| 00:03.57 | *** join/#asterisk n3hxs (~Ed@pool-71-162-133-193.phlapa.fios.verizon.net) |
| 00:04.27 | *** join/#asterisk russellb (~russellb@redhat/russellb) |
| 00:04.27 | *** mode/#asterisk [+o russellb] by ChanServ |
| 00:25.35 | *** join/#asterisk TriJetScud (~TriJetScu@van-app-svr.ad.v10networks.ca) |
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| 00:35.32 | *** join/#asterisk jhlavacek (~jirka@84.19.95.180) |
| 00:38.48 | Micc | What is the application that allows a channel to join as an announcer/broadcaster to a conf bridge? I read about it somewhere but can't find it now. |
| 00:39.16 | Micc | Or maybe its part of Park or something. |
| 00:39.57 | Micc | nvm, I found it. Holding bridge. |
| 00:40.32 | *** part/#asterisk karamell (~maxi@46.45.42.197) |
| 00:41.52 | *** join/#asterisk wonderworld (~ww@ip-176-199-164-4.hsi06.unitymediagroup.de) |
| 00:50.20 | *** join/#asterisk HeN (uid3747@gateway/web/irccloud.com/x-gejsoppgrrgrmqhf) |
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| 01:55.05 | *** join/#asterisk trafalgargirl (~Adium@MTRLPQ4706W-LP130-02-1176239601.dsl.bell.ca) |
| 01:56.19 | trafalgargirl | hoping I can get some help â i am dealing with a new set up I am not used to working with and therefor not seeing what I would normally see. For instance, when I launch the CLI I am not seeing any call info (yes there are calls coming in) |
| 01:56.45 | trafalgargirl | also despite having changed the afterhours message, it seems to be playing the old one |
| 01:58.26 | trafalgargirl | tried a restart and reloaded the dialplan but still playing the wrong file. |
| 02:02.13 | [TK]D-Fender | Asterisk doesn't have "afterhours" unless you invented it in your dilplan |
| 02:02.47 | [TK]D-Fender | as for CLI, if you actually set the verbose level and something like SIP (or equivalent relevant debug) and see nothing.. then something is very wrong |
| 02:03.05 | [TK]D-Fender | Show us your having set those things and then a status dump while a call is actually in progress |
| 02:03.15 | [TK]D-Fender | ~pb |
| 02:03.15 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 02:03.16 | [TK]D-Fender | ^^^^ |
| 02:03.47 | [TK]D-Fender | And another thought is one I've seen here before "Oh shit I was logged into the wrong server altogether, nevermind!" |
| 02:03.55 | [TK]D-Fender | Make sure it's not one of those |
| 02:04.03 | [TK]D-Fender | They are by far the most embarassing |
| 02:10.00 | trafalgargirl | luckily i am sure that i am indeed on the correct server :) |
| 02:10.33 | trafalgargirl | sorry afterhours is what we call our afterhours sound file |
| 02:10.43 | trafalgargirl | sorry about that (newbie here) |
| 02:12.41 | trafalgargirl | i did not set this up - so therefor unsure about the verbosity. Am happy to check whatever you advise me to |
| 02:13.02 | [TK]D-Fender | "core set verbose 10" |
| 02:13.51 | trafalgargirl | Verbosity was 0 and is now 10 |
| 02:17.49 | [TK]D-Fender | Considerably better... |
| 02:18.57 | trafalgargirl | much. though now i see other problems (like why is there no called id being captured.. but one thing at a time) |
| 02:19.04 | trafalgargirl | trying the pb now |
| 02:22.54 | trafalgargirl | http://pastebin.ca/2880730 |
| 02:24.58 | trafalgargirl | i put a new recording (but kept the same name) but I am still getting the old recording playing. |
| 02:25.05 | trafalgargirl | (not sure if this is making sense) |
| 02:27.44 | [TK]D-Fender | <PROTECTED> |
| 02:27.50 | [TK]D-Fender | well we see which file it's playing. |
| 02:28.10 | [TK]D-Fender | So the easy guess is you either put the wrong file in, or it's in the wrong place |
| 02:28.26 | [TK]D-Fender | because the contents of what that's playing is just that... |
| 02:29.35 | *** join/#asterisk andresmujica (~andresmmu@ubuntu/member/andresmujica) |
| 02:30.49 | trafalgargirl | this is why i am confused. i have checked both of those things |
| 02:32.08 | trafalgargirl | the file is in /var/lib/asterisk/sounds/mate1 |
| 02:34.29 | trafalgargirl | when i transferred the file today to my desktop it is the correct one |
| 02:35.28 | trafalgargirl | (i am not trying to be obtuse just really at a loss hereâ¦. ) |
| 02:39.19 | *** join/#asterisk Navion (~navion@rivendell/users/navion) |
| 02:43.13 | [TK]D-Fender | Then try renaming the file out from under it and see if it errors out |
| 02:43.29 | [TK]D-Fender | because if it does then tht is at least the file and you'll need a lot better proof of its contents |
| 02:44.27 | trafalgargirl | so do a rename of the file I think it should be? or change the name of the file it is pulling? |
| 02:44.54 | Navion | Anyone know of an open source appliance I can use to answer voip calls and play whatever is attached to an audio input? Needs to answer many simultaneous calls. Like attached to a radio that people can call and listen to (bad example...). |
| 02:44.56 | trafalgargirl | ie: change mate1/openinghours to something else? |
| 02:46.17 | [TK]D-Fender | yes |
| 02:46.18 | [TK]D-Fender | mate1/openinghours |
| 02:46.39 | [TK]D-Fender | "ls -la /var/lib/asterisk/sounds/mate1" |
| 02:46.42 | [TK]D-Fender | PB ^^^^^ |
| 02:47.07 | trafalgargirl | i did that earlier and it errored out (change it to pull from 444 which is the file that is the orginal recording the new file |
| 02:47.10 | trafalgargirl | k |
| 02:48.52 | trafalgargirl | http://pastebin.ca/2880740 |
| 02:51.09 | *** part/#asterisk Navion (~navion@rivendell/users/navion) |
| 02:54.44 | Penguin | Your complaint is that asterisk is playing a file that exists? |
| 02:54.49 | Penguin | Change the name of the file. |
| 02:55.38 | Penguin | mv openinghours.wav openinghours-disabled.wav |
| 02:57.05 | trafalgargirl | http://pastebin.ca/2880746 |
| 02:57.48 | trafalgargirl | no - it's playing an OLD version of the file - i think it's playing the .bak not the current version |
| 02:58.09 | Penguin | mv openinghours.bak openinghours-disabled.wav |
| 02:58.44 | [TK]D-Fender | never leave it with the same main filename |
| 02:59.05 | trafalgargirl | (i just errored out and pb it) |
| 02:59.14 | trafalgargirl | ok |
| 02:59.47 | Penguin | This pastebin says it can't play 444 |
| 03:00.31 | Penguin | What is the username that asterisk runs as? 'ps -C asterisk' |
| 03:00.47 | trafalgargirl | ok did the rename |
| 03:01.19 | Penguin | Sorry... 'ps -C asterisk u' |
| 03:01.23 | trafalgargirl | root@pbx:/var/lib/asterisk/sounds/mate1# ps -C asterisk |
| 03:01.23 | trafalgargirl | <PROTECTED> |
| 03:01.23 | trafalgargirl | 22343 ? 00:00:01 asterisk |
| 03:01.50 | Penguin | I forgot the u. ps -C asterisk u |
| 03:01.54 | trafalgargirl | asterisk 22343 0.0 0.2 810032 30896 ? Ssl 21:15 0:01 /usr/sbin/asterisk -p -U asterisk |
| 03:02.02 | Penguin | Okay, asterisk username. |
| 03:02.45 | Penguin | So the permission on 444.wav should be suitable for asterisk to read... |
| 03:02.50 | Penguin | 'file 444.wav' |
| 03:02.59 | Penguin | Check what type/format of file it is. |
| 03:03.35 | trafalgargirl | i444.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz |
| 03:03.38 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 03:03.48 | Penguin | i444.wav? |
| 03:04.13 | trafalgargirl | sorry that must be somethig extra from my cut and paste |
| 03:04.26 | Penguin | 16 bit mono 8K is good, so that's probably not the problem. |
