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| 01:41.55 | UncleKiwi | hi there, just wondering why parts of my Master.csv log file are logging the calls in correct 24 format and others are not |
| 01:42.03 | UncleKiwi | its 13:41 here now |
| 01:42.26 | UncleKiwi | but when i make a call to teh box it logs it as 01:41:00 |
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| 01:42.55 | UncleKiwi | however there is some situations where the calls are logged as 23:41:212 |
| 01:44.50 | UncleKiwi | but it would seem there is no consistancy |
| 01:46.33 | UncleKiwi | just wondering where asterisk is getting its time |
| 01:46.34 | UncleKiwi | from |
| 01:47.48 | UncleKiwi | for the logging |
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| 02:28.10 | UncleKiwi | can anyone explain what im seeing in the Master.csv file with the times being out by 12 hrs |
| 02:30.43 | UncleKiwi | sorry guys - see what it is |
| 02:30.52 | UncleKiwi | tims is in GMT |
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| 02:31.01 | UncleKiwi | in the Master.csv |
| 02:31.28 | WIMPy | Some times? |
| 02:31.44 | UncleKiwi | i think all times are in GMT |
| 02:31.51 | UncleKiwi | in the Master.csv |
| 02:32.06 | UncleKiwi | ie 2:30am |
| 02:32.24 | UncleKiwi | NZ time is 14:30 |
| 02:32.43 | UncleKiwi | this was casuing me confusion |
| 02:32.45 | WIMPy | If wouldn't have said it happens sometimes, I'd have asked if you timezome is correct or if you configured utc logging. |
| 02:33.11 | UncleKiwi | sorry my fault |
| 02:33.18 | WIMPy | Bad typing today :-( |
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| 02:34.00 | UncleKiwi | i think the Master.csv is the raw data |
| 02:34.11 | UncleKiwi | and you can't change the timezone in that |
| 02:35.14 | WIMPy | cdr.conf: usegmtime |
| 02:36.00 | UncleKiwi | hmm thats strange |
| 02:36.06 | UncleKiwi | I must have neabled it |
| 02:36.40 | UncleKiwi | i wonder what i was thinking |
| 02:37.32 | nightwolf_ | Don't want to interrupt but got a question regarding MixMonitor. |
| 02:37.38 | nightwolf_ | Is it possible to prevent MixMonitor from downsampling to 8kHz? I'm trying to make wideband recordings and according to here (https://reviewboard.asterisk.org/r/1314/) it should be possible but I've tried several different codecs (like SILK) yet the resulting file is still downsampled to 8kHz. Is this most likely a codec configuration issue or is there a way to specifically set the sample rate of the output audio file? |
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| 02:40.35 | UncleKiwi | all sorted thanks WIMPy |
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| 04:14.28 | fling | I have a sip client behind a nat |
| 04:14.29 | fling | 133/133 192.168.0.3 D 5060 UNREACHABLE |
| 04:14.36 | fling | I can call but I have no sound. |
| 04:14.55 | fling | also it's local net ip address is used instead of the nat address hmm hmmm |
| 04:14.58 | fling | How to fix? |
| 04:20.30 | fling | ok, used nat=yes |
| 04:20.44 | fling | yay! got sounds |
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| 05:48.58 | jzu_ | http://pastebin.com/fi68Qa0q - well that doesn't look bad, riight? |
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| 06:03.49 | WIMPy | JitBuf doesn't look sensible. |
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| 06:16.56 | jzu_ | what that would be telling? |
| 06:17.30 | jzu_ | that's call that hasn't yet been answered, it |
| 06:17.34 | jzu_ | it's just ringing |
| 06:18.11 | WIMPy | In that case the results might be meeaningless. |
| 06:18.28 | WIMPy | But you surely don't want a jitter buffer there. |
| 06:18.35 | WIMPy | But maybe FreePBX thinks otherwise. |
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| 06:24.00 | jzu_ | should I disable the jitterbuffer permanently? |
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| 06:24.35 | jzu_ | I had UNREACHABLE's last night with that IAX2 Trunk between two Asterisks, then I changed qualify to 60000 and UNREACHABLE's stopped :P |
| 06:25.13 | WIMPy | Jitter buffers should only be used at the endpoints. |
| 06:25.28 | jzu_ | endpoint meaning the phones? |
| 06:25.59 | WIMPy | And if you need to increase the timeout to 6s you probably don't want to try to use that connection for voip. |
| 06:26.09 | WIMPy | Wherever the call terminates. |
