IRC log for #asterisk on 20141023

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00:20.10pederindi[TK]D-Fender: You mean that the nat of the asterisk is not needed. Internally, works. And externally, is used the asterisk processing (directmedia=no)
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02:58.27jdzielnyis there anyone here who's familiar with Asterisk CDR?  I can't get it to work--at all
02:58.27jdzielnyLiterally nothing
02:58.33jdzielnythe db is empty set
03:14.10cmendes0101jdzielny: what database are you using?
03:16.20jdzielnyForgive me if this sounds ridiculously newbish--I'm a new user--but I don't know
03:16.27jdzielnyI've got FreePBX as a frontend
03:16.42jdzielnyI've tried without success to get some support over there
03:16.48jdzielnyIt's like a ghost town
03:17.16cmendes0101Gotcha, I'm not familiar with Freepbx so might not be able to help with that
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03:17.51jdzielnywell perhaps you could at least give me an overview of how the system works.  Searching google has just confused me to no end because there's so many posts that all mention different files and "not working"
03:17.53WIMPyjdzielny: You won't have more luck in here.
03:17.58WIMPy~freepbx
03:17.58infobot[~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
03:18.51WIMPyjdzielny: FreePBX has its own structure that goes beyond that of (plain) Asterisk.
03:24.03jdzielnyAt the risk of sounding like I disregarded the bot (I didn't), is there any place I could get just a broad quick explanation of how the CDR system works normally?
03:24.29WIMPydefine "normally"
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03:24.44WIMPyWheer do you even try to put them?
03:25.03WIMPyWhere *
03:29.14jdzielnyThis is the problem -- I don't know how it works
03:29.16jdzielnyAt all
03:29.30WIMPyThere are meny options.
03:29.34jdzielnyI've tried and tried to find an overview of how Asterisk stores call records, and I keep going in circles
03:29.45WIMPyWhere do you want them to go?
03:29.57jdzielnyHonestly, I don't particularly care, as long as they go somewhere
03:30.12WIMPyYou can write them to (a) file(s), to syslog, to AMI or to a database. for a start.
03:30.13jdzielnyother than /dev/null, which is where they seem to be going
03:30.57jdzielnyokay, well I don't expect a large volume, a file would be fine.  as would a database
03:30.58WIMPyStart with cdr.conf.
03:31.32jdzielnycdr.conf is empty
03:31.51WIMPyWell, that surely explains why nothing happens.
03:32.55jdzielnyokay...
03:33.10jdzielnybut what is it supposed to contain?  And how do I generate it
03:33.11jdzielny?
03:33.28jdzielnyI realize these may seem like basic questions, but I'm not stupid, just very new to this
03:34.10WIMPyTake the cdr.conf.sample file and check if the contents make sense to you or the wiki should haopfully have an explanation as well.
03:35.36WIMPyIf you need some general starting point try one of these two:
03:35.39WIMPy~primer
03:35.39infobotNew to asterisk configuration? Check out this primer to get started.  http://burner.com/asterisk-primer
03:35.44WIMPy~book
03:35.44infobotAsterisk: 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
03:36.39WIMPy(or both, off course :-) )
03:38.38dralepage 643 of that book
03:40.33dralei was looking at the pdf version
03:41.39jdzielnyWell this is definitely a good place to start
03:42.07jdzielnyWhy in the world there's no cdr.conf but there IS a cdr_mysql.conf (and a .bak) is a mystery
03:42.16jdzielnyfreepbx is an odd fkng duck
03:42.45WIMPyIf you're using FreePBX, don't mess with the config files. It will take revenge.
03:43.02jdzielnyCan't possibly take revenge beyond doing nothing
03:43.14WIMPyEither find out how to clicl around or get rid of it.
03:43.22WIMPyclick
03:45.22WIMPyAnd if you can read and aren't afraid of using a text editor, you probably don't want to use FreePBX.
