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00:46.49 | hackish | Every time my internet goes down or my asterisk server is down my iax2 won't auto register with the provider. |
00:47.29 | hackish | iax2 show registry Give me: (null) N 981999999 <Unregistered> 60 Unregistered |
00:47.49 | hackish | obviously I changed the real phone number... Google isn't giving any hints. Ideas? |
00:48.52 | hackish | if I try to call me I get the dreaded call can't be completed as dialled. |
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01:14.43 | cneb3000 | Hi everyone.. hate to be the guy who comes in the room and barks out for some advice but could do with some help! |
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01:15.14 | cneb3000 | I have an issue involving media not being correctly established on a call. here's a flow diagram of the call - http://imgur.com/0Nh6JyO |
01:15.30 | cneb3000 | as you can see, when we are asked to make the call, we immediately respond with a 100 trying, as normal |
01:15.47 | cneb3000 | but then we immediately, send a 200OK, at which point the person calling acknowledges with an ack |
01:15.50 | cneb3000 | and media is sent |
01:16.24 | cneb3000 | in the mean time, our carrier has informed us that the call is now rining. and when it is eventually answered (183 in progress) the media is all over the place. and eventually the call just gets cancelled |
01:16.52 | cneb3000 | we have another caller using the same trunk, same details etc. with no issues. and both clients have the same config |
01:17.14 | cneb3000 | I didn't know whether there was something in this clients invite, that was provoking us effectively telling them to send media early |
01:17.40 | cneb3000 | ..any advice would be appreciated, cause im stumped if theres some config in asterisk to help get around this! |
02:02.48 | cneb3000 | bah.. thanks anyway everyone :( |
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02:37.24 | wrouesnel | has anyone encountered an issue where a Sipura SPA-3000 or similar will drop call audio if it tries to reregister with Asterisk when the call is picked up? |
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04:05.01 | ChannelZ | cneb3000: would rather see a SIP debug of the call with verbose console as well |
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04:39.17 | wrouesnel | ok is there a way to filter manager logons out of the cli and verbose logging? I don't need to see "manager logon/logoff" every few seconds |
04:39.29 | wrouesnel | setting displayconnects=no doesn't seem to work |
04:39.36 | sawgood | wrouesnel: good question and point ... |
04:39.54 | sawgood | wrouesnel: you'll see that is more out of scope (could be) FPBX related ... |
04:40.22 | wrouesnel | it's coming from a small web app I use to monitor phone connections |
04:40.53 | sawgood | Oh for me, I have a different experience (sorry) |
04:41.12 | wrouesnel | if i drop the verbosity way down (to 1) it's fine, but defeats the point of being able to log my dialplans progress |
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05:01.51 | kaldemar | wrouesnel: the setting won't take effect on a "manager reload". have you restarted asterisk? |
05:05.49 | enyawix | is ignorecryptolifetime=yes supported in Asterisk 11? was a patch to get option in 1.8 |
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05:22.26 | wrouesnel | kaldemar: not yet. it's a work day so I'll try it this evening i guess. |
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06:32.17 | bulkorok | hi :-) |
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07:00.15 | AndyN_ | Hi guys, is it ok if I ask questions here? |
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07:05.32 | bulkorok | ~ask |
07:05.32 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
07:06.53 | AndyN_ | I'm having issue with Snom phone cannot pick up another extension call after upgrading from 1.2 to 1.8. |
07:07.09 | AndyN_ | My set up: Asterisk 1.8.22.0, Snom 360 firmware v5 |
07:08.13 | AndyN_ | I can see the BLF light blinking, but pushing it show 'connected' but the other phone will just keep ringing and basically nothing happens |
07:09.02 | AndyN_ | extra info please let me know |
07:10.25 | kaldemar | you should tell how you have implemented pickup, what the button on the phone is supposed to do and show an asterisk CLI output of all that with verbosity and sip debug enabled. |
07:12.50 | AndyN_ | roger |
07:13.04 | AndyN_ | In asterisk sip.conf I used this |
07:13.06 | AndyN_ | notifyringing=yes notifyhold=yes allowsubscribe=yes callcounter=yes counteronpeer=yes call-limit=99 |
07:13.26 | AndyN_ | then use Hint() in extensions.conf |
07:14.06 | AndyN_ | on Snom 360 I set up Function keys to monitor the other exts. This works perfect in asterik 1.2 |
07:15.21 | AndyN_ | By 'monitor' I meant if a call comes direct onto ext 100, a person sitting at his desk with ext 101 could push the Function key and 'steal' that call from 100 |
07:16.38 | kaldemar | counteronpeer is not a valid option for anything in sip.conf (unless your asterisk has been patched), what's that about? |
07:17.14 | kaldemar | by "monitor" you really mean a subscription from the phone to asterisk, if we speak in SIP terms. :) |
07:19.40 | AndyN_ | yes kaldemar, that's subscription, ple pardon my ignorance |
07:20.51 | AndyN_ | I picked up counteronpeer in an online guide, should I remove it? |
07:21.21 | kaldemar | feel free to remove it. |
07:21.44 | AndyN_ | yes will do |
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07:24.12 | AndyN_ | kaldemar anything else I could double-check? |
