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| 00:52.45 | qakhan | hi all. is there any way to get active call duration in dial plan or php script |
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| 02:27.35 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: Asterisk 11.10.2 (2014/06/13), 1.8.28.2 (2014/06/13); Standard: Asterisk 12.3.2 (2014/06/13); DAHDI: DAHDI-linux 2.9.0 (2014/01/30), DAHDI-tools 2.9.0.1 (2014/01/31); libpri 1.4.15 (2014/06/16) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- Read the Code of Conduct bit.ly/1hH6P22 |
| 02:27.41 | kunwon1 | i'm using asterisk 12.3.2, i have security => security in logger.conf, but the file doesn't get logged to. in "logger show channels" i see /var/log/asterisk/security but it has no configuration listed. has the security loglevel been deprecated or something? |
| 07:40.40 | *** join/#asterisk infobot (ibot@rikers.org) |
| 07:40.40 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: Asterisk 11.10.2 (2014/06/13), 1.8.28.2 (2014/06/13); Standard: Asterisk 12.3.2 (2014/06/13); DAHDI: DAHDI-linux 2.9.0 (2014/01/30), DAHDI-tools 2.9.0.1 (2014/01/31); 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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| 07:52.50 | puzzled | morning |
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| 08:32.21 | michael_work | gavimobile, seems like extension is unreachable or the dialstring is wrong one |
| 08:33.11 | gavimobile | michael_work: I think it has sopmething to do with host=dynamic |
| 08:33.16 | gavimobile | it checks to see if the user is registered |
| 08:33.34 | gavimobile | if not registered, than the error is reported |
| 08:33.39 | gavimobile | but everything works fine |
| 08:33.55 | michael_work | by not registered you mean unrechable, right :P |
| 08:34.22 | gavimobile | yap |
| 08:34.56 | gavimobile | I think its safe if I ignore it |
| 08:35.05 | gavimobile | even though I hate ignoring shyt |
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| 08:43.04 | BarthezZ | Hmm... I'm having a channel which is forwarded multiple times and finally turns to zombie on asterisk 1.8, so I'm trying to trace the actual events... But I'm looking at the CEL logs right now, but what does consistently uniquely identifies the channel? the linkedId or the uniqueId? |
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| 11:53.31 | Stefan27 | what kind of snom-phone-settings makes the snom-phone respond to asterisk's invites with "403 Use Proxy" |
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| 12:15.56 | qakhan | hi all. is there any way to get active call duration in dial plan or php script |
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| 12:26.42 | [TK]D-Fender | qakhan: "core show function CDR" |
| 12:27.00 | [TK]D-Fender | qakhan: "core show channels concise" |
| 12:27.21 | [TK]D-Fender | qakhan: Some AMI commands. |
| 12:27.26 | [TK]D-Fender | qakhan: And other stuff |
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| 12:30.41 | qakhan | [TK]D-Fender how to get current call duration in AMI> |
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| 12:30.56 | [TK]D-Fender | qakhan: Go look at the command list |
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| 12:32.57 | Apteryx | Hello. Is Google Voice for 3rd party (such as an Asterisk PBX) not working at this time? I read they were supposed to cease offering third party connections but nothing suggests so in the Asterisk docs. |
| 12:33.40 | file | We don't document that, but it may or may not be continuing to work. Who knows how long it will stay like that. |
| 12:34.03 | Apteryx | @file: ok, thanks |
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| 12:36.10 | Apteryx | Anyone would know of a VoIP trunk service provider offering TLS/SRTP support? I am looking at voip.ms right now but they don't do that. |
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| 12:43.53 | [TK]D-Fender | Apteryx: Who else have you looked at? |
| 12:48.09 | Apteryx | [TK]D-Fender: FreePhoneLine.ca |
| 12:48.18 | Apteryx | They also don't offer TLS. |
