SIP Trunk" on the page the follows. You will now have to fill out the folowing information:

Outbound Caller ID: House Line

Never Override Caller ID: leave unchecked

Maximum Channels: 1 (this must be set to 1 as a pstn line can only handle 1 call at a time)

Disable Trunk: leave unchecked

Monitor Trunk Failures: leave the Enable checkbox unchecked

Dial Rules: Leave blank

Dial rules wizards: leave this alone

Outbound Dial Prefix: leave blank

Trunk Name: House Line

Peer Details:
allow=ulaw
canreinvite=no
context=from-trunk
disallow=all
dtmfmode=rfc2833
host=dynamic
incominglimit=1
nat=never
port=5061
qualify=yes
secret=lmce
type=friend
username=House Line

USER Context: leave blank

USER Details: leave blank

Register String: Leave blank

go ahead and save the changes now!

Next, we are going to setup the Outbound Routes. In FreePBX, on the left, click on "Outbound Routes", and add the following for the new route:

Route Name: House Line Out

Route Password: leave blank

Emergency Dialing: leave unchecked

Intra Company Route: leave unchecked

Music On Hold?: leave at default

Dial Paterns:
112
411
911
9|.XXX

Dial Patterns Wizards: leave this alone

Trunk Sequence: select the SIP/House Line option from the drop down at the top position (this is the trunk we created earlier). If you already have VOIP setup, and you are adding the pstn as a second line, you may want to change the sequence order to suit your needs. All this sequence does, is if a call fails while trying the first sequence, then the call will be tried with the next one, and so on.

Submit your changes!

Next, we will setup the inbound routes. In FreePBX, click on "Inbound Routes", and add the following for the new route:

Description: Leave blank

DID Number : your line phone number, I.e. 800-555-1212

Caller ID Number: leave blank

Zaptel Channel: Leave blank

Fax Handling Section - leave these as-is

Privacy Manager : No

Options Settings: Leave as-is

lastly, in the Set Destination settings, select the last radio button, Custom App. Type the following in the text box for Custom App: custom-linuxmce,102,1

submit your changes.

One last thing to do while we are in FreePBX. You should see towards the top of the page an orange bar that should say "Apply Configuration Changes". Click this, then when prompted, select to continue with reload.

Next, go to Wizard->Devices->Phone Lines (on the left pane in the LMCE web admin, not in FreePBX). We are going to add a dummy line (NOTE: this is a temporary hack for now! I won't go into too many details other than saying that it will allow you to use the "Settings" link next to the listing to do some call routing on your pstn line!)
Use the dropdown to select broadvoice. Once you do this, you will see a form to fill in some data. Just put whatever you want in the fields, they won't be used with this hack! After you are done, you will see it listed as a phone line - use the "settings" link next to it to do call routing depending on security mode!
(Note: After creating this "dummy" phone line, go back to the FreePBX admin, and look at the Incoming Route for the broadvoice line. Look at the Custom App option at the bottom of the page. It should contain that same custom-linuxmce,102,1. If it does not, go back to the Incoming Route for the House Line, and change its custom app line to be the same as this one! From my tests, it should be the same (though the 3 digit number can change, so check this to be sure).

'''Step 2 - Adding the device template for the FXS port (your internal phone/phones)'''
In the web admin, on the left pane under devices, select Phones. On the resulting page, you should see all of the Orbiter Embedded Phones that your system already has. At the bottom of this page, click the "Add Device" button, then select "Generic Phone" from the Device Template picker. Click the "Pick device template" button.
Now notice that you have a new phone device in you list. Change the name to something like "House Line", set the PK_FloorplanObjectType to Pluto Telephone, set the Phone Type to SIP. Notice that the PhoneNumber is already filled out. LEAVE THIS ALONE AND REMEMBER IT! This will be the extension of your internal analog phones (or whatever single phone you have plugged into the FXS Port!) Also, you can assign this phone a room (it really doesn't matter which one you choose) and make sure the "Controlled By" column says Asterisk. If it does not say "Asterisk", click on it and choose Asterisk (Asterisk) - from the dropdown. When finished, Hit the Update button at the bottom of the screen to save these changes. You will have to reload your router, and do a quick regen on all orbiters manually (from the Devices->Orbiters section)

'''Step 3 - Configuring the spa3000 (almost done!)'''
This part assumes a factory default starting point. To do this, hook up an analog phone to the FXS (phone) port, and dial ****
When prompted, enter 73738# from the telephone keypad. This will reset the unit to factory defaults.

Now go ahead and connect the spa3000's LAN port to your internal network switch. We have to assign it a static IP address, so again dial **** from the connected handset. When prompted, enter 110# and the IP address will be spoken back to you. Write it down. Open up a web browser on the network, and enter the IP address to go to the spa3000 admin page. In the upper right hand corner, click on the links for both Admin Login and Advanced.

Go to the System tab. Under "Internet Connection Type" do the following:

DHCP: no
Static IP: enter a static IP address. I used 192.168.80.253
Netmask: 255.255.255.0
Gateway: 192.168.80.1

Update for SPA-3102 
Router -> WAN Setup 
set WAN Web Server to YES 
disconnect the LAN port from the network and connect the WAN port to the network. 