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| 06:26.50 | jzu_ | well funny thing is that iax2 show peers states that ping is 27 ms and still I experienced those UNREACHABLE's |
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| 06:27.04 | jzu_ | connection is quite goot for WAN link actually |
| 06:27.49 | WIMPy | Packet loss, or if it really was increasing the timeout that fixed it, extreme jitter. |
| 06:28.22 | jzu_ | hmm, do you think I should try SIP instead of IAX2, would it change anything?-) |
| 06:28.30 | jzu_ | s/goot/good/g |
| 06:28.35 | WIMPy | no |
| 06:29.09 | jzu_ | could it be 1st Gen Raspberry Pi that just can't cope with IAX2 Trunking.. hmm |
| 06:30.13 | WIMPy | Depends on how much you make it do. |
| 06:30.20 | jzu_ | single call :) |
| 06:30.42 | jzu_ | I checked load of the Linux during that call and it was minimum |
| 06:30.59 | WIMPy | Don't know what happens if you want to record it, but otherwise it will surely cope. |
| 06:32.13 | jzu_ | yeah, thats what I thought also |
| 06:33.16 | UncleKiwi | are u using an sdcard |
| 06:33.37 | WIMPy | Hard to avoid on the RPi. |
| 06:33.42 | jzu_ | :-) |
| 06:33.47 | UncleKiwi | nah its not |
| 06:33.57 | jzu_ | I tested the trunking inside my local LAN and calls worked OK |
| 06:34.03 | UncleKiwi | you can have the /boot on the sdcard only |
| 06:34.06 | WIMPy | Tell us how. |
| 06:34.13 | jzu_ | so it's something to do with the WAN link between the two PBX's |
| 06:34.18 | WIMPy | Sure. But you still need the card. |
| 06:34.51 | UncleKiwi | i guess im just suggesting maybe the write speed to the scrd might be casuing some problem |
| 06:34.54 | WIMPy | But using USB instead of the SD card surely won't make things better. |
| 06:35.22 | jzu_ | UncleKiwi: AFAIK Asterisk doesn't need to write to disk when passing call through a Trunk |
| 06:35.42 | jzu_ | atleast it would be stuuuupid way to handle calls :) |
| 06:35.48 | UncleKiwi | i thought you were recording it |
| 06:35.54 | jzu_ | no, I'm not |
| 06:35.57 | WIMPy | No, it doesn't. |
| 06:35.58 | UncleKiwi | ah ok |
| 06:36.05 | jzu_ | that's why I'm puzzled |
| 06:36.58 | jzu_ | does iax2 test losspct reside active between Asterisk reloads? |
| 06:37.17 | jzu_ | it MIGHT be that I had iax2 tesst losspct active on some level, ie. 50% last night |
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| 06:37.39 | jzu_ | I remember trying the command during the evening, I just cannot remember was it after I started having call quality issues or not :) |
| 06:38.14 | jzu_ | seems that you can't query current status of the test variable |
| 06:38.59 | WIMPy | Netstats are only available per call and only while it exists. |
| 06:39.19 | jzu_ | but yeah anyways - what commands I should run and on what ends when I experience these issues again? |
| 06:39.39 | jzu_ | the call was "stuttering" certain bits of voice were repeated few times within a second or two |
| 06:40.01 | jzu_ | actually I might have recording of it on my own PBX... |
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| 07:19.25 | jzu_ | yes well not it's looking better, I disabled jitterbuffer. |
| 07:19.28 | jzu_ | <PROTECTED> |
| 07:19.31 | jzu_ | Channel RTT Jit Del Lost % Drop OOO Kpkts Jit Del Lost % Drop OOO Kpkts FirstMsg LastMsg |
| 07:19.34 | jzu_ | IAX2/Joensuu-31376 27 -1 0 -1 -1 0 -1 0 0 40 0 0 0 0 0 Tx:NEW Rx:ACK |
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| 09:07.51 | Rad | Hello , curreny i'm having 716 active channels and 359 active calls on my PBX (Asterisk V13) when launching core show channels command. Apparently memory usage keeps increasing , actually 5% (keep increasing until 11%), majority of the active calls are outbounded calls placed via auto dial call files. Once rebooting asterisk, pid MEM% gets back to 0 until , when i start programming vocal |
| 09:07.51 | Rad | campaigns (.call files) then it keeps increasing once again. When having no active calls , the memory is stuck to its high value, and doesnt decrease until i reboot asterisk. I know that this is related to Linux not to free Memory but only replacing it, but related to your experiance is a MEM usage of 10% normal for a server running about 400 simultenous calls ? |
| 09:08.13 | Rad | I am running Asterisk 13.2.0, on Debian squeeze |