07:24.36 | kaldemar | the CLI output with verbosity and sip debug. |
07:24.40 | kaldemar | ~pb |
07:24.40 | 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. |
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07:31.14 | AndyN_ | Thank you. Got it - http://pastebin.com/b7sdPEX7 |
07:31.41 | AndyN_ | notice how it shows SIP/2.0 401 Unauthorized. What do you think? |
07:35.22 | kaldemar | 401 is a normal part of authentication |
07:36.34 | kaldemar | i'd be more concerned about the 404 on the subscription. |
07:37.03 | kaldemar | "Looking for 789 in internal-extensions (domain )" => "404 Not Found" |
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07:39.05 | AndyN_ | ok. 789 is no longer used |
07:41.37 | kaldemar | that's what something is still subscribing for. is it the phone that's at 192.168.3.30? there's no "User-Agent" header in the request. |
07:42.42 | kaldemar | if the phone is at 192.168.3.65, your pastebin contains no useful information. just a single qualify (keep-alive) with OPTIONS. |
07:43.50 | AndyN_ | kaldemar, Thank you. I gotta leave now but will jump back onto this later. Really appreciate your help |
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10:24.11 | cneb3000 | Hey everyone. this sounds stupid - but never had to post on an asterisk forum for help. Either get it from here on the book, but both have let me down ;) |
10:24.18 | cneb3000 | what is the go to forum for asterisk to ask tech questions? |
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10:36.30 | wdoekes | ~ask |
10:36.31 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
10:38.13 | cneb3000 | sure I get that, I asked last night and I don't think anyone could help :) but thanks, I'm just goign to use the digium forums |
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11:16.37 | hrolf | Hey, I'm getting a tick (sort of) sound in the start of each file, whenever I Playback. |
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11:17.12 | hrolf | When I listen to the same files in a player, they are fine. But in Asterisk I hear this sound before a file starts playing back. |
11:17.18 | hrolf | How do I debug this? |
11:20.55 | cneb3000 | hrolf: when you play them back in a media player |
11:21.02 | cneb3000 | are they EXACTLY the same files that are in asterisk? |
11:21.18 | cneb3000 | for example, most people will get a file as an mp3. and when they put it in asterisk,. covert it gsm.. g711.. wav etc. |
11:21.51 | cneb3000 | because when you convert to gsm etc. it for the most past is converting it to lower quality audio. and this might be bringing out problems in the audio that you may have otherwise not seen |
11:22.33 | cneb3000 | for example, if you use an audio editor to cut a song in half. It actually will leave a little 'spike' on the end/beginning. but you wont be able to hear this, because the spike is represented with all of its frequencies, and sort of blends in |
11:22.45 | cneb3000 | but if you compress the same file, it removes a bunch of the frequencies, and the spike is more audible |
11:22.53 | cneb3000 | it might be something else though :) |
11:22.57 | hrolf | cneb3000: Yes, they are the same files. .wav files to be specific. |
11:23.14 | hrolf | cneb3000: Bit-rate: 128kbps |
11:23.35 | hrolf | cneb3000: Sort of a like click sound |
11:24.05 | cneb3000 | would you mind sharing the file? |
11:24.11 | hrolf | Yes |
11:24.16 | hrolf | cneb3000: Let me do it. |
11:24.17 | WIMPy | Metadata? |
11:24.44 | hrolf | cneb3000: Where do I upload it? (If it matters.) |
11:25.29 | cneb3000 | don't matter lol try http://vocaroo.com/?upload ? |
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11:26.53 | hrolf | cneb3000: http://vocaroo.com/i/s1t95LV6A0lW |
11:27.27 | cneb3000 | and u hear the click at the beginning yes? |
11:27.31 | hrolf | cneb3000: Yes. |
11:27.40 | cneb3000 | ok let me have a look. i cant promise anything :) |
11:27.50 | hrolf | cneb3000: Each time a file is played. It is same with all my files. |
11:28.04 | hrolf | cneb3000: All are .wav, 128kbps. |
11:28.10 | hrolf | cneb3000: Yeah okay, no problem. |
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11:30.00 | hrolf | cneb3000: http://stackoverflow.com/questions/5326135/artificat-on-playing-a-audio-file-on-a-phone-from-a-asterisk-server |
11:30.18 | cneb3000 | yea, thats the same thing i'm talking about |
11:30.24 | hrolf | cneb3000: I think it is the same problem as above, but I don't understand what it is saying, anyone clarify? |
11:30.35 | cneb3000 | sure |
11:31.03 | cneb3000 | basically, like I said, you may have a small bit of audio at the very beginning of the file. and whatever is playing the audio, is aplifying it. and you get a click |
11:31.17 | cneb3000 | im just going to upload the file again for you. try it and see if its any better |
11:31.25 | hrolf | cneb3000: Okay. Thanks. |
11:31.46 | cneb3000 | hrolf: http://vocaroo.com/i/s1i5zF43rGpM try that :) |
11:31.56 | cneb3000 | i've added a very quick 'fade in' at the beginning |
11:32.04 | cneb3000 | if it doesnt work then I'm not sure what the problem is sorry :( |
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11:34.35 | hrolf | cneb3000: Dropping incompatible voice frame on SIP/999-0000001e of format ulaw since our native format has changed to 0x2 (gsm) |
11:34.50 | hrolf | cneb3000: WARNING[2396]: format_wav.c:103 check_header_fmt: Unexpected frequency 22050 |
11:34.53 | cneb3000 | oh sh** i didnt convert it to a proper asterisk friendly format |