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03:46.15jdzielnyI'm open to using another tool.  Frankly I really don't care.  I need to set up 100 DIDs so that they all go automatically to voicemail after a few rings
03:46.34jdzielnyAnd we need to be able to call out from them if necessary.  It's not a complicated setup
03:47.22jdzielnyIt's for a controlled study in how employers respond to certain key words in applications
03:48.33jdzielnyI like FreePBX mostly because it's got a slick GUI and is easy to use (except for CDR, apparently)
03:50.06jdzielnyIf you have another suggestion, I'd be open to looking and wiping out freepbx.  I'm running on an EC2 server anyway so it's trivial to set up a new machine
03:50.58WIMPyIf all you want is ring some device and go to voicemail after some time, just write extensions.conf yourself.
03:51.29WIMPyOr generate extensions.conf and voicemail.conf from a simple script.
03:52.16jdzielnyI also have to do logging, recording, etc.  And if the project is successful, it's gonna be expanded about 50-fold, at which point having a GUI will be essential
03:52.34jdzielnyI can manage 100 extensions manually, there's no way I can do it with 1500
03:53.08WIMPyI can manage hundreds with a text file, but I can't click 50 with a mouse.
03:53.25jdzielnyYou and I both know how to OPEN a text file, tho
03:53.26jdzielnylol
03:57.18jdzielnyThat's more than I can say for the people controllign the money here
03:57.20jdzielny:-\
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05:27.08Penguinburnbrighter: Yes.
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05:37.17ruben23hi guys any idae what should be done troubleshooting no outbound calls for an ISDN and pSTN channels on asterisk..? im suign Dahdi
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05:38.44WIMPyWrong Type Of Number, as usual?
05:39.27WIMPyhttp://voice.yeti.dk/Asterisk_vs_ISDN/5
05:42.23ruben23how about congestions..?
05:42.53[TK]D-Fenderruben23: I don't see you showing us the call.  Asking without showing the call is a waste of everyone's time
05:43.07WIMPyWhat about congestions?
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05:43.46ruben23sorry guys ill capture the call..
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06:00.16[TK]D-Fenderruns out of air while holding his breath
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06:03.57WIMPyBlue Fender :-)
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07:04.49stevenmAnyone know of an analogue phone that'll auto answer any incoming call?
07:06.32WIMPystevenm: Any one with built in answering machine?
07:07.10stevenmWIMPy, no not answer & record - just answer
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07:16.10ruben23WIMPy: hi there..
07:16.45WIMPyruben23: Where's your log. The fender already turned blue while waiting.
07:16.55stevenmsomething like this would be handy... http://oldphoneguy.net/BookletPDF/Visio-SCRcontrol5.pdf
07:17.14ruben23http://pastebin.com/6kmCcmwF
07:17.38ruben23they say incoming calls comes in and drop to busy tone
07:17.42stevenmjust a little box that answers any incoming analog call - and you can connect a speaker or something to it so you can listen to the audio
07:18.25WIMPystevenm: Probably not the best idea to do it analog and rely on somethign like busy tone detection.
07:20.09WIMPyruben23: Looks like you start an AGI that hangs up the calls.
07:23.23WIMPyOr rather immediately after that AGI.
07:24.41ruben23WIMPy: what could i do somehow..?
07:25.14WIMPyWell, I don't see anything usefull in that log. More of a lot of crap.
07:25.34WIMPyMaybe you should debug that AGI?
07:25.53WIMPyAnd timing information would definitely help to get a sense of what's going on.
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07:59.13bright_hello
07:59.26bright_i have a problem with remote extension behind nat. calls are passing through but asterisk is sending rtp to ip behind nat and in few secs it just drops a call
07:59.53bright_what could cause it?
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09:16.42MythosGRscenario: [SIP Videophone h.264] -- IAX trunk -- [SIP Videophone h.264] : video call immediately hangs up when answered. audio-only call works ok. chan_iax2.c: Can't compress subclass 2097217. Any ideas?
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11:26.44marceloamorimguys, anynone already get option h on the dial application when you didn`t set this option?
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11:35.04marceloamorimthere is a dialplan called ael-builtin-h-bubble that have this h option
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11:53.50Pernathow to play background music and second audio with speech in IVR?
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12:12.25erick2014Does anyone have advice for getting started with Asterisk 12 or 13 and pjsip?  I've been having troubles with a clean install of Ubuntu, Asterisk, and pjsip, and there's been no logging information to help.
12:12.42erick2014I'm not under any time crunch, and I can spin up a new VM to do anything I need.