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| 09:14.57 | Greenlight | I guess the % memory usage depends on how much total memory you have in the box |
| 09:17.57 | Rad | free -m |
| 09:17.57 | Rad | <PROTECTED> |
| 09:17.57 | Rad | Mem: 128965 14962 114003 0 241 4961 |
| 09:17.57 | Rad | -/+ buffers/cache: 9759 119206 |
| 09:17.57 | Rad | Swap: 1021 0 1021 |
| 09:18.02 | Rad | Actually MEM is 6.4% |
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| 12:55.02 | y_ateya | Is DAHDI timing resource better than timerfd? If I am using IAX channel on linux and not using diguim hardware. |
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| 12:57.02 | Greenlight | If you don't have any hardware then I would recommend timerfd |
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| 13:37.33 | [TK]D-Fender | ~book |
| 13:37.33 | 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 |
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| 13:44.23 | jepperl | hey @all , can anyone tell me how to retrieve the space on which a channel was parked? After my call to Park() i wanna know what space was used (and dictating it with PARKEXTEN is not an option for me). Is there any way i can retrieve the space after the Park? |
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| 13:45.31 | WIMPy | Ask a historian. In the 70s they knew about that stuff :-) |
| 13:47.41 | [TK]D-Fender | jepperl: "parkedcalls show" |
| 13:50.30 | jepperl | #asterisk:[TK]D-Fender: yea, but i need to retrieve it from my .ael file, right after the call to Park(). I need to know the specific parking space from the park that just took action. |
| 13:51.32 | [TK]D-Fender | There is no "after" park ... because that call sits in Park. It doesn't continu on. |
| 13:52.48 | [TK]D-Fender | You either have to parse out that CLI command, or inquire via AMI |
| 13:52.57 | jepperl | [TK]D-Fender , i see, brainfart on my part.. So i have to dictate it to know what space will be used. but then i need to make sure that space is available before setting PARKEXTEN and calling Park() |
| 13:53.59 | jepperl | [TK]D-Fender: i think ill go parse some CLI output beforehand to find an available space :) thanks for the help, bud. appreciate it |
| 13:54.11 | [TK]D-Fender | You're welcome |
| 13:55.04 | WIMPy | Don't parkingspaces have devicestates? |
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| 13:58.24 | [TK]D-Fender | WIMPy: Yes, it does report presence, just not reporting back who is in what slot |
| 13:58.42 | WIMPy | But you can find a free one. |
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| 14:01.03 | [TK]D-Fender | WIMPy: that's one way I guess. |
| 14:07.42 | Demon_VoIP | Hello. Asterisk 13, sip.conf. Setting the option "port" in the peer doesn't change anything. Requesting "sip show peer" i see ":5060" in the host and the address. Where am I wrong? Why do we need this option? |
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| 14:12.26 | Compy | Is it possible to have asterisk (running on FreePBX distro) log into a linphone SIP account and forward incoming calls to an extension (handset) already tied to the asterisk box? |
| 14:13.06 | Compy | thinks this would be configured in sip.conf |
| 14:13.16 | WIMPy | Compy: I'd assume that round about what it's there for. |
| 14:13.39 | Compy | Yep, cool, I'll experiment on. Just wanted to run it by the gurus. Thanks! |
| 14:16.22 | [TK]D-Fender | You should be asking in #freepbx for all things management |
| 14:16.25 | [TK]D-Fender | ~freepbx |
| 14:16.33 | 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 |
| 14:17.10 | Compy | Will do in the future for anything related to FreePBX and its UI (i agree its pretty heavy). D'oh! Thanks for the pointer. |
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| 14:34.36 | wanda_ | hello, I would like to know the best hours to speak here? :) |
| 14:36.13 | [TK]D-Fender | ALL OF THE HOURS |
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| 14:52.18 | file | squirrel |
| 14:53.21 | [TK]D-Fender | file: http://www.alaskacommons.com/wp-content/uploads/2013/02/adhd-highway-to-hey-look-a-squirrel.png |
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| 15:30.36 | *** join/#asterisk hayonj (4077d488@gateway/web/freenode/ip.64.119.212.136) |