11:35.17 | cneb3000 | ttry http://my.digium.com/en/products/ivr/audio-converter/ |
11:35.24 | hrolf | http://my.digium.com/en/products/ivr/audio-converter/ |
11:35.58 | hrolf | cneb3000: Which format do I convert it to in order to be "Asterisk-friendly" ? |
11:36.27 | cneb3000 | go to that site, convert it to 8khz |
11:37.18 | hrolf | cneb3000: Yes, it is gone now. |
11:37.26 | cneb3000 | ahh that fixed it? :) |
11:37.30 | hrolf | cneb3000: What did you do? And how do I make the rest of my files? |
11:37.33 | hrolf | cneb3000: Yes. |
11:37.38 | cneb3000 | awesome, to secs ill find a link |
11:37.54 | cneb3000 | have a look at this http://www.youtube.com/watch?v=Xa9S-bel9P4 |
11:38.02 | cneb3000 | i basically used a piece of audio software called audacity |
11:38.10 | cneb3000 | and added a quick fade in to the beginning |
11:38.28 | cneb3000 | the fade in, got rid of that little click at the beginning, cause it made it quiet |
11:38.46 | hrolf | cneb3000: Okay, let me give it a look. |
11:38.52 | cneb3000 | good luck! |
11:39.11 | kaldemar | sox can be used to batch-process files using the "fade" filter. |
11:39.42 | kaldemar | maybe a little easier and faster than editing a bunch of files by hand using audacity. |
11:40.09 | hrolf | kaldemar: Thanks, I'll also give it a try. |
11:44.18 | hrolf | cneb3000, kaldemar: What length of fade-in do I give? |
11:47.32 | kaldemar | hrolf: try what works for you. |
12:04.48 | cneb3000 | hrolf: a very small one |
12:04.59 | cneb3000 | like, i faded in to where it actually starts if that makes sense |
12:05.33 | hrolf | cneb3000: Tried 170 ms and it worked fine. |
12:05.42 | hrolf | cneb3000: Thanks. |
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13:19.27 | jmetro | .net .split |
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13:19.38 | jmetro | maybe rivaling peer in terms of disconnects |
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13:22.08 | Katty | morning |
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14:37.24 | Katty | pesters Qwell |
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14:37.54 | Katty | hi tzafrir |
14:38.21 | tzafrir | hi |
14:38.27 | Katty | how're you dear |
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14:39.24 | tzafrir | you're not confusing me with tzanger, right? |
14:39.24 | tzafrir | and quite OK |
14:39.38 | Katty | no i'm not (= |
14:39.44 | Katty | and i'm glad you're doing just peachy! |
14:39.56 | Katty | tzanger: but hello to you too :P |
14:40.17 | tzafrir | enjoying the weekend |
14:40.24 | Katty | you get a 4 day weekend? |
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14:40.31 | Katty | hugs slav3_kitten |
14:41.01 | slav3_kitten | yo Katty |
14:41.04 | slav3_kitten | snug purrs |
14:41.25 | Katty | slav3_kitten: what's shakin |
14:41.44 | slav3_kitten | sitting in pittsburgh at anthrocon |
14:42.46 | Katty | what's anthrocon? |
14:42.53 | jmetro | furry convention |
14:43.16 | Katty | ah right. |
14:43.19 | sruffell | slav3_kitten: really? |
14:43.28 | Katty | slav3_kitten: hope you're having a grand ol time! |
14:43.33 | Katty | slav3_kitten: each some junk food for me |
14:43.47 | Katty | hugs jmetro |
14:43.49 | jmetro | i have a friend that goes to MFF |
14:43.50 | jmetro | Hi katty :3 |
14:44.10 | Katty | jmetro: i see you survived the fireworks |
14:44.16 | Katty | fistbump jmetro |
14:44.38 | Katty | gmorning sruffell (= |
14:44.48 | slav3_kitten | yea sruffell |
14:44.49 | sruffell | waves |
14:45.01 | slav3_kitten | i'll eat all the junkfood for you katty |
14:45.05 | slav3_kitten | ALL OF IT! |
14:45.06 | Katty | woohoo! |
14:45.08 | sruffell | hah…that's cool. For work or just personal interest? |
14:45.59 | sruffell | Probably one of the only convention with a page on Behavior Around Fursuiters. http://www.anthrocon.org/behavior-around-fursuiters |
14:47.14 | jmetro | Katty: I did! *fistbumps* My doggies almost didnt. |
14:47.23 | jmetro | They were shaking so hard. |
14:48.32 | slav3_kitten | personal interest sruffell |
14:48.58 | Katty | jmetro: poor things )= |
14:49.04 | slav3_kitten | been a furry for a long time |
14:49.08 | Katty | slav3_kitten: did you get all dressed up for it? |
14:49.24 | jmetro | Katty: We wrapped them in blankets and made a fort for them =) |
14:49.32 | slav3_kitten | i got my skirt, hosiery, tail, and ears on |
14:49.41 | slav3_kitten | so yea totally dressed up for it |
14:49.47 | Katty | slav3_kitten: woohoo!! |
14:49.53 | sruffell | nice! |
14:49.58 | Katty | slav3_kitten: how long did the tail and ears take you to make? |
14:50.03 | Katty | slav3_kitten: or were they premade? |
14:50.19 | slav3_kitten | couple hours |
14:50.24 | Katty | jmetro: that was nice of you :> i wear that they make a sort of snuggly fit doggy wrap for such things. |
14:50.29 | Katty | err i hear. |
14:50.35 | slav3_kitten | long as hell pruple feathered fur with a green tip |
14:50.46 | slav3_kitten | i loves them so much |
14:50.51 | Katty | jmetro: i've seen them at petco for plane trips and what not |
14:50.59 | slav3_kitten | mistress is also here from england |
14:51.05 | Katty | slav3_kitten: that sounds pretty elaborate! |
14:51.09 | slav3_kitten | so it's been pretty fun so far |
14:51.49 | jmetro | Katty: They do, its called a Thundershirt. |
14:52.37 | Katty | jmetro: ah right. it's nice to know that a Thundershirt (or blanket wrap) will help |
14:52.46 | Katty | jmetro: should i ever get another doggy |
14:53.15 | slav3_kitten | Katty, yea it is. i may get a picture taken for you if you want |
14:53.25 | Katty | slav3_kitten: sure if you want! |
