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| 00:22.05 | Yinzara | I'm attempting to compile Asterisk with WebRTC support. I'm specifically trying to use it with OSDial which was designed to work with asterisk 11.13 however I'm willing to upgrade if necessary. I followed all the compilation instructions on the Wiki and the server starts up fine, however when you attempt to connect a call, channel information isn't available in the real time database so OSDial isn't functioning. I turned on debug |
| 00:22.35 | Yinzara | I can see there are no insert statements into the real time database which do occur in the prepacked version of asterisk with OSDial |
| 00:22.49 | Yinzara | Does anyone have any ideas what could cause that? |
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| 01:16.26 | pools0n | dudes ! |
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| 01:26.56 | tompaw | Anyone Are there any reasons not to deploy Asterisk @ EC2? |
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| 01:40.23 | MaliutaLap | chemtrails? that sounds like a good reason |
| 01:40.59 | tompaw | well, old (2012-2013) posts complain about bandwidth issues / not enough cpu power / funky kernels |
| 01:41.23 | MaliutaLap | maybe deploying asterisk in ec2 can give you autism ... there's no evidence to show it doesn't, I think it's been covered up by "big data" |
| 01:42.15 | MaliutaLap | tompaw: "funky kernels" did they have james brown do their kernel compilation? |
| 01:42.39 | tompaw | mostly, and then some asian guys kept releasing a "1000 Hz" kernels designed to work with asterisks |
| 01:42.47 | tompaw | but that's all up to 2013, no recent complaints. |
| 01:43.48 | MaliutaLap | not even about the chemtrails? |
| 01:44.04 | tompaw | suprisingly, not. |
| 01:44.16 | MaliutaLap | the autism? |
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| 01:44.41 | MaliutaLap | was ec2 taking their guns away? |
| 01:45.05 | tompaw | I feel like this conversation can give me autism, so I'll take it as a "no" :> |
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| 01:52.02 | tompaw | Right, so I'd have to turn my neat transport configuration based on 2 separate (lan/wan) network interfaces: https://bpaste.net/show/a9723502c560 |
| 01:52.16 | tompaw | Into a EC2-NAT one... |
| 02:01.05 | tompaw | I guess in this case I need to use one transport for both lan and wan endpoints, like this: https://bpaste.net/show/d71bcdf99e3a ? |
| 02:01.27 | tompaw | (not sure if local_net accepts comma separated networks) |
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| 02:18.43 | MaliutaLap | tompaw: it doesn't |
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| 03:51.33 | tompaw | MaliutaLap: it seems that it does. |
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| 05:11.29 | BitSlayer | can anyone help me with new setup, I've read docs, googled my brains out and still can't get incoming or outgoing calls working. i'm using a sipgateway as a trunk. the docs examples use a "context" which I changed to match my extension but it still doesnt work, error logs say the incoming call can't go to the extention because it's not in the context⦠augh |
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| 05:31.25 | pools0n | make some routes |
| 05:33.18 | [TK]D-Fender | He's gone |
| 05:33.27 | [TK]D-Fender | and the concept of "routes" should not be a thing |
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| 06:25.09 | pools0n | its a thing ! |
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| 08:26.40 | guest8831 | when changing from mysql to odbc, do I need to restart asterisk? |
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| 08:30.38 | Chainsaw | guest8831: I would, just to ensure all database connections are freshly remade with the new parameters. |
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| 10:02.39 | enoch | hi all, is it possible to pass a variable to "channel originate" and then pass that variable to an agi script? |
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| 11:15.03 | enoch | channel originate sip/Test/<number> application EAGI test ?params |
| 11:15.11 | enoch | how to pass params to my test application? |