| 15:34.23 | hayonj | Hi there, I'm trying to follow this tutorial https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial but can't find the folder /usr/src/asterisk can anyone point me towards the proper direction ? |
| 15:35.18 | *** join/#asterisk russlevy (d539e73c@gateway/web/freenode/ip.213.57.231.60) |
| 15:38.30 | *** join/#asterisk sudharshaw (~sudharsha@175.157.25.209) |
| 15:41.00 | [TK]D-Fender | hayonj: That is assuming you extracted the source tarball there when you installed * |
| 15:42.44 | hayonj | [TK]D-Fender : I installed starit from an ISO image downloaded from the website. Does that mean I have to compile Elastix from scratch ? |
| 15:44.05 | [TK]D-Fender | hayonj: We don't know if this is even REQUIRED. It depends what they pre-compiled into their packages. You should be asking on their channel for this. |
| 15:45.45 | hayonj | [TK]D-Fender : I dont use IRC regurlary, what is the most active channel for elastix ? I though coming to the Asterisk channel would have helped. |
| 15:46.25 | sudharshaw | what is the elastix version you downloaded |
| 15:46.27 | WIMPy | #elastix , but I wouldn't expect much activity there. |
| 15:46.47 | [TK]D-Fender | hayonj: We don't support their packages. |
| 15:47.10 | sudharshaw | I installed elastix once and it installed the whole package from the ISO |
| 15:47.15 | WIMPy | Maybe you should reconsider using Elastix before continuing. |
| 15:47.15 | [TK]D-Fender | hayonj: Also I see no reference to src anywhere on that page you linked |
| 15:47.32 | [TK]D-Fender | Definitely |
| 15:47.45 | [TK]D-Fender | hacked up FreePBX to boot... |
| 15:47.53 | [TK]D-Fender | feels a little sick... |
| 15:48.58 | sudharshaw | personal openion // install asterisks & freepbx much better and flexible than Elastix |
| 15:49.53 | hayonj | Is there a easy way to respond back to a particular person using the webchat.freenode.net or just simply copy paste username ? |
| 15:50.39 | hayonj | [TK]D-Fender : Haha sorry, Look at this youtube video, https://www.youtube.com/watch?v=QbGglbeV-kw [1:57 min ] |
| 15:50.40 | [TK]D-Fender | hayonj: Standard auto-complete is to type a partial nick and hit <tab> |
| 15:51.01 | hayonj | [TK]D-Fender: cool thanks that helped |
| 15:51.48 | hayonj | sudharshaw: freePBX 2.8.1 release 16 |
| 15:52.16 | hayonj | sudharshaw: elastix 2.4.0 release 1 |
| 15:53.05 | hayonj | sudharshaw: Asterisk and FreePBX, compiled from scratch ? |
| 15:53.38 | sudharshaw | hayonj, elastix has version 3 if I am not mistaken which is an ISO installation |
| 15:53.53 | sudharshaw | this I think I remember isntalls the whole works |
| 15:54.34 | hayonj | WIMPy: I've become used to this version, as this is the version used for all my live deployments, why do you say I should reconsider it ? |
| 15:55.11 | WIMPy | Because you probably won't find much support when using it. |
| 15:55.36 | [TK]D-Fender | hayonj: I doubt that underlying FreePBX engine of yours even supports this |
| 15:57.25 | hayonj | sudharshaw: version 3 ( I tried it, it's multitenant version, can't get the trunks to work because our provider doesn't require a "peer/username/secret" ) Still trying to get that to work. They changed a lot of things because of security. |
| 15:57.38 | hayonj | sudharshaw: version 4 is still in beta |
| 15:59.15 | *** join/#asterisk D-Boy (~D-Boy@unaffiliated/cain) |
| 16:00.29 | hayonj | I'm ready to change my habits if the community can help out. I'm new the linux world (prefered distro is CentOS about 2years) and voip (using Elastix only for about 6 months) |
| 16:02.02 | WIMPy | There is some learning to do. Best to read the book to find out if you're up to it or if you live with the limitations of 3rd party config tools. |
| 16:02.06 | WIMPy | ~book |
| 16:02.06 | 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 |
| 16:02.22 | *** join/#asterisk babak (uid19622@gateway/web/irccloud.com/x-mmunqvjjwzitergo) |
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| 16:05.53 | [TK]D-Fender | hayonj: hayonj If you require a GUI then at a minimum I'd recommend using FreePBX's own baked all-in-one ISO. At lease the core Asterisk and FreePBX GUI parts will be largely stock |
| 16:07.50 | *** join/#asterisk camerin (hoax@elite.bshellz.net) |
| 16:07.54 | hayonj | Ok thanks for everything, I will try to install a new centos distribution and install asterisk from scratch using this link, http://www.powerpbx.org/content/asterisk-freepbx-install-guide-centos-v6-asterisk-v13-freepbx-v12 |