Next, go to the SIP tab. Change the following:

RTP Packet Size: 0.020

Next, go to the Line 1 tab. Change the following:

Line Enable: Yes

SIP Port: 5060

Proxy: 192.168.80.1

Register: Yes

Register Expires: 300

Make Call Without Reg: yes

Ans Call Without Reg: yes

Display Name: <extension number LMCE assigned your Generic Phone device> (mine was 206 for example)

UserID: <again, extension number LMCE assigned your Generic Phone device>

Password: <once again, extension number LMCE assigned your Generic Phone device>

Use AuthID: no

Preferred Codec: make sure its set to G711u

Lastly, go to the PSTN Line tab. Ensure the following settings:

Line Enable: Yes

SIP Port: 5061

Proxy: 192.168.80.1

Register: yes

Register Expires: 300

Make Calls Without Reg: Yes

Ans Calls Without Reg: Yes

Display Name: PSTN Call (this will be displayed in place of unknown caller ID's. You may want to name this "Unknown Call" or similar)

user ID: House Line (this must match the trunk name you did in the FreePBX setup!)

Password: lmce

Use Auth ID: no

Preferred Codec: make sure it is set to G711u

DTMF Process INFO: yes

DTMF Process AVT: yes

DTMF TxMethod: Auto

Dial Plan 2: (S0<:phone_number_here>) i.e. (S0<:800-555-1212>) (this is very important - it must match the DID number entered in FreePBX)

VoiP-To-PSTN Gateway Enable: yes

Voip Caller Auth Method: None

Voip PIN Max Retry: 3

One Stage Dialing: yes

Line1 VoIP Caller DP: None

ViOP Valler Default DP: None

Line1 Fallback DP: None (this may need changed to make sure incoming calls come in when the power goes out - will experiment)

PSTN-To-VoIP Gateway Enable: yes

PSTN Caller Auth Method: none

PSTN Right Thru Line 1: no

PSTN Pin Max Retry: 3

PSTN CID for VioP CID: yes

PSTN Caller Default DP: 2 (this is the dialplan 2 that we set up above!)

Off Hook While Calling VoIP: no

Line 1 Signal Hook Flash to PSTN: Disabled

VoIP Answer Delay: 0

PSTN Answer Delay: 3 (this allows enough time for the caller ID information to be passed along)

SPA to PSTN Gain and PSTN to SPA Gain - these can be adjusted to amplify the incoming/outgoing volume to a comfortable level. I like mine set at 6

That should be it! Enjoy LMCE Telecom on your old analog phone line!

== Disconnect Tone ==
If your line doesn't hang up after a call then you may need to change the disconnect tone. Different countries have different disconnect tones. Go to the SPA3000's configuration page, click on Admin Login and then Advanced. Now head to the PSTN Line tab. Scroll down to the PSTN Disconnect Detection. There is a setting called "Disconnect Tone" and that's what needs to change.

The format entered needs to be of the following: X@-30,X@-30;Y(A/B/1+2)

where:

X = The frequency in Hz.

Y = The cycles the pattern repeats

A = Time in seconds for the frequency

B = Time in seconds for the silence

The -30 is the level in dB.

The 1+2 is saying cycle the first and second tone. So for the example below you can just have 1 instead of 1+2.

The default is 480@-30,620@-30;4(.25/.25/1+2) however this didn't work for me. I live in Switzerland where the disconnect tone is at 425Hz for 200ms followed by a silence at 200ms.
So my settings is at 425@-30,425-30;4(.20/.20/1+2). Due to the simplicity of the Swiss disconnect tone I may not have needed to display the same Hz twice nor have a 4 repeat cycle pattern but it works this way and I haven't tested further.

== BUG REPORTS ==
Please note any bugs, improvements, fixes or Suggestions below so I can make sure it gets included in the final plug-and-play setup!

- Upon power outtage, or network failure/core crash, the FXO and FXS lines are bridged so you can still use the phones. Dialing out works fine (just unplug the power and/or network cable from your spa3000 to see!). However, incoming calls in this failsafe mode only ring once. (not yet fixed)

- Incoming calls don't directly ring the house phone extension. All orbiters alert of the call, and the call can be directed to go to the house line, but I would have expected that the houseline would ring and could be picked up on a call with no orbiter interaction at all. (not quite sure if this a bug or not, of if this is how LMCE should handle it. Can anyone with VoIP confirm the behavior?) (UPDATE: Found the problem. This will not work until do the web admin portion of this - it will essentiall fix its self when I implement that code)

Linksys SPA3000

2012-01-16T15:45:11Z

Tred: /* Setup */

[[Category:Hardware]]
[[Category:Telecom]]
[[Category:VoIP]]
[[Category:Tutorials]]
[[Category:ATA]]
{{merge|Linksys SPA3102}}
== General Information ==

The Sipura (now bought out by Linksys) spa 3000 is a pstn (analog phone line) to Asterisk gateway device that provides a SIP interface for 1 FXO port (analog phone line) and 1 FXS port (analog phone). This allows you to use the Telecom section of LMCE, and use Asterisk to send/recieve/route phone calls.