| 12:48.23 | [TK]D-Fender | Apteryx: And that's all? |
| 12:48.43 | [TK]D-Fender | Don't expect great features from a "free" service like that |
| 12:48.47 | Apteryx | Although I must say these are really budget providers ;) |
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| 12:50.12 | Apteryx | What happens to my peer secret when I connect using an unencrypted TCP connection? Is the password at least hashed or its tranmitted in clear? |
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| 12:52.11 | [TK]D-Fender | hashed |
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| 13:15.01 | Apteryx | [TK]D-Fender: OK, thanks :) |
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| 13:32.17 | marceloamorim | guys, anyone knows if there is any kind of problem at the asterisk 11.6-cert2 on debian 7.4 64bits? |
| 13:33.20 | file | can you be more specific? |
| 13:33.22 | WIMPy | issues.asterisk.org |
| 13:33.42 | marceloamorim | the asterisk just crash, I join on CLI, but there is no response, I need to kill the process and then start again |
| 13:34.05 | marceloamorim | the problem is, I didn't find any kind on /var/log/asterisk/full yet |
| 13:35.39 | MaliutaLap | marceloamorim: not as far as I know, and I have 2 instances running on separate machines |
| 13:36.51 | marceloamorim | I had a problem when I install asterisk and load the modules of mysql, I can't start asterisk after the mysql |
| 13:37.12 | marceloamorim | but I just remove those modules because I couldn't fix |
| 13:37.40 | marceloamorim | but just that, and I don't know if there is any connection with those problems |
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| 14:19.58 | FuriousGeorge | hey all |
| 14:20.36 | FuriousGeorge | is there any way to prevent dahdi from building pciradio.o? it keeps failing |
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| 14:51.13 | Stefan27 | If I use externhost="xyz" and externrefresh=10 in sip.conf and my asterisk is registered at some SIP-provider Y. will asterisk re-register itself with Y automatically whenever the dns-look-up for xyz changes value? |
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| 15:00.57 | Stefan27 | from code looks like network_change_sched_cb would do something like that, but is that really called if the dns-look-up that happens every 10 seconds due to externrefresh=10 has a different value than last lookup? I dont have the tools atm to test this |
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| 15:05.25 | _omer | Question related to ConfBridge: Users join the conference and record their names. Is this possible to play all the names at once when "admin" joins the conference. Conference is already marked to start when admin joins (wait_marked). |
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| 15:46.39 | lvlinux | I need some help with echo cancellation---I am trying to use OSLEC, and have it set to echocancel=256 in chan_dahdi.conf, but the client is still complaining of echo. What should I try next? Sangoma PSTN card BTW. (B600) |
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| 16:08.51 | WIMPy | _omer: Sure. But not automagic. |
| 16:09.02 | _omer | any clue ? |
| 16:09.22 | WIMPy | Dialplan. As usual. |
| 16:09.36 | WIMPy | You have to do it all manually. |
| 16:10.04 | _omer | I have no clue to broadcast a file to all the users in a conference. |
| 16:10.58 | WIMPy | originate a call to the conference that will playback the file. |
| 16:11.10 | WIMPy | But that was another question? |
| 16:12.20 | _omer | well. I can originate a call to the conference but beore that I need to mix all Users Names sounds files into one file? |
| 16:12.32 | _omer | or someting like that |
| 16:16.07 | WIMPy | Or just play them one at a time. |
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| 16:19.21 | _omer | yes. that's a bit tricky |
| 16:19.28 | [TK]D-Fender | WIMPy: where would he get those to be able to play them back? I suppose this would have to be recorded before confbridge.... |
| 16:21.00 | Qwell | Katty: I may make your latest recipe thingie. |
