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00:20.10 | pederindi | [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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02:58.27 | jdzielny | is there anyone here who's familiar with Asterisk CDR? I can't get it to work--at all |
02:58.27 | jdzielny | Literally nothing |
02:58.33 | jdzielny | the db is empty set |
03:14.10 | cmendes0101 | jdzielny: what database are you using? |
03:16.20 | jdzielny | Forgive me if this sounds ridiculously newbish--I'm a new user--but I don't know |
03:16.27 | jdzielny | I've got FreePBX as a frontend |
03:16.42 | jdzielny | I've tried without success to get some support over there |
03:16.48 | jdzielny | It's like a ghost town |
03:17.16 | cmendes0101 | Gotcha, I'm not familiar with Freepbx so might not be able to help with that |
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03:17.51 | jdzielny | well 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" |
03:17.53 | WIMPy | jdzielny: You won't have more luck in here. |
03:17.58 | WIMPy | ~freepbx |
03:17.58 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
03:18.51 | WIMPy | jdzielny: FreePBX has its own structure that goes beyond that of (plain) Asterisk. |
03:24.03 | jdzielny | At 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? |
03:24.29 | WIMPy | define "normally" |
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03:24.44 | WIMPy | Wheer do you even try to put them? |
03:25.03 | WIMPy | Where * |
03:29.14 | jdzielny | This is the problem -- I don't know how it works |
03:29.16 | jdzielny | At all |
03:29.30 | WIMPy | There are meny options. |
03:29.34 | jdzielny | I've tried and tried to find an overview of how Asterisk stores call records, and I keep going in circles |
03:29.45 | WIMPy | Where do you want them to go? |
03:29.57 | jdzielny | Honestly, I don't particularly care, as long as they go somewhere |
03:30.12 | WIMPy | You can write them to (a) file(s), to syslog, to AMI or to a database. for a start. |
03:30.13 | jdzielny | other than /dev/null, which is where they seem to be going |
03:30.57 | jdzielny | okay, well I don't expect a large volume, a file would be fine. as would a database |
03:30.58 | WIMPy | Start with cdr.conf. |
03:31.32 | jdzielny | cdr.conf is empty |
03:31.51 | WIMPy | Well, that surely explains why nothing happens. |
03:32.55 | jdzielny | okay... |
03:33.10 | jdzielny | but what is it supposed to contain? And how do I generate it |
03:33.11 | jdzielny | ? |
03:33.28 | jdzielny | I realize these may seem like basic questions, but I'm not stupid, just very new to this |
03:34.10 | WIMPy | Take the cdr.conf.sample file and check if the contents make sense to you or the wiki should haopfully have an explanation as well. |
03:35.36 | WIMPy | If you need some general starting point try one of these two: |
03:35.39 | WIMPy | ~primer |
03:35.39 | infobot | New to asterisk configuration? Check out this primer to get started. http://burner.com/asterisk-primer |
03:35.44 | WIMPy | ~book |
03:35.44 | 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 |
03:36.39 | WIMPy | (or both, off course :-) ) |
03:38.38 | drale | page 643 of that book |
03:40.33 | drale | i was looking at the pdf version |
03:41.39 | jdzielny | Well this is definitely a good place to start |
03:42.07 | jdzielny | Why in the world there's no cdr.conf but there IS a cdr_mysql.conf (and a .bak) is a mystery |
03:42.16 | jdzielny | freepbx is an odd fkng duck |
03:42.45 | WIMPy | If you're using FreePBX, don't mess with the config files. It will take revenge. |
03:43.02 | jdzielny | Can't possibly take revenge beyond doing nothing |
03:43.14 | WIMPy | Either find out how to clicl around or get rid of it. |
03:43.22 | WIMPy | click |
03:45.22 | WIMPy | And 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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03:46.15 | jdzielny | I'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 |
03:46.34 | jdzielny | And we need to be able to call out from them if necessary. It's not a complicated setup |
03:47.22 | jdzielny | It's for a controlled study in how employers respond to certain key words in applications |
03:48.33 | jdzielny | I like FreePBX mostly because it's got a slick GUI and is easy to use (except for CDR, apparently) |
03:50.06 | jdzielny | If 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 |
03:50.58 | WIMPy | If all you want is ring some device and go to voicemail after some time, just write extensions.conf yourself. |
03:51.29 | WIMPy | Or generate extensions.conf and voicemail.conf from a simple script. |
03:52.16 | jdzielny | I 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 |
03:52.34 | jdzielny | I can manage 100 extensions manually, there's no way I can do it with 1500 |
03:53.08 | WIMPy | I can manage hundreds with a text file, but I can't click 50 with a mouse. |