| 03:04.51 | [TK]D-Fender | Ok, go copy another replacement file in there, and test it on a desktop or something first to prove it actually is right |
| 03:05.13 | [TK]D-Fender | Moving it out of the way seems to prove you're looking at the right file in the right place |
| 03:05.24 | [TK]D-Fender | So now I'm on to solidly distrusting the files contents |
| 03:06.42 | trafalgargirl | just to be clear (cause i am new) we recorded the new recording on our system - and called it 444. Then I moved 444 to sounds/mate1 ⦠then I bk up the old openinghours, and cp 444 to openinghours |
| 03:06.44 | Penguin | I've had this same problem before. There's no reason it should do this, but it would not play a wav file when using ulaw. I created a ulaw version of the same file and the file played. This isn't a fix, but a workaround. |
| 03:08.05 | [TK]D-Fender | I'd go prove this process directly again... |
| 03:08.15 | [TK]D-Fender | Because I'm smelling SNAFU here.. |
| 03:09.54 | trafalgargirl | sorry what should I do? |
| 03:10.33 | [TK]D-Fender | go prove your supposed good file (444) is working, and kill off the other openinghours and cp it again. |
| 03:11.02 | trafalgargirl | put it on my desktop and play it? |
| 03:11.03 | [TK]D-Fender | I'd be wondering if this was a case of bad aim on your CP |
| 03:11.07 | [TK]D-Fender | Sure that too |
| 03:11.17 | [TK]D-Fender | You want to prove a thing... prove a thing |
| 03:11.20 | [TK]D-Fender | the hard way. |
| 03:11.25 | [TK]D-Fender | Nothing beats brute force |
| 03:11.35 | [TK]D-Fender | Go empirical on it... |
| 03:12.36 | trafalgargirl | gotcha. now i just have to figure out how to get that file to my desktop :/ |
| 03:14.16 | [TK]D-Fender | Windows? |
| 03:14.19 | Penguin | Windows? |
| 03:14.20 | trafalgargirl | mac |
| 03:14.34 | Penguin | I'm guessing there's an sftp client in Mac OS. |
| 03:14.49 | [TK]D-Fender | yup |
| 03:15.27 | trafalgargirl | usually i just scp the file to my desktop â but having trouble |
| 03:15.41 | [TK]D-Fender | http://sourceforge.net/projects/filezilla/files/FileZilla_Client/3.9.0.6/FileZilla_3.9.0.6_macosx-x86.app.tar.bz2/download |
| 03:15.58 | [TK]D-Fender | The more trouble you're haing the less I trust your files.... |
| 03:16.26 | trafalgargirl | (part of this is a newbie issue) but i am more than happy to do a screenshare etc. withyou |
| 03:18.57 | [TK]D-Fender | Give filezilla a try first |
| 03:19.09 | [TK]D-Fender | very solid app |
| 03:20.46 | trafalgargirl | i have it - since i need to be root to get the pbx not sure how to navigte through that using it (currently just ssh to a server as a jump off) |
| 03:21.26 | Penguin | You don't connect directly to your asterisk box from your desktop computer? |
| 03:24.25 | trafalgargirl | no |
| 03:24.34 | trafalgargirl | not when i am at home (where I am) |
| 03:24.50 | trafalgargirl | and we are currently sans vpn |
| 03:25.40 | Penguin | Whichever host you ssh to, that's where you'll have to put the file to get it to your computer. (there are other ways, but it's already complicated enough.) |
| 03:25.56 | trafalgargirl | k |
| 03:26.13 | Penguin | So if you ssh to host1 and then from there you ssh to asterisk machine, sftp the file to host1, then sftp/scp it from host1 to your computer. |
| 03:29.35 | trafalgargirl | trying⦠|
| 03:32.06 | trafalgargirl | what i am doing should be working â sorry I am trying to get this right |
| 03:33.52 | trafalgargirl | oh ffs - whatever i tried to do seems to have written files to that dir usern@home.dom.com |
| 03:35.23 | *** join/#asterisk CeBe1 (~CeBe@port-92-200-92-160.dynamic.qsc.de) |
| 03:35.36 | trafalgargirl | ok i am now at a loss. |
| 03:37.40 | Penguin | sftp user@remotehost:/path/to/file |
| 03:37.48 | Penguin | scp user@remotehost:/path/to/file . |
| 03:37.57 | Penguin | Use either one. |
| 03:38.11 | Penguin | That dot is important, by the way. |
| 03:43.37 | trafalgargirl | Connected to server. |
| 03:43.37 | trafalgargirl | Couldn't stat remote file: No such file or directory |
| 03:43.56 | trafalgargirl | (I am logged in as root on the pbx) |
| 03:43.57 | Penguin | That would be a major problem. |
| 03:44.32 | trafalgargirl | should i be doing this from the pbx or from my host1? |
| 03:44.36 | Penguin | Wait... |
| 03:44.41 | Penguin | Login to your jump host. |
| 03:44.50 | trafalgargirl | ok i have a new window |
| 03:45.01 | Penguin | sftp root@asterisk:/path/to/your/file |
| 03:46.41 | trafalgargirl | oh we did it! |
| 03:46.45 | trafalgargirl | playing file |
| 03:48.51 | trafalgargirl | ok the file is on host1 â now just having issues getting it from there to my desktop |
| 03:49.11 | Penguin | Now you can use your sftp or scp client to fetch it from the jump box. |
| 03:49.32 | trafalgargirl | if i do an ls i see it but not via the sftp |
| 03:55.45 | trafalgargirl | user@server:~$ ls |
| 03:55.45 | trafalgargirl | openinghours.wav |
| 03:55.53 | trafalgargirl | so the file is there |
| 03:57.36 | Penguin | And that's the jump host? |
| 03:57.42 | trafalgargirl | yes |
| 03:57.56 | Penguin | Be sure you sftp to the jump host using the right username. |
| 03:58.10 | trafalgargirl | yep |
| 03:58.21 | trafalgargirl | (only have one option) |
| 03:59.45 | trafalgargirl | (Im sorry this should not be so hard) |
| 03:59.58 | Penguin | No argument from me. |
| 04:00.42 | Penguin | We're at 45 minutes on a 20 second task. |
| 04:01.03 | Penguin | 30 seconds if you type really slow. |
| 04:01.38 | trafalgargirl | i am so sorry |
| 04:01.46 | trafalgargirl | i don't why this isn't working |
| 04:03.50 | trafalgargirl | ok file on my desltop and it's the right one |
| 04:03.55 | trafalgargirl | openinghours.wav |
| 04:04.28 | trafalgargirl | i can repeat with the 444 file if you want |
| 04:04.49 | Penguin | This was [tk]d-fender's request, so I'll defer to him. |
| 04:08.21 | trafalgargirl | anyhing else you want me to do in the meantime? |
| 04:09.52 | Penguin | Does the file play on the PBX like you want it to now? |
| 04:11.17 | trafalgargirl | nope |
| 04:11.29 | Penguin | (2146.41) <trafalgargirl> oh we did it! |
| 04:11.29 | Penguin | (2146.45) <trafalgargirl> playing file |
| 04:11.36 | Penguin | That was kind of misleading, then. |
| 04:12.01 | trafalgargirl | sorry that was about me moving the file to host1 then to my desktop and playing it there |
| 04:12.04 | trafalgargirl | my apolgis |
| 04:12.07 | trafalgargirl | apologies |
| 04:12.21 | Penguin | The file plays fine on your computer? |
| 04:12.57 | trafalgargirl | yes - i copied the sound file from pbx ->host1->my desktop |
| 04:13.09 | trafalgargirl | and it is indeed the correct messsage |
| 04:13.47 | *** join/#asterisk Cynagen (~cynagen@ip72-208-223-60.ph.ph.cox.net) |
| 04:19.31 | trafalgargirl | http://pastebin.ca/2880772 - new test |
| 04:22.15 | *** join/#asterisk Orbixx (~orbixx@freenode/sponsor/orbixx) |
| 04:22.50 | *** join/#asterisk king1337-2 (~king1337@S0106c8fb2641d848.vs.shawcable.net) |
| 04:23.03 | trafalgargirl | something else you want me to try? |
| 04:27.51 | [TK]D-Fender | ok, now we see it playing a SLIN file. |
| 04:27.59 | [TK]D-Fender | is that a newly converted one you just brought over? |
| 04:28.03 | [TK]D-Fender | (I stepped out for a bit) |
| 04:28.31 | trafalgargirl | nope - that is the recording only |
| 04:28.48 | trafalgargirl | oops- the original recording that is stil playing *the one i am trying to replace |