14:53.30 | Katty | slav3_kitten: you can send it to me in /query |
14:53.36 | slav3_kitten | tit for tat though :P |
14:53.44 | Katty | lol |
14:54.10 | slav3_kitten | and i'd not have it for a while. need to get all the pictures an shit sorted after con is over |
14:54.41 | Katty | whenever is fine, i just like crafty things! |
14:54.57 | slav3_kitten | so, want me in clothes or out? |
14:55.11 | Katty | you can just take a picture of the costume if you want. |
14:55.34 | Katty | or whatever crafty bits you're particularly proud of |
14:58.39 | slav3_kitten | well i'm quite fond of my crafty bits :P |
14:59.00 | Katty | ^_^ |
14:59.11 | slav3_kitten | if you know what i man |
14:59.13 | slav3_kitten | mean |
14:59.25 | slav3_kitten | anyhow. off to con! |
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15:13.57 | aberrios | slaps Rumbles |
15:14.10 | jmetro | ^ with a large trout? |
15:14.15 | aberrios | and that |
15:16.44 | Katty | ahh, the classic Dead Fish Greeting of IRC |
15:17.39 | WIMPy | Who said it has to be dead? |
15:18.31 | aberrios | usually wet, being a fish.. |
15:18.44 | jmetro | the fish flop adds extra force! |
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15:19.17 | Katty | WIMPy: well considering we started trout slapping in say, the early 90s... |
15:19.37 | Katty | WIMPy: either the fish is either dead, or older than some of the asterisk folks |
15:20.50 | WIMPy | Ohhhh, now it even has to be the same fish all the time. |
15:20.58 | Katty | hehehehe |
15:21.22 | [TK]D-Fender | FEELING LIKE A DEAD DUCK! |
15:21.28 | [TK]D-Fender | </tull> |
15:21.42 | Katty | good morning fenderbender. |
15:23.10 | jmetro | genderfender? |
15:24.12 | [TK]D-Fender | Katty: Mew. |
15:24.20 | Katty | [TK]D-Fender: how're you dear? |
15:25.07 | [TK]D-Fender | Katty: TGIF ... doing OK. Work running smooth, had meeting with band last night and ironed out or new material process, trimmed our existing list, and planned out the marketing responsibilities |
15:26.17 | Katty | [TK]D-Fender: wow, sounds like a full week |
15:26.47 | [TK]D-Fender | that was last night :) |
15:26.47 | Katty | [TK]D-Fender: we don't have a lot of band marketing out here, just fliers and what not posted about |
15:27.41 | Katty | good lord, do you never sleep!? |
15:28.01 | [TK]D-Fender | Katty: We ironed out the process for getting "staged" pics" for posters, getting multiple proposed official logo samples, etc. Then we'll be able to finalize the image on our FB page, build up a website, make posters & business cards, etc |
15:28.10 | WIMPy | You get enough sleep when you're dead. |
15:28.21 | Katty | [TK]D-Fender: ahh. |
15:28.37 | Katty | [TK]D-Fender: i hope your stage photos are humorious |
15:28.39 | [TK]D-Fender | Katty: We didn't actually DO the work, jsut laid out the path |
15:28.59 | Katty | always like seeing folks with a sense of humor |
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15:43.57 | tzanger | Katty: hello |
15:48.10 | igcewieling | screams obscenities at PHP |
15:50.33 | jmetro | igcewieling: Dont you talk that way to him! He has done nothing wrong! |
15:53.12 | cneb3000 | whats wrong with PHP? |
15:53.40 | WIMPy | prepares for a big spike in used bandwidth. |
15:54.12 | jmetro | a lot of people call it a "double claw hammer" because it "does everything wrong" |
15:54.26 | jmetro | but i call them "c++ addicts that like unneccessary outdated code languages" |
15:54.42 | Katty | stuffs a burger in igcewieling's mouth |
15:55.25 | igcewieling | cneb3000: I'm sure we'll make up soon the makeup coding will be awesome |
15:55.51 | jmetro | ^ seee |
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15:57.24 | igcewieling | PHP is like using a C Hammer, a POSIX screwdriver, a Perl saw all in one tool. It may not do anything perfect, but it does lots of stuff well. |
15:58.52 | [TK]D-Fender | more like "well .... it does lots of stuff" :) |
15:59.07 | jmetro | I'd rather code in PHP than look at C++ code. |
15:59.39 | jmetro | As "bad" as php is, keep in mind you could be working in c++ and feel good about yourself |
15:59.41 | WIMPy | The chinese multitool from the DIY maket. |
16:00.37 | igcewieling | "strings" (or their non-existence) in C is why I hate C. |
16:00.39 | mattp | When I join the conference (provided by MeetMe application), I get a short ~100ms jingle (it sounds a bit like a trumpet). Where does this come from? I've not found anything likely in the sound directory (but also don't know what I'm looking for) |
16:00.59 | jmetro | mattp: does it show in the console when you join? |
16:01.00 | igcewieling | mattp: either meetme.conf or the options passed to MeetMe |
16:01.13 | WIMPy | igcewieling: That was an easy one. But a massive one as well. |
16:01.40 | WIMPy | Probably the reason for at least 50% OF ALL CRASHES AND SECURITY ISSUES WE SEE. |
16:01.43 | WIMPy | oops |
16:01.47 | igcewieling | Hmm? |
16:02.08 | WIMPy | The other half is the pointer in general. |
16:02.18 | igcewieling | *nod* |
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16:04.28 | mattp | jmetro: no. |
16:05.51 | mattp | igcewieling: neither of those. But interestingly, I just got a spam call from an autodialler. When the human came on, I heard the same mini jingle. Coincidence that they use asterisk, perhaps, but it's an indication that the sound is one which comes bundled with it. |
16:09.33 | *** join/#asterisk teff (~teff@client-86-23-75-235.brhm.adsl.virginmedia.com) |
16:09.59 | *** join/#asterisk evilman_home (kvirc@89-179-77-66.broadband.corbina.ru) |
16:10.59 | *** join/#asterisk srulik (~sruli@82.152.156.64) |