Most office environments have a phone in each room with their own extension number. Most homes, however, are set up with 1 common line (extension) that all phones are connected to - when a call comes in, all phones ring together. LMCE will allow you to have either setup (if you have enough FXS ports/SIP Phones to support the number of extensions you want). Since the spa 3000 has only 1 FXS port, I am going to show you how to set it up so that all phones on the main house phone line will be treated as one extension and ring together. The picture below illustrates the installation method that I'm going to outline here.
[[Image:spa3000_setup.png]]

[[Image:spa3000_setup_DSL.png]]

Another interesting feature of the spa3000, is that if the LMCE network goes down, or if there is a loss of power, the spa3000 will bridge the FXO and FXS ports, bypassing LMCE and Asterisk altogether, so that you will still have use of your phones in such emergencies.

Also, please realize that these instructions are for manually setting up the spa3000. In the near future, the process will be automated, and if this feature is available at the time you are reading this, then you should go the automatic route.

Please set aside some time to do this manual setup - as you will see, there are a lot of steps involved and it may take you quite some time!

:-:These Instructions are obsolete. Review the Linksys SPA3102 for current ones:-:

== Setup ==
We are going to accomplish this setup in 3 steps:
1- Configuring the Phone Line in the web admin (FreePBX)
2- Adding the device for the internal phone line (FXS port)
3) Configuring the spa3000 (as we will need information from the previous 2 steps to complete)

'''Step 1 - Configuring the Phone Line'''

First, log into the web admin at 192.168.80.1, then navigate to Advanced->Configuration->Phones Setup. This will bring you to the FreePBX Admin page
On the left hand side, click on "Trunks", then click "Add SIP Trunk" on the page the follows. You will now have to fill out the folowing information:

Outbound Caller ID: House Line

Never Override Caller ID: leave unchecked

Maximum Channels: 1 (this must be set to 1 as a pstn line can only handle 1 call at a time)

Disable Trunk: leave unchecked

Monitor Trunk Failures: leave the Enable checkbox unchecked

Dial Rules: Leave blank

Dial rules wizards: leave this alone

Outbound Dial Prefix: leave blank

Trunk Name: House Line

Peer Details:
allow=ulaw
canreinvite=no
context=from-trunk
disallow=all
dtmfmode=rfc2833
host=dynamic
incominglimit=1
nat=never
port=5061
qualify=yes
secret=lmce
type=friend
username=House Line

USER Context: leave blank

USER Details: leave blank

Register String: Leave blank

go ahead and save the changes now!

Next, we are going to setup the Outbound Routes. In FreePBX, on the left, click on "Outbound Routes", and add the following for the new route:

Route Name: House Line Out

Route Password: leave blank

Emergency Dialing: leave unchecked

Intra Company Route: leave unchecked

Music On Hold?: leave at default

Dial Paterns:
112
411
911
9|.XXX

Dial Patterns Wizards: leave this alone

Trunk Sequence: select the SIP/House Line option from the drop down at the top position (this is the trunk we created earlier). If you already have VOIP setup, and you are adding the pstn as a second line, you may want to change the sequence order to suit your needs. All this sequence does, is if a call fails while trying the first sequence, then the call will be tried with the next one, and so on.

Submit your changes!

Next, we will setup the inbound routes. In FreePBX, click on "Inbound Routes", and add the following for the new route:

Description: Leave blank

DID Number : your line phone number, I.e. 800-555-1212

Caller ID Number: leave blank

Zaptel Channel: Leave blank

Fax Handling Section - leave these as-is

Privacy Manager : No

Options Settings: Leave as-is

lastly, in the Set Destination settings, select the last radio button, Custom App. Type the following in the text box for Custom App: custom-linuxmce,102,1

submit your changes.

One last thing to do while we are in FreePBX. You should see towards the top of the page an orange bar that should say "Apply Configuration Changes". Click this, then when prompted, select to continue with reload.

Next, go to Wizard->Devices->Phone Lines (on the left pane in the LMCE web admin, not in FreePBX). We are going to add a dummy line (NOTE: this is a temporary hack for now! I won't go into too many details other than saying that it will allow you to use the "Settings" link next to the listing to do some call routing on your pstn line!)
Use the dropdown to select broadvoice. Once you do this, you will see a form to fill in some data. Just put whatever you want in the fields, they won't be used with this hack! After you are done, you will see it listed as a phone line - use the "settings" link next to it to do call routing depending on security mode!
(Note: After creating this "dummy" phone line, go back to the FreePBX admin, and look at the Incoming Route for the broadvoice line. Look at the Custom App option at the bottom of the page. It should contain that same custom-linuxmce,102,1. If it does not, go back to the Incoming Route for the House Line, and change its custom app line to be the same as this one! From my tests, it should be the same (though the 3 digit number can change, so check this to be sure).