| 04:38.52 | trafalgargirl | this whole thing makes no sense to me - it should be straight forward - our set up is not that diff. |
| 04:39.53 | *** join/#asterisk bkruse (~Adium@24.42.207.11) |
| 04:44.19 | *** join/#asterisk timahvo1 (~rogue@196.216.71.122) |
| 04:44.21 | trafalgargirl | TK]D-Fender should i try something else? |
| 04:57.49 | *** join/#asterisk trafalgargirl (~Adium@MTRLPQ4706W-LP130-02-1176239601.dsl.bell.ca) |
| 04:58.34 | trafalgargirl | Penguin? anything else i should try> |
| 04:58.35 | trafalgargirl | ? |
| 05:02.37 | trafalgargirl | open to anything .. |
| 05:12.22 | ChannelZ | Without scrolling back 5 pages to read the whole thing.. what is the issue? I don't see a failure in your last pastebin but obviously something is not working right that you're asking :) |
| 05:13.58 | trafalgargirl | the file that is playing when I call is the wrong file. Right name - wrong sound file. |
| 05:15.10 | ChannelZ | /var/lib/asterisk/sounds/en/mate1/openinghours.*** ? |
| 05:17.22 | Penguin | How many files in that directory have the name matching openinghours.* ? |
| 05:17.29 | trafalgargirl | just the one |
| 05:17.46 | Penguin | Show me. |
| 05:17.55 | Penguin | The last time you showed it, there were two. |
| 05:18.09 | trafalgargirl | right but then you had me rename |
| 05:18.18 | trafalgargirl | will do again if you would like? |
| 05:18.23 | Penguin | Show me that it has been done. |
| 05:18.28 | trafalgargirl | (i also restarted after that just be safe) |
| 05:18.31 | trafalgargirl | one sec |
| 05:18.34 | Penguin | ls -la /var/lib/asterisk/sounds/en/mate1/ |
| 05:21.21 | ChannelZ | Guessing either you have two files, one in the en/ and one not and you're thinking one of them should be the other, or you've got two in different formats/with different extensions and the 'wrong' one is being played |
| 05:21.27 | trafalgargirl | http://pastebin.ca/2880778 |
| 05:21.51 | trafalgargirl | that is asterisk/sounds/mate1 |
| 05:22.24 | ChannelZ | not en/mate1 ? Is Language Prefix turned off on your system? (core show settings) |
| 05:22.26 | Penguin | That's a good thought. Let's check for others: find /var/lib/asterisk -iname openinghours\* |
| 05:23.28 | ChannelZ | cuz if Language Prefix is on, it's looking in asterisk/sounds/en/mate1 and not asterisk/sounds/mate1 |
| 05:23.52 | ChannelZ | which it seems to be: Playing 'mate1/openinghours.slin' (language 'en') |
| 05:24.02 | ChannelZ | the last bit being important |
| 05:24.44 | trafalgargirl | root@pbx:/var/lib/asterisk/sounds/mate1# find /var/lib/asterisk -iname openinghours\* |
| 05:24.44 | trafalgargirl | /var/lib/asterisk/sounds/mate1/openinghours.wav |
| 05:24.44 | trafalgargirl | /var/lib/asterisk/sounds/mate1/openinghours-disabled.wav |
| 05:24.57 | Penguin | hmms |
| 05:26.31 | *** join/#asterisk fergus (~sergei@mail.55h.by) |
| 05:28.16 | trafalgargirl | default lang - |
| 05:28.17 | trafalgargirl | Default language: en |
| 05:28.39 | Penguin | <PROTECTED> |
| 05:28.42 | ChannelZ | ....and.... |
| 05:28.42 | Penguin | This? |
| 05:28.45 | ChannelZ | ^^ |
| 05:29.29 | trafalgargirl | <PROTECTED> |
| 05:29.33 | Penguin | tada |
| 05:29.44 | ChannelZ | So, like I said... |
| 05:29.55 | Penguin | But wait, there's more! |
| 05:30.23 | Penguin | 'find' doesn't know about the openinghours file(s) existing in a subdirectory of en. |
| 05:31.43 | ChannelZ | then maybe the data directory is not /var/lib/asterisk and is living elsewhere? What does 'core show settings' say the Data directory is? |
| 05:32.14 | Penguin | This seems oddly familiar. :( |
| 05:32.51 | ChannelZ | a bit |
| 05:33.31 | trafalgargirl | Data directory: /usr/share/asterisk |
| 05:33.34 | Penguin | I will have you know that all those problems got ironed out and I have been using the new CentOS/Asterisk11 box for several days now. |
| 05:33.54 | Penguin | find /usr/share/asterisk -iname openinghours\* |
| 05:34.04 | ChannelZ | so there's your problem right thaarrr. |
| 05:34.23 | Penguin | This is exactly the same problem I just went through. |
| 05:34.36 | Penguin | The cause... |
| 05:34.44 | Penguin | two different packagers. |
| 05:34.57 | ChannelZ | fucking packages, moving things all over the place |
| 05:35.08 | Penguin | packageRs |
| 05:35.15 | Penguin | The people who did it caused the problem. |
| 05:35.41 | Penguin | The official RPM had a data directory of the conventional /var/lib/asterisk. |
| 05:36.23 | Penguin | The sounds packages from EPEL were in /usr/share/asterisk. I'm assuming that was to avoid conflicts with the official package. |
| 05:37.41 | [TK]D-Fender | Penguin, I'm SSH'd in and looking at it |
| 05:37.51 | Penguin | It's all solved, so now I can sit back and laugh about it. |
| 05:38.05 | *** join/#asterisk gryphon (~gryphon@82.140.120.164) |
| 05:39.25 | trafalgargirl | except it causes errors? |
| 05:39.41 | Penguin | Nope. Totally fixed. It's in production now. |
| 05:46.28 | [TK]D-Fender | Penguin, No, not solved. Somoething is FUBAR'd BIG TIME |
| 05:46.37 | [TK]D-Fender | Penguin, I swear it's fucking with me... |
| 05:46.40 | Penguin | No, I assure you it's solved. |
| 05:46.59 | [TK]D-Fender | It will be.. it's a question of how big an aneurism I'm headed for first |
| 05:47.16 | Penguin | It seems like maybe you're talking about something other than what I'm talking about. |
| 05:47.54 | [TK]D-Fender | Penguin, Talking about sound file issue? |
| 05:48.15 | ChannelZ | Oh yay, email says my phone number port request will go through next Monday. Finally byebye to the POTS |
| 05:48.33 | Penguin | Sort of. I'm talking about how I just went through this exact same problem on my new CentOS and Asterisk 11 box. |
| 05:48.45 | Penguin | With a little help, I got it all fixed up. |
| 05:49.23 | Penguin | trafalgargirl: What distro is your asterisk on? |
| 05:51.24 | [TK]D-Fender | Debian |
| 05:51.25 | trafalgargirl | linux (debian i believe) |
| 05:52.02 | Penguin | Linux isn't a distribution, but Debian is. |
| 05:52.28 | trafalgargirl | understood. |
| 05:52.32 | [TK]D-Fender | ok, I see somehting |
| 05:52.37 | [TK]D-Fender | SNEAKY... |
| 05:52.58 | [TK]D-Fender | Actually... not 100 sure |
| 05:53.13 | Penguin | trafalgargirl: Did you ever 'find /usr/share/asterisk -iname openinghours\*' like I asked? |
| 05:53.27 | Penguin | I don't see the results if you did it. |
| 05:53.41 | [TK]D-Fender | I just saw something messed up. |
| 05:53.50 | [TK]D-Fender | I did a CP to copy the file and the TYPE CHANGED |
| 05:53.57 | [TK]D-Fender | same file size, but check this : |
| 05:54.12 | [TK]D-Fender | root@pbx:/var/lib/asterisk/sounds/mate1# file openinghours.wav |
| 05:54.12 | [TK]D-Fender | openinghours.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz |
| 05:54.12 | [TK]D-Fender | root@pbx:/var/lib/asterisk/sounds/mate1# file 444.wav |
| 05:54.12 | [TK]D-Fender | 444.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz |
| 05:54.12 | [TK]D-Fender | root@pbx:/var/lib/asterisk/sounds/mate1# |
| 05:54.22 | [TK]D-Fender | ummm.. |
| 05:54.28 | [TK]D-Fender | maybe I'm just corss-eyed |
| 05:54.28 | Penguin | It seems the same to me. |
| 05:54.38 | Penguin | Both PCM 16 bit mono 8k |
| 05:54.51 | [TK]D-Fender | -rw-r--r-- 1 asterisk asterisk 441164 Nov 28 14:39 444.wav |
| 05:55.00 | [TK]D-Fender | -rw-r--r-- 1 asterisk asterisk 441164 Dec 2 00:41 openinghours.wav |
| 05:55.12 | [TK]D-Fender | I fixed perms, but : |