| 11:27.44 | enoch | ... |
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| 12:17.16 | enoch | the best text to speech module to use? |
| 12:23.34 | Tim_Toady | asterisk comes with festival support, and also there are other 3rd party options |
| 12:23.45 | Tim_Toady | how would you define best? |
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| 12:34.30 | enoch | Tim_Toady: something like the google quality? |
| 12:34.40 | enoch | how can i improve festival quality |
| 12:34.41 | enoch | ? |
| 12:35.22 | Tim_Toady | well, there is this: http://zaf.github.io/asterisk-googletts/ |
| 12:35.31 | Tim_Toady | but you have to be aware of the limitations |
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| 12:35.49 | Tim_Toady | at the end of the page it has some more links |
| 12:42.45 | enoch | i tried it but i want something "official" |
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| 12:55.49 | jr219 | Good morning! I am working on an application which uses WaitForNoise, and I cannot figure out why it is not working as expected.. |
| 12:57.21 | jr219 | Even with constant talking on the stream, it seems to never keep counting the ms of noise. There is always periods of silence detected even when there are none.. Iâve looked at changing the dsp time up to 250ms, but that makes no difference. Iâve looked at the code and it seems that sometimes no audio frames are delivered and it counts that as silence and thinking that perhaps it shouldnât? |
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| 15:31.35 | huxley | So a collegue of mine, has encountered random clock-drifting, when trying to run an Asterisk server (5000 connected calls at peak times), inside LinuxKVM/qemu and Vmware |
| 15:31.53 | huxley | And I've been trying to help him come up with work-arounds or solutions |
| 15:32.59 | huxley | Everything works flawlessly on bare-metal OS installs |
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| 15:33.36 | huxley | but it would be better from a business perspective, if he could virtualize his configuration |
| 15:33.58 | huxley | Love to hear people's thoughts on this :) |
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| 16:01.12 | stefan27 | regarding joinempty option on queues, what am I supposed to set joinempty to if I want callers to enter the queue no matter what. I tried setting joinempty to the empty string and I seem to get the desired effect but app_queue warns me WARNING[6624]: app_queue.c:2950 parse_empty_options: Unknown option for 'joinempty' |
| 16:02.59 | [TK]D-Fender | Show us the config |
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| 16:08.04 | anthonym | Hey guys, I can't find much documentation on this.. core show file formats clearly shows mp3 is supported, MixMonitor() records flawlessly with .wav extension, change it to .mp3 however and the filesize is always zero. I can't find any documentation on why this would be happening |
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| 16:08.33 | file | format_mp3 only allows reading |
| 16:08.46 | anthonym | ahh, that'd explain that then |
| 16:08.58 | anthonym | so i'd just need to push the file off to a converter after the recording is complete |
| 16:09.07 | [TK]D-Fender | Correct |
| 16:09.14 | [TK]D-Fender | format_ = reads |
| 16:09.18 | [TK]D-Fender | codec_ = transcodes to |
| 16:10.06 | anthonym | excellent .. Another question, my old call recorder the .wav files are much smaller in ratio to call length, on my new recorder that I've just configured, the same length call .wav file is much much larger. Is there any direction you could point me to for changing settings to reduce the file size? |
| 16:10.40 | [TK]D-Fender | File size is directly from the format you choose |
| 16:10.48 | anthonym | basically got a 4 port digium card, 30chan pri, 10chan pri, DAHDI\ -> MixMonitor -> PBX and reversed, exact same hardware setup as the old recorder |
| 16:10.48 | [TK]D-Fender | Asterisk has not changed specs on this |
| 16:10.53 | anthonym | ah okay cool |
| 16:11.00 | [TK]D-Fender | wav != WAV |
| 16:11.07 | anthonym | gotcha. |
| 16:12.43 | anthonym | Perfect, I changed the file extension from wav to WAV, and it's much much smaller. |
