Messages

46

Users / Re: Anyone know how to move video files in 7.10 Raid 1 array?

« on: June 22, 2010, 03:43:47 pm »

OK, thanks, so do you know what is in there from linuxmce? Is it possible to delete some of the files there or is my only choice to try and enlarge this partition? A little guidance would be greatly appreciated.

47

Users / Re: Anyone know how to move video files in 7.10 Raid 1 array?

« on: June 21, 2010, 09:16:11 pm »

Sure!

Here is the output:

Filesystem            Size  Used Avail Use% Mounted on
rootfs                 65G   59G  2.3G  97% /
udev                  506M  120K  506M   1% /dev
/dev/disk/by-uuid/81a8f990-8d10-437a-8c11-8df9482ffd8c
                       65G   59G  2.3G  97% /
/dev/disk/by-uuid/81a8f990-8d10-437a-8c11-8df9482ffd8c
                       65G   59G  2.3G  97% /dev/.static/dev
tmpfs                 506M  920K  505M   1% /var/run
tmpfs                 506M     0  506M   0% /var/lock
tmpfs                 506M   38M  469M   8% /lib/modules/2.6.22-14-generic/volatile
tmpfs                 506M     0  506M   0% /dev/shm
tmpfs                 506M  920K  505M   1% /var/run
tmpfs                 506M     0  506M   0% /var/lock
/dev/sda6             9.2G  4.5G  4.3G  52% /mnt/recovery
//192.168.80.8/Storage217$
                       13G  2.1G   10G  18% /mnt/device/217
/dev/md1              151G  108G   35G  76% /mnt/device/26

Many thanks for looking at this!

48

Users / Anyone know how to move video files in 7.10 Raid 1 array?

« on: June 21, 2010, 04:34:50 pm »

Hi,
I'm a bit befuddled by this as I keep getting the "your system is running low on disk space" message despite my 153 gb 2-disk raid 1 array on the core.

I've deleted all my log files, even got rid of all of 2009's flicker files.

I have very few videos on this array, but from within linuxmce all it will allow me to do is delete them--"move" just looks back at me highlighted in red. All the rest of my video, music etc. is pulled from a NAS. So I can't figure out how linuxmce could be occupying the whole space--but I will move all the "data" files if I can find a way to get to them:

From ssh I can go into the directory structure, but since the raid drives are aliased I can not get into the folder. I don't know enough to understand how to into this aliased folder from the terminal.

Can someone help or am I trying to do the impossible and I should just go delete all media on the drive and then rip it to the NAS?

Any help greatly appreciated.

49

Installation issues / Re: Orbiter phone not setting up sip account correctly in 8.10?

« on: March 28, 2010, 04:13:28 pm »

I'm certainly happy to help test.
Here is what I did to fix it:

Figured what the heck and did a complete reinstall to see if I could duplicate it. BTW, this is a big mistake because once 8.10 is installed in my case I had to completely wipe the disk to the point that LMCE thought I was from "Eastern Kentucky." Also kind of tricky if you are trying to set it up with a fixed ip at ETH0--it needs a dhcp id there so it can download stuff.

Finally got LMCE to install. Did the Wizard only to the point of having a user and a room. Went into network and straightened out ETH0. Then went into Wizard and set up Broadvoice. Bingo, system worked as it was supposed to although the settings for SIP did not show the secret and were no longer SIP5061 but just SIP.

My first thought is that this version is incredibly hardware dependent in getting it to work. The first time around I did not have a sound card and tried to use snd-dummy after asterisk was set up. This may have caused hiccups as in this install I had the sound card in from the start.

During install, I also froze the whole system twice when it went to look at the PCI bus to determine what is there until I removed a firewire card that had been successfully installed in 7.10. Perhaps all the devices have not yet been included...