'''Step 2 - Adding the device template for the FXS port (your internal phone/phones)'''
In the web admin, on the left pane under devices, select Phones. On the resulting page, you should see all of the Orbiter Embedded Phones that your system already has. At the bottom of this page, click the "Add Device" button, then select "Generic Phone" from the Device Template picker. Click the "Pick device template" button.
Now notice that you have a new phone device in you list. Change the name to something like "House Line", set the PK_FloorplanObjectType to Pluto Telephone, set the Phone Type to SIP. Notice that the PhoneNumber is already filled out. LEAVE THIS ALONE AND REMEMBER IT! This will be the extension of your internal analog phones (or whatever single phone you have plugged into the FXS Port!) Also, you can assign this phone a room (it really doesn't matter which one you choose) and make sure the "Controlled By" column says Asterisk. If it does not say "Asterisk", click on it and choose Asterisk (Asterisk) - from the dropdown. When finished, Hit the Update button at the bottom of the screen to save these changes. You will have to reload your router, and do a quick regen on all orbiters manually (from the Devices->Orbiters section)

'''Step 3 - Configuring the spa3000 (almost done!)'''
This part assumes a factory default starting point. To do this, hook up an analog phone to the FXS (phone) port, and dial ****
When prompted, enter 73738# from the telephone keypad. This will reset the unit to factory defaults.

Now go ahead and connect the spa3000's LAN port to your internal network switch. We have to assign it a static IP address, so again dial **** from the connected handset. When prompted, enter 110# and the IP address will be spoken back to you. Write it down. Open up a web browser on the network, and enter the IP address to go to the spa3000 admin page. In the upper right hand corner, click on the links for both Admin Login and Advanced.

Go to the System tab. Under "Internet Connection Type" do the following:

DHCP: no
Static IP: enter a static IP address. I used 192.168.80.253
Netmask: 255.255.255.0
Gateway: 192.168.80.1

Update for SPA-3102 
Router -> WAN Setup 
set WAN Web Server to YES 
disconnect the LAN port from the network and connect the WAN port to the network. 

Next, go to the SIP tab. Change the following:

RTP Packet Size: 0.020

Next, go to the Line 1 tab. Change the following:

Line Enable: Yes

SIP Port: 5060

Proxy: 192.168.80.1

Register: Yes

Register Expires: 300

Make Call Without Reg: yes

Ans Call Without Reg: yes

Display Name: <extension number LMCE assigned your Generic Phone device> (mine was 206 for example)

UserID: <again, extension number LMCE assigned your Generic Phone device>

Password: <once again, extension number LMCE assigned your Generic Phone device>

Use AuthID: no

Preferred Codec: make sure its set to G711u

Lastly, go to the PSTN Line tab. Ensure the following settings:

Line Enable: Yes

SIP Port: 5061

Proxy: 192.168.80.1

Register: yes

Register Expires: 300

Make Calls Without Reg: Yes

Ans Calls Without Reg: Yes

Display Name: PSTN Call (this will be displayed in place of unknown caller ID's. You may want to name this "Unknown Call" or similar)

user ID: House Line (this must match the trunk name you did in the FreePBX setup!)

Password: lmce

Use Auth ID: no

Preferred Codec: make sure it is set to G711u

DTMF Process INFO: yes

DTMF Process AVT: yes

DTMF TxMethod: Auto

Dial Plan 2: (S0<:phone_number_here>) i.e. (S0<:800-555-1212>) (this is very important - it must match the DID number entered in FreePBX)

VoiP-To-PSTN Gateway Enable: yes

Voip Caller Auth Method: None

Voip PIN Max Retry: 3

One Stage Dialing: yes

Line1 VoIP Caller DP: None

ViOP Valler Default DP: None

Line1 Fallback DP: None (this may need changed to make sure incoming calls come in when the power goes out - will experiment)

PSTN-To-VoIP Gateway Enable: yes

PSTN Caller Auth Method: none

PSTN Right Thru Line 1: no

PSTN Pin Max Retry: 3

PSTN CID for VioP CID: yes

PSTN Caller Default DP: 2 (this is the dialplan 2 that we set up above!)

Off Hook While Calling VoIP: no

Line 1 Signal Hook Flash to PSTN: Disabled

VoIP Answer Delay: 0

PSTN Answer Delay: 3 (this allows enough time for the caller ID information to be passed along)

SPA to PSTN Gain and PSTN to SPA Gain - these can be adjusted to amplify the incoming/outgoing volume to a comfortable level. I like mine set at 6

That should be it! Enjoy LMCE Telecom on your old analog phone line!

== Disconnect Tone ==
If your line doesn't hang up after a call then you may need to change the disconnect tone. Different countries have different disconnect tones. Go to the SPA3000's configuration page, click on Admin Login and then Advanced. Now head to the PSTN Line tab. Scroll down to the PSTN Disconnect Detection. There is a setting called "Disconnect Tone" and that's what needs to change.

The format entered needs to be of the following: X@-30,X@-30;Y(A/B/1+2)

where:

X = The frequency in Hz.

Y = The cycles the pattern repeats

A = Time in seconds for the frequency

B = Time in seconds for the silence

The -30 is the level in dB.