| 05:55.15 | Penguin | trafalgargirl: How about that find already? |
| 05:55.26 | [TK]D-Fender | -- <IAX2/209.217.98.158:4569-5394> Playing 'mate1/openinghours.slin' (language 'en') |
| 05:55.29 | ChannelZ | didn't we establish /var/lib/asterisk is not where that should all be living (according to the config) |
| 05:55.34 | Penguin | Yes we did. |
| 05:55.35 | trafalgargirl | yes posted earlier |
| 05:55.40 | Penguin | I'm still waiting on the results. |
| 05:55.40 | [TK]D-Fender | -rw-r--r-- 1 asterisk asterisk 441164 Dec 2 00:41 openinghours.wav |
| 05:55.45 | Penguin | It was not put in this channel. |
| 05:55.47 | [TK]D-Fender | Penguin, which find? |
| 05:55.53 | Penguin | (2353.13) <Penguin> trafalgargirl: Did you ever 'find /usr/share/asterisk -iname openinghours\*' like I asked? |
| 05:55.54 | [TK]D-Fender | Thre's nothing in the en folder BTW |
| 05:55.55 | *** join/#asterisk ThatDamnRanga (~wiretap@unaffiliated/wiretap) |
| 05:56.07 | [TK]D-Fender | seems to path out properly for astvarlibdir |
| 05:56.08 | ChannelZ | <trafalgargirl> Data directory: /usr/share/asterisk |
| 05:56.35 | Penguin | Still waiting to see if a file of the same name exists in the real sounds path. |
| 05:56.44 | [TK]D-Fender | VarLib directory: /var/lib/asterisk |
| 05:56.46 | trafalgargirl | '/usr/share/asterisk/sounds/mate1/openinghours.bak' |
| 05:56.46 | trafalgargirl | '/usr/share/asterisk/sounds/mate1/openinghours.wav' |
| 05:56.52 | [TK]D-Fender | never seen "data" before... |
| 05:56.55 | Penguin | There's the file it's playing. |
| 05:57.17 | trafalgargirl | its playing the one in the share? |
| 05:57.24 | Penguin | correct |
| 05:57.27 | [TK]D-Fender | I guess so |
| 05:57.35 | Penguin | Can you tell me why you would have put the files in that path if that weren't the case? |
| 05:57.36 | [TK]D-Fender | I see most of Share is symlinks |
| 05:57.39 | [TK]D-Fender | EXCEPT THIS FOLDER |
| 05:57.45 | ChannelZ | Hurray! |
| 05:57.49 | [TK]D-Fender | but core show settings lists the varlib |
| 05:57.53 | trafalgargirl | WTF would it do that? I always thought it was in /var/lib |
| 05:57.54 | [TK]D-Fender | WTH is this "data"? |
| 05:58.04 | [TK]D-Fender | astvarlibdir |
| 05:58.13 | Penguin | You're probably using asterisk from a repo that isn't the digium repo. |
| 05:58.17 | ChannelZ | it's the root dir sounds/... gets tacked onto |
| 05:58.29 | trafalgargirl | just fyi - I did not set this up - I am new to this set up |
| 05:58.50 | [TK]D-Fender | Yes, but I've never seen * grab relate off anything but astvarlibdir which is NOT "share" |
| 05:59.02 | [TK]D-Fender | So the only thing that is listed as that is this "data" entry. Here: |
| 05:59.03 | trafalgargirl | it was set up by our old sys admin who admitedly said he knew nothing about asterisk |
| 05:59.04 | Penguin | As we determined with mine, when you're using an unofficial asteirsk build, they'll put your stuff in the unconventional path. |
| 05:59.40 | Penguin | So he probably just installed whatever asterisk was available in whatever repo he already had configured. |
| 05:59.46 | [TK]D-Fender | http://pastebin.com/ygYCMCnf |
| 05:59.49 | Penguin | It's not the end of the world. |
| 06:00.03 | [TK]D-Fender | trafalgargirl, Its' cleay using that other folder. |
| 06:00.12 | [TK]D-Fender | trafalgargirl, It's clearly using that other folder. |
| 06:00.21 | [TK]D-Fender | trafalgargirl, I'm at a loss as to why. |
| 06:00.26 | ChannelZ | well I looked at the source, and it pulls from AST_DATA_DIR.. though I've never seen it documented what varlib/datadir are definitively, what else data is used in. They are usually the same path so I never knew. |
| 06:00.27 | Penguin | I just explained why. |
| 06:01.14 | [TK]D-Fender | " astdatadir for example determines where asterisk looks for the sound files." |
| 06:01.22 | [TK]D-Fender | was sure that was varlib;... |
| 06:01.35 | [TK]D-Fender | but yeah, it is that folder pointed wrong relative I guess |
| 06:01.42 | [TK]D-Fender | news to me. |
| 06:01.54 | [TK]D-Fender | guess it was an assumed heiracy thing on my side |
| 06:01.56 | Penguin | It was compiled that way. |
| 06:02.14 | [TK]D-Fender | I've never seen ONLY that one out though |
| 06:02.33 | [TK]D-Fender | maybe the guy just did an incomplete job and I never noticed this commonality all this time |
| 06:02.49 | Penguin | I doubt he's the one who compiled it. |
| 06:03.07 | Penguin | Normal sysadmins use their package managers to install things. apt-get install asterisk |
| 06:03.12 | *** join/#asterisk caveat- (hoax@2a01:4f8:191:9111:30::10) |
| 06:03.29 | trafalgargirl | that is what i would have expected. but apparently this has been quite the 'history' with this |
| 06:11.04 | Penguin | trafalgargirl: apt-cache showpkg asterisk |
| 06:11.23 | [TK]D-Fender | Penguin, It is actually just what it says... |
| 06:11.34 | [TK]D-Fender | previous guy did a botched transplant I'm betting |
| 06:11.55 | *** join/#asterisk caveat- (hoax@2a01:4f8:191:9111:30::10) |
| 06:12.27 | ChannelZ | the host rejected the donor organ |
| 06:12.51 | Penguin | I've had that happen a couple times. |
| 06:13.48 | trafalgargirl | you guys have been awesome (and patient) |
| 06:16.23 | trafalgargirl | :) |
| 06:16.59 | Penguin | Seriously, apt-cache showpkg asterisk |
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| 06:29.58 | ChannelZ | I don't want to see your package. |
| 06:31.24 | Penguin | That's what she said? |
| 06:32.14 | trafalgargirl | oops sorry - was in the other windw |
| 06:32.15 | trafalgargirl | Package: asterisk |
| 06:32.15 | trafalgargirl | Versions: |
| 06:32.15 | trafalgargirl | 1:11.11.0~dfsg-2 |
| 06:32.32 | Penguin | pastebin the entire output |
| 06:32.47 | trafalgargirl | k |
| 06:32.56 | Penguin | There's more than just those three lines. |
| 06:33.45 | trafalgargirl | http://pastebin.ca/2880788 |
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| 06:38.36 | trafalgargirl | peguin / ChannelZ - gracias gracias gracias |
| 06:38.38 | trafalgargirl | :) |
| 06:50.17 | [TK]D-Fender | heads to bed... |
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| 08:13.31 | Jymmm | Could someone suggest a GSM gateway with FXS? |
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| 08:56.26 | kristoffelloos | Hi anyone knows the Yeastar PBX? |
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| 10:53.10 | tomodachi | how can i list settings for a in place confbridge conference in asterisk console? |
| 10:53.16 | tomodachi | or at least the state |
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| 11:55.02 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.0.1 (2014/11/20), 11.14.1 (2014/11/20), 1.8.32.1 (2014/11/20); Standard: 12.7.1 (2014/11/20); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- Read the Code of Conduct bit.ly/1hH6P22 |
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| 12:53.26 | angryuser | hello, i am searching for a pro who knows well rtpproxy |
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| 14:08.49 | tomodachi | hmm im having torubles getting my confbridge room to take any settings |
| 14:09.01 | tomodachi | the documentation mentions two ways of using settings |
| 14:09.17 | tomodachi | either adding it to confbridge.conf , allowing a specific profile |
| 14:09.44 | tomodachi | or doing it in extensions.conf directly |
| 14:10.17 | tomodachi | im trying max_members=1 in confbridge.conf, but i can still call in more tha two people to my conf room ;( |