| 16:21.53 | _omer | [TK]D-Fender: Yes. Files need to be mixed before Admin Joins ....but.... how would I get admin's name recorded |
| 16:22.22 | [TK]D-Fender | _omer: You'd have to record them OUTSIDE of confbridge |
| 16:22.46 | _omer | then how to play to all the users |
| 16:23.26 | [TK]D-Fender | Same thing WIMPy said. Originate a call into the conference and PLAYBE them |
| 16:23.30 | [TK]D-Fender | PLAYBACK |
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| 16:31.36 | nix8n82 | Anyone know of an active and up to date AGI and AMI library for python? |
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| 17:09.17 | ChkDigit | I have an extension call a macro, which figures out how to best route a call to someone. One way is to call ParkAndAnnounce(), but when this happens, and times out instead of returning to priority+1 it returns to priority. Also, putting anything in the return context appears to be ignored. Am I doing something wrong? |
| 17:11.25 | ChkDigit | Asterisk 11.7.0... not 12. |
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| 17:29.49 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: Asterisk 11.10.2 (2014/06/13), 1.8.28.2 (2014/06/13); Standard: Asterisk 12.3.2 (2014/06/13); DAHDI: DAHDI-linux 2.9.0 (2014/01/30), DAHDI-tools 2.9.0.1 (2014/01/31); libpri 1.4.15 (2014/06/16) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- Read the Code of Conduct bit.ly/1hH6P22 |
| 17:29.49 | Katty | Qwell: the corn chowder thingy? |
| 17:29.57 | Qwell | Katty: yar |
| 17:30.05 | Katty | it was tasty (= |
| 17:30.16 | Katty | may i highly recommend velveeta |
| 17:30.21 | Katty | unless you're watching your health, of course ;) |
| 17:30.25 | Qwell | meh |
| 17:30.33 | Qwell | velveeta == :( |
| 17:30.56 | Katty | i agree that it's not a proper cheese |
| 17:31.03 | Katty | but it does have quite the melty effect |
| 17:31.25 | file | melt my heartttt |
| 17:35.43 | mbowie | "Proper cheese"? It's artificial cheese product substitute... it has as much to do with cheese as beer does with mayonnaise. |
| 17:40.11 | coppice | Katty: Dali has quite the melty effect, but isn't high on realism |
| 17:40.36 | mbowie | I can see some parallels there. |
| 17:40.59 | mbowie | Melty + not real = Dali = Not Cheese |
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| 17:53.27 | haroldp | My asterisk server doesnât want to connect to my voip providor this morning. Iâm seeing âchan_sip.c:13761 sip_reg_timeout: -- Registration for âusername@losangeles.voip.ms' timed outâ on the console. Any ideas how I can track down the problem? |
| 17:54.20 | haroldp | If I test with a simple SIP client from my desktop, it seems to work fine |
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| 18:52.40 | Jacoby6000 | When specifying timezones in a dialplan on a gotoiftime will the timezone of the server offset the selected time zone by the selected timezone? |
| 18:53.53 | Jacoby6000 | https://gist.github.com/Jacoby6000/5df77a264a4bc99cd0d9 |
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| 19:36.32 | TazzNZ | "morning" all |
| 19:37.05 | raspberrypifan | hi |
| 19:38.36 | mbowie | Moin TazzNZ |
| 19:41.14 | *** join/#asterisk sgriepentrog (~sgriepent@nat/digium/x-novqxnmisxjnprjz) |
| 19:41.46 | lvlinux | hey raspberrypifan. ru in Ecuador now? |
| 19:41.52 | raspberrypifan | yup |
| 19:42.01 | lvlinux | get ur phone setup? |
| 19:42.31 | raspberrypifan | nop, i havent really had internet at my apartment. when i come to see my grandma for lunch ive been playing with freeswitch |
| 19:42.35 | raspberrypifan | and i finally got that ATA unlocked |
| 19:43.14 | lvlinux | ah cool. was it harder or easier than u thot? |
| 19:43.58 | raspberrypifan | the ata unlock was super simple |
| 19:44.02 | TazzNZ | who sells locked ATA's...... |
| 19:44.06 | lvlinux | that's nice. |
| 19:44.07 | raspberrypifan | id recommend anyone who needs an fxo/fxs ata to buy this one |
| 19:44.13 | lvlinux | TazzNZ: they're all over ebay |
| 19:44.21 | raspberrypifan | it only cost me 10 buck |
| 19:44.44 | lvlinux | can't beat that. |
| 19:44.54 | TazzNZ | yeah - can't really argue with that :) |
| 19:44.55 | [TK]D-Fender | 9$ <- |