| 16:08.13 | hayonj | I will also go on amazon and get myself a hardcopy of the book. |
| 16:09.30 | [TK]D-Fender | hayonj: Use the guides provided from FreePBX themselves and not all these 3rd parties you keep going to. |
| 16:10.29 | hayonj | [TK]D-Fender: I have a tendency to google things up, best to follow official documentation you are absolutely right. |
| 16:11.14 | [TK]D-Fender | hayonj: The downside to taking guides from "some guy" is how often "some guy" .... is incompetent. |
| 16:12.34 | hayonj | so I should follow this wiki for (I believe it's the official one) for installing asterisk https://wiki.asterisk.org/wiki/display/AST/Installing+Asterisk |
| 16:13.26 | hayonj | and this one for FreePBX http://wiki.freepbx.org/pages/viewpage.action?pageId=1048598 |
| 16:13.39 | jzu_ | WIMPy: beats me, the IAX2 has been working A-OK today |
| 16:14.06 | jzu_ | WIMPy: Asterik<--->Powerline adapter<--->7960G had 5ms ping |
| 16:14.30 | jzu_ | WIMPy: and IAX2 Trunk has average ping on 27ms, geographical distance is 500km's |
| 16:14.35 | jzu_ | s/on/of |
| 16:14.38 | [TK]D-Fender | hayonj: http://wiki.freepbx.org/display/HTGS/Installing+FreePBX+12+on+CentOS+6.5 |
| 16:14.42 | [TK]D-Fender | hayonj: that guide is complete |
| 16:15.15 | [TK]D-Fender | hayonj: Assuming you want to compile it all yourself. Their ISO would be a valide quick-start since they do a rather complete job |
| 16:15.27 | bmurt | anyone know if you can includ miliseconds for the dateformat in v1.4? |
| 16:15.36 | bmurt | (ik now 1.4 is old as shit) |
| 16:15.49 | bmurt | %3q displays literal |
| 16:16.59 | hayonj | [TK]D-Fender: You are a great help on all of this advice. I want to compile it all from scratch just for the knowledge I would gain from it, but a quicker way would also be appreciated |
| 16:18.30 | [TK]D-Fender | hayonj: both are options and up to you |
| 16:18.39 | [TK]D-Fender | hayonj: Judge for your needs accordingly |
| 16:19.20 | [TK]D-Fender | hayonj: either is far more supportable than elastix |
| 16:19.38 | hayonj | [TK]D-Fender: I stumbled on this link earlier, which points out to the fact only a select few modules can be recompiled without recompiling everything http://blog.michael.kuron-germany.de/2011/04/asterisk-compile-srtp-module-without-recompiling-asterisk/ |
| 16:20.19 | [TK]D-Fender | hayonj: Some Guy. 2011. |
| 16:20.31 | hayonj | [TK]D-Fender: I also wanted to make sure that the reason I don't see the /usr/src/asterisk folder is because I installed straight from a DVD instead of compiling from sources. |
| 16:20.57 | *** join/#asterisk dimitry7 (~dimitry7@gate.aaamerica.com.mx) |
| 16:21.24 | dimitry7 | Hi guys, Do you have some pre-made IPtables rules that work to give priorization to asterisk traffic? specially port 4569 |
| 16:21.34 | [TK]D-Fender | hayonj: Correct, and your distro didn't include them there |
| 16:21.37 | dimitry7 | I've been googling a loooot.... |
| 16:21.47 | dimitry7 | but nothing seems to work |
| 16:21.55 | [TK]D-Fender | dimitry7: Any prioiritization rule will do... just change the port |
| 16:22.01 | [TK]D-Fender | dimitry7: IPTABLES is IPTABLES |
| 16:22.09 | dimitry7 | [TK]D-Fender, really? okay :) |
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| 16:27.06 | hayonj | [TK]D-Fender: I also saw a few other versions out there (that come precompiled, although I haven't tested them) i.e. AsteriskNOW , PIAF... as per what you are saying ... all of these are not good as well ? |
| 16:27.45 | [TK]D-Fender | Recently AsteriskNOW is pretty much a re-badged FreePBX ISO last I heard. |
| 16:28.06 | [TK]D-Fender | And.... I'd rather get it from at least the maker of ONE of the 2 key parts |
| 16:28.16 | hayonj | [TK]D-Fender: Ok I have a lot of work to do, since I'll be compiling from scratch for the first time. Will you be here if I need further assistance ? |
| 16:28.38 | [TK]D-Fender | PIAF color-codes their stuff. That's for kiddies. |
| 16:28.53 | [TK]D-Fender | People are here and in #freePBX for this when you have questions. |
| 16:29.06 | [TK]D-Fender | At least you'll be working in environments most have seen and can support |
| 16:31.27 | babak | Hi,is it possible to we create a uuid and put it in originate in AMI ? |
