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18
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LinuxMCE / Users / Re: Orbiter with knock off web pads
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on: April 25, 2012, 10:38:51 am
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Hi,
I'm also taking close look to those tablets... But have bought one about year ago and it wasn't responsive as expected...
Therefore would be more than happy if we exchange our experiences and see if we can find suitable one for good price...
Thanks in advance for any experience,
regards,
Bulek.
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19
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LinuxMCE / Users / Re: [Dianemo S] Phone system configuration
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on: April 21, 2012, 06:42:53 pm
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Hi, I have phone system "kind of working". My major problem is that I use 3 Cisco 7970 phones and chan_sccp_b currently doesn't work for me (only svn version for Asterisk 1.  . Other extensions seems to be working, although have some weird problems (one of the embedded ORbiters looses connections and I get no audio on calls). But also, phone system still uses Freepbx, but you cannot change anything regarding extensions in Freepbx, cause that causes troubles with Perl scripts - so all extensions are on trivial passwords - not quite secure... I'm not sure if I can help much, but maybe you can try to post more specific questions.... Basically you have to add extensions through web Admin page, I only use Freepbx to setup some specific things for trunks, incoming calls etc... The rest can then be done on Telephony page on web admin... Regards, Bulek.
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21
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LinuxMCE / Developers / Re: Marantz SR over RS232
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on: April 16, 2012, 07:56:42 am
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Hi,
I have it working under Dianemo with whisperer Perl script. But I guess this is not of much help for you.... Selecting all those inputs in template is too much work, Dianemo has a bit easier system, but you have to know more to use it...
Regards,
Bulek.
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23
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LinuxMCE / Users / [Dianemo] Problems with speech announcements, delay command ....
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on: March 29, 2012, 08:58:26 pm
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Hi, I'm trying to setup speech announcements under Dianemo 11.10 and have problems. I remember from 10.10 and LMCE 7.10 that similar problems were present, perhaps they exist also on recent releases of LMCE. I have following questions and would be glad to hear your experiences/opinions : 1. It seems that tts wav files are properly generated, but it seems like they are shortened when played as audio... Also if another speech announcement is triggered it overlaps with the first one - it seems like no queue is used here ? I'm not sure, but maybe generation of wav file and play media command are too close in time (?) : 08 03/29/12 21:42:10.596 Received Message from 0 (unknown / ) to 18 (Text To Speech / Living Room), type 1 id 253 Command:Send Audio To Device, retry none, parameters: <0x5373ab70> 08 03/29/12 21:42:10.596 Parameter 9(Text): This is a test and wonderfull time. <0x5373ab70> 08 03/29/12 21:42:10.596 Parameter 103(List PK Device): 22 <0x5373ab70> 08 03/29/12 21:42:10.596 Parameter 254(Bypass Event): <0x5373ab70> 08 03/29/12 21:42:10.596 Parameter 276(Dont Setup AV): <0x5373ab70>
08 03/29/12 21:42:10.627 Received Message from 18 (Text To Speech / Living Room) to 10 (Media Plug-in / Living Room), type 1 id 43 Command:MH Play Media, retry none, parameters: <0x45506b70> 08 03/29/12 21:42:10.627 Parameter 2(PK_Device): 22 <0x45506b70> 08 03/29/12 21:42:10.627 Parameter 13(Filename): /home/public/data/tts/0.wav <0x45506b70> 08 03/29/12 21:42:10.627 Parameter 29(PK_MediaType): 0 <0x45506b70> 08 03/29/12 21:42:10.627 Parameter 44(PK_DeviceTemplate): 0 <0x45506b70> 08 03/29/12 21:42:10.627 Parameter 45(PK_EntertainArea): <0x45506b70> 08 03/29/12 21:42:10.627 Parameter 116(Resume): 1 <0x45506b70> 08 03/29/12 21:42:10.627 Parameter 117(Repeat): 0 <0x45506b70> 08 03/29/12 21:42:10.627 Parameter 253(Queue): 0 <0x45506b70> 08 03/29/12 21:42:10.627 Parameter 254(Bypass Event): 0 <0x45506b70> 08 03/29/12 21:42:10.627 Parameter 276(Dont Setup AV): 0 <0x45506b70>