| 19:45.11 | raspberrypifan | fxo? |
| 19:45.14 | [TK]D-Fender | 9$ < 10$ |
| 19:46.28 | lvlinux | which ones are $9 TK? |
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| 19:54.09 | lvlinux | well back to work for me... |
| 19:54.14 | [TK]D-Fender | Which ones are $10? |
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| 19:55.14 | lvlinux | lots on ebay |
| 19:55.22 | lvlinux | locked of course |
| 19:55.30 | [TK]D-Fender | Everything "depends" |
| 19:55.32 | lvlinux | and of course most of them can't be unlocked reasonably |
| 19:55.49 | lvlinux | but sounds like raspberrypifan got lucky and hit an easy unlocker. |
| 19:58.31 | [TK]D-Fender | BBL |
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| 20:53.55 | mic_ | are there any MOH numbers one can buy that sound good on mobiles? |
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| 21:04.04 | mic_ | it sounds awful regardless of used trickery. |
| 21:04.29 | mic_ | as long as G711 is involved - sounds great - call from a cell phone -> really bad. |
| 21:06.55 | WIMPy | Are you still on 2G? |
| 21:07.17 | mic_ | most of the country has 3G |
| 21:07.23 | mic_ | but you never know... |
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| 21:07.31 | TazzNZ | 2G shouldn't make a diffs |
| 21:07.31 | mic_ | (where people call from - physically) |
| 21:07.39 | TazzNZ | it's more signal |
| 21:08.14 | TazzNZ | mic_: does your codec change on a mobile call ? |
| 21:08.17 | WIMPy | 2G only has rather bad codecs. |
| 21:08.35 | mic_ | TazzNZ: nope, I push everything out to the operator via SIP G711 |
| 21:08.48 | dimitry7 | Hi guys. I need to do asterisk pagging, but I get this error: Unable to re-open DSP device /dev/dsp: No such file or directory. I have this device for audio .... Bus 002 Device 004: ID 0d8c:000c C-Media Electronics, Inc. Audio Adapter |
| 21:08.50 | mic_ | TazzNZ: on the cell phone itself - no idea (not sure I can control that) |
| 21:08.53 | TazzNZ | WIMPy: last time I checked it was still all the same |
| 21:09.05 | dimitry7 | and have enabled load => chan_oss.so and load => chan_alsa.so modules |
| 21:09.11 | TazzNZ | that was close to 10 years ago thou |
| 21:09.18 | dimitry7 | but it still can't find the device. why? |
| 21:09.43 | mic_ | dimitry7: unable to re-open |
| 21:09.53 | WIMPy | dimitry7: You can only use one of them. And it looks like oss is not available on your system. |
| 21:09.53 | mic_ | dimitry7: I would say your card does not have a hw mixer |
| 21:10.19 | WIMPy | TazzNZ: 3G offers a lot more. |
| 21:10.21 | TazzNZ | mic_: I would report the issue to the upstream provider |
| 21:10.30 | mic_ | TazzNZ: that is the plan for tomorrow |
| 21:10.43 | WIMPy | At least in theory. How much of it really gets used is another question, off course. |
| 21:10.47 | mic_ | TazzNZ: I just wanted to hear other opinions just to rule out possible ignorance on my side |
| 21:11.40 | TazzNZ | GSM has used a variety of voice codecs to squeeze 3.1 kHz audio into between 6.5 and 13 kbit/s. Originally, two codecs, named after the types of data channel they were allocated, were used, called Half Rate (6.5 kbit/s) and Full Rate (13 kbit/s). These used a system based on linear predictive coding (LPC). In addition to being efficient with bitrates, these codecs also made it easier to identify more |
| 21:11.42 | TazzNZ | important parts of the audio, allowing the air interface layer to prioritize and better protect these parts of the signal. |
| 21:11.46 | TazzNZ | As GSM was further enhanced in 1997[14] with the Enhanced Full Rate (EFR) codec, a 12.2 kbit/s codec that uses a full-rate channel. Finally, with the development of UMTS, EFR was refactored into a variable-rate codec called AMR-Narrowband, which is high quality and robust against interference when used on full-rate channels, or less robust but still relatively high quality when used in good radio conditions |
| 21:11.56 | TazzNZ | on half-rate channel. |
| 21:12.07 | dimitry7 | WIMPy, Okay I have disabled OSS (noload)... still the same.. |
| 21:12.15 | dimitry7 | mic_, do I need a hw mixer? |