| 16:31.49 | [TK]D-Fender | sure |
| 16:32.48 | babak | [TK]D-Fender: if possible an example or link for reading please |
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| 16:33.28 | [TK]D-Fender | babak: making up whatever ID you want.. is your job. putting in the originate.... SetVar <- |
| 16:33.33 | hayonj | [TK]D-Fender: Cool, I'll keep you posted on my assension thank you very much for giving me proper advice |
| 16:34.15 | [TK]D-Fender | hayonj: Saves you a lot of pain when you realize that other distros often use hacked up / ancient junk. Working around their nonsense is just asking for pain |
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| 16:43.39 | hayonj | [TK]D-Fender: Would know where I can find a map of all the system and how things interact with each other ? I have some knowledge of PHP and MySQL ... I was able to figure some things out by installing PHPMYADMIN and looking at databases. But I want a more in depth knowledge. ( For example I know there is a few database in MySQL and a few for SQLite3 ) |
| 16:44.10 | [TK]D-Fender | Not sure what direction you're headed with that... |
| 16:46.29 | hayonj | [TK]D-Fender: Just for debugging purposes, for example I know the file /var/lib/asterisk/bin/retrieve_conf is the link between my system files and my databases. |
| 16:47.24 | [TK]D-Fender | hayonj: Kinda far-reaching... |
| 16:48.05 | [TK]D-Fender | The only SQLite really is for AstDB which is Asterisk itself, not freepbx |
| 16:48.16 | [TK]D-Fender | that's the new backend that replaces BDB2 |
| 16:48.28 | [TK]D-Fender | Which it still emulates oddly enough |
| 16:49.03 | [TK]D-Fender | Everything else is pretty much PHP to generate * configs, AstDB values for dialplan flow control, and then back--end configs mostly |
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| 17:03.51 | mjordan | [TK]D-Fender: it mostly emulates BDB because that way we (and users) didn't have to update everything that used the DB family of functions. There was always a notion in the back of our heads that one day we'd expose the AstDB in a more SQL-esque kind of way; it just never materialized. |
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| 17:43.46 | OurRoyalGabe | !next |
| 17:44.41 | russlevy | hi, i'm trying to create a simple dialplan from one PJSIP extension to another, and then to call using a SIP trunk |
| 17:47.27 | russlevy | in both cases i'm getting the following error (this is calling PSTN): |
| 17:47.28 | russlevy | <PROTECTED> |
| 17:47.50 | file | you want to use PJSIP/4167777777@voipms |
| 17:48.36 | russlevy | OK -- is this new for PJSIP? I was pretty sure it was the other way with SIP |
| 17:50.24 | file | chan_sip allows either way |
| 17:50.28 | file | chan_pjsip does not |
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| 17:53.50 | russlevy | understood -- thanks! now the other side is giving me congestion for every number i call, even though i'm registered. is that probably a configuration problem in my settings? |
| 17:54.58 | RoyK | hrmf - http://paste.debian.net/172125/ |
| 17:58.01 | *** join/#asterisk karelk (~karel@84-72-164-65.dclient.hispeed.ch) |
| 17:59.16 | RoyK | tried to upgrade to 13.3.2, but still same error |
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| 18:11.59 | [TK]D-Fender | RoyK: -- Error initializing Bluetooth device RoysSGS5. |
| 18:12.10 | russlevy | how do you see which codec a PJSIP channel is using? |
| 18:12.16 | [TK]D-Fender | RoyK: Give this error you have... I see nothign indicating a code issue with Asterisk. |
| 18:12.35 | [TK]D-Fender | RoyK: You haven't provided any other debug to show anything meaningful to assist with. |
| 18:12.54 | RoyK | [TK]D-Fender: Even with full debug on, I can't seem to get much from it |
| 18:13.30 | [TK]D-Fender | RoyK: we still need something useful to go on. |
| 18:13.39 | RoyK | I know |
| 18:16.30 | *** join/#asterisk areski (~areski@80.174.128.114.dyn.user.ono.com) |
| 18:17.18 | RoyK | hm... [May 6 18:15:14] WARNING[10361]: chan_mobile.c:3093 sdp_register: Failed to connect sdp and create session. |
| 18:19.38 | RoyK | [TK]D-Fender: I get that during startup - however - if I'm on the phone, asterisk steals the audio channel without an issue |
| 18:19.47 | RoyK | [TK]D-Fender: any idea how to debug this further? |