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12:45.51erick2014does anyone have a working install of asterisk and pjsip?
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12:49.05rmassihHey Guys, silly question im sure but here we go.  Callcentric has got me to put a ton of enteries in my sip_custom_post.conf
12:49.15rmassihthe entries look like this:  [callcentric1](Callcentric);
12:49.16rmassihhost=alpha1.callcentric.com
12:49.46rmassihwhen I do a sip show peers I see all the entries, is there anyway around this? other than removing them as it makes my inbound calls from callcentric work
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12:54.34[TK]D-Fenderrmassih: If they made entries for multiple sending servers then that's wha you'll have to do if you want them to match a peer.  Otherwise you'll have to allow for un-authed calls and process them that way.
12:54.41[TK]D-Fenderrmassih: Take your pick.
12:56.50erick2014can anyone see my messages?
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13:01.10erick2014Hey, just testing.  If you can see this, can you respond?  I'm not sure my messages are coming through.  I'm not getting ANY responses to my messages.
13:01.19rmassihThanks Fender, im already allowing anon and sip guests.  To be honest im not clear about the effects of these as ive got an any/any rule
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13:14.21erick2014hey, rmassih or fender, can either of you see my messages?
13:14.35rmassihyep I see them Erick
13:15.10erick2014ok, thanks.  I've asked a few questions over the past few days with no response - I just wanted to make sure they were coming through.
13:15.10kchehabhi , how can i use  DBdeltree(test) as i have before it 678,n,WaitMusicOnHold(3600)  in dialplan ,when client hangup the dialplan order will not reach DBdeltree(test)
13:16.16kchehabmy dialplan can be found http://pastebin.com/auCsjzVV
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13:19.24marceloamorimGuys, I want to say I figured out what happen with the transfer drop call on my asterisk
13:19.56*** join/#asterisk serafie (~erin@nat/digium/x-iptgdchqyicyofok)
13:23.35marceloamorimI put a comment with " ; " in one line, so the pbx it was sending to ael-builtin-h-bubble
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13:25.33jdzielnyhi there, has anyone ever seen this? "AOR [extension number] has no configured mailboxes. MWI subscription failed"
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13:27.08kchehabfrom where i can add an action when a client hangup ?please can you advise by link to read
13:29.20[TK]D-Fender[08:45]erick2014does anyone have a working install of asterisk and pjsip? <- yes.
13:29.27[TK]D-Fender~poll
13:29.27infobotScript for automating Fidonet polls. URL: http://www.drmach.demon.co.uk/vashti/software/index.html
13:29.29[TK]D-Fender~polls
13:29.30infobot"Does anyone have X or use Y?" is taking a poll, not asking a valid question. Don't do it or our army of rabid weasels will hurt you. Usually, people other than those with the exact same set up can help you and those who have sometimes will not be able to help you. Also see <ask>
13:29.32[TK]D-Fender^^^^^
13:30.27[TK]D-Fendererick2014: Yes many people use it, even in production.  Yes it work. No, I don't personally use it.  If you have a real question don't waste time on meta questions.  Most people don't waste time answering those.
13:32.02[TK]D-Fender[09:27]kchehabfrom where i can add an action when a client hangup ?please can you advise by link to read <- go read up on your Asterisk Standard Extensions.  You have missed the basics....
13:32.08[TK]D-Fenderkchehab: "h" <--------
13:32.13MaliutaLapkchehab: https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Dial
13:32.25erick2014Gotcha.  I was having a problem yesterday and started to get an answer, but then the conversation sort of stopped.  I asked again today and didn't hear back at all and started to wonder if it was an issue with nobody having any experience with pjsip, or not being able to see my messages.  Understanding that this is a community chat room, I wanted to be respectful and not push if nobody had any experience with pjsip, but
13:32.25erick2014<PROTECTED>
13:33.11CuznerLOL Fidonet... that's a name i haven't heard in a while
13:33.14MaliutaLap"Carefully explaining your problem is half the solution." :)
13:33.16Cuznerused to have a node
13:33.32MaliutaLapCuzner: I hope you had it removed properly ;)
13:33.46[TK]D-FenderCuzner: Same here
13:33.48Cuznerlasers, how do they work?