| 21:12.17 | TazzNZ | I didn't know about the 3.1 KHz one |
| 21:12.39 | dimitry7 | only alsa is enabled now |
| 21:13.41 | WIMPy | dimitry7: Do you have a 3rd one? like chan_console or what its name was? |
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| 21:14.49 | dimitry7 | WIMPy, well I have this configuration: exten => _*51,1,Dial(console/dsp,20,A(beep)) |
| 21:15.22 | dimitry7 | WIMPy, yes /dev/dsp does not exist :-S |
| 21:15.33 | mic_ | TazzNZ: I did read all wikipedia stuff etc. |
| 21:15.49 | dimitry7 | WIMPy, ls: cannot access /dev/dsp: No such file or directory |
| 21:15.53 | mic_ | TazzNZ: but I am also thinking - could it be, that these bastards use say G729 between operators |
| 21:16.02 | mic_ | TazzNZ: to optimize the amount of traffic? |
| 21:16.10 | WIMPy | dimitry7: yes, that device existed on OSS, not with ALSA. |
| 21:17.00 | WIMPy | I'm not sure if the dialstring is even used. |
| 21:18.11 | Katty | OH WHERE IS MY HAIRBRUSH |
| 21:18.42 | mic_ | where is my food... |
| 21:18.48 | Katty | infobot: ohwhereismyhairbrush? |
| 21:18.54 | mic_ | HA! |
| 21:19.02 | Katty | infobot: ohwhereismyhairbrush |
| 21:19.04 | dimitry7 | WIMPy, I have OSS module compiled |
| 21:19.05 | mic_ | never underestimate geek's freezer. Found 2x ice creams :D |
| 21:19.09 | Katty | infobot: whereismyhairbrush? |
| 21:19.10 | infobot | i heard whereismyhairbrush is not fair, my poor hairbrush! |
| 21:19.22 | Katty | infobot: forget whereismyhairbrush |
| 21:19.22 | infobot | Katty: i forgot whereismyhairbrush |
| 21:19.27 | WIMPy | dimitry7: Noone should need OSS any more. |
| 21:19.32 | Katty | infobot: ohwhereismyhairbrush is NOT FAIR, MY POOR HAIRBRUSH! |
| 21:19.32 | infobot | Katty: okay |
| 21:19.45 | dimitry7 | no? and how do they configure paging? |
| 21:20.12 | WIMPy | dimitry7: With ALSA. |
| 21:20.34 | Tim_Toady | dimitry7: you ca still get /dev/dsp with alsa, u will need to install the alsa-oos wrapper |
| 21:20.44 | Tim_Toady | but its better to move on and use chan_alsa |
| 21:21.08 | dimitry7 | WIMPy, okay, Tim_Toady perfect. let me try it |
| 21:21.10 | Katty | SO. my asterisk broke today. what's wrong with it? |
| 21:21.38 | WIMPy | Katty: Why do you think there's something wrong? |
| 21:21.39 | Tim_Toady | some missing hairbrush i guess |
| 21:21.57 | Katty | WIMPy: it's quacking. |
| 21:22.04 | Katty | WIMPy: QUACK. |
| 21:22.10 | WIMPy | res_hairbrush not loaded? |
| 21:22.15 | Katty | probably not. |
| 21:22.39 | WIMPy | Is it an OSS or an ALSA hairbrush? |
| 21:22.49 | Katty | it's a hipaa hairbrush! |
| 21:22.58 | TazzNZ | I worry about Katty some days.... |
| 21:23.17 | WIMPy | I think that's not supported. |
| 21:23.18 | Katty | me too. |
| 21:23.28 | mic_ | TazzNZ: we will be contacting the provider tomorrow |
| 21:23.35 | Katty | WIMPy: you know a lot of IT vendors say that about hipaa |
| 21:23.44 | Katty | WIMPy: oh you have to be hipaa compliant? we don't support that! |
| 21:23.45 | TazzNZ | mic_: yeah - that might be best |
| 21:26.01 | dimitry7 | how do you do a console dial in asterisk 1.8.28? |
| 21:26.41 | WIMPy | does console/alsa, but as I said, I'm not sure the part after the / is used at all. |
| 21:27.20 | dimitry7 | console dial 4567 |
| 21:27.29 | dimitry7 | asterisk > console dial 4567 |
| 21:27.51 | dimitry7 | show ring the extension... this version does not have the 'console' command :-S |
| 21:27.53 | WIMPy | Oh, that way. |
| 21:28.01 | dimitry7 | yup |
| 21:28.12 | WIMPy | Never did that. |
| 21:28.39 | mic_ | I love those 17h work days |
| 21:28.59 | dimitry7 | WIMPy, try it, its good :-) |
| 21:29.03 | mic_ | thanks for your help guys, I am off to bed. Will share info about that GSM + MOH issue if they tell me anything useful. |
| 21:29.38 | dimitry7 | with chan_oss.so enabled is the trick |
| 21:29.44 | Chainsaw | mic_: The simpler the melody the better it fares. |
| 21:29.46 | dimitry7 | I now can use console dial XXXX :-) |
| 21:30.02 | WIMPy | I don't have any OSS anymore anywhere. |
| 21:30.24 | dimitry7 | WIMPy, why? |
| 21:30.25 | mic_ | Chainsaw: we noticed, that vocals go very well |