03:53.25 | jdzielny | You and I both know how to OPEN a text file, tho |
03:53.26 | jdzielny | lol |
03:57.18 | jdzielny | That's more than I can say for the people controllign the money here |
03:57.20 | jdzielny | :-\ |
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05:27.08 | Penguin | burnbrighter: Yes. |
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05:37.17 | ruben23 | hi 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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05:38.44 | WIMPy | Wrong Type Of Number, as usual? |
05:39.27 | WIMPy | http://voice.yeti.dk/Asterisk_vs_ISDN/5 |
05:42.23 | ruben23 | how about congestions..? |
05:42.53 | [TK]D-Fender | ruben23: I don't see you showing us the call. Asking without showing the call is a waste of everyone's time |
05:43.07 | WIMPy | What about congestions? |
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05:43.46 | ruben23 | sorry guys ill capture the call.. |
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06:00.16 | [TK]D-Fender | runs out of air while holding his breath |
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06:03.57 | WIMPy | Blue Fender :-) |
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07:04.49 | stevenm | Anyone know of an analogue phone that'll auto answer any incoming call? |
07:06.32 | WIMPy | stevenm: Any one with built in answering machine? |
07:07.10 | stevenm | WIMPy, no not answer & record - just answer |
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07:16.10 | ruben23 | WIMPy: hi there.. |
07:16.45 | WIMPy | ruben23: Where's your log. The fender already turned blue while waiting. |
07:16.55 | stevenm | something like this would be handy... http://oldphoneguy.net/BookletPDF/Visio-SCRcontrol5.pdf |
07:17.14 | ruben23 | http://pastebin.com/6kmCcmwF |
07:17.38 | ruben23 | they say incoming calls comes in and drop to busy tone |
07:17.42 | stevenm | just 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 |
07:18.25 | WIMPy | stevenm: Probably not the best idea to do it analog and rely on somethign like busy tone detection. |
07:20.09 | WIMPy | ruben23: Looks like you start an AGI that hangs up the calls. |
07:23.23 | WIMPy | Or rather immediately after that AGI. |
07:24.41 | ruben23 | WIMPy: what could i do somehow..? |
07:25.14 | WIMPy | Well, I don't see anything usefull in that log. More of a lot of crap. |
07:25.34 | WIMPy | Maybe you should debug that AGI? |
07:25.53 | WIMPy | And timing information would definitely help to get a sense of what's going on. |
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07:59.13 | bright_ | hello |
07:59.26 | bright_ | 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 |
07:59.53 | bright_ | what could cause it? |
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09:16.42 | MythosGR | scenario: [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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11:26.44 | marceloamorim | guys, anynone already get option h on the dial application when you didn`t set this option? |
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11:35.04 | marceloamorim | there is a dialplan called ael-builtin-h-bubble that have this h option |
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11:53.50 | Pernat | how to play background music and second audio with speech in IVR? |
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12:12.25 | erick2014 | Does 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. |
12:12.42 | erick2014 | I'm not under any time crunch, and I can spin up a new VM to do anything I need. |
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12:45.51 | erick2014 | does anyone have a working install of asterisk and pjsip? |
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12:49.05 | rmassih | Hey 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 |
12:49.15 | rmassih | the entries look like this: [callcentric1](Callcentric); |
12:49.16 | rmassih | host=alpha1.callcentric.com |
12:49.46 | rmassih | when 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 |
12:51.29 | *** join/#asterisk theron (~theron@199.201.65.135) |
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12:54.34 | [TK]D-Fender | rmassih: 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-Fender | rmassih: Take your pick. |
12:56.50 | erick2014 | can anyone see my messages? |
12:59.29 | *** join/#asterisk D-Boy (~D-Boy@unaffiliated/cain) |
13:01.10 | erick2014 | Hey, 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. |
13:01.19 | rmassih | Thanks 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 |
13:09.31 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
13:13.39 | *** join/#asterisk evilman_work (~evilman@87.244.6.228) |
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13:14.21 | erick2014 | hey, rmassih or fender, can either of you see my messages? |
13:14.35 | rmassih | yep I see them Erick |
13:15.10 | erick2014 | ok, 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. |