13:34.00MaliutaLapoff the heads of sharks!
13:34.13[TK]D-Fenderstrokes Mr. Bigglesworth
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13:35.12MaliutaLap[TK]D-Fender: well that makes a change from what you normally stroke :P
13:35.32filefalls over
13:35.40[TK]D-Fendergrabs his pen and stabs MaliutaLap with it.
13:36.04[TK]D-FenderMaliutaLap: Would you like another stroke?
13:36.05MaliutaLap[TK]D-Fender: you don't normally play with the stroke on your engine?
13:36.23MaliutaLap[TK]D-Fender: please sir, can I have some more? :P
13:36.29[TK]D-FenderMaliutaLap: My sister just got a Tesla this year... it doesn't have one ;)
13:36.45MaliutaLapno stroke to bore!
13:37.11MaliutaLap[TK]D-Fender: so your sister stopped stroking - that doesn't mean you have :)
13:38.21jdzielnyAnyone who can help, what does AOR stand for in asterisk?
13:38.27MaliutaLapshould off to bed. Got to be at the hospital for most of tomorrow afternoon
13:38.52fileaddress of record
13:39.01fileit's a SIP term
13:39.27jdzielnyOkay
13:39.43jdzielnyfile, any idea what would cause this warning message? "res_pjsip_mwi.c:661 mwi_subscribe_single: AOR 2001 has no configured mailboxes. MWI subscription failed"
13:39.51jdzielny2001 is an extension btw
13:40.21fileWhat is registering to AOR '2001' is attempting to subscribe to get voicemail message waiting information but there is no mailbox configured on the AOR
13:40.40ChainsawNo mailbox, no status.
13:40.59[TK]D-Fenderjdzielny: Address Of Record
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13:41.15jdzielnyI think I'm misunderstanding something here.  Extension 2001 does have a mailbox
13:41.29fileit may have one, but it may not be configured in the pjsip.conf AOR
13:41.59jdzielnyI'm new to this, file, can you explain what that means in a bit more detail?
13:42.04[TK]D-Fenderjdzielny: And don't call SIP config entries "extensions"
13:42.28[TK]D-Fenderjdzielny: Show us your config
13:42.30[TK]D-Fender~pb
13:42.31infobotA "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.
13:42.32[TK]D-Fender^^^
13:42.52CuznerAren't some of you guys in vegas? Why you up so early?
13:43.03jdzielny[TK]D-Fender, which config?
13:43.05jdzielnypjsip?
13:43.08Cuzneris jelly btw
13:43.18[TK]D-Fenderjdzielny: Clearly
13:43.47fileCuzner, good question - why am I up so early...
13:44.04Cuznerfile: seriously... you think leif is up right now? fat chance :P
13:44.24Cuzneror alan heh
13:44.27fileCuzner, he tweeted an hour ago
13:44.35Cuznermarquis, maybe...
13:44.39jdzielny[TK]D-Fender, there's a bunch of pjsip.whatever.conf files. which one did you want to see in particular?
13:44.44file@leifmadsen "Can't sleep. Clown will eat me. #astricon"
13:45.01Cuznerhah, like i follow that nerd.
13:45.57[TK]D-Fenderjdzielny: pjsip.conf clearly...
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13:46.06Cuznersurprised he's up though, unless like he says in his tweet, he just didn't sleep.
13:46.13filejdzielny, is something else (like FreePBX) managing configs?
13:46.20[TK]D-Fenderfile: SHHH!!!
13:46.27[TK]D-Fenderfile: I'm handing out rope
13:46.44Cuznerteach a man to fish... something, something
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13:46.55[TK]D-FenderCuzner: We're gonna need a bigger boat
13:47.41jdzielnyfile, yes
13:47.46jdzielnyhttp://paste.ubuntu.com/8640803/
13:47.57[TK]D-Fenderjdzielny: This is NOT the place for FreePBX support
13:48.09[TK]D-Fender~freepbx
13:48.10infobot[~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
13:48.11[TK]D-Fender^^^
13:48.38Cuzner~trixbox
13:48.39infobotDelving into Trixbox is like exploring a pyramid; it's ancient, forgotten, dark, and dangerous.  Trixbox was one of the earliest complete PBX distros and a relic of a bygone era.  While it was a great idea, it was implemented by a horrible group of Wizards from an evil, barren wasteland that stuffed it full of black magic and FUD.  Also, an example of how not to run a business.