| 21:30.30 | Chainsaw | mic_: The simplest one I have, Blue Valley S3M, makes it across most of the time. But even that will do the white noise distortions of doom. |
| 21:30.43 | WIMPy | Because ALSA is much better. |
| 21:30.54 | mic_ | Chainsaw: it's the instruments every now and then result in "snowy sound" like from a TV without antenna |
| 21:30.59 | Chainsaw | mic_: That's not outside the line of expectations, that's what the codec was designed for. Violins and other string instruments do particularly badly. |
| 21:31.02 | dimitry7 | WIMPy, good! I will start using ALSA too . |
| 21:31.48 | mic_ | Chainsaw: I have a chart with frequency map of all instruments |
| 21:32.02 | WIMPy | But ALSA guves timing issues with Asterisk, just as OSS did. |
| 21:32.11 | Chainsaw | mic_: Try "Blue Valley", which you may know as the Uplink theme. |
| 21:32.23 | mic_ | Chainsaw: I did find some music, that "fits" more or less - resample, compress, EQ... but every now and then - "compression artifacts" come up |
| 21:32.32 | mic_ | Chainsaw: checking right away |
| 21:32.41 | Chainsaw | mic_: It's a 12 minute piece, some do better then others. |
| 21:33.10 | Chainsaw | mic_: With a bit of luck you might be able to get 4 or so minutes of GSM-compatible hold music out of it. But try with a good mobile signal. Under bad coverage your "TV noise" is going to return no matter what. |
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| 21:33.15 | Chainsaw | Hi Fender. |
| 21:34.28 | dimitry7 | WIMPy, how can I compile the alsa module? |
| 21:34.40 | dimitry7 | [Jul 7 16:34:35] WARNING[22880]: loader.c:434 load_dynamic_module: Error loading module 'chan_alsa.so': /usr/lib/asterisk/modules/chan_alsa.so: cannot open shared object file: No such file or directory |
| 21:35.17 | WIMPy | Same as any other module. |
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| 21:35.25 | mic_ | Chainsaw: thanks a lot. I found the s3m - I would love to try it out now, but I am falling off due to a long day |
| 21:35.31 | WIMPy | Check it in make menuselect. |
| 21:35.45 | Chainsaw | mic_: Let me know how it does :) |
| 21:35.49 | dimitry7 | WIMPy, ohhh you're right! I did that before :-) |
| 21:35.51 | Chainsaw | mic_: And good night. |
| 21:36.06 | mic_ | Chainsaw: will do. Tomorro I' |
| 21:36.29 | mic_ | Chainsaw: I'll be going there again and will set this up and do some testing |
| 21:36.53 | dimitry7 | why is there like two different versions of asterisk? There is asterisk 1.8, asterisk 10 , 11 and 12 |
| 21:37.48 | WIMPy | Why do you say two and list 4? |
| 21:38.20 | Chainsaw | WIMPy: No, it's teenager-speak. "like two" is anywhere between 2 and 6. |
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| 21:39.03 | Chainsaw | dimitry7: 1.8 is for when you want stability and no features, like Debian stable. 11 is for when you want to be where the action is but without the real breakage, like Debian testing. |
| 21:39.05 | malcolmd | the version after 1.8 was 10. the version after 10 was 11. the version after 11 is 12. |
| 21:39.15 | Chainsaw | dimitry7: And 12 is if you like to be on the bleeding edge. The blood will be yours. |
| 21:40.00 | malcolmd | prior to 1.8 there were 1.6.2, 1.6.1, 1.6.0, 1.4, 1.2, 1.0 and 0.x. following 1.8, and in lieu of 1.10, we dropped the 1. |
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| 22:08.17 | dimitry7 | Chainsaw, ohh perfect!! I got it :-) |
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| 22:14.36 | Chainsaw | dimitry7: 10 is out of support though, so you shouldn't try & run that. Your choice is really between 1.8 & 11. |
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| 23:07.31 | kunwon1 | i'm using asterisk 12.3.2, i have security => security in logger.conf, but the file doesn't get logged to. in "logger show channels" i see /var/log/asterisk/security but it has no configuration listed. When asterisk loads, it prints 'Asterisk 12.3.2 built by..' etc, to the security log, so i know asterisk can write to it. What else can I check to figure out why security log isn't being written? |
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