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| 14:12.23 | [TK]D-Fender | pastebin is your friend. Show us. |
| 14:12.24 | [TK]D-Fender | ~pb |
| 14:12.25 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 14:12.26 | [TK]D-Fender | ^^^ |
| 14:12.52 | [TK]D-Fender | That means your dialplan, the configs, and the call debug showing them going in. |
| 14:13.43 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 14:16.33 | marceloamorim | guys, there is some way to detect why my sound files, when I try to play a playback-sound, are failing at the beginning of the audio? |
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| 14:20.42 | [TK]D-Fender | marceloamorim: what does ""failing at the beginning" mean? |
| 14:21.31 | marceloamorim | like, the audio seems to start but there is no audio and then come back starting the audio again |
| 14:22.47 | [TK]D-Fender | You can't have "seems to start" and then say "but there is no audio" |
| 14:23.07 | [TK]D-Fender | Does your sound file start from the BEGINNING or not? |
| 14:23.51 | [TK]D-Fender | Does a second sound file played back right after that one play in its entirety? |
| 14:24.10 | marceloamorim | I heard like 0.3secs of the audio then no audio after that and 2secs later the audio restart again |
| 14:25.32 | [TK]D-Fender | And for the next audio file? |
| 14:25.50 | marceloamorim | http://pastebin.com/KZNQCAby |
| 14:26.02 | marceloamorim | I`m trying to hit this all-circuirs-busy-now |
| 14:26.22 | marceloamorim | there is just one playback sound |
| 14:28.46 | [TK]D-Fender | I just asked you what happens when you play back another one after this |
| 14:28.52 | [TK]D-Fender | Preferrably the SAME file |
| 14:29.04 | marceloamorim | I heard the all then there is almost 2secs after that and then I heard the continuation of the sound |
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| 14:38.36 | marceloamorim | Ill post the full debug |
| 14:39.27 | marceloamorim | this is a new instalation for 13.0.1 isn`t in production yet, so its easier to get =) |
| 14:39.46 | angryuser | hello, i am searching for a person who knows well rtpproxy |
| 14:40.33 | angryuser | willing to pay for the info* |
| 14:40.39 | [TK]D-Fender | MarI've just asked you to play the same sound file back to back. Twice. |
| 14:40.45 | [TK]D-Fender | marceloamorim: I've just asked you to play the same sound file back to back. Twice. |
| 14:42.21 | marceloamorim | back to back means put another playback after this one? |
| 14:42.25 | *** join/#asterisk XATRIX (~xatrix@77.88.209.171) |
| 14:42.31 | [TK]D-Fender | [09:28][TK]D-FenderI just asked you what happens when you play back another one after this |
| 14:42.39 | [TK]D-Fender | [09:23][TK]D-FenderDoes a second sound file played back right after that one play in its entirety? |
| 14:43.08 | XATRIX | Hi, how can i find out what channel for SIP do i have ? I tried 'sip show channels' - it says i have 0 , but 'sip show registry' shows i have one trunk |
| 14:43.24 | XATRIX | Sorry, i'm new to PBX |
| 14:44.07 | WIMPy | "channels" is active channels. |
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| 14:44.48 | marceloamorim | yeah [TK]D-Fender, if I put the second playback and a third one, all those play correctly |
| 14:45.28 | XATRIX | WIMPy: em... it means active calls ? |
| 14:45.30 | marceloamorim | I set Playback(all-circuits-busy-now), next line the same Playback(all-circuits-busy-now) and a third with playback(vm-isunavail) |
| 14:45.35 | [TK]D-Fender | marceloamorim: Then the issue is that your audio path isn't getting set up very fast. Go play something like 2 seconds of silence before you actually start playing back that first file you wanted. |
| 14:46.04 | [TK]D-Fender | XATRIX: Yes |
| 14:46.12 | WIMPy | XATRIX: yes |
| 14:46.35 | marceloamorim | ok, but why there is delay to go there? |
| 14:47.03 | [TK]D-Fender | becasue whatever you are communicating with doesn't set up very quickly. |
| 14:47.14 | marceloamorim | well, is the same machine =( |
| 14:47.43 | marceloamorim | /etc/asterisk/extension.conf to /var/lib/asterisk/sounds/en/files |
| 14:47.50 | [TK]D-Fender | That means nothing. |
| 14:47.59 | [TK]D-Fender | this has NOTHING to do with your sound files |
| 14:48.13 | [TK]D-Fender | your CHANNEL does not set up the audio path very quickly. |
| 14:49.23 | marceloamorim | do you know why? or this isn`t important? |
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| 14:50.05 | [TK]D-Fender | MarI don't even know what kind of channel you are using, or any of the process it goes through. |
| 14:50.55 | marceloamorim | ok then, ty |
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| 15:00.35 | XATRIX | Ok, but how can i find a channel to set in "call-file" ? "Channel: SIP/$number@????????" |
| 15:01.07 | XATRIX | What is my actuall sip-provider name for ? |
| 15:01.52 | [TK]D-Fender | XATRIX: You put whatever you would put in a DIal() to actually call out |
| 15:02.26 | XATRIX | It's my first time i do this. I never tried |
| 15:02.42 | XATRIX | I have only a register string |
| 15:04.25 | [TK]D-Fender | You shouldn't be doing call files if you don't even understand your dialplan basics yet |
| 15:04.47 | [TK]D-Fender | ~book |
| 15:04.47 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 15:05.00 | XATRIX | Yes, but i'm just experimenting. Ok, let's go find some info about this |
| 15:05.01 | *** join/#asterisk u0m3 (~u0m3@92.80.68.195) |
| 15:05.21 | [TK]D-Fender | And you should have a a peer created to use in dialing whatever it is you are registering to. |
| 15:06.41 | XATRIX | Yes, i have a peer |
| 15:07.10 | [TK]D-Fender | [10:02]XATRIXI have only a register string |
| 15:08.04 | XATRIX | Em.. looks like i have 6 peers |
| 15:08.26 | XATRIX | 5 local phones extensions, and one from my provider |
| 15:12.25 | tomodachi | im having problems getting confbridge settings to stick , any settings actually |
| 15:12.35 | tomodachi | as an example i tried to set max_members=1 |
| 15:12.43 | tomodachi | put i can call in 2 people whatever i try |
| 15:12.48 | tomodachi | http://pastebin.com/PK1TjKLL |
| 15:13.17 | tomodachi | its a fairly short txt file showing the extension part for the confbridge application , along with confbridge.conf and the output from a call in the asterisk console |
| 15:13.24 | [TK]D-Fender | [Dec 2 16:14:33] ERROR[9305]: pbx.c:3978 ast_func_write: Function CONFBRIDGE not registered |
| 15:13.25 | [TK]D-Fender | ^ |
| 15:13.39 | [TK]D-Fender | perhaps you'll have noticed taht none of those SET's you did actually worked |
| 15:14.03 | tomodachi | [TK]D-Fender: yeah , i tried to google what that actually meant (not registered) |
| 15:14.11 | [TK]D-Fender | "Does not exist" |
| 15:14.32 | tomodachi | ok so im basically failing to understand the syntax? |
| 15:15.00 | tomodachi | or using bad / outdated sources for it |
| 15:15.24 | tomodachi | <PROTECTED> |
| 15:15.30 | tomodachi | http://www.voip-info.org/wiki/view/Asterisk+cmd+ConfBridge |
| 15:15.34 | WIMPy | Or not loading the module. |
| 15:15.37 | [TK]D-Fender | ^ |
| 15:15.47 | tomodachi | WIMPy: but the conference works so i presume that confbridge is present |
| 15:16.21 | tomodachi | hmm the wiki mentions confbridge 10 , maybe im on another version |
| 15:16.38 | [TK]D-Fender | tomodachi: It would be good if you knoew what you were running |