The 1+2 is saying cycle the first and second tone. So for the example below you can just have 1 instead of 1+2.

The default is 480@-30,620@-30;4(.25/.25/1+2) however this didn't work for me. I live in Switzerland where the disconnect tone is at 425Hz for 200ms followed by a silence at 200ms.
So my settings is at 425@-30,425-30;4(.20/.20/1+2). Due to the simplicity of the Swiss disconnect tone I may not have needed to display the same Hz twice nor have a 4 repeat cycle pattern but it works this way and I haven't tested further.

== BUG REPORTS ==
Please note any bugs, improvements, fixes or Suggestions below so I can make sure it gets included in the final plug-and-play setup!

- Upon power outtage, or network failure/core crash, the FXO and FXS lines are bridged so you can still use the phones. Dialing out works fine (just unplug the power and/or network cable from your spa3000 to see!). However, incoming calls in this failsafe mode only ring once. (not yet fixed)

- Incoming calls don't directly ring the house phone extension. All orbiters alert of the call, and the call can be directed to go to the house line, but I would have expected that the houseline would ring and could be picked up on a call with no orbiter interaction at all. (not quite sure if this a bug or not, of if this is how LMCE should handle it. Can anyone with VoIP confirm the behavior?) (UPDATE: Found the problem. This will not work until do the web admin portion of this - it will essentiall fix its self when I implement that code)

RoamingOrb

2010-12-12T18:47:49Z

Tred: /* Configuration */

[[Category:Orbiters]]
[[Image:splash.png|left]]
[[Image:logo.jpg|right]]
{{versioninfo|810Status=works|810UpdatedDate=5th October 2010|810UpdatedBy=Foxi352}}

 
== About ==
[[Image:orbiter.png|left]]
RoamingOrb is a touch orbiter port for iOS using the lightweight touch orbiter concept based on the proxy orbiter plugin. It uses the same techniques as WebOrbiter 2.0.
The app is available through the official Apple app store.

 
== Installation ==
Installation is done the same way as any other app.
*Connect to the app store
*Search for the app "RoamingOrb"
*On the description page click on "Free" to install it

== Configuration ==
[[Image:preferences.jpg|left]][[Image:settings.jpg]]
 Firstly, make sure that you have generated a [[Web Orbiter 2.0]] in WebAdmin. Some users have reported more success with the Basic skin and iPhone resolution than any other skin or resolution.
RoamingOrb requires two values to be defined before it can be used. To input these values, use the Settings app on your iOS device and go to the subsection titled "Roaming Orb" There are several different versions of these values.
If you would like to use the Orbiter over 3G, then you must enter the public IP address of your CORE (the one assigned to you by your ISP) and open the orbiter's listening port on any firewalls between it and the core (this includes the core's)
If you would just like to connect through your local network, input the cores local IP (usually 192.168.80.1) and the listening port defined by your Web Orbiter.

== Intent ==
Please note, this is not a full Orbiter with media playback capabilities. It only has the same remote control features as the WebOrbiter 2.0.

RoamingOrb

2010-12-12T18:46:55Z

Tred: /* Configuration */

Apple Iphone

2010-12-12T18:32:33Z

Tred:

[[Category: Hardware]]
{{versioninfo}}
[[Category:Orbiters]]
[[Image:Iphone.gif]]
==About this device==
* 3.5-inch (diagonal) widescreen multi-touch display
** 480-by-320-pixel resolution at 163 ppi
** Support for display of multiple languages and characters simultaneously
* OS X
* Quad-band (850, 900, 1800, 1900 MHz)
* Wi-Fi (802.11b/g)
* EDGE
* Bluetooth 2.0+EDR
* 2.0 megapixels

This is a very popular device that many users use as an Orbiter.

==Lmce related applications running on iPhone==
Roaming Orb- Touch Orbiter app for iPhone
Dianemo- Touch orbiter for Dianemo distributions of LinuxMCE

Apple Iphone

2010-12-12T18:30:49Z

Tred: /* About this device */

Apple Iphone

2010-12-12T18:30:29Z

Tred: /* Lmce related applications running on iPhone */

[[Category: Hardware]]
{{versioninfo}}
[[Category:Orbiters]]
[[Image:Iphone.gif]]
==About this device==
* 3.5-inch (diagonal) widescreen multi-touch display
** 480-by-320-pixel resolution at 163 ppi
** Support for display of multiple languages and characters simultaneously
* OS X
* Quad-band (850, 900, 1800, 1900 MHz)
* Wi-Fi (802.11b/g)
* EDGE
* Bluetooth 2.0+EDR
* 2.0 megapixels

This is a very popular device that many users would prefer to use as an Orbiter.