| 18:20.22 | [TK]D-Fender | nope. |
| 18:26.20 | *** join/#asterisk nightwolf_ (4055f7c7@gateway/web/freenode/ip.64.85.247.199) |
| 18:27.08 | nightwolf_ | Hey I dropped in last night and wasn't able to get an answer then but does anyone know if it's possible to configure the sample rate of recordings made Asterisk using ConfBridge, MixMonitr, or Monitor? |
| 18:32.21 | [TK]D-Fender | nightwolf_: Those are intrinsic in the format you are choosing for your recording. |
| 18:33.31 | nightwolf_ | I've been using 24kHz and 48kHz codecs and requested they be mixed at 48kHz (using internal_sample_rate in confbridge) yet the resulting file is still downsampled to 8kHz. Is there anything that could cause this? |
| 18:40.17 | nightwolf_ | Is there a more specific way to specify the format of the file? |
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| 18:47.33 | nightwolf_ | [TK]D-Fender: Could this be an issue the available file formts for asterisk? |
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| 19:15.28 | [TK]D-Fender | nightwolf_I've been using 24kHz and 48kHz codecs and requested they be mixed at 48kHz (using internal_sample_rate in confbridge) yet the resulting file is still downsampled to 8kHz. Is there anything that could cause this? <- the FORMAT you are using |
| 19:16.40 | nightwolf_ | [TK]D-Fender: Ok got it. Where/how can I set the format to something different then? |
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| 19:22.13 | exuberocity | Afternoon all. Question -- So I am wanting to use custom states to differentiate between lunch, break, etc on my FreePBX system. I'm pretty sure I can do it with custom states but never have. Thoughts? |
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| 19:31.21 | Cyford33_u | how can i make a + addition for the following dialplan |
| 19:31.23 | Cyford33_u | exten => loop,1,Set(LOOPCOUNT=${LOOPCOUNT} + 1) |
| 19:32.24 | [TK]D-Fender | exuberocity: Nope. |
| 19:32.35 | [TK]D-Fender | exuberocity: as in "nothing offered by FreePBX" |
| 19:33.00 | [TK]D-Fender | exuberocity: Cyford33_u Actually make a proper expression. |
| 19:33.04 | [TK]D-Fender | Cyford33_u Actually make a proper expression. |
| 19:33.27 | Cyford33_u | thats my question reworded lol |
| 19:33.38 | Cyford33_u | whats the proper expression |
| 19:33.44 | [TK]D-Fender | https://wiki.asterisk.org/wiki/display/AST/Expressions |
| 19:33.49 | nightwolf_ | I've been looking at the documentation for ConfBridge and it says the default format for the recorded file 8kHz but it doesn't say how to change the format. Any ideas? Thanks in advance. |
| 19:34.04 | [TK]D-Fender | Dialplan basics |
| 19:34.07 | Cyford33_u | exten => 321,n,Set(NEWCOUNT=$[${COUNT} + 1]) |
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| 19:34.48 | [TK]D-Fender | nightwolf_: There are very obvious setting where you tell it the file format.... |
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| 19:34.55 | [TK]D-Fender | nightwolf_: read the sample config |
| 19:35.05 | exuberocity | Ok thanks I will take a look. |
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| 19:35.19 | [TK]D-Fender | exuberocity: Nothing to look for. |
| 19:36.24 | Cyford33_u | this is how i originallyt had it exten => loop,1,Set(LOOPCOUNT=$[${LOOPCOUNT} + 1]) |
| 19:36.29 | Cyford33_u | is this correct? |
| 19:36.58 | [TK]D-Fender | Go try it |
| 19:37.16 | Cyford33_u | i did an it doesnt work |
| 19:37.22 | Cyford33_u | <PROTECTED> |
| 19:37.26 | [TK]D-Fender | Then clearly something is wrong.... |
| 19:37.44 | Cyford33_u | me askin here should tell u its down |
| 19:37.45 | [TK]D-Fender | What is LOOPCOUNT when you're running it? |
| 19:37.48 | Cyford33_u | <PROTECTED> |
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| 19:40.15 | Cyford33_u | i am trying to limit the ivr loop |
| 19:40.36 | nightwolf_ | [TK]D-Fender: I'm sorry I feel really silly I can't find the setting you're talking about. All I see is record_conference and record_file. What's the one you're referring to? |
| 19:40.37 | Cyford33_u | <PROTECTED> |
| 19:40.57 | Cyford33_u | [2015-05-06 15:40:15] WARNING[13341][C-0000011e]: ast_expr2.fl:470 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected '+', expecting $end; Input: |
| 19:41.22 | Cyford33_u | ohh nvm i found it |
| 19:41.24 | Cyford33_u | lol |