13:48.47Cuzner^_^
13:48.58Cuzneri used to run so many of those things hehe
13:48.59[TK]D-FenderCuzner: Not quite pertinent... but still somewhat funny
13:49.00jdzielnyreally, [TK]D-Fender, that was what that whole conversation was about?
13:49.13[TK]D-Fenderjdzielny: What conversation?
13:49.32[TK]D-Fenderjdzielny: You had mailbox questions... but you are not in charge of your own configs, FreePBX is.
13:49.37[TK]D-Fenderjdzielny: it OWNS you.
13:49.42Cuzner[TK]D-Fender: well, when i started playing with asterisk, freepbx was just a configuration gui, but apparently it's a whole 'distribution' now... does it include FOP!? :D
13:49.44jdzielnyThank you for the lesson in slavery
13:49.55[TK]D-Fenderjdzielny: So if that GUI isn't generating configs the right way, then it is at fault, not asterisk
13:50.02fileany changes you make will be overwritten the next time it writes the configs, so while you could fix it... the fix is not permanent
13:50.08[TK]D-Fenderjdzielny: ^^^
13:50.18filethe only permanent thing is to do it from within the FreePBX system
13:50.25CuznerFOP was the bees knees
13:50.27file(if it permits it, I'm unsure of that)
13:50.27Cuznerhides
13:51.10jdzielnyThe fact that FreePBX may be screwing up doesn't mean that asterisk isn't ultimately reading the config files written by it.  And if it's a temporary fix, that's fine, so long as I can get it working temporarily so I can try to fix the problem with FreePBX
13:52.21[TK]D-Fenderjdzielny: Well * told you the AOR didn't have a mailbox defined in it.
13:52.58[TK]D-Fenderjdzielny: #include pjsip.aor.conf <--- so this probably shouldn't have been a question
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13:55.01jdzielnyI hope this doesn't sound like I'm being rude, because I'm not, but what may seem obvious to an experienced asterisk user can seem like greek to a new one.  I'm trying to learn how asterisk works.  That I've got a GUI to help because I'm a clueless newb doesn't mean I'm incapable of reading config files and figuring them out, provided I have some help on what asterisk actually DOES with them
13:55.07[TK]D-Fenderjdzielny: read the sample configs to see where these parameters belong
13:57.02jdzielnyand fwiw, I have multiple devices connected to the system, all are making and receiving calls fine, and voicemail is working fine for all of them, but only ONE is giving me that error
13:57.22Chainsawjdzielny: That's good news. That means that only one is misconfigured.
13:57.25jdzielnyand the entries in pjsip.aor.conf are the same for all of them
13:57.34filesubscribing for voicemail mwi is not mandatory, other stuff may not use that mechanism
13:57.43filethe other way is to simply tell the device the information without it asking
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14:01.55fileit's the "mailboxes" option in the type=aor for the device, what that value should be you could pull from the type=endpoint elsewhere itself - cause I don't know how FreePBX does naming
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14:03.00RadJacksonHello ,  we have a got a good internet connection speed. a Windows 2003 server (lets call it Xserver) Sometimes when an user close his windows session , the active directory launches the synchronisation , at this time pricesely i notice that my sip peers (connected via Xserver) start to Lag , (Stauts LAGGED or UNREACHABLE) , ive got qualify=yes nat aswell
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14:03.48RadJackson<PROTECTED>
14:04.00RadJacksononce i reboot internet , everything gets back to normal , any idea ?
14:04.51RadJacksonWhat can i do to avoid such problems? Call a network guy ? create aVLAN specially for my softphones ?
14:05.53[TK]D-FenderRadJackson: Sounds like its routing for your subnet.  When it stops routing, * stops getting responses to its OPTIOSN packets and it starts thinking your peers have dropped off.