==Lmce related applications running on iPhone==
Roaming Orb- Touch Orbiter App for iPhone

Insteon - Setting Up PLM Template

2010-06-21T23:00:28Z

Tred: /* Supported Devices */

{{Versioninfo}}
[[Category:Tutorials]]
[[Category:Power Line Control]]
[[Category:GSD| ]]
{| align="right"
| __TOC__
|}

:''' UPDATED: 11-MAR-08 Many bugs fixed, X10 fixes...
:''' UPDATED: 13-FEB-08 X10 Support fixed!
'''
:--ddamron

Here are the steps to create an Insteon PLM Template. This driver is written entirely in Ruby. Please consider this driver complete, and is in Release Candidate 1. I'm sure there are still a few bugs (although we've ironed most of them out now). The more people testing this code, the more bugs we can find, and get this to a release state. Once it's in release state, it will be added to sqlcvs and offered via a 'quick' update.

==Supported Devices==
This driver supports ONLY the Powerlinc Modem model (PLM)#2412S and #2413S. The code will NOT work for a model #2414 also known as a Powerlinc CONTROLLER (PLC).

NOTE: I am expecting a PLC to arrive in the mail in the next few days. Support for that device will be sooner than later.

Simplehomenet's EZBridge/EZServ products will be added once this code is stable. These products are a 'wrapper' for the PLM, and hence, use the same code. Integration will be a matter of adding the 'wrapper' code. (which is done)

==Setting up the Template==
* Goto [[LinuxMCE Admin Website]]-->Wizard-->Devices-->Interfaces
* Add device-->Add Manufacturer
* type 'Smarthome' (and click Add)
*Click Close

===Adding a New Template===
:*Press F5 to refresh the page.
:*In the Manufacturer field, select 'Smarthome'.
:*Under Device Category, select 'Lighting Interface <-- Interfaces'.
:*Add device Template
:* select the LAST option:

<nowiki>"It's a simple, generic serial device controlled by RS232, Ethernet or USB, which you'll control with Pluto's GSD and embedded Ruby snippets (very easy). Examples: Lighting control, alarm panels, surveillance cameras, thermostats, weather stations, a/v equipment with RS232 ports, etc"</nowiki>

:* Type in 'Insteon PLM' in the name or model#.
:* Click Add
There will not be a confirmation screen, instead the page will be blank
:* Close the Add device Template window.

===modifying the new Template===
* In Webadmin, Select Advanced, Configuration, Device Templates (from the top menu)
* In the Device Template field, select 'Insteon PLM'
* click 'Pick Device Template' (loads Edit Device Template Popup)
:* NOTE the Template Number! write this down. (it will be needed for Plug N Play)
:* Make sure 'Implements DCE' is checked (should be)
:* In the Comm Method, select RS232
===Adding a Code Group===
:* Click 'Edit Ruby Codes' ( loads Edit Ruby Codes Popup)
::* In the field to the LEFT of 'Create new group', type in 'Insteon PLM'
::* click 'Create new group'

----
b4 is broken have to run sql manually to add new group
close 'Edit Ruby codes' page. Look at 'Edit Device Template' popup page and
note '''Device_Category''' and '''Manufacturer''' you will need to set them in sql code below

''INSERT INTO InfraredGroup (FK_DeviceCategory,FK_Manufacturer,Description,FK_CommMethod,psc_id,psc_batch,psc_user,psc_frozen,psc_mod,psc_restrict) VALUES ('''Device_Category''','''Manufacturer''',"Insteon PLM",2,NULL,NULL,NULL,NULL,NULL,NULL);''

run this sql and reload 'Edit Ruby codes' page and you should see 'Insteon PLM' in 'Uses Group/Codeset'
----

::* If the 'Uses Group/Codeset' does not change to 'Insteon PLM', repeat the previous 2 steps again.
::* Once the 'Uses Group/Codeset' says 'Insteon PLM' Check 'Discoverable Interfaces'
::* Check 'Ruby Internal Commands'
::* click Update

===Adding Ruby Code===
::* In the #756 Report Child Devices field (the big one) copy and paste this code:
:::[[Insteon PLM Ruby Code 756]]
::* In the #776 Reset field, copy and paste this code:
:::[[Insteon PLM Ruby Code 776]]
::* In the #760 Send Command to Child field, copy and paste this code:
:::[[Insteon PLM Ruby Code 760]]
::* In the #788 StatusReport field, copy and paste this code:
:::[[Insteon PLM Ruby Code 788]]
::* In the #373 Private Method Listing field, copy and paste this code:
:::[[Insteon PLM Ruby Code 373]]
::* In the #351 Process IDLE field, copy and paste this code:
:::[[Insteon PLM Ruby Code 351]]
::* In the #350 Process Incoming Data field, copy and paste this code:
:::[[Insteon PLM Ruby Code 350]]
::* In the #355 Process Initialize field, copy and paste this code:
:::[[Insteon PLM Ruby Code 355]]
::* In the #384 Process Receive Command for Child field, copy and paste this code:
:::[[Insteon PLM Ruby Code 384]]
::* In the #356 Process Release, leave it blank. (no code for now)
::* Click Update
::* Click Close
'''UPDATE''' I had to remove the ruby code from 141 & 161. The PLM wouldn't start due to errors from these two. 8/9/08 williammanda