Finally, perhaps in my trying to switch the network around before the system had gone through all its background processes, mythtv did not fully install to the point that I was getting:

ERROR 1049 (42000): Unknown database 'mythconverg'

after searching around a bit, I decided what the heck again and did a

apt-get dist-upgrade

which identified that mythtv needed to be installed again

after that did a
/etc/init.d/mythtv-backend stop


and then

 mysqlcheck -c -uroot -p mythconverg

and everything looked fine.

Hope this helps someone else.

50

Installation issues / Orbiter phone not setting up sip account correctly in 8.10?

« on: March 26, 2010, 11:36:01 pm »

I hate to pile on with possible problems because I just loaded 8.10 and have seen glimpses of what it could be--and I really like it.

Having been impressed with the you tube video about using the phone from the orbiter I ponied up for one. It indeed installed itself. Also impressed with how easily my broadvoice account went in. However, when I dialed in from outside, I kept getting a series of recorded messages from asterisk telling me to leave a message and nothing happening on the orbiter screen.

Went to look at the orbiter in the Wizard \ Devices \phones and they looked OK. Then went to Advanced \Configuration \Phone Setup and looked at their extensions. Confusingly, there was no sip set up "This phone is using "" Technology" with no fields to fill in--even though it showed it in the Wizard file. Tested with asterisk, looked at the devices database under asterisk table in mysql, tried to call from a generic sip phone on the system to the orbiter and concluded they had never been set up properly. Strangely, you could call the sip phone from the orbiter or even call out from the orbitor through broadvoice but you could not call it from anywhere.

Went back to the Wizard, deleted  the orbiter and added another one associated with that MD. Now you can call it from SIP phone and if you designate it as the extension to call on the inbound route from broadvoice it will ring.

However, if you go back to using the custom app on the inbound route of linuxmce-custom,102,1 it goes back to dropping you into voicemail when you call from the outside.

Went and looked at that /etc/asterisk/extensions_pluto_dial.conf where that app takes you and indeed it is trying to dump you into "Local/@trusted,15" instead of "voice-menu-pluto-custom,s,1"

Is anyone else experiencing this and is there a work around? Or is that just where things stand for now?

Many thanks.

51

Users / Re: Ruby code in wiki tutorial for LG 42LB5D #355 (Solved)

« on: March 23, 2010, 08:46:21 pm »

I took some time to delve into how Ruby works and found where the (my) error was in the script. You need to look up the device id number of your LG TV from the Wizard \ Devices \ AV Equipment in the upper left hand corner and replace the number "158" in all the scripts with this number (for example, in my case 310). This allows the script to process successfully.

52

Users / Re: Cisco 7970 and Error Message

« on: March 20, 2010, 06:21:11 pm »

I'm not sure what you mean by group ranges for the 7970(1). The template has to have a range based on your cisco phone's mac address which you must convert as per the instructions in the wiki. If by ranges you mean ip addresses, my understanding is that the initialization of the cisco phone is supposed to be pnp and that the dhcp server will assign and (hopefully) remember the correct ip for your phone.

In case pnp doesn't completely finish, as was my case,  and the cisco extension fails to register in freepbx (Advanced \Configurations \Phones Setup) you cut to the chase and go to my solution at the end of this thread:

http://forum.linuxmce.org/index.php?topic=8029.0

This also provides you links to what you are supposed to do to make the phone pnp.

Good luck with your new phone.

53

Users / Re: Cisco 7970 and Error Message (solved for me)

« on: March 19, 2010, 04:47:46 pm »

I started getting this error as well on 2 of my 3 cisco 7970 phones. Since it was working on one, I knew that something must have changed on the other two but I didn't know where to look. I logged into the terminal and ran

tail -f /var/log/pluto/DCERouter.log

as I tried to get the orbiter to connect. This gave me the cisco orbitor device number that was trying to log in but not much more.