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| 19:48.52 | [TK]D-Fender | nightwolf_: You set the format when you set the file... |
| 19:49.05 | Cyford33_u | wow the answer was putting the number before the variable lol |
| 19:49.34 | Cyford33_u | this worked lol exten => loop,1,Set(LOOPCOUNT=$[1 + ${LOOPCOUNT}]) |
| 19:50.06 | [TK]D-Fender | no |
| 19:51.21 | nightwolf_ | [TK]D-Fender: Right but wav files can have different sample rates despite being the same format. Is there a way to specifically get a 48kHz format? Because right now when I set the file to wave, it's signed linear at 8kHz. |
| 19:51.44 | nightwolf_ | 48kHz sample rate* , sorry |
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| 19:54.51 | [TK]D-Fender | You don't get a different rate WITHIN that format |
| 19:54.55 | [TK]D-Fender | the format itself IS the limit |
| 19:56.21 | Cyford33_u | my ivr loop limit |
| 19:56.24 | Cyford33_u | exten => loop,1,Set(LOOPCOUNT=$[1 + ${LOOPCOUNT}]) |
| 19:56.24 | Cyford33_u | exten => loop,n,GotoIf($[${LOOPCOUNT} > 2]?hangup:) |
| 19:56.24 | Cyford33_u | exten => loop,n,Goto(18775743418,1) |
| 19:56.24 | Cyford33_u | exten => loop(hangup),n,Hangup |
| 19:57.47 | Cyford33_u | exten => t,1,Goto(loop,1) |
| 19:59.07 | [TK]D-Fender | Yup, automatic mistake |
| 19:59.41 | [TK]D-Fender | you didn't set an INITIAL VALUE |
| 20:02.15 | [TK]D-Fender | (anywhere we see) |
| 20:03.58 | Cyford33_u | yes, i would think null would be 0 |
| 20:04.35 | Cyford33_u | the fist loop shows as 0 though, so it hangs up after 3 round trips |
| 20:05.28 | Cyford33_u | but this didnt work at all LOOPCOUNT=$[${LOOPCOUNT + 1} |
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| 20:06.32 | Katty | hi fenderbender. |
| 20:06.55 | [TK]D-Fender | Cyford33_uyes, i would think null would be <- wrong |
| 20:07.05 | [TK]D-Fender | Cyford33_u[2015-05-06 15:40:15] WARNING[13341][C-0000011e]: ast_expr2.fl:470 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected '+', expecting $end; Input: |
| 20:07.10 | [TK]D-Fender | it is BLANK |
| 20:07.17 | [TK]D-Fender | hence the error you first got |
| 20:07.23 | Katty | blank like my face. |
| 20:11.35 | [TK]D-Fender | Not EXACTLY like that. |
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| 20:29.39 | *** join/#asterisk tompaw (~tompaw@tompaw.xxx) |
| 20:29.42 | tompaw | Hello. |
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| 20:30.20 | tompaw | I have a problem with res_pjsip_endpoint_identifier_ip.c - even though it comes up in the list of loaded modules, it's not being used, i.e. doesn't show up in DEBUG at all for inbound calls. |
| 20:30.45 | tompaw | As a consequence, my identify entries in pjsip.conf are totally ignored. |
| 20:31.35 | tompaw | 13.3.2 + pjsip |
| 20:32.22 | tompaw | ooops, actually it says not running |
| 20:34.04 | *** part/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 20:35.03 | [TK]D-Fender | packs up to head home |
| 20:35.53 | tompaw | ah, it's probably the sorcery |
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| 20:48.44 | tompaw | yep, fixed. |
| 20:51.21 | tompaw | another question - how can I access source endpoint in asterisk dialplan? |
| 20:52.01 | tompaw | I don't want to have to identify it by CLID, which changes with destinations, is there a variable that stores the actual source endpoint name? |
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| 20:54.42 | file | a dialplan function, yes, ${CHANNEL(endpoint)} |
| 20:54.53 | tompaw | file: thank you |
| 20:55.02 | tompaw | couldn't find it anywhere in docs :( |
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| 20:57.04 | file | https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Function_CHANNEL for more info |
| 20:57.43 | tompaw | file: much appreciated. |
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| 21:11.51 | overyander | why does System(/var/lib/asterisk/scripts/foo.sh) run the shell script, but System(python /var/lib/asterisk/scripts/foo.py) doesn't run? |
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| 21:22.38 | ChannelZ | is 'python' in the path of the environment of the user that asterisk was run as? |
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| 22:03.16 | RoyK | is the asterisk-user ml moderated? |
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