| 16:13.37 | anthonym | thank you for your assistance, it is greatly appreciated. KR. |
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| 16:51.46 | Get_The_Fish | For music to sound decent when played through the PSTN, it needs to be "encoded" at 44.1 khz 8 bit mono, correct? |
| 16:52.12 | Get_The_Fish | "encoded" for lack of proper terminology |
| 16:52.18 | [TK]D-Fender | Clearly not |
| 16:53.08 | Get_The_Fish | Uh, ok?? |
| 16:53.26 | [TK]D-Fender | What is the sampling rate of what you use to communicate with the PSTN? |
| 16:53.48 | [TK]D-Fender | There is a CODEC used in that call.... |
| 16:54.00 | [TK]D-Fender | Go look at that spec |
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| 16:56.45 | Get_The_Fish | However once its on the PSTN it's G711. |
| 16:57.06 | [TK]D-Fender | Then you have your answer |
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| 17:54.34 | DivideBy0 | tompaw: did ec2 work out ok? |
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| 18:50.33 | tompaw | DivideBy0: still testing it. |
| 18:51.10 | tompaw | DivideBy0: My pjsip config seemed to worked fine, I am now looking at psql latency for real time pjsip configuration (I store Contacts, Endpoints, AORs, etc. in a remote database) |
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| 18:53.18 | tompaw | DivideBy0: it will take a few days of live testing with production agents before I can draw any conclusions, though. I am considering a "cluster" setup if psql latency turns out to be an issue (i.e. each geographical location will have its own local db). Also, I am looking into Postgresql BDR at the same time. |
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| 19:27.53 | huxley | Has anyone here had experience configuring a multi-tenant Asterisk solution? Either through containerization or virtualization of separate Asterisk instances/servers on the same hardware? |
| 19:41.09 | [TK]D-Fender | VM = VM |
| 19:41.22 | [TK]D-Fender | It works if you do thing right and the loads are reasonable |
| 19:41.41 | [TK]D-Fender | Forget the idea of multiple instances on the same |
| 19:48.38 | tompaw | In most cases you don't really know how many instances are running on a given piece of hardware. |
| 19:49.05 | tompaw | Say you get an EC2 instance, unless it's a dedicated host, you can't control what other people are doing next to you. |
| 19:49.56 | huxley | its all physical bare-metal boxes in a colo |
| 19:50.17 | huxley | there's an average of 5000 connected calls, and about 10000 active dials going on |
| 19:50.21 | tompaw | [TK]D-Fender: when you say multiple instances on the same, do you mean multiple VMs on the same server, or multiple asterisks on the same VM? |
| 19:52.02 | huxley | I'm new to Asterisk, but the gentleman who configured and setup everything - and has been running it for a few years - seems to feel it is an above-average workload for asterisk |
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| 19:52.17 | huxley | and he claims to have tried to virtualize in the past |
| 19:52.24 | huxley | without success |
| 19:52.53 | huxley | clock drifting and a synchonrization of the audio becoming the biggest challenge, both of which are not an issue with a single OS installed onto the bare metal |
| 19:53.10 | huxley | but even that same server with a single VM instance running with all available memory and resource, is an issue |
| 19:54.05 | huxley | I was just hoping to get some ideas from everyone if he could migrate into a multi-tenant solution - because of the several Asterisk servers he has, they are call-centers from different parts of the world, and lots of CPU and memory resources are being wasted sitting idle |
| 19:58.35 | tompaw | I'm in a slightly different situation: I have a cluster of Asterisk servers as a part of our ARI-based call center solution. I'm looking for help in profiling my 13.5 configuration, as the performance I am getting from them is quite poor. |
| 20:00.27 | huxley | Cool - I'm not familiar with ARI, but I hope you get the profiling figured out |