I then went to Wizard \ Devices \Orbiters in the web admin and found the device number (which is listed next to the description field). Then clicked on advanced and near the bottom of the page found

Remote Phone IP

For some reason I do not understand, it was off by one digit: 192.168.80.14 instead of 192.168.80.13
I corrected this field. Saved the page and then did a Wizard \Restart \Quick Reload Router from inside the web admin.

Exiting the orbiter on the phone and then reinitializing it under the Services Menu while the DCERouter.log was running, I no longer got the invalid ip message (who would of thought that it actually was telling you the truth) and saw from the log that the orbiter was actually loading.

My suspicion is that this ip gets coded into that field the first time the phone is loaded and when you change your subnet or for some other reason restart the system after a catastrophe, the DHCP server gives the device a new ip but the old ip is still in that field. Just guessing.

Hope this helps someone who has run into a dead end with this problem.

54

Users / STB --> Firewire --> Core works; RS232 controls TV How to coordinate remote?

« on: March 17, 2010, 11:39:13 pm »

I must be missing something. It seems like I have all the hard parts done but the remote refuses to follow along. Anyone had this problem?

STB works beautifully with DCT 6412 template--usbuirt codes do exactly what asked when tested in webadmin or with remote.

HD output from STB through firewire to core/hybrid works and changes channels from mythTV but the image is not 1080i with current video card. Also a bit slow and sometimes flaky with current hardware.

Temporary solution: I want to go directly from STB through HDMI to TV and have Linuxmce just handle the pass off using the linuxmce usbuirt remote all the way: Connection wizard shows audio and video output from STB to HDMI input on TV (which is controlled by RS232).

What happens when I go to media\ LiveTV DCT6412 in linuxmce is that it recognizes I want to go to HDMI on the TV and switches to that input on the TV using rs232. Also, if I hit the "windows or green button" on my remote it recognizes that I want to go back to Linuxmce interface and takes me there where basically all I can do is turn "off" the LiveTV connection.

The problem seems to be that the STB is controlled by usbuirt infrared codes and once you have switched to the TV on the HDMI port Linuxmce seems to be trying to use the RS232 codes through the remote when what you want is the remote to switch back to the STB codes to control everything.

I've read everything I can find in the wiki and the forums but I must be missing something.  One thing I have considered--but don't want to try unless it is confirmed as the way to go--is  to delete all the rs232 codes for the TV except the inputs so that Linuxmce is forced to use the STB codes by default once the input is changed. I just don't know enough about how the system works and it seems to be a pretty ruthless way to get this accomplished.

Any advice or experience with this problem greatly appreciated.

55

Users / Ruby code in wiki tutorial for LG 42LB5D #355 Process Initialize breaks compile

« on: March 15, 2010, 06:13:00 pm »

So that no one else has to suffer through figuring this out:

On this link http://wiki.linuxmce.org/index.php/LG_42LB5D
under "Add Ruby Snippets to Commands" you will find this code
----------------------------
#355 Process Initialize
   for iRetry in 0...4
          print "Initializing unit\n"
          conn_.Send("ke 01 01\r") # Send UnMute Command
          buf = conn_.Recv(30,200) # Expected Return # ke 01 01\r\ne 00 OK01x\r\n
          if !buf.nil? && buf.include?("OK")
            print "Initialized ok\n"
            print "Setting volume to 30%\n"
            cmd_313(15)
            SetDeviceDataInDB( device_.devid_, 158, "15" )
            # 158 = DEVICEDATA_Volume_Level_CONST
            return
          end
          print "Failed to initialize. Wait 1 secs and try again\n"
          sleep(1)
        end
#DisableDevice( device_.devid_, true )
#print "The device would not respond. Disabling it.\n"    
----------------------------
Unfortunately it will cause a compile error in 7.10 in the "follow log" on the "advanced" link of the A/V device and will not allow your template to run. The compile error will state that it found a "kend" and was expecting "$end".
If you delete this code snippet from the template--leaving #355 blank---and restart the router, it will compile with no problem and the "send command to device" Link on the "advanced link of the GSD device seems to work through rs232 .