14:06.05[TK]D-FenderRadJackson: Lesson: Don't use Windows as a router
14:06.51jdzielnyalright I'll figure it out.  If all else fails I'll just delete it and put it back in
14:06.58jdzielnyMaybe that'll do the trick
14:08.19RadJackson[TK]D-Fender The router isnt Windows its Juniper, its the domain controller that is WIndows Server 2003.
14:09.05[TK]D-FenderRadJackson: Seems your packets are going through it...
14:09.21RadJacksonok , shud i call a network guy ?
14:09.59[TK]D-FenderRadJackson: What is this "connected via Xserver" you mentioned actually mean?
14:10.43RadJacksoni didnt explain well , i meant that the windows server 2003 has DHCP role
14:11.30RadJacksonit isnt a sip client problem , ive got Eyebeam softphones and CISCO IP PHONE SPA504G aswell
14:11.38RadJacksoni start getting the lag on both
14:12.10RadJacksoni'm not much of a network guy...
14:12.41RadJacksonEverytime a session is closed , a file is beign downloaded, all my sip clients start lagging and keep lagging minutes later , i always have to reboot internet
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14:14.16[TK]D-FenderRadJackson: Flooded connection means slow responses as well...
14:14.23[TK]D-FenderRadJackson: Consider traffic shaping.
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14:19.55RadJackson[TK]D-Fender ok thank you
14:22.29jdzielnyfile, this might seem overly simplistic, but how does asterisk know whether or not a "mailbox has been configured" within the context of the mwi warning I'm having?
14:22.52[TK]D-FenderjzdAOR's have a mailbox parameter as shown in the sample config
14:22.55fileif there is no mailboxes option set for the AOR, then it's not set
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14:25.12jdzielnyfile, the mailboxes option for each extension ( [TK]D-Fender said "extension" is the wrong term, but I don't know the correct one) shows "mailboxes=XXXX@device", where XXXX is the extension number
14:25.58filethat statement doesn't have context for me as I don't know where - if it's in the endpoint then it won't work for that device, because that's not an AOR
14:26.43jdzielnyit's in pjsip.endpoint.conf
14:27.02[TK]D-Fenderjdzielny:doesn't sound like "AOR"
14:27.15[TK]D-Fenderjdzielny: And we have no prrof of what's actually in that file
14:27.20[TK]D-Fenderjdzielny: Location counts
14:28.41jdzielnyhttp://paste.ubuntu.com/8641210/ <--pjsip.aor.conf
14:28.42jdzielnyhttp://paste.ubuntu.com/8641211/ <--pjsip.endpoint.conf
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14:29.07filethere is no mailboxes= in the AOR
14:29.14filethat's why it is not working
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14:35.09jdzielnyhm
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14:43.10R1ppaTime showing wrong on all phones after a power outage, some route missing somewhere?
14:47.14[TK]D-FenderR1ppa: The phones haven't refreshed from whatever time server they were using.  Either waiting for a resync period, or some networking issue, etc
14:47.20ChannelZWell they either get the time from an ntp server somewhere, or perhaps from a SIP header which means the time on the server is probably wrong
14:47.31[TK]D-FenderSIP doesn't distribute time
14:48.06R1ppa[TK]D-Fender, yeah we were experiencing some network issues, found a loop! woot, routes seem ok, can phone in and out, just no phone has any idea what time it is lol, they have been this way overnight
14:48.43[TK]D-Fenderreboot them now.  Maybe the phone came up before they were able to reach their time server
14:48.52[TK]D-FenderAnd gave up
14:48.59R1ppa[TK]D-Fender, have been rebooted several times, will try again
14:49.04Cuznersounds like the most plausible expolaination
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14:52.39CuznerR1ppa: you should consider an on premise ntp server that syncs from the 'net to avoid the problem in the future, aim all your local devices at it.
14:52.46ChannelZhmm asterisk.org says the latest version is 11.13.0 on the Downloads page but the folder in the tar is actually 11.13.1
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15:14.38ChainsawChannelZ: It's correct on the "all Asterisk versions" though.
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15:22.20olspookishmagusis there any live feed from Astricon?
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15:25.10[TK]D-Fenderolspookishmagus: The food is typically killed before serving
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15:28.47olspookishmagusso we have to wait
15:28.57mbowielolz
15:29.28Cuzner[TK]D-Fender: but gagh is best served live!
