IRC log for #asterisk on 20170106

00:00.13WIMPyverbose is local to a specific console.
00:00.29Prelude2004cohhh
00:00.35WIMPySo it wont have any effect when you use it from AMI.
00:00.36Prelude2004cso to the socket
00:00.53Prelude2004cwhat i actually want to do is " sip prune peer xxxx
00:01.06*** join/#asterisk Dovid (~dovid@static-173-63-105-210.nwrknj.fios.verizon.net)
00:01.37wyoungWIMPy, WIMPy, WIMPy!!!!!1
00:02.27WIMPyPrelude2004c: You can test with debug. That's global.
00:03.08WIMPywyoung: You're suffering from remature release of the shift key :-)
00:06.58Prelude2004cOMG .. it works with just core set debug unreal
00:07.05wyoungWIMPy: It's a gaming thing
00:07.19wyoungalthough I do it on purpose for the gaming effect :)
00:07.27wyoungsame with pwn
00:07.41wyoungand teh
00:11.48*** join/#asterisk igcewieling (~ewieling@ip98-170-211-145.pn.at.cox.net)
00:25.59*** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com)
00:42.41*** join/#asterisk KaliLinuxGR (~KaliLinux@unaffiliated/kalilinuxgr)
00:44.16*** join/#asterisk newtonr (~newtonr@173-17-133-211.client.mchsi.com)
00:44.16*** mode/#asterisk [+o newtonr] by ChanServ
00:49.06drmessanoOMG ITS 2004 AGAIN!!!!11!!!!11!!!!ONES!!!11!!1!!!1!!!ELEVENTY!!!11!!!!!11!!!!1!
01:02.52WIMPyWe are no longer stuck in the early 90s? What happened?
01:04.54robmalThe Soviet Union collapsed after that wall in Berlin collapsed.
15:44.46*** join/#asterisk infobot (ibot@rikers.org)
15:44.46*** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.13.1 (2016/12/08), 11.25.1 (2016/12/08), Standard: 14.2.1 (2016/12/08); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.5.0 (2016/03/28) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu
15:52.22*** join/#asterisk anthm (~anthm@freeswitch/developer/anthm)
15:54.03tompawHey guys, I'm experiencing a weird issue: when setting up an ARI bridge and adding two channels to it, the second channel gets the caller ID (From: field in SIP INVITE) from the 1st channel despite being explicitly told not to do so. I use both .originate(callerId) parameter *AND* Set(CALLERID(num)...), which are both executed properly, but asterisk still ignores them and uses wrong "From:" value.
15:54.29tompawMy dialplan, .originate calls and the debug log are here: https://hastebin.com/omudijiyoh.rb
15:57.04*** join/#asterisk simplydrew (~simplydre@unaffiliated/simplydrew)
15:57.12WIMPyCALLERID is for Dial. If you connect two existing calls, look at CONNECTEDLINE.
15:57.34tompawWIMPy: it is a Dial(), have you seen my dialplan?
15:57.42tompawline 8
15:57.57WIMPyNope. Can't see it.
15:58.11tompawWould you prefer a different pastebin?
15:58.26WIMPyBut you said you add the calls to a bridge. That's somethign different.
15:58.57tompawWell, not really, I still originate them using Dial().
15:59.26tompawLet me google CONNECTEDLINE - although this setup of mine worked fine in previous 13.x builds.
15:59.57WIMPyYes. And at that time they should come from the CALLERID, but when you add them to a bridge it will change to the CONNECTEDLINE.
16:00.27WIMPyAnd that usually happens automagically.
16:00.52tompawOooh, I see
16:00.56tompawShould I set both then?
16:01.01tompawCALLERID and CONNECTEDLINE?
16:02.03WIMPyI'm not even sure how the priorities are, i.e. if the bridging action would overwrite whatever you set before.
16:02.17WIMPyMaybe someone else knows in more detail. Or you just try it.
16:02.47tompawtrying it as we speak!
16:04.31WIMPyOr if you never want it, you can disble connectedline updates per peer IIRC.
16:05.24tompawI added CONNECTEDLINE and it's now also being ignored, just like CALLERID
16:05.52tompawI can see both are being properly Set(), but From: is still showing the bridge's first channel Caller ID
16:07.43*** join/#asterisk Oatmeal (~Suzeanne@cpe-65-185-34-151.columbus.res.rr.com)
16:08.52tompawWIMPy: are you saying there is a race condition between my Dial() and bridge.addChannel()?