16:14.21 | srulik | hi everyone, i am after a PBX with a particular feature, can anyone tell me if asterisk has this feature? |
16:14.21 | srulik | I want to call into my sip account, get a prompt for a pin and then get a dial tone to dial an external number (like a calling card) does asterisk have this feature? |
16:15.19 | *** join/#asterisk hehol (~Adium@2a01:198:71d:0:5c75:425a:4cc6:9e76) |
16:16.30 | WIMPy | srulik: No, Asterisk is not a PBX, but such a thing is easy to configure. |
16:16.32 | cneb3000 | hey srulik, you mean DISA? |
16:16.36 | *** join/#asterisk ipiera (~Paul@ipiera.plus.com) |
16:16.47 | srulik | i dont know what is DISA ;-) |
16:17.16 | cneb3000 | http://www.voip-info.org/wiki/view/Asterisk+cmd+DISA <--- 'direct inward system access' |
16:17.41 | srulik | cneb3000: thanks i will check that |
16:17.41 | cneb3000 | or https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_DISA |
16:17.42 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
16:17.51 | cneb3000 | ^^^ that will probably be more up to date |
16:18.09 | ChannelZ | core show application disa |
16:18.17 | ChannelZ | even more |
16:19.28 | *** join/#asterisk elmargol (~elmargol@host232-104-dynamic.249-95-r.retail.telecomitalia.it) |
16:19.34 | cneb3000 | hm. if I type that into the asterisk11 box right in front of me, it's not there. probably wasn't compiled in? |
16:20.40 | srulik | cneb3000: can this be setup from the web GUI? |
16:20.56 | cneb3000 | srulik: which web GUI? |
16:21.05 | cneb3000 | srulik: freepbx or something? |
16:21.06 | srulik | asterisk |
16:21.14 | srulik | yes freepbx |
16:21.27 | WIMPy | Ask the ones who did the GUI. Asterisk doesn't have one. |
16:21.36 | cneb3000 | srulik: asterisk is kind of like the underlying software as it were.. it does all the dialling and stuff. FreePBX is just a website to make it easier to use |
16:22.07 | *** join/#asterisk aruntomar (~Thunderbi@49.248.156.149) |
16:22.13 | WIMPy | "claims to make it easier" |
16:22.18 | cneb3000 | sure lol |
16:22.24 | cneb3000 | havn't used one in a while myself |
16:22.26 | srulik | i understand that, i have only started using linux 6 months ago (what a divorce from MS) so i am not yet good enough with command line |
16:22.36 | cneb3000 | haha srulik we have to start somehwere |
16:22.43 | cneb3000 | in Linux, get used to reading manuals |
16:22.54 | cneb3000 | because, there's a culture to learning to doing things yourself :) |
16:23.06 | cneb3000 | for example - http://wiki.freepbx.org/dashboard.action |
16:23.07 | srulik | am reading, still have a day job and family, cant read all in 1 day |
16:23.11 | cneb3000 | haha |
16:23.20 | cneb3000 | read that website. your DISA answers are in there somewhere :) |
16:26.10 | cneb3000 | srulik: I've never used freeswitch until recently. that also has a learning curve.. trust me, learning to read manuals is a good skill. |
16:26.33 | cneb3000 | espiecally since.. alot of the time.. some of this software is free, and the developers dont always have the time to delve into how to actually use the software :) |
16:26.53 | cneb3000 | (not referring to asterisk.. dont hit me..) |
16:29.49 | *** join/#asterisk aruntomar (~Thunderbi@49.248.156.149) |
16:31.55 | *** join/#asterisk peetaur2 (~peter@x2f10bec.dyn.telefonica.de) |
16:39.27 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.186) |
16:40.26 | cneb3000 | generally speaking - what would be the most efficient way of storing asterisk CDRs on a remote server. Assuming the server was on the same network (i.e. ping of < 1ms). |
16:40.37 | cneb3000 | am i right in saying storing them in a database as opposed to a csv? |
16:41.01 | *** join/#asterisk teff (~teff@client-82-26-202-36.pete.adsl.virginmedia.com) |
16:41.03 | cneb3000 | this is generally speaking.. i'm sure one type of database would be better than another. just looking for a push in the right direction :) |
16:43.00 | Greenlight | Depends on your requirements |
16:43.24 | Greenlight | If you need to access them and search them regularly then, sure a database is best |
16:43.45 | Greenlight | If you just need to know you have them somewhere, then archiving the csv's might be better |
16:44.41 | *** join/#asterisk Mattski (~MattN@cpc11-whit4-2-0-cust235.5-2.cable.virginmedia.com) |
16:44.49 | Greenlight | On a billing gateway, we store to local csv, local mysql, and remote mssql |
16:45.30 | Mattski | Good day, all. I hope you're well, and that your tankards are suitably filled. |
16:46.17 | ChannelZ | My whatnows? |
16:46.28 | cneb3000 | cheers greenlight |
16:46.30 | Mattski | Tankards, steiners, pint glasses...? |
16:46.32 | Mattski | ;) |
16:46.37 | Greenlight | Ahh pint glasses |
16:46.38 | ChannelZ | Oh. |
16:48.08 | Mattski | Nuts, forgot I left the ironing on - brb |
16:48.18 | ChannelZ | Ye Olde Water Cup next to me is, anyway. |
16:49.08 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.186) |
16:50.53 | *** join/#asterisk vlad_sta_ (~vlad_star@nat.canmos.ru) |
16:51.35 | Katty | Mattski: my tankard is empty :< |
16:51.47 | Katty | Mattski: but i did run across a hobnob recipe, so that makes it a bit brighter at least :> |
16:55.00 | *** join/#asterisk threesome (~threesome@ip-94-113-12-233.net.upcbroadband.cz) |
16:55.02 | *** join/#asterisk Defraz (~Defraz@24-116-129-18.cpe.cableone.net) |
17:08.07 | *** join/#asterisk YoMomma (~YoMomma@cpe-75-84-153-163.socal.res.rr.com) |
17:08.19 | *** join/#asterisk peetaur2 (~peter@x2f10bec.dyn.telefonica.de) |
17:16.37 | Mattski | Hobnobs! Chocolate ones would get me into a beartrap tbh |