2. I've setup two speech announcements and delay command for DCERouter for 1min in between, but speech commands are send one after another : 08 03/29/12 21:16:14.415 Received Message from 62 (Windows XP PC/tablet (Horiz) / Living Room) to 18 (Text To Speech / Living Room), type 1 id 253 Command:Send Audio To Device, retry none, parameters: <0x4b40cb70> 08 03/29/12 21:16:14.415 Parameter 9(Text): This is a test of speech synthesis. I hope you can hear and understand me clearly. <0x4b40cb70> 08 03/29/12 21:16:14.415 Parameter 103(List PK Device): 22,356 <0x4b40cb70> 08 03/29/12 21:16:14.415 Parameter 254(Bypass Event): <0x4b40cb70> 08 03/29/12 21:16:14.415 Parameter 276(Dont Setup AV): <0x4b40cb70> 08 03/29/12 21:16:14.417 Received Message from 62 (Windows XP PC/tablet (Horiz) / Living Room) to 2 (DCERouter / Living Room), type 1 id 257 Command:Delay, retry none, parameters: <0x4b40cb70> 08 03/29/12 21:16:14.417 Parameter 102(Time): 300000 <0x4b40cb70> 08 03/29/12 21:16:14.419 Received Message from 62 (Windows XP PC/tablet (Horiz) / Living Room) to 18 (Text To Speech / Living Room), type 1 id 253 Command:Send Audio To Device, retry none, parameters: <0x4b40cb70> 08 03/29/12 21:16:14.419 Parameter 9(Text): I'm speaking again after 5 mins. I hope you can hear me. <0x4b40cb70> 08 03/29/12 21:16:14.419 Parameter 103(List PK Device): 22,356 <0x4b40cb70> 08 03/29/12 21:16:14.419 Parameter 254(Bypass Event): <0x4b40cb70> 08 03/29/12 21:16:14.419 Parameter 276(Dont Setup AV): <0x4b40cb70>
3. "Send Audio to Device" command for Text&Speech device has parameters I don't clearly understand : # 254 Bypass Event (bool) # 276 Dont Setup AV (bool)
Anyone knows some more details about this ? 4. When I try to make speech announcement during media play, then things get pretty messy. Audio stops playing, speech announcement is not heard at all, then media continues but media screen gets empty (no files in playlist), media floorplan has some empty colour rectangles and media button shows like nothing is played. Anyone with similar experience ? It also seems that media is not paused, but after short interval, next song from playlist is played - with not enough time to play announcement: 08 03/29/12 21:42:10.956 Received Message from 23 (Xine Plug-in / Living Room) to 22 (Xine Player / Living Room), type 1 id 37 Command:Play Media, retry none, parameters: <0x4a901b70> 08 03/29/12 21:42:10.956 Parameter 29(PK_MediaType): 4 <0x4a901b70> 08 03/29/12 21:42:10.956 Parameter 41(StreamID): 1002 <0x4a901b70> 08 03/29/12 21:42:10.956 Parameter 42(MediaPosition): <0x4a901b70> 08 03/29/12 21:42:10.956 Parameter 59(MediaURL): /home/public/data/tts/0.wav <0x4a901b70>
.... and then really soon next song ....
08 03/29/12 21:42:10.996 Received Message from 23 (Xine Plug-in / Living Room) to 22 (Xine Player / Living Room), type 1 id 37 Command:Play Media, retry none, parameters: <0x4a901b70> 08 03/29/12 21:42:10.996 Parameter 29(PK_MediaType): 4 <0x4a901b70> 08 03/29/12 21:42:10.996 Parameter 41(StreamID): 1001 <0x4a901b70> 08 03/29/12 21:42:10.997 Parameter 42(MediaPosition): <0x4a901b70> 08 03/29/12 21:42:10.997 Parameter 59(MediaURL): /home/public/data/audio/Media [30]/FAMILY/Lesnik Marjana/Best of/Lesnik Marjana - Best of - Zasnubi me.mp3 <0x4a901b70>
40 ms is clearly not enough to play speech announcement. Thanks in advance, regards, Bulek.
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24
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LinuxMCE / Marketplace / Re: New Update Released - Improved Orbiter Progress Screen
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on: March 29, 2012, 10:16:02 am
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We have released an update that replaces the Orbiters progress screen with a much improved one;
We replaced the X11 progress screen/bars with SDL progress bars similar to the one you see on startup on UI1 already (as you probably noticed, you don't see it on UI2 - it was disabled in code). Now in UI1 & UI2 we have a single progress screen design that has a much improved visual layout and is based on SDL rather than the old 'bolted' on X11 progress window. These progress windows are displayed by the Orbiter during a regen or a reload router.
More updates on the way... new ZWave binary & a new Dianemo UI Skin...
All the best
Andrew
Hi, thanks for efforts. If I may add, there are still some things that at least I'd like to have them differently regading UI (I'm using Dianemo (dark) skin) : 1. UI1: Media button that displays current media name is in gray color on virtually black background - and cannot be seen... 2. On interactive floorplans all texts are written in white, so one should have really dark backgrounds to see anything... Any chance to give choice of black or white to user ? I had black colour on 7.10 and visibility was better. 3. I miss more floorplans on UI1 (only 4 currently - there are approx 350 parameters flowing from home automation to Dianemo and I miss more floorplans - there are 8 floorplans on UI2) 4. On UI2 floorplans are still overlapped (this is the same as it was for me on 7.10). That means that when you click on some light on floorplan, you get "fast movie of all floorplans" and then the right one shows.... Hope this helps a bit, regards, Bulek.