| 15:16.48 | tomodachi | :) i know what asterisk version |
| 15:16.53 | WIMPy | Are you using an old Asterisk version with the proof-of-concept prerelease version of ConfBridge? |
| 15:16.55 | [TK]D-Fender | Which is? |
| 15:17.02 | [TK]D-Fender | 1.8 does NOT have that function |
| 15:17.07 | tomodachi | Asterisk 1.8.13.1~dfsg1-3+deb7u3 |
| 15:17.11 | [TK]D-Fender | There you go |
| 15:17.13 | tomodachi | debian stable |
| 15:17.18 | [TK]D-Fender | No good |
| 15:17.21 | [TK]D-Fender | (for that function |
| 15:17.31 | WIMPy | That's too old. |
| 15:18.03 | WIMPy | The current stable is 13. Or if you want old stuff, you could use 11, but don;t use anything older. |
| 15:18.04 | tomodachi | to old :) i just switched from 1.4 like a yearago :p |
| 15:18.27 | WIMPy | That was current 6 years ago or so. |
| 15:18.28 | [TK]D-Fender | tomodachi: Bad move |
| 15:18.55 | tomodachi | so how do you recomend , finding a repo , or from src?! |
| 15:19.07 | [TK]D-Fender | tomodachi: Try not to be staying 4 branches behind current |
| 15:19.18 | [TK]D-Fender | tomodachi: Doesn't matter how you install it. |
| 15:19.33 | [TK]D-Fender | tomodachi: As long as you can maintain it |
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| 15:20.19 | tomodachi | do you think i will have any interpoerability issues with using 13 on one and 1.6 on the other pbx (we have 2 in different offices) |
| 15:21.27 | [TK]D-Fender | Shouldn't |
| 15:21.30 | babak | Hi, performance capacity of Asterisk increase 1.8 vs 11 vs 13 ? |
| 15:21.50 | [TK]D-Fender | and by that I mean "no" and "you shouldn't be using 1.6 either" |
| 15:21.58 | tomodachi | ok thanks for your help guys |
| 15:22.16 | [TK]D-Fender | babak: Generally a little better every time |
| 15:22.22 | WIMPy | Definitely not. The 1.6 versions haven;t been that good IMHO. |
| 15:22.30 | [TK]D-Fender | And are no longer supported |
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| 15:29.45 | babak | I want to use Asterisk as costume ring back tone player, Asterisk has 8or 16E1 just plays custome music as ring back tone and go our of path by ECT(2BCT) average music time is 12 seconds so CAPS is high for 8E1=20 caps 16E1=40CAPS is it possible ? what version you recommend ? |
| 15:30.56 | WIMPy | babak: Have you tried the concept? |
| 15:31.30 | babak | yes |
| 15:31.46 | WIMPy | ok, just asking. |
| 15:31.47 | babak | one call ok |
| 15:32.43 | babak | but for high capacity not sure |
| 15:33.34 | WIMPy | As long as you can handle the IRQs, the rest should be easy going. |
| 15:35.06 | babak | I read many improvement in threading in new versions so expect increase in CAPS |
| 15:35.39 | WIMPy | Won't help you with dahdi. |
| 15:36.20 | WIMPy | But then the bandwidth still isn't that much for modern hardware. |
| 15:36.34 | WIMPy | It's all down to IRQ handling. |
| 15:38.41 | babak | WIMPy: you recomend 2 cheaper server each 4E1 or one good server with 8E1 ? |
| 15:39.56 | WIMPy | I'd hope that the number of interrupts depents on the number of cards, not the number of channels. |
| 15:40.17 | WIMPy | But make sure you don't have any SWEC running. That might be lethal. |
| 15:40.33 | *** join/#asterisk CustosL1men (~CustosLim@unaffiliated/cust0slim3n) |
| 15:40.50 | [TK]D-Fender | Don't forget to transcode everything to G.729 while you're at it! |
| 15:41.04 | [TK]D-Fender | #earthshatteringkaboom |
| 15:41.17 | *** join/#asterisk eject_ck (~Adium@62.159.86.18) |
| 15:41.22 | babak | no it is fixed line and prefer G.711 |
| 15:41.40 | eject_ck | What's best place to find DID provider from exotic country ? |
| 15:41.41 | WIMPy | Indeed. Make sure you have your files in the the right format. |
| 15:41.52 | [TK]D-Fender | eject_ckWhat's best place to find DID provider from exotic country ? <- from that country |
| 15:43.52 | babak | WIMPy: yes music saved in G.711 , increasing RAM and saving all musics in RAM will help ? |
| 15:45.12 | WIMPy | Yes, it will help, but I'm sure yo'd have a hard time finding a harddrie that couldn't deliver 2MB/s even from random locations. |
| 15:45.44 | eject_ck | [TK]D-Fender: in need get DID for Virgin Islands British |
| 15:46.03 | eject_ck | can't find any DID provider |
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| 16:00.22 | dft | elastix or freepbx that is the question |
| 16:00.45 | dft | currently running trixbox ce and trying to decide on what to migrate to |
| 16:00.56 | WIMPy | ~elastix |
| 16:00.56 | infobot | hmm... elastix is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes Elastix VERY difficult to support, and is not supported in #asterisk. Try asking in #Elastix or on their forums at http://www.elastix.org |
| 16:00.58 | WIMPy | ~freepbx |
| 16:00.58 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 16:01.03 | WIMPy | There's your choice. |
| 16:01.54 | dft | WIMPy: lol, thank yo |
| 16:01.55 | dft | u |
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| 16:30.27 | WIMPy | ipengineer: Was it you who had the issue with los digits? |
| 16:30.47 | ipengineer | WIMPy: Not that I can recall |
| 16:31.08 | WIMPy | With WaitExte or Background? |
| 16:31.22 | ipengineer | No I dont think that was me |
| 16:31.23 | WIMPy | +n |
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| 16:32.06 | WIMPy | Ok, bad luck. |
| 16:32.44 | WIMPy | If the one with that issue is reading, give me a ping. |
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| 16:42.16 | Kobaz | anyone know where can i get a list of up to date area codes and city/locations ? |
| 16:42.54 | WIMPy | From the regulator for the country you're searching them for. |
| 16:43.17 | Kobaz | soooo |
| 16:43.24 | Kobaz | in the US that would be? |
| 16:43.56 | WIMPy | The FCC, I guess. |
| 16:44.06 | Kobaz | mm, makes sense |
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| 17:19.19 | pabelanger | Is there actually a way to delete a channel variable in asterisk? Aside from setting it to null |
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| 18:51.22 | marceloamorim | [TK]D-Fender: do you remember about the delay from audio early ? |
| 18:52.04 | marceloamorim | I get the same problem with transfer audio, I press the button to transfer and there is a delay |
| 18:53.33 | marceloamorim | and I`m using sip to sip |
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| 19:01.28 | Milenco | Hmm, is there any alternative to busy-level |
| 19:01.57 | Milenco | I want to restrict incoming calls to 1 for callcenter phones, but call-limit set to 1 means they cant transfer |
| 19:02.25 | bluesoul|work | so i've got asterisk in a VM that was provisioned with an overly small disk. i've expanded the disk in ESXi but FreePBX is still showing 95% full on disk / |
| 19:02.37 | Milenco | i know i can use busy-level but i need to adjust my dialplan heavily to check the devicestate of each sip-account in order for it to work |
| 19:02.41 | bluesoul|work | i don't know if it's a linux quirk or an asterisk quirk |
| 19:03.04 | Milenco | how much MB's are left and how much do you log? |
| 19:03.59 | bluesoul|work | verbose logging is on, i'm not seeing where to determine free space |
| 19:04.15 | [TK]D-Fender | This isn't an * |
| 19:04.17 | [TK]D-Fender | <PROTECTED> |
| 19:04.30 | [TK]D-Fender | And just because you make a disk bigger doesn't mak the partition bigger |