16:09.09tompawand if the latter executes first, it destroys the Dial's CLI?
16:09.24tompawbecause that would explain why it happens ~80% of the time
16:09.30tompawand not always
16:12.47tompawif that's the case, that's nasty stuff.
16:12.55*** join/#asterisk Sprocks (~Sprocks@bmtnon3746w-lp140-05-65-92-121-159.dsl.bell.ca)
16:14.14tompawThere's not a lot in bridge.addChannel to control it tho.
16:16.37tompawHey, I remember what triggered this situation.
16:17.31tompawBefore, the 1st channel (the one with the Agent's internal CLI) had to be answered by the agents manually, which usually took 2-3 seconds. In the meantime, the 2nd channel Dial() always managed to execute before they answered it.
16:18.09tompawRecently I've changed it so the CRM answers the call automatically, which means the 1st channel is answered before the 2nd channel's Dial() gets executed.
16:19.37tompawWIMPy: how do I "disble connectedline updates per peer"? This might be the key.
16:23.24tompawhttps://issues.asterisk.org/jira/browse/ASTERISK-18999 << is that it?
16:23.41WIMPyI haven't used Bridge much. And nor ARI at all.
16:24.04WIMPyBut rpid_update = no might help. (assuming sip)
16:24.21tompawpjsip
16:25.09tompawlemme try it
16:28.21*** join/#asterisk Alblasco1702 (~Alblasco1@ip5456b46b.speed.planet.nl)
16:28.28*** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com)
16:28.39*** join/#asterisk Sprocks (~Sprocks@bmtnon3746w-lp140-05-65-92-121-159.dsl.bell.ca)
16:31.50*** join/#asterisk Y04NN (~y04nn@178.18.54.206)
16:32.09tompawnope
16:32.28tompawI guess the only dirty way to "fix" it would be to delay the auto-answer of channel 1 by let's say 1 second
16:32.41tompawThat gives enough time for the second .originate to process the Dial()
16:45.30*** join/#asterisk mutex7c (~mutex7c@p4FE5C0A8.dip0.t-ipconnect.de)
16:49.39*** join/#asterisk gtjoseph (~gtjoseph@unaffiliated/gtj)
16:49.39*** mode/#asterisk [+o gtjoseph] by ChanServ
16:54.52*** join/#asterisk simplydrew (~simplydre@unaffiliated/simplydrew)
16:59.36*** join/#asterisk simplydrew_ (~simplydre@unaffiliated/simplydrew)
17:07.31*** join/#asterisk rwb (~Thunderbi@204.13.43.166)
17:14.51*** join/#asterisk rwb (~Thunderbi@204.13.43.166)
17:45.54*** join/#asterisk Sprocks (~Sprocks@bmtnon3746w-lp140-05-65-92-121-159.dsl.bell.ca)
17:51.06*** join/#asterisk simplydrew (~simplydre@unaffiliated/simplydrew)
17:58.15*** join/#asterisk cresl1n (Adium@asterisk/libpri-and-libss7-expert/Cresl1n)
17:58.15*** mode/#asterisk [+o cresl1n] by ChanServ
18:02.21*** join/#asterisk [TK]D-Fender (~joe@216.191.106.165)
18:10.14*** join/#asterisk sekil (~sekil@cable-89-216-223-142.dynamic.sbb.rs)
18:16.11*** join/#asterisk pdugas (~retentive@2601:cf:4400:d8e4:7e8d:d78f:1264:a1ac)
18:16.40*** join/#asterisk Sprocks (~Sprocks@bmtnon3746w-lp140-05-65-92-121-159.dsl.bell.ca)
18:26.30*** join/#asterisk mcargile (~mikec@rrcs-97-76-33-146.se.biz.rr.com)
18:29.03*** join/#asterisk Sprocks (~Sprocks@bmtnon3746w-lp140-05-65-92-121-159.dsl.bell.ca)
18:37.27*** join/#asterisk magellanicus (be522922@gateway/web/freenode/ip.190.82.41.34)
18:37.30magellanicushello
18:37.38magellanicuseveryone
18:38.23magellanicuswhat can i do when i dont see sip show peers comands in asterisk cli
18:38.29*** join/#asterisk sekil (~sekil@cable-89-216-223-142.dynamic.sbb.rs)
18:39.34newtonrmagellanicus, Yo!