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25
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LinuxMCE / Users / Re: Command Line System Queries
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on: March 16, 2012, 11:51:55 am
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Thanks for wiki page.
I'd like to ask additional question : can I check what is happening in certain Entertainment zone (whether media is playing, or no,...) ?
Thanks in advance,
regards,
Bulek.
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27
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LinuxMCE / Users / Re: [Dianemo S] Phone system configuration
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on: March 15, 2012, 11:25:06 am
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My biggest problem is getting pluto/dianemo to understand what I did in FreeBPX! I'll wait for the wiki, hopefully that will help ...
You shouldn't change too much in Freepbx as parts are generated by Dianemo/LMCE scripts. I remember Radu said not to touch extensions (although passwords are trivial and make system vulnerable)... What I changed in Freepbx is only trunk settings and took care so incoming phone calls come to proper extension. The rest is up to Dianemo.... HTH, regards, Bulek.
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29
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LinuxMCE / Users / Can we get another Dianemo centric board under LinuxMCE
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on: March 13, 2012, 10:23:24 am
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Hi,
those two systems are different in many details, so I think it would be good to separate Dianemo specific discussions into separate board not to confuse all other less experienced users...
Can we make new Dianemo board under LinuxMCE ? I think it would be better for all of us.
I just found out that I can do it - but would still like to hear other member's opinions.
Regards,
Bulek.
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30
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LinuxMCE / Users / Re: [Dianemo S] Phone system configuration
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on: March 13, 2012, 10:20:54 am
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Hi, I remember that I had to fix few things and got telephony in some kind of "working state". I think that some of basic Asterisk debugging skills should be used... First of all, enter asterisk shell with "asterisk -vvvvvvvvvvvvvvvrgc" and from there you can check "sip show registry" state of your SIP trunks, then "sip show peers" your extensions, etc... Then also in shell you can get debug messages when you try to make a call - then try to locate warnings, errors. You can even get more messages by "tail -f /var/log/asterisk/messages" I've found my notes and instructions what I needed to fix my system - hope they will help, although they should get into updates soon : had to change : [19.2.2012 3:12:08] tiniahouse: add this below to /etc/asterisk/extensions_custom.conf [19.2.2012 3:12:11] tiniahouse: [incoming-sip-calls] include => ext-did
#include extensions_pluto_dial.conf
; ########################################################################### ; PLUTOs "trusted" context ; ########################################################################### [trusted] include => from-internal [19.2.2012 3:12:25] tiniahouse: comments : [19.2.2012 3:14:05] tiniahouse: I have SIP trunk to my provider on dedicated network card... For some reason, all incoming calls (even from trunk) end in from-sip-external context that is basically meant for anonymous incoming calls... so it's pretty restricted and all I got was "out of service" annoucement... [19.2.2012 3:15:47] tiniahouse: So I created my own context (first two lines) and also made this context default. That lowers security level, but we already have passwords being the same as extension number... Now, when call comes in, if it's for DID, then it goes to coresponding incoming route - that's first step I made [19.2.2012 3:17:15] tiniahouse: Now, my telephony is organized in this manner : I route all incoming calls to one dummy Dianemo user (housephone)... Why ? Cause I don't need to have more users and then I have to switch statuses for all of them... So we have housephone and calls are routed according to its state... [19.2.2012 3:19:18] tiniahouse: so at the end of incoming route, there is a dial for Local channel 307@.... and 307 is virtual extension for user housephone... That was not working cause 307 and all other user extensions were not visible in dialplan... For some reason file with them is not included in any other config files.... So that explains 3rd line [19.2.2012 3:21:18] tiniahouse: Then I got to proper housephone user, but calling more users at once (according to routing setup in admin page) was not working, cause calls are made as 202@trusted - trusted context was unknown in my system (it's only synonim for context from-internal - so that explains last lines [19.2.2012 3:22:33] tiniahouse: I also had to change /etc/asterisk/sip_general_custom.conf ,where I changed from-sip-external to my new context incoming [19.2.2012 3:22:38] tiniahouse: -sip-calls.... [19.2.2012 3:23:34] tiniahouse: Also, there are files missing in /usr/share/asterisk/agi-bin - those with pluto-... in front of them (I compared to my 7.10 system)... [19.2.2012 3:23:57] tiniahouse: Now also incoming calls are working ok in my house... Did you follow me ?
Sometimes I just wonder how Dianemo users use telephony...  HTH, regards Bulek.
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