| 20:03.14 | tompaw | I am trying to pin point which part of my process might be the bottleneck. It's all based on ARI-controlled bridges with audio recording. There's no transcoding (apart from the bridge itself) (the load hasn't changed between me using different codecs in the bridge vs using the same one, though). |
| 20:03.55 | tompaw | At the moment I am hitting 50% load with 10 calls on a 6c E5, which is really really bad. |
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| 20:52.10 | huxley | tompaw: hmm, that does indeed sound like low performance |
| 20:52.49 | huxley | Have you investigated the general memory and cpu usage statistics during this time? and looked at 'vmstat' and 'iostat' to determine where some of the bottlenecks might be indicating a problem? |
| 20:53.58 | tompaw | huxley: yes, it doesn't seem to be related to any particular resources like ram or io |
| 20:55.01 | huxley | what does vmstat show? how many processes are in wait and how many in sleep? |
| 20:55.22 | huxley | where are you measuring the '50%' load ? |
| 20:55.48 | tompaw | All i see is a nice asterisk process eating up CPU. It works fine, segfaulting ~ once / 48hours (looking at the dump, it seems like jasson might be the cause, we'll be testing with stock packages vs source builds next week). |
| 20:55.54 | tompaw | huxley: let me grab that for you |
| 20:56.21 | tompaw | huxley: https://bpaste.net/show/7ca2e61080ed |
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| 21:00.01 | tompaw | huxley: this is what one of these systems looks like: http://prntscr.com/a5ghl6 |
| 21:00.37 | tompaw | as you can see, there's hardly any ram/io activity, it's all CPU |
| 21:02.53 | huxley | interesting |
| 21:03.18 | huxley | I would strace the process then, and see what system calls are occuring to get a better idea what is happening |
| 21:03.44 | huxley | it seems very odd |
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| 21:04.36 | huxley | https://newspaint.wordpress.com/2013/07/24/how-to-diagnose-high-sys-cpu-on-linux/ is a clever way to get some details about what is happening |
| 21:04.48 | tompaw | This is what all my calls look like: https://bpaste.net/show/abde9d01df09 |
| 21:05.08 | tompaw | I wonder if this WriteTranscode/ReadTranscode has something to do with it. |
| 21:05.51 | tompaw | (I assume transoding is there because all my calls are in multi-party bridges, even if there's only 2 people talking - it's set up this way so that supervisors can easily jump on a call or pull another agent in) |
| 21:06.22 | huxley | Yeah you should get the list of the process's threads using that technique or something similar, in the aforementioned wordpress blog post |
| 21:06.40 | tompaw | huxley: thanks, I will try - do you know if strace can distrupt the operation of a process? My admin guru is on a plane crossing the atlantic now and we'll be experimenting next week :-) |
| 21:09.07 | huxley | It certainly can and does affect the process - http://www.brendangregg.com/blog/2014-05-11/strace-wow-much-syscall.html -- I would be careful using it in production of course, but it should help to tell you precisely where the problem lies. |
| 21:11.14 | huxley | I have to idle for a while, but I wish you the best of luck in figuring out your situation! |
| 21:11.25 | tompaw | https://bpaste.net/show/e48797ad2c2e - this is what is showed me, 99% "poll" |
| 21:11.29 | tompaw | huxley: thank you, happy idling! |
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| 21:37.13 | DivideBy0 | thanks tompaw |
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| 22:02.55 | BitSlayer | can anyone help out a newbie? yes I've read the docs til my eyes bleed⦠|
| 22:04.55 | BitSlayer | need help with the whole trunk to extension thing, it says context is bad on incoming calls.. |
| 22:05.54 | [TK]D-Fender | Almost certainly doesn't say that |
| 22:06.16 | [TK]D-Fender | It may say that it can't find a matching EXTENSION in the context the call is falling under |
| 22:06.31 | [TK]D-Fender | So look at where the call is falling |
| 22:06.44 | [TK]D-Fender | And show us what you think it should have succeeding in matching |
| 22:06.47 | [TK]D-Fender | ~pb |
| 22:06.47 | 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. |
| 22:06.49 | [TK]D-Fender | ^^^ |