15:30.24[TK]D-FenderMORE BLOOD-WINE!!!
15:30.31WIMPyDoes anyone know what the difference is between Asterisk following a "Registration for ... timed out" message by a "Probably a DNS error" message or not?
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15:50.06nicknam1232#JP Konichiwa
15:50.37nicknam1232Consider° operator ** FR - france **
15:59.26Cuznerdamned FR locale with all their damned emergency numbers...
15:59.43Cuzneri swear france has an emergency short code for stubbing your toe on a fire hydrant
16:00.43WIMPyYou need a manual for making an emergency call there?
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16:14.04nicknam1232consider{[é]
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16:25.02nicknam1232Souix Exchange - McC........, nombre.  : - )    :-)   *  _  * [xnr2] 757
16:25.28ChainsawI think it wants to communicate.
16:26.09[TK]D-FenderProbably just an autonomic response.
16:26.16[TK]D-FenderPoke it with a bigger stick
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16:35.03nicknam1232... Danke Scheun          -                                          ^  -  ^
16:35.12filefalls over
16:35.17nicknam1232* Merci. Beaucoup.
16:35.32nicknam1232? Ville danke .
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16:47.49boatmn810ok, stupid question  - make menuselect  - any difference between ncurses and newt versions ?
16:47.56boatmn810and how to choose which one ?
16:48.31PenguinI'd imagine there has to be a difference or there wouldn't be two names.
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17:02.07Stefan27asterisk has opened 400+ fds related to rtp (udp sockets) and they seem to persist even though i have no active calls anymore. I use 12.6.1 now, and installed on linux fedora 20.
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17:03.20Stefan27How can I debug this, or how can i give you relevant info? here's a thread dump http://pastebin.com/RxTKXJJh
17:04.13WIMPyboatmn810: If you have newt, that will be used. The advantage is color, the disadvantage is keyboard settings.
17:04.44WIMPyStefan27: There's a debug fd leaks option.
17:05.03WIMPyBut I haven't tried to use it.
17:05.13boatmn810i have newt, but its still not using the color screens - so far thats the only diff between 2 systems that I can find -
17:05.37*** join/#asterisk kirilvalchev (~KirilValc@ns.atsoftconsult-bg.com)
17:05.48boatmn810in which the logger is giving me problems on the one without newt
17:05.58Stefan27stracing the asterisk pid shows a spam of system calls ##4979  nanosleep({0, 10000000}, NULL)    = 0## related to the thread 116 which was called from  timer_worker_thread() in res_rtp_asterisk.c
17:06.15boatmn810correction - with newt
17:06.37*** part/#asterisk kirilvalchev (~KirilValc@ns.atsoftconsult-bg.com)
17:08.31Stefan27looks like some thread is stuck in an infinite while loop on line 1706 in res_rtp_asterisk.c?
17:09.14Stefan27but what puzzles me is that that function is surrounded by a block #ifdef HAVE_PJPROJECT ... #endif
17:09.37Penguin~spam
17:09.37infobotmethinks spam is probably a preferred environment. SPAM; Shut up, You damn Vikings! SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM SPAM, or to destroy it, try spamassassin (spamd+spamc) and razor.
17:09.54Penguinuseless
17:10.12Stefan27i didn't even load any asterisk module with the name "pj" in it
17:10.16Stefan27i use chan_sip
17:10.29boatmn810do you have it all turned off ?
17:10.48PenguinIt has to be related to pjsip.
17:10.58Stefan27I did compile asterisk with pjprojects
17:11.20Stefan27but i decided only to load chan_sip and make no reference to pjsip stuff from dialplan
17:11.20WIMPyMaybe you should try without.
17:11.58Stefan27I will - but it took me a long time to produce this bug
17:12.12Stefan27more than 10000 calls
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17:12.26boatmn810thats about 1 1/2 days for me
17:13.01boatmn810so box has been up awhile ?
17:13.03boatmn810how logn /
17:13.07boatmn810how long ?
17:13.08Stefan27debug-fd-leak option requires special compilation flags?
17:14.15WIMPyIt's in menuselect
17:15.41Stefan27can i from the OS force a close of these FDs without restarting the asterisk process?