13:15.10 | kchehab | hi , 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) |
13:16.16 | kchehab | my dialplan can be found http://pastebin.com/auCsjzVV |
13:18.40 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
13:19.24 | marceloamorim | Guys, 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) |
13:23.35 | marceloamorim | I put a comment with " ; " in one line, so the pbx it was sending to ael-builtin-h-bubble |
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13:25.33 | jdzielny | hi there, has anyone ever seen this? "AOR [extension number] has no configured mailboxes. MWI subscription failed" |
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13:27.08 | kchehab | from 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 |
13:29.27 | infobot | Script for automating Fidonet polls. URL: http://www.drmach.demon.co.uk/vashti/software/index.html |
13:29.29 | [TK]D-Fender | ~polls |
13:29.30 | infobot | "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-Fender | erick2014: 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-Fender | kchehab: "h" <-------- |
13:32.13 | MaliutaLap | kchehab: https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Dial |
13:32.25 | erick2014 | Gotcha. 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 |
13:32.25 | erick2014 | <PROTECTED> |
13:33.11 | Cuzner | LOL Fidonet... that's a name i haven't heard in a while |
13:33.14 | MaliutaLap | "Carefully explaining your problem is half the solution." :) |
13:33.16 | Cuzner | used to have a node |
13:33.32 | MaliutaLap | Cuzner: I hope you had it removed properly ;) |
13:33.46 | [TK]D-Fender | Cuzner: Same here |
13:33.48 | Cuzner | lasers, how do they work? |
13:34.00 | MaliutaLap | off the heads of sharks! |
13:34.13 | [TK]D-Fender | strokes Mr. Bigglesworth |
13:34.25 | *** join/#asterisk Draecos (~Draecos@106-69-9-235.dyn.iinet.net.au) |
13:35.12 | MaliutaLap | [TK]D-Fender: well that makes a change from what you normally stroke :P |
13:35.32 | file | falls over |
13:35.40 | [TK]D-Fender | grabs his pen and stabs MaliutaLap with it. |
13:36.04 | [TK]D-Fender | MaliutaLap: Would you like another stroke? |
13:36.05 | MaliutaLap | [TK]D-Fender: you don't normally play with the stroke on your engine? |
13:36.23 | MaliutaLap | [TK]D-Fender: please sir, can I have some more? :P |
13:36.29 | [TK]D-Fender | MaliutaLap: My sister just got a Tesla this year... it doesn't have one ;) |
13:36.45 | MaliutaLap | no stroke to bore! |
13:37.11 | MaliutaLap | [TK]D-Fender: so your sister stopped stroking - that doesn't mean you have :) |
13:38.21 | jdzielny | Anyone who can help, what does AOR stand for in asterisk? |
13:38.27 | MaliutaLap | should off to bed. Got to be at the hospital for most of tomorrow afternoon |
13:38.52 | file | address of record |
13:39.01 | file | it's a SIP term |
13:39.27 | jdzielny | Okay |
13:39.43 | jdzielny | file, 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" |
13:39.51 | jdzielny | 2001 is an extension btw |
13:40.21 | file | What is registering to AOR '2001' is attempting to subscribe to get voicemail message waiting information but there is no mailbox configured on the AOR |
13:40.40 | Chainsaw | No mailbox, no status. |
13:40.59 | [TK]D-Fender | jdzielny: Address Of Record |
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13:41.15 | jdzielny | I think I'm misunderstanding something here. Extension 2001 does have a mailbox |
13:41.29 | file | it may have one, but it may not be configured in the pjsip.conf AOR |
13:41.59 | jdzielny | I'm new to this, file, can you explain what that means in a bit more detail? |
13:42.04 | [TK]D-Fender | jdzielny: And don't call SIP config entries "extensions" |
13:42.28 | [TK]D-Fender | jdzielny: Show us your config |
13:42.30 | [TK]D-Fender | ~pb |
13:42.31 | 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. |
13:42.32 | [TK]D-Fender | ^^^ |
13:42.52 | Cuzner | Aren't some of you guys in vegas? Why you up so early? |
13:43.03 | jdzielny | [TK]D-Fender, which config? |
13:43.05 | jdzielny | pjsip? |
13:43.08 | Cuzner | is jelly btw |
13:43.18 | [TK]D-Fender | jdzielny: Clearly |
13:43.47 | file | Cuzner, good question - why am I up so early... |
13:44.04 | Cuzner | file: seriously... you think leif is up right now? fat chance :P |
13:44.24 | Cuzner | or alan heh |
13:44.27 | file | Cuzner, he tweeted an hour ago |
13:44.35 | Cuzner | marquis, maybe... |
13:44.39 | jdzielny | [TK]D-Fender, there's a bunch of pjsip.whatever.conf files. which one did you want to see in particular? |
13:44.44 | file | @leifmadsen "Can't sleep. Clown will eat me. #astricon" |
13:45.01 | Cuzner | hah, like i follow that nerd. |
13:45.57 | [TK]D-Fender | jdzielny: pjsip.conf clearly... |
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13:46.06 | Cuzner | surprised he's up though, unless like he says in his tweet, he just didn't sleep. |