18:40.03newtonrmagellanicus, first, make yourself a drink.
18:40.28newtonrThen you can check to see if the SIP channel driver is loaded. What version of Asterisk are you using? "core show version"
18:42.23*** join/#asterisk Cuzner (~ccuzner@74.117.140.46)
18:49.09*** join/#asterisk follow-me (~follow-me@212.34.23.67)
18:49.40follow-me[TK]D-Fender thanks
18:49.44follow-meit worked now
18:49.52follow-meSamot same for u
18:49.58follow-methanks
18:52.20follow-meafter  doing originate how can i wait(7) and and do bridge in case of no answer
18:52.36[TK]D-FenderThis is STILL not a job for orignate
18:52.46[TK]D-FenderNot sure if it's going to take another WEEK to beat this into you
18:52.59[TK]D-FenderYou have a USER placing a SINGLE call.
18:53.21[TK]D-FenderTher i no need for conferencing, Bridge(), Originate or ANY of that nonsense
18:53.27follow-meman its working like a charm
18:53.34follow-meuser
18:53.46follow-mei have 500 concurent man
18:54.05[TK]D-FenderIt doesn't mean you aren't doing it in a pointless painful way with no retun on investment and possible race-conditions.
18:54.39follow-me[TK]D-Fender i want to do the check within 7 seconds
18:55.04[TK]D-Fenderwait != check
18:55.20[TK]D-Fenderthose 2 things are not compatible
18:55.23follow-mebut after 7 seconds idont want to check and
18:55.43follow-methe problem is i need it to turn the detection on within 7 seconds only
18:55.44[TK]D-FenderYour description is still terrible and you've shown nothing.
18:55.48[TK]D-FenderPASTEBIN <-
18:55.53follow-mealright
18:56.00follow-me~pb
18:56.00infobotwell, 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://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude.
18:57.36SamotWouldn't it just be easier to have processes in place to not have the "tier 1 provider" play messages back?
19:01.48igcewielingah, this again.
19:01.52*** part/#asterisk igcewieling (~ewieling@ip98-170-211-145.pn.at.cox.net)
19:01.56SamotLol.
19:08.55follow-mehttps://gist.github.com/anonymous/1ba49c92bbb09996c602a795777cc54d
19:10.56follow-mehttps://gist.github.com/anonymous/0acf9504fc166d300fb50468732a6763
19:11.04follow-methis is better [TK]D-Fender
19:11.36follow-meit is working good , but i want it to do check only within 7 seconds after that bridge and dont do chekcs
19:11.45follow-meanyway to do the test within 7 seconds
19:12.07follow-meas after 7 seconds im sure no that stupid message again
19:12.18follow-meno more messages after 7 seconds
19:18.08[TK]D-Fenderhttps://gist.github.com/anonymous/0acf9504fc166d300fb50468732a6763 <- all you had to do was DIAL.
19:18.17[TK]D-FenderTold you a dozen times
19:19.00[TK]D-Fenderand obviously STILL didn't use the damn macro like you were told
19:19.14follow-mei tried couldnt do it
19:19.36[TK]D-Fendersame => n,GotoIf($[ "${AMDSTATUS}" = "notsure" ]?zaid)
19:19.52[TK]D-Fenderbroken Goto. there is no LABEL for that to land on
19:20.28[TK]D-Fenderexten => _!,1,Noop(AMD check ${CHANNEL} ${CH})
19:20.56follow-meok is there anyway to do test for 7 seconds only
19:21.01[TK]D-FenderThis will LOOP on any non-unique thing tthat may be executing.
19:21.10follow-mewill fix that
19:21.12follow-methanks mate
19:21.16[TK]D-FenderFortunately yuo had a "h" in ther.. which you SHOULDN'T need in the first place
19:21.47*** join/#asterisk Gugge (gugge@92.246.2.105)
19:22.05*** join/#asterisk TandyUK2 (~admin@87.252.44.195)
19:22.09*** join/#asterisk Juggie (~Juggie@unaffiliated/juggie)
19:23.55*** join/#asterisk sh_smith (~sh_smith@cpe-76-174-26-91.socal.res.rr.com)
19:24.27follow-me[TK]D-Fender anyway to do test within 7 seconds
19:24.39follow-meif within 7seconds no answer just bridge
19:24.53follow-meany possibility to do condition based on time
19:24.54follow-mewith asterisk
19:25.19[TK]D-Fender"core show application amd" <--------------------
19:25.49SamotI'm still kind of confused why a tech solution is being used for a business problem.