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17:23.05Stefan27Yeah I closed them, so if they were not actual leaks asterisk will likely seg fault later
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17:48.53jeevis asterisk on switchvox any different than asterisk opensource?
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17:52.35R1ppahad some network issues yesterday, all is well with Asterisk except all phones time wrong, if I can route calls fine and dhcp seems to be working fine, whats stopping it from getting the time?
17:55.01[TK]D-FenderR1ppa: Exactly what we've already told you hours ago
17:55.42R1ppa[TK]D-Fender, "or some networking issue" is what I am working on
17:56.26[TK]D-FenderHave you dumped traffic to see if it's looking?
17:56.34[TK]D-FenderBecause this isn't an Asterisk issue...
17:56.59R1ppa[TK]D-Fender, yes agreed, this is more likely a network issue on my part, just thought friendly folks may have some advice
17:57.29[TK]D-FenderWe advise you look at your phone configs, test the places it's trying to look, look at firewall dumps to see if it's really getting that far, etc
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18:15.54R1ppa[TK]D-Fender, I can see the network traffic coming in from phone IP to time server, configs were fine before so they are fine now, I am sure this is network issue, but its not obvious to me
18:16.51PenguinIs the time some hours and some minutes wrong, or just hours?
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18:17.28R1ppaPenguin, everything lol Dec 31st
18:17.43PenguinIP phones?
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18:18.23R1ppaPenguin, yarrr Polycom soundpoint 430
18:18.26PenguinThey will receive calls now?
18:18.33[TK]D-Fender[14:15]R1ppa[TK]D-Fender, I can see the network traffic coming in from phone IP to time server, configs were fine before so they are fine now, I am sure this is network issue, but its not obvious to me <- some of this doesn't sound like double-checking
18:18.39R1ppaPenguin, yes, calls inbound and out are fine
18:20.47R1ppa[TK]D-Fender, I checked ntp service, dhcp service, dns, tcpdumped and can see traffic coming into ntp yet all phones are still wrong, dood I am not asking for spoonfeeding so can you stop treating me so?
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18:33.38[TK]D-FenderR1ppa: have you tried changing the resync time on it?  Confirmed where it's actually applying settings from?
18:34.16[TK]D-FenderR1ppa: Polycoms can grab from multiple places but may use just the one specified in its configs
18:36.59PenguinCan't you set it manually from the phone's menu?  I thought menu configured settings would override the config file settings.
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18:38.19R1ppaPenguin, have not tried override yet because it looks to me like things should be working, just not enough experience to troubleshoot network routes, I can see traffic coming in and going to the IP phones but they are not updating
18:38.59PenguinHave you restarted any phone to see if it gets time when it starts up?
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19:19.47WIMPyWhen EXACTELY does Asterisk say "Spawn extension (...) exited non-zero"? I always thought it would only happen when it runs out of dialplan, but I see it on Hangup().
19:20.14[TK]D-Fenderthat's whenever the channel dies
19:20.23[TK]D-Fenderboth are reasons
19:21.15WIMPyIt doesn't seem consistant to me.
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19:46.39R1ppaPenguin, first thing I tried was a reboot, but having a 24hour resync time, lol , well maybe I should wait out the 24 hours or shorten the time to test....at least I am hoping thats the case
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19:47.48[TK]D-FenderChange the resync delay
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21:21.50*** join/#asterisk Simon-- (~sim@2606:6a00:0:28:5604:a6ff:fe02:702b)
21:22.16Simon--is there a manager command that (intentionally or not) emits a log entry? just for debugging if something is working..
21:23.01Simon--I found QueueLog(), but that isn't inband with what needs debugging
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21:27.25ipengineerDoes anyone know what would cause these channels to hang in the “Ring” state like this? This is v13: https://gist.github.com/zconkle/06cbab4eff3aba9e28e7
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21:42.16Simon--command agi exec verbose..sounds disgusting :)
21:43.03WIMPyAnd probably has side-effects.
21:52.54Simon--scanning source for ast_log()..
21:54.44Simon--[Oct 23 14:54:38] NOTICE[4240] manager.c: Invalid 'Cause: log me please' in manager action Hangup
21:54.45Simon--yay
21:54.56Simon--that was a bitch
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