13:46.13 | file | jdzielny, is something else (like FreePBX) managing configs? |
13:46.20 | [TK]D-Fender | file: SHHH!!! |
13:46.27 | [TK]D-Fender | file: I'm handing out rope |
13:46.44 | Cuzner | teach a man to fish... something, something |
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13:46.55 | [TK]D-Fender | Cuzner: We're gonna need a bigger boat |
13:47.41 | jdzielny | file, yes |
13:47.46 | jdzielny | http://paste.ubuntu.com/8640803/ |
13:47.57 | [TK]D-Fender | jdzielny: This is NOT the place for FreePBX support |
13:48.09 | [TK]D-Fender | ~freepbx |
13:48.10 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
13:48.11 | [TK]D-Fender | ^^^ |
13:48.38 | Cuzner | ~trixbox |
13:48.39 | infobot | Delving 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. |
13:48.47 | Cuzner | ^_^ |
13:48.58 | Cuzner | i used to run so many of those things hehe |
13:48.59 | [TK]D-Fender | Cuzner: Not quite pertinent... but still somewhat funny |
13:49.00 | jdzielny | really, [TK]D-Fender, that was what that whole conversation was about? |
13:49.13 | [TK]D-Fender | jdzielny: What conversation? |
13:49.32 | [TK]D-Fender | jdzielny: You had mailbox questions... but you are not in charge of your own configs, FreePBX is. |
13:49.37 | [TK]D-Fender | jdzielny: it OWNS you. |
13:49.42 | Cuzner | [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 |
13:49.44 | jdzielny | Thank you for the lesson in slavery |
13:49.55 | [TK]D-Fender | jdzielny: So if that GUI isn't generating configs the right way, then it is at fault, not asterisk |
13:50.02 | file | any 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-Fender | jdzielny: ^^^ |
13:50.18 | file | the only permanent thing is to do it from within the FreePBX system |
13:50.25 | Cuzner | FOP was the bees knees |
13:50.27 | file | (if it permits it, I'm unsure of that) |
13:50.27 | Cuzner | hides |
13:51.10 | jdzielny | The 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-Fender | jdzielny: Well * told you the AOR didn't have a mailbox defined in it. |
13:52.58 | [TK]D-Fender | jdzielny: #include pjsip.aor.conf <--- so this probably shouldn't have been a question |
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13:55.01 | jdzielny | I 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-Fender | jdzielny: read the sample configs to see where these parameters belong |
13:57.02 | jdzielny | and 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 |
13:57.22 | Chainsaw | jdzielny: That's good news. That means that only one is misconfigured. |
13:57.25 | jdzielny | and the entries in pjsip.aor.conf are the same for all of them |
13:57.34 | file | subscribing for voicemail mwi is not mandatory, other stuff may not use that mechanism |
13:57.43 | file | the other way is to simply tell the device the information without it asking |
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14:01.55 | file | it'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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14:03.00 | RadJackson | Hello , 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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14:03.48 | RadJackson | <PROTECTED> |
14:04.00 | RadJackson | once i reboot internet , everything gets back to normal , any idea ? |
14:04.51 | RadJackson | What can i do to avoid such problems? Call a network guy ? create aVLAN specially for my softphones ? |
14:05.53 | [TK]D-Fender | RadJackson: 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-Fender | RadJackson: Lesson: Don't use Windows as a router |
14:06.51 | jdzielny | alright I'll figure it out. If all else fails I'll just delete it and put it back in |
14:06.58 | jdzielny | Maybe that'll do the trick |
14:08.19 | RadJackson | [TK]D-Fender The router isnt Windows its Juniper, its the domain controller that is WIndows Server 2003. |
14:09.05 | [TK]D-Fender | RadJackson: Seems your packets are going through it... |
14:09.21 | RadJackson | ok , shud i call a network guy ? |
14:09.59 | [TK]D-Fender | RadJackson: What is this "connected via Xserver" you mentioned actually mean? |
14:10.43 | RadJackson | i didnt explain well , i meant that the windows server 2003 has DHCP role |
14:11.30 | RadJackson | it isnt a sip client problem , ive got Eyebeam softphones and CISCO IP PHONE SPA504G aswell |
14:11.38 | RadJackson | i start getting the lag on both |
14:12.10 | RadJackson | i'm not much of a network guy... |
14:12.41 | RadJackson | Everytime 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-Fender | RadJackson: Flooded connection means slow responses as well... |
14:14.23 | [TK]D-Fender | RadJackson: Consider traffic shaping. |
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14:19.55 | RadJackson | [TK]D-Fender ok thank you |
14:22.29 | jdzielny | file, 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-Fender | jzdAOR's have a mailbox parameter as shown in the sample config |