19:26.07SamotOne that could be solved without any tech solutions needed.
19:28.13follow-meno i meant to stay in amd for 7 seconds if no answer go back to the originate or dial command
19:29.25SamotDoes it matter what we say?
19:29.41SamotSo far you haven't done pretty much one thing you've been told to fix this issue.
19:30.10SamotYou were told Originate is not the way to do this, yet you've added Originate commands to the dialplan.
19:30.40SamotYou were told to fix your Gotosub, you added Originate and bridge commands.
19:31.39SamotYou were told just to keep funds in the account and not do things that violate the providers TOS so you wouldn't get these messages and thus not even need this hack solution.
19:31.41SamotNope.
19:31.43SamotNot doing that.
19:32.39follow-meSamot lick my .....
19:32.43follow-meshut the fuck up
19:32.48follow-meyou dont know how to do a shit
19:32.55follow-meyou were always a loser
19:33.08follow-meand still , and it doesnt work with dial because it will open the line for sure
19:33.09SamotOK.
19:33.12follow-mewith originate it doesnt
19:33.27SamotOriginate opens a line/channel.
19:33.33follow-meyes
19:33.34SamotHow do you think it makes a call?!
19:33.45follow-mebut it doesnt open on the caller side
19:34.13SamotCallers shouldn't need an originate command.
19:34.17follow-meman you using asterisk to call your neighbours or girlfriends giving them some sips
19:34.23follow-mewe use asterisk for real business
19:34.28SamotOK.
19:34.32follow-meyou do code for yourself
19:34.39follow-mewe do the code for large operators
19:34.42SamotYes, I do.
19:34.52follow-mewell done
19:34.53SamotMainly for the voice networks I run for people.
19:35.04follow-mestupid networks having 4 5 people
19:35.13follow-mewe do it for  most of the world
19:35.24follow-mefor the humanity not for small corps looking to call for free
19:35.33SamotNone of my clients call for free.
19:35.39follow-meso please when i ask a question here shut the fuck up its either you try to help or shutup
19:35.48SamotWe have tried to help
19:35.50SamotFor days.
19:35.55follow-meyou charge them and end up with 2 3 k and you call that a business
19:35.59SamotYou refuse to listen or follow the help that has been given.
19:36.04follow-mecome to dubai and let me show you what u can do with asterisk
19:36.09follow-meand how many operators depend on you
19:36.15follow-meman
19:36.18follow-mefor god sake
19:36.21SamotMy clients are ITSP/MSPs.
19:36.31follow-meyou only gave me hints  i felt like in a  puzzle
19:36.48SamotRight, because you were begging for people to give you the code.
19:37.03SamotI told you, this isn't a place for people to "do for you"
19:37.03SamotThey will help.
19:37.14SamotGuide you, look at debugs to help figure out the issue..
19:37.18SamotMake suggestions.
19:37.20follow-mebegging because this is business
19:37.22SamotNo one is writing the code for you.
19:37.31follow-mei dont mind begging people to solve a problem
19:37.40follow-mei shouldnt be shy to ask some experts here
19:37.45SamotBut you didn't listen to anything given to you.
19:37.49follow-mebut not greedy people who keep the codes
19:38.05follow-meand wwant everyone to spend days to look for a thing that could be given
19:38.07Samot[TK]D-Fender has given you multiple suggestions and pointers and you haven't done any of them basically.
19:38.33follow-mei have i tried the dial before it worked but  when it is detected as machine it also open the lines
19:38.36follow-meso i had to use the originate
19:38.56follow-menow GUY please is theere anyway to do a timeout code
19:39.01follow-melike a condition with 7 seconds
19:39.19follow-mewithin 7 seconds if call answred do this if more do that ?
19:39.40SamotThat's why dial and Originate have timeout settings.
19:39.57SamotHow long to dial the call.
19:40.02follow-mewhat if i put 7 seconds timeout i will lose  the call if not answered
19:40.18follow-mei want it to dial for 7 seconds but if within 7 seconds no body answer dont do amd check
19:40.23SamotIf the Dial times out, continue in dialplan.