| 22:07.56 | BitSlayer | D-Fender yes, the extention doesnt match the context. i dont understand how the trunk should be setup to be related to the extension. I want incoming calls to go to IVR not directly to extension. when i setup the extension it says it should use the default "dont change this unless you know what you're doing" and i certainly dont |
| 22:08.12 | BitSlayer | yes i know about pastebin |
| 22:08.18 | [TK]D-Fender | It's your dialplan... |
| 22:08.23 | [TK]D-Fender | it has whatever you made in there |
| 22:09.16 | [TK]D-Fender | Show us what it is looking for... and what you made to match it |
| 22:12.45 | BitSlayer | ok i think that's where it's failing. i setup a trunk in isolation, and ivr and an extension, but I havent done any "glue" to connection them. and i don't understand how that works. i'm looking at sample dialplans that dont make any sense to me |
| 22:13.12 | [TK]D-Fender | Dialplan is 90% of Asterisk |
| 22:13.31 | [TK]D-Fender | If you don't understand that... then you're close to nowhere at all |
| 22:13.43 | BitSlayer | yep i dont understand that all :( |
| 22:14.07 | [TK]D-Fender | Setting up a SIP section entry = peanuts. How do you handle all of the calls you'll be getting? That's dialplan and that's where all your work really ends up going |
| 22:14.27 | [TK]D-Fender | Well... you don't even have a specific question about it... so it sounds like you need your basics |
| 22:14.28 | [TK]D-Fender | ~book |
| 22:14.28 | 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 |
| 22:14.29 | [TK]D-Fender | ^^^ |
| 22:16.26 | BitSlayer | ok, well that's going to take me forever :( |
| 22:17.01 | [TK]D-Fender | You don't have even a specific question. How can we help with that? |
| 22:18.05 | [TK]D-Fender | From what you've said I would be assuming you have no dialplan whatsoever and a completely non-functioning system |
| 22:22.02 | BitSlayer | ok, how do I create the dialplans for what I need then? |
| 22:22.49 | BitSlayer | i thought this would be easier. i just wanted calls to come into ivr then dial an extension. and dialing out, extension to dial like normal |
| 22:22.57 | BitSlayer | but there seems to be a lot of "glue" that goes into that |
| 22:23.25 | BitSlayer | anyone willing to take a look? |
| 22:23.47 | [TK]D-Fender | What are you even calling an "extension" |
| 22:23.59 | [TK]D-Fender | I've already asked you to SHOW us what you make |
| 22:24.12 | [TK]D-Fender | Why are you asking us to take a look .. and not SHOWING us what you made like I asked? |
| 22:28.51 | BitSlayer | extensions = Applications >> extensions. which files do you want me to post to pastebin? sip.conf, extensions.conf, and asterisk.conf? |
| 22:29.24 | [TK]D-Fender | That is NOT Asterisk |
| 22:29.26 | [TK]D-Fender | That is a GUI |
| 22:29.30 | [TK]D-Fender | which is not supported here |
| 22:29.40 | [TK]D-Fender | This is something you should have specified walking in the door |
| 22:29.53 | [TK]D-Fender | ~freepbx |
| 22:29.53 | 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 |
| 22:30.03 | [TK]D-Fender | Bring your questions over there |
| 22:30.24 | BitSlayer | ok thanks for your help though, sorry i'm very new, been trying to read |
| 22:30.24 | [TK]D-Fender | we can pick this up in there |
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| 22:43.43 | anahata | Is it possible to set default parameters for all pjsp AORs? Otherwise, is it possible to add to an AOR configuration file with a syntax similar to [0101](+) in a file that gets loaded later (using freepbx) |
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| 23:10.14 | [TK]D-Fender | that isn't "default" |
| 23:10.25 | [TK]D-Fender | You should be able to use templates AFAIK |
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| 23:49.30 | anahata | You talking to me D-fender? |
| 23:50.28 | anahata | Does anyone know how to set a JITTERBUFFER in all communications |
| 23:50.31 | anahata | Does anyone know how to set a JITTERBUFFER in all communications? |
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| 23:55.46 | anahata | Or how to set JITTERBUFFER on a cofenrence room? |
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