17:17.54 | jmetro | Thin mints. |
17:19.13 | Mattski | They'd get me into a nuclear reactor. Thank God I've got pizza otherwise I'd be drooling right now.. |
17:19.20 | [TK]D-Fender | Thin Mints : Staple food of fat people.... |
17:21.22 | jmetro | http://weknowmemes.com/wp-content/uploads/2012/04/ate-4-boxes-of-thin-mints.jpg |
17:22.11 | *** part/#asterisk ipiera (~Paul@ipiera.plus.com) |
17:25.34 | *** join/#asterisk jagster` (~chatzilla@unaffiliated/jagster/x-9084543) |
17:29.30 | *** part/#asterisk srulik (~sruli@82.152.156.64) |
17:29.59 | Mattski | Wanna be that cat. |
17:34.59 | *** join/#asterisk dlynes_laptop (~dlynes@65.94.155.230) |
17:36.00 | *** join/#asterisk Addisk (~WhyYouLoo@pool-173-60-252-44.lsanca.fios.verizon.net) |
17:42.14 | Mattski | Do any of you fine people have a few minutes to hear me ask a question you've probably heard a million times...? You can beat me with sticks afterwards if it helps! |
17:44.37 | WIMPy | Do we have to wait til then? |
17:44.39 | WIMPy | ~ask |
17:44.39 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
17:48.13 | Mattski | Apologies. Here goes: I'm using Asterisk 1.8.21.0. I've got the server talking happily to Sipgate.co.uk and incoming calls work, but I can't place any outbound calls. It looks like I've fudged authentication somewhere along the line. Setup is: home PC -> Asterisk box -> Sipgate.co.uk. Debug shows failure to authenticate on invite. |
17:48.36 | *** join/#asterisk vlad_starkov (~vlad_star@nat.canmos.ru) |
17:48.57 | Mattski | I've been Googling it for a couple of days, but I'm missing something. (Probably something small and obvious.) |
17:50.43 | jagster` | mattski, download wireshark |
17:51.12 | jagster` | you can get a packet capture going, filter to sip and see what the response is to your invite |
17:51.28 | Mattski | Cheers, I'll give that a go now. :] |
17:51.50 | jagster` | once you know the problem, you can start working on the fix ;) |
17:51.51 | [TK]D-Fender | "sip set debug on" <-- from * CLI |
17:51.56 | [TK]D-Fender | do not waste time on external tools |
17:52.06 | WIMPy | Or upgrade to a version that tells you the result. |
17:53.37 | Mattski | [TK]D-Fender: I've set debug mode to on, and I see the following in the headers: SIP/2.0 407 Proxy Authentication Required |
17:54.05 | Mattski | I'm thinking I've got to tweak auth details somewhere, but I'm not entirely sure where. :< |
17:56.33 | [TK]D-Fender | Mattski: pastebin the ENTIRE failed attempt from the very beginning to the end. |
17:56.35 | [TK]D-Fender | ~pb |
17:56.35 | 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. |
17:56.37 | [TK]D-Fender | ^^^ |
17:56.51 | Mattski | Gotcha. 2 sec.. |
17:56.59 | cneb3000 | Mattski: have you seen this buddy - http://www.sipgate.com/faq/article/394/How_do_I_configure_Asterisk |
17:57.12 | jmetro | cneb3000: have you seen this buddy - |
17:57.13 | jmetro | ~book |
17:57.13 | 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 |
17:57.34 | cneb3000 | woah what did i do? |
17:57.53 | Mattski | I have seen that, and the config I'm using is based on that; I just can't make outbound calls for some odd reason. Back to PB.. |
17:58.07 | jmetro | i dunno i just like using the book link |
17:58.14 | cneb3000 | lol |
17:58.49 | cneb3000 | Mattski: you see the top line. "register => SIP-ID:SIP-Password@sipgate/SIP-ID". where did you put that in the sip.conf file? |
17:59.59 | cneb3000 | NORMALLY, most people put it in under the [general] context. so.. something like - http://pastebin.com/dDuVqLB5 |
18:00.11 | cneb3000 | but on that help link, i guess it doesn't really tell you that |
18:00.17 | [TK]D-Fender | Registration is not the issue |
18:00.20 | [TK]D-Fender | The call is ARRIVING. |
18:00.25 | [TK]D-Fender | Lets just get the debug |
18:00.48 | cneb3000 | you owe me a snickers if I'm right. |
18:02.00 | [TK]D-Fender | cneb3000: He already confirmed the call was coming in and getting a 407. |
18:02.07 | [TK]D-Fender | cneb3000: You lost that bet the moment you made it |
18:02.51 | cneb3000 | \0/ |
18:03.15 | Mattski | Sorry for the delay, here's what I could get: http://pastebin.com/4gFcbmgB |
18:03.47 | [TK]D-Fender | not the whole call. |
18:03.55 | [TK]D-Fender | "core set verbose 10" |
18:04.02 | [TK]D-Fender | We need to see it from the very start |
18:04.08 | Mattski | Ok, 2 ticks |
18:04.44 | Mattski | Oh arse, that's the Putty buffer misbehaving.. |
18:05.32 | Katty | takes the snickers from [TK]D-Fender and eats it |
18:05.44 | Katty | [TK]D-Fender: so sweet, you shouldn't have! |
18:06.20 | [TK]D-Fender | waits for the tranquilizers to work their way through Katty's system.... |
18:06.29 | [TK]D-Fender | idsposes of the needle |
18:06.31 | Mattski | H'okay, try this: http://pastebin.com/1qh4u1Zj |
18:06.59 | Katty | [TK]D-Fender: i could go for a nap. |
18:07.18 | [TK]D-Fender | Mattski: Still not the entire call. We don't see the start of the INVITE |
18:07.57 | *** join/#asterisk votetrev (~votetrev@8.29.129.84) |
18:08.23 | [TK]D-Fender | Mattski: the ENTIRE call. |
18:08.40 | [TK]D-Fender | Mattski: if this is you using another device to place a call... we should see THAT debug along with it. |
18:08.51 | cneb3000 | im off - good luck mattski |
18:08.51 | [TK]D-Fender | Mattski: When I say "entire call" I mean ENTIRE CALL. |
18:11.37 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
18:11.39 | Mattski | [TK]D-Fender: I'm using Linphone from a W7 box, there's nothing in the debug window except for keepalive messages. Hopefully this has got everything this time: http://pastebin.com/y5iaDx4s |