19:40.38SamotAMD only happens when the call is answered.
19:40.48SamotAMD can't detect shit if there is no answered call.
19:40.52follow-meyes but i dont want the amd if call is ringing more than 5
19:40.56[TK]D-Fender<follow-me> i have i tried the dial before it worked but  when it is detected as machine it also open the lines <- you did it wrong.  Over and over and over.  You consistently failed to follow instructions.
19:41.08follow-mebecause the ivr will open within 5 seconds if more than 7 seconds no ivr will be played
19:41.16SamotTHERE IS NO IVR
19:41.22SamotSTOP USING THE TERM
19:41.28follow-meno messsage
19:41.34follow-meno damn message
19:41.51follow-me[TK]D-Fender i tried the u option you refered but the line were opening anyway
19:42.02[TK]D-Fenderyou did it wrong
19:42.14[TK]D-FenderI told you what to do and you did NOT do it
19:42.20[TK]D-FenderMass incomeptence
19:42.29follow-meexcuse you mate
19:42.34follow-mei did all my best
19:42.35[TK]D-FenderI don't need excuses
19:42.39Samotfollow-me: When a client/enduser doesn't pay their bill and their service is suspended, I send their outbound calls to an Asterisk server that plays an announcement.
19:42.40follow-meman
19:42.50SamotThis is what is happening to you
19:43.10SamotYou are trying to not have that message played to *your* users.
19:43.10follow-meman dont even compare your fucking clients like ours ,
19:43.26follow-meyou have  maybe 23 clients
19:43.26SamotNo, I'm comparing what your Tier 1 provider is doing to you.
19:43.34follow-mewe have 5 operators with 5534042040230400230432 users
19:43.43SamotYeah, I'm sure you do.
19:43.46follow-mewe have 78 asterisk server
19:43.50SamotOK
19:43.51follow-meservers
19:43.56SamotI have that many for 1 client.
19:44.08SamotKeep going.
19:44.18follow-mewe sell 16million minute a day
19:44.23SamotOK.
19:44.40*** join/#asterisk evil_gordita (robert@ip70-188-41-127.rn.hr.cox.net)
19:44.47follow-mei dont have money to pay bills as you like
19:44.55SamotMy users don't have to worry about *my* upstream providers playing them recordings.
19:44.59follow-me[TK]D-Fender you are a progammer
19:45.03follow-meim a footballer
19:45.05follow-melol
19:45.15SamotHow can you have such a large business and not be able to pay your bills?
19:45.16follow-meshow me that its done with the dial
19:45.20follow-mei doubt though
19:45.45SamotQuestioning our skills is not going to get us to "show you" that you are wrong
19:45.48follow-meSamot do you wwant to visit us in dubai
19:45.49follow-meor london
19:45.50SamotAnd just produce the answer for you.
19:45.52follow-meim inviting you
19:46.01follow-meon a first class
19:46.18follow-meskillss my ass
19:46.22SamotNaw, take that money and pay your Tier 1 Provider so you don't get messages from them.
19:46.38follow-meim going to explain it for you
19:46.41follow-mefor the last time
19:46.46follow-mewe buy cheap routes
19:46.49follow-meto resell it
19:46.59follow-mecheap routes  they dont charge the sims always
19:47.13follow-meand the tier1 wont buy routes thats has tthe stupid messages
19:47.15follow-methank god
19:47.19follow-mei pay my bills
19:47.52[TK]D-FenderYou're working for a telco with tools you can't read instructions for.
19:48.01Samot^ that.
19:48.52SamotAs a reseller of these services, using a vendor that constantly causes you issues means get a new vendor.
19:52.57*** join/#asterisk follow-me (~follow-me@212.34.20.82)
19:53.29follow-meok anyway to  do dial or originate and if call were  answered before 7 seconds to hangup A SIDE
19:53.48follow-mei only want calls are ringing for more than 7 seconds
19:54.32SamotWow.