| 19:04.36 | bluesoul|work | yea i didn't really think so but it's so easy to join #asterisk hehe |
| 19:04.40 | [TK]D-Fender | this is also linux basics |
| 19:04.42 | [TK]D-Fender | df |
| 19:04.43 | [TK]D-Fender | ^ |
| 19:04.46 | Milenco | true :) |
| 19:05.14 | WIMPy | And just because you make the partition bigger doesn't make the filesystem bigger. |
| 19:07.20 | Milenco | is there an option for asterisk to not dial a sip account when device state is busy? |
| 19:08.00 | [TK]D-Fender | Busy is a result, not a state |
| 19:08.22 | [TK]D-Fender | You could always check if it's doing ANYTHING at all and consider taht "enough". Or keep a counter |
| 19:08.49 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 19:08.53 | Milenco | well, im setting busylevel to 1, and when i try to get the device state after one call it returns busy |
| 19:09.18 | Milenco | i could easily extend my extensions so it wont dial if the state is busy, but im dialing a group of phones |
| 19:09.48 | WIMPy | That's theway it is. |
| 19:10.16 | Milenco | too bad :( |
| 19:10.32 | Milenco | seems like my phones cant send back busy either automatically |
| 19:11.05 | WIMPy | What kind of phone can't do that? |
| 19:11.43 | Milenco | got Grandstream phones here |
| 19:11.51 | Milenco | all GXP2140's, pretty shiny and nice ones |
| 19:12.17 | WIMPy | shiny on the outside? |
| 19:12.23 | Milenco | when they get a call and answer that, they still get offered new calls (because call-limit is >1) |
| 19:12.28 | bluesoul|work | literally made of gold |
| 19:12.30 | Milenco | :D |
| 19:13.46 | Milenco | of course i can press the deny button on the phone once it rings, but i rather not send new calls to the phone if its already handling a call |
| 19:13.54 | Milenco | call-limit 1 would do that, but that also disable tranfers |
| 19:14.24 | WIMPy | Check the state(s) before dialling. |
| 19:14.41 | WIMPy | Or get phones that have the option to disable call waiting. |
| 19:15.04 | zaf | Milenco, you probably want to use queues |
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| 19:15.12 | Milenco | yes zaf! :) |
| 19:15.14 | Milenco | we really do |
| 19:15.25 | Milenco | but im forced into a rushed deployment for a remote callcenter |
| 19:15.30 | Milenco | management gone wild :( |
| 19:15.39 | zaf | Milenco, i set ours up yesterday within about an hour |
| 19:15.45 | Milenco | yes queues are easy |
| 19:15.57 | Milenco | but the control needs to be integrated into our current systems |
| 19:16.18 | zaf | and just set it up as ringall so it would behave the way they're used to |
| 19:16.28 | Milenco | agents logging on/assigning usernames to sip-accounts, agents need to pause themselves, managers need to put people in queues based on rosters, etc. |
| 19:16.52 | Milenco | hmm |
| 19:16.58 | Milenco | thanks a pretty good idea zaf, thanks! |
| 19:17.15 | Milenco | just use queues with static members |
| 19:17.26 | Milenco | would be the same as dial() basicly but with better handling |
| 19:17.36 | zaf | yep |
| 19:17.47 | Milenco | thanks for the tip, i'll take that route :) |
| 19:17.58 | WIMPy | That's a way to do it. |
| 19:18.06 | Milenco | thanks again :) |
| 19:18.07 | WIMPy | If you don't have too many groups :-) |
| 19:18.54 | Milenco | nah, only 5 incoming lines with need queues |
| 19:18.58 | Milenco | and about 40-50 phones |
| 19:19.06 | Milenco | should be easy enough :) |
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| 19:23.02 | marceloamorim | [TK]D-Fender: lol , the problem is the hostname of the machine |
| 19:23.03 | marceloamorim | o.O |
| 19:23.15 | marceloamorim | sorry to bother you early man |
| 19:23.57 | Milenco | oke, even faster to fix it...disable call waiting |
| 19:24.01 | Milenco | thanks WIMPy |
| 19:24.06 | Milenco | that was the term i was looking for |
| 19:24.15 | Milenco | its right there and i looked over it the whole time.. :P |
| 19:28.22 | WIMPy | hah |
| 19:28.34 | WIMPy | That sounds like less work :-) |
| 19:28.46 | Milenco | indeed, especially since we provision the config on these phones |
| 19:28.58 | Milenco | so im already editing the prov-database :) |
| 19:29.14 | WIMPy | I'd surely hope so if you have many of them. |
| 19:29.23 | *** join/#asterisk e4voip (uid13742@gateway/web/irccloud.com/x-rlgyacdkhpinxtql) |
| 19:29.26 | Milenco | yes, but only recently |
| 19:29.36 | Milenco | we been expanded dramatically |
| 19:29.45 | Milenco | 2 years ago we had 40 phones in total |
| 19:29.50 | Milenco | now we manage about 400 |
| 19:30.10 | WIMPy | I wouldn't manage 40 manually. |
| 19:30.38 | Milenco | true, but then we didnt do much with them and developing a provision tool would have cost too much time |
| 19:30.41 | Milenco | but in hindsight.. |
| 19:30.59 | Milenco | i'm glad i finally got approval for my windows server proposal |
| 19:31.14 | WIMPy | Well, I don't know why some people still think using VOIP is cheap. |
| 19:31.15 | Milenco | last month i had to update virusscanner license on 250 pc's :(( |
| 19:32.09 | WIMPy | Luckily I'm not involved in anything Windows. |
| 19:32.12 | Milenco | well voip is a lot cheaper for us making outgoing calls |
| 19:32.36 | Milenco | just 2 more weeks for my holiday.. :) |
| 19:32.46 | Milenco | almost going crazy all the stuff i need to take care of |
| 19:32.46 | WIMPy | Not even that is true here. |
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| 19:33.07 | Milenco | we pay around 1-2 for outgoing calls |
| 19:33.14 | Milenco | mobile around 3-5 cent p/m |
| 19:33.39 | Milenco | landlines would be 7-8c for normal lines, 10-15c p/m mobile |
| 19:34.00 | WIMPy | That's what you can get fromthe chinese ITSPs, but on a real line you can get mobile calls for 0.7-2¢. |
| 19:34.27 | Milenco | thats amazingly cheap! :) |
| 19:34.41 | WIMPy | It is. |
| 19:34.43 | Milenco | what are the startup costs and monthly fees? |
| 19:35.05 | WIMPy | I was very astounded after I hadn't checked it for a while. |
| 19:35.11 | *** join/#asterisk yokel (~yokel@unaffiliated/contempt) |
| 19:35.12 | WIMPy | none |
| 19:36.00 | Milenco | so its just like prefix-dialing? |
| 19:36.14 | WIMPy | yes |
| 19:36.18 | Milenco | we use some cheap fax providers which get dialed with a certain prefix |
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| 19:37.01 | Milenco | i should notify some people then, to look into that again |
| 19:37.13 | bluesoul|work | WIMPy resizing a hot partition is terrifying what the hell |
| 19:37.39 | Milenco | ext2 should be able to extend without issues |
| 19:37.47 | bluesoul|work | this is ext4 |
| 19:37.49 | Milenco | but on a mounted fs that might be difficult indeed |
| 19:37.56 | bluesoul|work | and it's the root partition |
| 19:38.06 | Milenco | ext4 should be able to extend while mounted i think |
| 19:38.14 | WIMPy | IIRC you need ext3 (or ext4) to do it online. |
| 19:38.20 | bluesoul|work | asterisk didn't do lvs |
| 19:38.23 | bluesoul|work | so eh |
| 19:38.50 | Milenco | http://askubuntu.com/questions/24027/how-can-i-resize-an-ext-root-partition-at-runtime |
| 19:39.12 | bluesoul|work | ah hell lol |
| 19:39.23 | bluesoul|work | i'm weighing just taking a backup and reinstalling |
| 19:39.45 | Milenco | resize2fs should be able to resize mounted root partitions on ext4 |
| 19:39.58 | bluesoul|work | it said "nothing to do" |
| 19:40.09 | Milenco | you should always backup when doing these things anyway :) |