Since this page has been copied as a tutorial for other LG TVs and a request was put in a couple of weeks ago to have the code included as a pnp device for 8.10
http://forum.linuxmce.org/index.php?topic=9729
many would assume that it has been tested and works. This is not true.
It is probably very easy to fix--looks to me like the definitions at the top of this snippet were not copied into the tutorial, but I know nothing about Ruby or I would attempt it.
BTW, otherwise this seems to be a good template for LG TVs with serial ports. Some of the command need to be changed for later models <$xb 01 90\r$> instead of <$kb 01 07\r$> for inputing HDMI 1 for example, but generally a good guide for adapting it to your TV.

If the author of the tutorial or someone who has access to this page on the wiki could fix the code it would be greatly appreciated.

56

Installation issues / Re: LG LCD Template

« on: March 14, 2010, 01:55:28 pm »

Should I be looking here or elsewhere for the post that it has been done? Also any help on how to retrieve where you post it and install it would be greatly appreciated as I don't have experience with what you posted as the destination.

I'm having exactly the same problem getting rs232 to work on my LG TV--despite the fact that the commands work beautifully from the command line--a functional template would be great. The LG template description in the wiki andincluded in 7.10 gives me compile errors about expecting a $end and finding a kend. Even when I create a new template with only a single command and it says it compiles OK, when you test the command inside the web admin page they do not show up as being sent in the "follow log" for the TV (the send command on the advanced page for the AV device does show up). Any help greatly appreciated

57

Users / Re: Anyone know how to change the default custom-linuxmce script announcement?

« on: March 06, 2010, 05:53:27 pm »

Many thanks again! It took some digging this morning on my own to find the directory that you so graciously posted--somehow I knew the answer would be waiting for me when I logged back in! As a side note, it would be great if we could have a central listing of where everything is stored in the directory structure for each device/application in linuxmce. This is often half the battle in trying to get things to work.

My big question still remains how Linuxmce is creating the new announcement as you add users. It seems like it is re-recording the files in that directory as it needs them and then doing a text to speech of the users names in the order they are registered with phone extensions. Anyone that could shed light on how this happens would be a true guru of Linuxmce in my eyes. Lacking that....

If you want to modify your intro on asterisk here is what I have found:

1) As pointed out above, all the voice-menu files are saved as gsm files and located in the directory /var/lib/asterisk/sounds/pluto. Asterisk seems to default to that directory so I didn't have any luck getting any of the files in /var/lib/asterisk/sounds to play with default conf script (listed below in 2). So, if you want to add your own voice menu you should probably consider putting it in there.

2) The file you need in order to tell asterisk where to locate the new voice menu is /etc/asterisk/extensions_pluto_dial.conf and the context is [voice-menu-pluto-custom]. The fourth line down in that context tells asterisk to play the voice menu file: exten => s,4,Background(pluto/pluto-default-voicemenu) and this is the one that has all the users names and #'s included. On my system, it is not really of great audio quality and sounds very computer-speech generated--which is a bit confusing to first-time callers because it is difficult to understand.

3) The other files in the directory which seem to be used to make this file are of quite good quality. If for example, you swap out the line above with the following examples you will get the messages listed:

"Thank you for calling":
exten => s,4,Background(pluto/pluto-default-voicemenu1)

"To call everbody in the house dial 0":
exten => s,4,Background(pluto/pluto-default-voicemenu2)

"If you know the extension, Dial it now":
exten => s,4,Background(pluto/pluto-default-voicemenu3)

"To leave a message press the # sign":
exten => s,4,Background(pluto/pluto-default-voicemenu4)

4) So, if you didn't care about listing the extensions of the users in your home, you could conceivably take these 4 files, join them together, and rename the new file something like /pluto/mashup-intro and use that as the voice intro once you replace the line listed above with your file name. I guess you might also consider just adding them one by one as arguments in the Background command: exten => s,4,Background(pluto/pluto-default-voicemenu1,pluto/pluto-default-voicemenu2, etc.) but I haven't tried that and don't know how that would affect the timing.