18:11.49 | Mattski | If not, make with the stick hitting. |
18:12.17 | [TK]D-Fender | What's with the ANSI junk in there? |
18:12.29 | [TK]D-Fender | It's polluted with crap... |
18:12.43 | Mattski | Sorry about that. Not having the best day ever, as you might've guessed |
18:14.41 | [TK]D-Fender | Mattski: idea : it seems to circle on that auth and you change the callerid prior. add "fromuser=[yourusername]", "sendrpid=yes", "trustrpid=yes" to you peer and retest |
18:16.07 | WIMPy | Or better set them to pai than yes. |
18:16.31 | [TK]D-Fender | Sure, why not... |
18:18.46 | Mattski | Still no joy. I'll just change the buffer settings for Putty while I'm at it |
18:20.25 | Mattski | Better. Latest paste -> http://pastebin.com/75kh10ys |
18:20.48 | WIMPy | Do you still (try to) change caller ID? |
18:21.55 | WIMPy | yes. |
18:22.07 | WIMPy | That doesn't work. |
18:22.17 | *** join/#asterisk italoros_ (~italoross@187.60.66.11) |
18:22.38 | WIMPy | Either you have to use SIPID/name or use P-preferred-identitiy. |
18:24.32 | Mattski | Ok, so what kind of syntax would I be looking at there? Set(CALLERID=SIPID) or similar? |
18:24.51 | WIMPy | Leave that part out until you've got your calls working. |
18:27.51 | Mattski | Ok, so how about: exten => _90[1-9].,1,Dial(SIP/SIPID-here@sipgate,30,trg) ? |
18:32.17 | *** join/#asterisk The_Miser (~miser@cblmdm72-240-216-74.buckeyecom.net) |
18:32.18 | [TK]D-Fender | no |
18:32.35 | [TK]D-Fender | you need to spccify the target # there |
18:32.39 | [TK]D-Fender | and you took that out\ |
18:32.57 | [TK]D-Fender | and r= evil |
18:35.45 | The_Miser | Greetings: I am using dahdi 2.3.0.1. Do recent versions have the ability to do per FXS port different ring frequencies on a TDM400P card? I have an old party line (54hz) phone I'd like to hook up, but before going through what is required (upgrading my install) want to be sure it will work. |
18:42.45 | [TK]D-Fender | The_Miser: Seriously doubt it... |
18:43.17 | [TK]D-Fender | the you might try to divvy up your system.conf and change the loadzone & tonezone between the ports you're specifying.... |
18:51.08 | jmetro | i love forging new territory with equipment that was designed to do what im trying to do but has 0 documentatin |
18:55.41 | The_Miser | Fender: Interesting. I did not realize that was possible. Time to read a bit. I could have *swore* I found a link that said it was possible with recent versions of dahdi and the wctdm module, but of course cannot now find it. (which is aggravating in itself). |
18:56.28 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.195) |
19:06.18 | *** join/#asterisk eject_ck (~Evgeniy@109.86.157.219) |
19:06.38 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.195) |
19:07.03 | eject_ck | Hi all, any suggestions for sip softphone for linux with g729 codec ? |
19:14.19 | *** join/#asterisk felipealmeida (~user@mvx-187-16-79-187.mundivox.com) |
19:17.45 | *** join/#asterisk caveat- (hoax@shell.bshellz.net) |
19:20.03 | *** join/#asterisk ujjain (ujjain@unaffiliated/ujjain) |
19:35.09 | ChannelZ | eject_ck: the non-free version of zoiper |
19:35.28 | ChannelZ | g729 being a licensed codec you're unlikely to find anything free |
19:36.15 | jmetro | convert it to g722 on your end, success. |
19:40.21 | *** join/#asterisk chris_n (~Chris@koha/developer/chris-n) |
19:42.49 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
19:47.52 | eject_ck | jmetro: what about bandwidth requirements? Only g729 works for me on 3G |
19:48.03 | eject_ck | I even ready to pay |
19:48.14 | *** join/#asterisk killown (~killown@pdpc/supporter/student/killown) |
19:48.29 | jmetro | probably the paid zoiper then. i only run g711 or g722 |
19:49.54 | ChannelZ | there's always gsm |
19:50.31 | [TK]D-Fender | [15:47]eject_ckjmetro: what about bandwidth requirements? Only g729 works for me on 3G <- iLBC does better on PLC for quality |
19:55.10 | eject_ck | ChannelZ: I'm using voip client over 3g data connection, why you are talking about gsm ? |
19:55.27 | eject_ck | [TK]D-Fender: what's PLC? |
19:56.03 | [TK]D-Fender | eject_ck: Googleable |
19:56.53 | *** join/#asterisk kresp0 (~kresp0@30.Red-88-6-105.staticIP.rima-tde.net) |
19:57.33 | jmetro | programmable logic controller? |
19:58.22 | eject_ck | <PROTECTED> |
19:59.02 | jmetro | pigmy life circus? |
19:59.04 | Katty | packet loss concealment. |
19:59.09 | Katty | that sounds like.. |
19:59.18 | Katty | you know what, nevermind. i'll just keep that one to myself ^_^ |
19:59.21 | eject_ck | well, what linux softphones support ilbc ? |
20:00.05 | *** join/#asterisk sgimeno (~sgmieno@22.Red-88-1-76.dynamicIP.rima-tde.net) |
20:07.23 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.184) |
20:08.05 | *** join/#asterisk Doom2k` (Doom2k@unaffiliated/doom2k) |
20:09.52 | eject_ck | what this warning means when I'm calling with iLBC codec via via external trunk which is using g729 codec |
20:10.08 | eject_ck | [2013-07-05 23:07:26] WARNING[2982][C-00000b43]: translate.c:343 framein: no samples for ilbctolin |
20:22.29 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
20:22.30 | jmetro | If i want my queue to ring an agent following rules i have to log them in using a local context right |
20:22.36 | jmetro | rather than logging in as a peername |
20:24.05 | Katty | i bet you'd love to ring an agent in the queue. |
20:24.50 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.189) |
20:33.16 | [TK]D-Fender | And .... it's checkout time... BBIAB |
20:35.19 | *** join/#asterisk [sr] (~kvirc@smtp.decimal.pt) |