| 19:40.20 | Milenco | "You can use fdisk to delete the root partion and then recreate it at the same starting block. fdisk will write out the change, but it won't take effect till after a reboot. after the reboot you can use the resize2fs program to send the disk to fill the partion. â Jbecwar Jun 14 '12 at 15:15" |
| 19:40.29 | WIMPy | You should always backup. |
| 19:40.36 | Milenco | indeed |
| 19:40.38 | bluesoul|work | i took a snapshot |
| 19:40.49 | Milenco | but especially when dealing with these tricky things |
| 19:42.08 | Milenco | if you have any partitions after your root partition then it will become more difficult |
| 19:53.46 | *** join/#asterisk talntid (~talntid@173-160-189-58-Washington.hfc.comcastbusiness.net) |
| 19:54.04 | talntid | is it trivial to log to a different database in asterisk? |
| 19:54.21 | [TK]D-Fender | Pick another driver. Set it up |
| 19:54.32 | talntid | there isn't currently a driver for what I am looking for |
| 19:54.52 | [TK]D-Fender | Namely? |
| 19:54.55 | talntid | Firebase |
| 19:55.41 | [TK]D-Fender | Firebird? |
| 19:55.47 | talntid | firebase |
| 19:55.59 | talntid | https://www.firebase.com/ |
| 19:56.14 | *** part/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 19:57.07 | [TK]D-Fender | https://www.google.ca/#q=JSON+unixodbc |
| 19:57.09 | [TK]D-Fender | ^^ |
| 19:57.27 | [TK]D-Fender | firebase isn't an engine, just an API from what they say |
| 19:57.34 | *** join/#asterisk MadHatter42 (~MadHatter@unaffiliated/madhatter42) |
| 19:58.06 | talntid | right, it is just an API. I can make the software to send the data to firebase, if I can figure out where to hook into * |
| 19:59.11 | [TK]D-Fender | ODBC <---------------- |
| 19:59.16 | talntid | roger that |
| 19:59.29 | [TK]D-Fender | That was implicit in the search I gave you |
| 20:02.53 | talntid | hm. writing a ODBC driver is not even remotely trivial. I will do this another way. thanks. |
| 20:03.27 | [TK]D-Fender | Who said you'd have to write one? |
| 20:04.42 | talntid | well I don't see anything that directly fits the bill |
| 20:06.16 | [TK]D-Fender | You might want to spend more than 2 minutes looking |
| 20:07.33 | bluesoul|work | i wussed out and used partedmagic |
| 20:07.37 | bluesoul|work | yaaaay |
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| 20:15.20 | Mango45 | I have written a small dial plan that uses WaitForNoise and WaitForSilence to play prompts to a caller, and record their responses. This works, except when a call is terminated and my Asterisk doesn't receive the BYE. |
| 20:15.55 | Mango45 | In that case, Asterisk keeps playing the prompts over and over, sending RTP to a nonexistant peer. |
| 20:16.08 | Mango45 | WaitForNoise curiously detects noise, even though there is no RTP arriving. |
| 20:16.39 | Mango45 | Does anyone have any suggestions? |
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| 20:16.50 | WIMPy | Apart from fixing your network issue, you may want to look at session-timers and rtptimeout. |
| 20:17.27 | Mango45 | rtptimeout sounds like just what I need. |
| 20:17.29 | Mango45 | Thanks. |
| 20:18.32 | [TK]D-Fender | What prompts? |
| 20:18.42 | [TK]D-Fender | Neither of those apps seem to involve prompts... |
| 20:19.34 | Mango45 | Playback is used for the prompts., WaitForNoise / WaitForSilence is used to detect when the caller has answered the question. |
| 20:19.49 | Mango45 | I'm still mildly curious that WaitForNoise fires, even though there is no incoming RTP. |
| 20:22.08 | [TK]D-Fender | well those are your prompts and if they keep goin around in circles then its your fault for not putting a loop counter on them to limit them |
| 20:22.31 | bluesoul|work | hilariously it looks like the problem wasn't because the drive was nearly full, but because the address of the DNS server changed and i needed to edit resolver.conf |
| 20:22.38 | bluesoul|work | oh well, i'm being proactive right? |
| 20:22.43 | Mango45 | It's intended to loop...if the caller keeps speaking. |
| 20:23.34 | [TK]D-Fender | perhaps you need to trim your waitfonoise ration |
| 20:23.42 | [TK]D-Fender | have you tried to spy on one taht's stuck? |
| 20:25.27 | Mango45 | as in, listen to the audio? |
| 20:26.26 | [TK]D-Fender | yes |
| 20:27.42 | Mango45 | Looks like the -in file ends at the moment the RTP stops. |
| 20:28.22 | Mango45 | Although I did just prove that rtptimeout works perfectly. Any side effects to using that? |
| 20:29.17 | [TK]D-Fender | https://www.youtube.com/watch?v=P1xdGCvNMEY&feature=player_detailpage#t=89 |
| 20:29.43 | Mango45 | grins |
| 20:31.00 | Mango45 | I genuinely hope the server won't explode |
| 20:31.50 | WIMPy | Put some popcorn on it, just in case. |
| 20:32.02 | Mango45 | :P |
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| 20:37.27 | Mango45 | Excellent, rtptimeout does not time out when audio is traveling directly between endpoints (i.e. when Asterisk is not supposed to be receiving rtp.) |
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| 21:40.27 | Dovid | Hi all. I have a phone number that plays white noise. When call via the PSTN or direct on SIP work as well. The issue seems to be with Mobile. I am testing in the US on Verizon Mobile and ATT and the audio randomly has hissing sounds and breaks in the audio. is this simply a limit of mobile devices and mobile networks? |
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| 22:05.39 | krapper | hello... trying to get SIPAddHeader for custom tagging on inbound calls from SIP provider... told to put in SIP 200 OK... any ideas? I have no problem setting the custom tag on outbound calls by putting SIPAddHeader line before Dial line. |
| 22:11.11 | zaf | Dovid, you mean it plays white noise on purpose? |
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| 22:26.51 | [TK]D-Fender | krapper, " told to put in SIP 200 OK" <- not going to happen |
| 22:27.00 | pigpen | #hamradio |
| 22:27.08 | [TK]D-Fender | krapper, that app only works on the invite sent out in a subsequent dial |
| 22:28.46 | Chainsaw | pigpen: ? |
| 22:28.57 | pigpen | yeah, forgot /join ;-) |
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| 22:31.50 | Dovid | Zaf: it's to help people fall asleep. |
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| 22:33.46 | danielbk | good evening everyone. I hope you might be able to ask. I'm using freepbx/asterisk 12.5 and I have the lifetime srtp issue where calls drop due to inability to accept lifetime srtp. I have browsed online and found various patches and instructions on editing .c files but really do not know how to do it. I would be very thankful if anyone can help |
| 22:36.48 | danielbk | (i.e to reach the point where I can use ignorecryptolifetime=yes) |
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| 22:40.55 | zaf | Dovid, yeah, sounds to me like the frequency ranges of the mobile codecs don't handle those of your white noise |
| 22:45.35 | Chainsaw | Just play some violins like your average hold music. |
| 22:45.42 | Chainsaw | Mobiles will turn that into white noise quite effectively. |
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| 23:22.48 | f0ner00t | Anybody know why I am getting this error when setting up google voice [2014-12-02 15:10:42] WARNING[10066]: res_xmpp.c:3573 xmpp_client_receive: Parsing failure: Hook returned an error. |
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