5. Alternatively, you could make your own recording based on the extension prompts already listed in the default "pluto/pluto-default-voicemenu" with a recording program that generates gsm or wav files and place it in the directory.

Any other suggestions greatly appreciated.

58

Users / Re: Anyone know how to change the default custom-linuxmce script announcement?

« on: March 05, 2010, 08:06:58 pm »

Thanks for the link. Went there and it looks like this is all straight asterisk stuff that would be used with the IVR. Which would be great if this is true.

However, since my FreePBX hooks into asterisk  (Advanced -> Configurations -> Phone Setup) having nothing setup for the Digital Receptionist (IVR) I assume (or perhaps remember reading somewhere?) that Linuxmce doesn't use this (and must not be touched or the whole asterisk house or cards will fall into the Linuxmce foundation).

Can someone confirm that this is true? Could you actually use asterisk's IVR instead of Linuxmce's [custom-linuxmce] context (really [from-pluto-custom] in /etc/asterisk/extensions_pluto_dial.conf? From my uneducated look there, I thought Linuxmce was calling an external program (text-to-speech engine?) to do the reading from a text file and then moving to the next step based on the caller's input. What I read of the setup in 8.10 using festival would also lead one to believe that something like this is the case in 7.10.

Anyone else have some clues/thoughts?

Again, thanks in advance.

59

Users / Anyone know how to change the default custom-linuxmce script announcement?

« on: March 05, 2010, 05:11:39 pm »

Hi,
I can't find how to do this in the forum or the wiki for 7.10. Currently it says, "Thank you for calling, to call everybody in the house press 0..." when you call in on Asterisk. I am assuming that there is a text-to-speech script written somewhere that could be modified but I can't find it. Any pointers greatly appreciated.

60

Users / Re: Problem with call routing

« on: September 21, 2009, 10:04:01 pm »

For anyone having this problem, as I did, and has found no love in the forum here is what I did to fix it:

1) Go to Telecom/ Call Routing in the Linuxmce Admin Website and under a user's name select a user mode/ call type.
2) Next to "add new step" select "transfer to outside number" from the drop down box, then type in your outside number preceded by a 9 such as 918885551212 if you live in the US. Click on the Add button.
3) You will see that it added the step you requested but puts the number 2147483647 in the box instead of the number you added. Even though this is editable, ignore it.
4) Open a terminal and ssh into linuxmce (ssh linuxmce@192.168.80.1 if you use the default ip) then get admin rights with a sudo -s
5) Open an editing program and the file this is supposed to be written to with a command such as "nano /etc/asterisk/extensions_pluto_dial.conf"
6) Search for "214" if it is there, replace the whole 2147483647 with the number you entered originally (preceded with 9 as described above), Save the file. If 214 is not there, search for the number you entered and make sure it is associated with the correct user extension such as 301. If neither are there try adding the outside number step again and proceed once one of them is there.
7) Reopen the file and check again, just to be sure. Exit.
from the terminal run "asterisk -vvvvvr"
9) once asterisk opens, type "restart gracefully"
10) open asterisk CLI again as described in 8
11) Dial into your phone system and go to user you want to have phone forwarded to by selecting it from the prompt. It should now dial out correctly. (Easy to test this using the cell phone you have listed as the number to dial"

I don't know how the script works on this web page in the admin website, but it sure seems like it would be an easy fix. Obviously the developer has put a place holder in there for testing and has forgotten to replace it with the variable you are entering. Hopefully once you fix this you don't need to go back and fix it every time asterisk restarts or another call type is modified, but I have not tested this yet.

This is in 7.10. Hopefully someone will take note of this for the next release. Sure makes your phone system much more useful.