===Adding Device Data===
:* Press F5 for refresh the 'Edit Device Template' popup
:* In the 'Device data' section, in 'Add a New Parameter' select 'COM Port on PC' and click Add.
:* in the Comments Field, type
COM Port
:* in the Default Value field, type
/dev/ttyS0
:* check the 'Required' and 'Allowed to Modify' boxes
:* In the 'Device data' section, in 'Add a New Parameter' select 'COM Port BaudRate' and click Add.
:* in the Comments Field, type
Baud Rate
:* in the Default Value field, type
19200
:* check the 'Required' and 'Use Master Device List Defaults' boxes
:* In the 'Device data' section, in 'Add a New Parameter' select 'Only One Per PC' and click Add.
:* in the Comments Field, type
Only One Per PC
:* in the Default Value field, type
1
:* In the 'Device data' section, in 'Add a New Parameter' select 'Auto-assign to parents room' and click Add.
:* in the Comments Field, type
Assign to Parents Room
:* in the Default Value field, type
1

===Making the Driver Plug & Play===
:* Scroll down to the 'Plug & Play' Section
:* Enter These values:
::From =
0
::To =
0
::Vendor Model ID =
2412S
::PNP protocol =
Proprietary
::Serial Number =
Insteon PLM
::Parms is blank
::PNP detection script is
88_InsteonPLM.sh
::Comment is
Insteon PLM

:* Scroll down to the bottom of the popup window and click Save
:* Click Close
:* Modify the SECOND LAST LINE of this script (88_InsteonPLM.sh) and change the LAST number to your Template ID.
:* Add this script file to /usr/pluto/pnp/ directory (88_InsteonPLM.sh). The script will not currently exist so it must be created. Depending on who you have logged in as you may need super user privileges.
:The command '''sudo vi /usr/pluto/pnp/88_InsteonPLM.sh''' will both create the file and allow you to edit. The super user password will be required.
:User Notes:
:'''UPDATE 8/9/08''' Use this format to execute the command '''Kate /usr/pluto/pnp/88_InsteonPLM.sh''' missing "/" after pnp. Williammanda
<pre>
#!/bin/bash

echo "Insteon Powerline Modem Detection Script queue $2"
/usr/pluto/bin/TestSerialPort -p $3 -P N81 -b 19200 -t "\02\60" -i 1 -s "\03\05"
if [[ "$?" -ne 0 ]]; then
echo "It's not an Insteon Powerline Modem"
/usr/pluto/bin/MessageSend dcerouter -r 0 $1 1 806 224 $2 13 "$4" 44 0
else
echo "It is an Insteon Powerline Modem"
/usr/pluto/bin/MessageSend dcerouter -r 0 $1 1 806 224 $2 13 "$4" 44 1901
fi
</pre>
:* chmod 755 /usr/pluto/pnp/88_InsteonPLM.sh (make it executable). The sudo command prefix will work here as well if you are not logged in as root.

=== Setting up the Generic Serial Device ===
:* on the top, select Wizard, Devices, Generic Serial Devices. You should now see the InsteonPLM device.
:* select the room it's in.
:* in the Controlled By box, select CORE
:* select Update (at the bottom)
:* Now, in the COM PORT ON PC, you can select the com port it's on. '''UPDATE''' I didn't see any "COM PORT ON PC" field. You wil need to reload the router first, then you see the "COM PORT ON PC" field. 8/9/08 Williammanda
'''TODO''' Other than looking on the motherboard for the com port label and guess and check methods, what else can be done to identify the correct com port?
:* select Update (at the bottom)

You are Done!

==Next Steps==
go to [[Running the Insteon PLM Driver for the First Time]]

Bittorrent Setup Guide

2010-04-24T20:34:36Z

Tred: /* Install uTorrent */

[[Category:Tutorials]]

This is a guide to help you set up the uTorrent bittorrent client running under wine on a LinuxMCE box. This setup has the following features:
* torrent client that is always running right on the core
* add new and monitor torrents from a web browser (i.e. works in a headless configuration)
* completed downloads automatically show up in media library
* partially downloaded torrents survive LinuxMCE reinstallation

This setup was tested on a hybrid/core running LinuxMCE 0710RC2. The same should work on a dedicated core.

===Storage Setup===
I recommend downloading torrents to a local hard drive that is not the system drive. If you are using a NAS, I find that manually moving completed downloads from the local drive to the appropriate NAS location works well.

For this guide I will refer to the secondary local hard drive used for downloading as the dl-drive. Such a drive installed in a core is automatically mounted by LinuxMCE/Kubuntu at /mnt/device/#. # is a number that changes from installation to installation.

===Install Wine===
[http://www.winehq.org/ Wine] provides the windows environment on linux. Wine allows you to run the windows-native uTorrent.exe on linux. Wine is not an emulator, rather it is a compatibility layer so there is no performance penalty for running uTorrent on linux.

1. First install the wine package:
sudo apt-get install wine

2. Run winecfg to set up the windows environment -- do not run this as super user otherwise you will run into permissions problems later.
winecfg

===Install uTorrent===
[http://www.utorrent.com/ uTorrent] is a widely-used and efficient bittorrent client. The following steps install it on the dl-drive itself rather than the system drive. The reason for this is so that the uTorrent configuration will survive a re-installation of LinuxMCE thereby allowing any in-progress downloads to continue after a re-install.