19:54.38*** join/#asterisk follow-me (~follow-me@212.34.20.82)
19:59.08*** join/#asterisk kambei (~kambei@unaffiliated/kambei)
20:08.14follow-mei dont want amd check
20:08.25follow-meonly if call is answered within 7 seconds i dont want it
20:08.42follow-meso can we test if call is answerd within 7 seconds hangup without opening channel A
20:10.36follow-melike gotoif ringing time  is 7 s
20:17.40*** join/#asterisk ppd1990 (~mfederle@p5B122B17.dip0.t-ipconnect.de)
20:21.00ppd1990hello. I have a seemingly simple question regarding current Asterisk 14. I have set voicemail's audio format to WAV and it records just fine (I can play those files on the computer). However the Voicemailmain application tries to play the message with the extension ".slin", which is obviously not gonna work. Do you have a tip for me where to look for the appropriate setting? Thank you very much
20:23.13SamotAsterisk transcodes the audio for playback.
20:23.35SamotFor the best possible format.
20:25.37ppd1990so it should transcode it on the fly then? It just outputs silence though :(
20:26.12SamotShow the call to voicemail.
20:26.19Samotasterisk -rvvvvvvv
20:26.21Samot?pb
20:26.23Samot~pb
20:26.24infobotmethinks 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://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude.
20:26.52ppd1990the recording call or the playback call?
20:28.36follow-meif ringingtime
20:28.49follow-memore than 7 do this if less do that
20:28.52follow-mepossible in asterisk
20:28.54follow-me?
20:30.15ppd1990Samot: http://pastebin.ca/3753921
20:30.40SamotPlayback call...
20:30.49SamotWhere it's playing back silence.
20:32.00SamotSo you're not hearing any of those audio files?
20:32.05ppd1990exactly
20:32.09ppd1990ah no
20:32.16ppd1990all except the message itself
20:32.22SamotSo only the message.
20:32.35SamotOK.
20:32.48ppd1990only the message does not work
20:32.50SamotShow the whole call please.
20:33.04SamotSo I can see what type of codecs are being used, etc.
20:35.52ppd1990what can I set to generate more log output? I have verbosity and debug level up to 7 for core, and that's everything it is outputting
20:36.42SamotOK, well what codec is the phone using?
20:38.44ppd1990I have set g722 for the endpoints and it supports that, so I hope it does use it. How can I verify?
20:39.21SamotTry ulaw
20:40.22Samotg722 is wideband and there are separate wideband soundfiles.
20:40.44ppd1990bloody hell, that works
20:40.46SamotI don't think it transcodes "up" so to speak.
20:40.57SamotIt transcode to the best format to support the codec.
20:41.01ppd1990you are a genius
20:41.15SamotTry setting the codec to allow=g722&ulaw
20:41.19SamotThen check again.
20:42.43ppd1990that works too. I had allow=g722 and allow=ulaw in different lines
20:43.21SamotYou can do it in one line.
20:43.26SamotIn order.
20:43.57SamotI, personally, always have a secondary codec for when g722 is used.
20:44.41ppd1990thank you so very much for your help. I think that'd have taken me quite some time to figure this out :)
20:44.56SamotNp.
20:45.06SamotJust a note, all playback generally shows up a .slin
20:51.48*** join/#asterisk Eloy_ (~Eloy@78.152.215.246)
20:53.08*** join/#asterisk jeffspeff (~Jeff@12.49.160.131)
21:16.54*** join/#asterisk Y04NN (~y04nn@2a01:e34:ef37:5870:99bc:47e0:ff1c:3ad5)
21:37.59[TK]D-Fenderpacks up to head home
22:06.42*** join/#asterisk igcewieling (~ewieling@ip98-170-211-145.pn.at.cox.net)
22:23.32*** join/#asterisk rwb (~Thunderbi@65-183-151-239-dhcp.burlingtontelecom.net)
22:25.59*** join/#asterisk [TK]D-Fender (~joe@64.235.216.2)
23:04.51*** join/#asterisk Y04NN (~y04nn@2a01:e34:ef37:5870:99bc:47e0:ff1c:3ad5)
23:10.57*** join/#asterisk eaxxae (~eaxxae@unaffiliated/eaxxae)
23:11.40*** join/#asterisk Maliuta (~nobusines@unaffiliated/maliuta)
23:11.43*** join/#asterisk MaliutaLap (~nobusines@unaffiliated/maliuta)
23:23.16*** join/#asterisk cmatei (~cmatei@95.76.45.223)
23:33.24*** join/#asterisk Y04NN (~y04nn@2a01:e34:ef37:5870:2d15:4cf0:73e6:ff39)
23:33.56*** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com)

Generated by irclog2html.pl Modified by Tim Riker to work with infobot.