20:38.17 | *** join/#asterisk qakhan (~qakhan@50-200-52-14-static.hfc.comcastbusiness.net) |
20:38.23 | qakhan | hi all |
20:39.12 | qakhan | i am recording calls on asterisk. i want to create folder each date recordings |
20:39.26 | qakhan | how can i make folder in dial plan? |
20:40.14 | jmetro | same => n,NoOp(SHELL(php dosomescript.php)) |
20:42.54 | qakhan | is there any fuction of veriable which can make date folder? |
20:44.41 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.189) |
20:45.51 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
20:49.05 | jmetro | well |
20:49.09 | jmetro | you can mkdir |
20:49.15 | jmetro | and timestamp |
20:49.19 | jmetro | and then mkdir with a timestamp. |
20:49.28 | jmetro | \o/ |
20:50.32 | *** join/#asterisk qmr (~qmr@50.116.18.140) |
20:50.55 | qmr | [Jul 5 16:50:39] NOTICE[12049]: chan_sip.c:20325 handle_request_invite: Call from 'qmr' to extension '1002' rejected because extension not found in context 'users' |
20:51.10 | jmetro | i think extension 1002 doesnt exist in the users context. |
20:51.33 | qakhan | jmetro what the the command> |
20:51.34 | qakhan | ? |
20:51.47 | jmetro | qakkhan lmgtfy.com |
20:51.58 | qmr | https://wiki.archlinux.org/index.php/Asterisk my setup looks pretty much like that |
20:52.06 | qmr | I have 2 softphones connected but they can't call each other |
20:52.36 | jmetro | what is their context |
20:52.47 | qmr | context=users |
20:53.03 | jmetro | does your extensions.conf have a [users] context |
20:53.25 | qmr | probably not - what kind of bits should I put in there? |
20:54.08 | jmetro | i think you need to set your [users] context in sip.conf to have them dial out in [INSERT-CONTEXT-NAME-TO-DIAL-OUT-ON-SOMETHING-UNIQUE-PROBABLY] |
20:54.50 | jmetro | then make that context in extensions.conf and have it be filled with stuff like exten => [INSERT-EXTENSION],1,Dial(SIP/[PEERNAME-OF-EXTENSION]) |
20:54.54 | jmetro | ~book |
20:54.55 | 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 |
20:55.05 | jmetro | ^ base your server off of stuff like that and it will work gooder. |
20:55.58 | jmetro | [the free online book is going to be your best resource on the internet btw] |
20:56.10 | *** join/#asterisk eaxxae (~eaxxae@unaffiliated/eaxxae) |
20:56.25 | qmr | last time I set this up I just added users and everything worked :\ |
20:56.46 | jmetro | contexts are like subroutines [if you program much] |
20:56.50 | jmetro | everything has to point the right way. |
20:57.05 | jmetro | phones are set up with a default dialing context and it sounds like they have the wrong one. |
20:59.47 | qmr | derp |
20:59.49 | qmr | thanks jmetro |
21:00.13 | jmetro | np |
21:00.28 | jmetro | was it in sip.conf? |
21:00.34 | qmr | any recommended connection to PSTN? something I can spend $5 on and not worry about. I won't be making many calls |
21:00.44 | qmr | yea - I had the group in sip.conf instead of extensions.conf |
21:00.50 | jmetro | :) |
21:01.40 | jmetro | im not too sure about connection to the pstn, but you can look at a SIP Trunking provider [i use that and it works pretty great] |
21:06.53 | qmr | any that you recommend? |
21:07.02 | jmetro | depends where you live. |
21:07.03 | *** join/#asterisk transfinite (~aleph_0@unaffiliated/transfinite) |
21:07.05 | jmetro | i think theres a list? |
21:07.10 | jmetro | ~SIPTrunking |
21:07.22 | jmetro | i know theres a list but i have no idea how to bring it up |
21:09.39 | qmr | I'm in .us |
21:10.12 | jmetro | well yeah |
21:10.20 | jmetro | im in the US as well but midwest varies from east coast, etc. |
21:10.38 | jmetro | google "[your region or state] SIP TRUNKING PROVIDER" |
21:10.53 | jmetro | they are usually per minute |
21:12.16 | eject_ck | well, I've paid for zoiper 29 EUR |
21:12.27 | eject_ck | their site sucks |
21:13.17 | eject_ck | I've downloaded some binary zoiperbizg729 which don't start :( |
21:13.31 | eject_ck | ubuntu x64 12.10 |
21:13.45 | jmetro | youre running ubntx64? |
21:13.48 | jmetro | and its for 64bit? |
21:16.11 | qmr | jmetro: Do you have any tips on recording all calls? |
21:17.19 | jmetro | mixmon is the best from what i know |
21:17.35 | jmetro | dump them into a harddrive that is NOT your main one |
21:17.53 | jmetro | or you can do like, packet captures. |
21:20.06 | *** join/#asterisk wolfmitchell (~wolfmitch@botters/wolfmitchell) |
21:26.49 | *** join/#asterisk Tuju (~tuju@214.204.50.195.sta.estpak.ee) |
21:40.58 | *** part/#asterisk eject_ck (~Evgeniy@109.86.157.219) |
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21:54.13 | qmr | jmetro: I almost have this working. I have a connection to localphone now, I just need to get outgoing calls to work |
21:54.29 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.198) |
21:55.13 | jmetro | qmr: that will all depend on connecting to your sip trunk provider |
21:55.16 | jmetro | or etc. |
21:56.18 | qmr | I am connected, just not sure how to dial out to the world |
21:56.25 | qmr | http://www.localphone.com/help/voip/device_guides/softphone/asterisk |
21:58.32 | qmr | looking at the consoel it looks like it tries to call an extension when I enter a phone number |
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23:43.35 | igcewieling | qmr: you need to read the book |
23:43.38 | igcewieling | !book |
23:43.42 | igcewieling | ~book |
23:43.43 | 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 |
23:48.14 | Katty | evening |
23:49.57 | WIMPy | morning |
23:50.29 | Katty | waves to WIMPy |
23:50.58 | WIMPy | waits for the video stream |
23:55.29 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.234) |