1. Create uTorrent folders on the dl-drive.
mkdir /mnt/device/#/uTorrent
mkdir /mnt/device/#/incomplete

2. Create a symlink to the uTorrent folder on the dl-drive from the Program Files folder.
cd ~/.wine/drive_c
cd "Program Files"
ln -s /mnt/device/#/uTorrent

3. Create a symlink to dl-drive. By mapping the uTorrent download directories using this symlink, in-progress downloads can continue after a LinuxMCE re-installation.
ln -s /mnt/device/# ~/dl-drive

4. Download uTorrent
cd /mnt/device/#/uTorrent
wget http://download.utorrent.com/2.0.1/utorrent.exe
wget http://www.utorrent.com/webui/webui.1220503364.zip
mv webui.1220503364.zip webui.zip

===Create the utorrent-startup Script===
In order to start uTorrent from a shortcut in KDE, a startup script is required.

1. Create the script file:
<editor-of-choice> /mnt/device/#/uTorrent/utorrent-startup

2. Paste this into the file:

#!/bin/bash
#
#

cd ~/.wine/drive_c
cd "Program Files"
cd uTorrent
wine utorrent.exe

===Create the uTorrent Shortcuts===
A desktop shortcut allows you to manually launch uTorrent. A copy of it is also used to automatically start uTorrent on startup.

1. Right-click the desktop and select Create New Link to Application. A dialog box is shown.

2. On the general tab type uTorrent beside the icon.

3. On the application tab type the following in the command text box:
/mnt/device/#/uTorrent/utorrent-startup

4. Click OK. You should now be able to run uTorrent using the newly created link.

5. Put a copy of the link into the Autostart folder at ~/.kde/Autostart.

===Make KDE Desktop Launch Automatically at Boot===
uTorrent launches automatically when KDE desktop starts. By default LinuxMCE does not start KDE desktop on boot. To have uTorrent start automatically on boot, you need to make KDE desktop start automatically.

1. Edit the media center startup script:
<editor-of-choice> /usr/pluto/bin/startup-script.sh

2. Add the following line to the very end of the file:
StartService "KDE Desktop" "/usr/pluto/bin/Start_KDE.sh"

Reboot LinuxMCE. You should see KDE desktop start just before the on-screen orbiter is shown. You should also see the green 'u' icon in the system tray.

===Enable WebUI===
The WebUI is handy for adding and monitoring torrents. It allows you to access uTorrent from any browser with access to the core. It needs to be enabled in order to access it.

1. In uTorrent check the '''Enable WebUI''' box and setup a username and password at Options=>Preferences=>WebUI.

2. Browse to the URL below. <incoming-port> is the port number at Options=>Preferences=>Connection.
http://dcerouter:<incoming-port>/gui/

===Configure uTorrent Directories===
Set up the the directories to keep the incomplete torrents out of the LinuxMCE media folders. Once a torrent completes however, uTorrent should move it to the video folder so that it is picked up automatically by LinuxMCE.

1. Go to Options=>Preferences=>Directories
2. Check '''Put new downloads in:''' and set the path to
z:\home\linuxmce\dl-drive\incomplete
3. Check '''Move completed downloads to:''' and set the path to
z:\home\linuxmce\dl-drive\public\data\videos
4. Check '''Store .torrents in:''' and set the path to
z:\home\linuxmce\dl-drive\incomplete
5. Check '''Move .torrents for finished jobs to''' and set the path to:
z:\home\linuxmce\dl-drive\tfarch

Hardware compatibility list

2010-02-07T22:00:43Z

Tred: /* PSTN */

[[Category: Compatibility]]
= Hardware Compatibility List =

== Climate ==
=== Thermostats ===

== Core/Media Directors ==
=== Motherboards ===
=== Optical Drives ===
==== Blu-Ray Drives ====
==== CD/DVD Drives ====
=== Sound Cards ===
=== Storage Devices ===
=== Video Cards ===
NVidia GForce 7600

== Lighting ==
=== Interfaces ===
[[CM11A]]
=== Switches ===
[[X10]]

[[Z-Wave]]

== Media ==
=== Audio ===
=== TV's ===
=== Video ===
[[Hauppauge WinTV-PVR-150 MCE | Hauppauge WinTV PVR-150 MCE]]

== Orbiters ==

== Telecommunications ==
=== Phones ===
[[Cisco 7970]]
=== PSTN ===
[[Sipura/Linksys spa3000 pstn interface | Linksys 3000 (formerly Sipura 3000)]]

X100P (Wiki Coming Soon!)

=== VOIP ===

== Security ==
=== Cameras ===
[[D-Link DCS-920]]

[[D-Link DCS-5300G]]

[[Linksys WVC54GCA]]

=== Interfaces ===
[[GlobalCache GC-100 | GC-100]]
=== Panels ===
[[Apex Destiny 6100]]

[[Caddx NX-8E]]
=== Sensors ===