14:22.55 | file | if there is no mailboxes option set for the AOR, then it's not set |
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14:25.12 | jdzielny | file, 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 |
14:25.58 | file | that 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 |
14:26.43 | jdzielny | it's in pjsip.endpoint.conf |
14:27.02 | [TK]D-Fender | jdzielny:doesn't sound like "AOR" |
14:27.15 | [TK]D-Fender | jdzielny: And we have no prrof of what's actually in that file |
14:27.20 | [TK]D-Fender | jdzielny: Location counts |
14:28.41 | jdzielny | http://paste.ubuntu.com/8641210/ <--pjsip.aor.conf |
14:28.42 | jdzielny | http://paste.ubuntu.com/8641211/ <--pjsip.endpoint.conf |
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14:29.07 | file | there is no mailboxes= in the AOR |
14:29.14 | file | that's why it is not working |
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14:35.09 | jdzielny | hm |
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14:43.10 | R1ppa | Time showing wrong on all phones after a power outage, some route missing somewhere? |
14:47.14 | [TK]D-Fender | R1ppa: The phones haven't refreshed from whatever time server they were using. Either waiting for a resync period, or some networking issue, etc |
14:47.20 | ChannelZ | Well 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-Fender | SIP doesn't distribute time |
14:48.06 | R1ppa | [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-Fender | reboot them now. Maybe the phone came up before they were able to reach their time server |
14:48.52 | [TK]D-Fender | And gave up |
14:48.59 | R1ppa | [TK]D-Fender, have been rebooted several times, will try again |
14:49.04 | Cuzner | sounds like the most plausible expolaination |
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14:52.39 | Cuzner | R1ppa: 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. |
14:52.46 | ChannelZ | hmm 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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15:14.38 | Chainsaw | ChannelZ: It's correct on the "all Asterisk versions" though. |
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15:22.20 | olspookishmagus | is there any live feed from Astricon? |
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15:25.10 | [TK]D-Fender | olspookishmagus: The food is typically killed before serving |
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15:28.47 | olspookishmagus | so we have to wait |
15:28.57 | mbowie | lolz |
15:29.28 | Cuzner | [TK]D-Fender: but gagh is best served live! |
15:30.24 | [TK]D-Fender | MORE BLOOD-WINE!!! |
15:30.31 | WIMPy | Does 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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15:50.06 | nicknam1232 | #JP Konichiwa |
15:50.37 | nicknam1232 | Consider° operator ** FR - france ** |
15:59.26 | Cuzner | damned FR locale with all their damned emergency numbers... |
15:59.43 | Cuzner | i swear france has an emergency short code for stubbing your toe on a fire hydrant |
16:00.43 | WIMPy | You need a manual for making an emergency call there? |
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16:14.04 | nicknam1232 | consider{[é] |
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16:25.02 | nicknam1232 | Souix Exchange - McC........, nombre. : - ) :-) * _ * [xnr2] 757 |
16:25.28 | Chainsaw | I think it wants to communicate. |
16:26.09 | [TK]D-Fender | Probably just an autonomic response. |
16:26.16 | [TK]D-Fender | Poke it with a bigger stick |
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16:35.03 | nicknam1232 | ... Danke Scheun - ^ - ^ |
16:35.12 | file | falls over |
16:35.17 | nicknam1232 | * Merci. Beaucoup. |
16:35.32 | nicknam1232 | ? Ville danke . |
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16:47.49 | boatmn810 | ok, stupid question - make menuselect - any difference between ncurses and newt versions ? |
16:47.56 | boatmn810 | and how to choose which one ? |
16:48.31 | Penguin | I'd imagine there has to be a difference or there wouldn't be two names. |
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17:02.07 | Stefan27 | asterisk 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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17:03.20 | Stefan27 | How can I debug this, or how can i give you relevant info? here's a thread dump http://pastebin.com/RxTKXJJh |
17:04.13 | WIMPy | boatmn810: If you have newt, that will be used. The advantage is color, the disadvantage is keyboard settings. |
17:04.44 | WIMPy | Stefan27: There's a debug fd leaks option. |
17:05.03 | WIMPy | But I haven't tried to use it. |
17:05.13 | boatmn810 | i have newt, but its still not using the color screens - so far thats the only diff between 2 systems that I can find - |
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17:05.48 | boatmn810 | in which the logger is giving me problems on the one without newt |
17:05.58 | Stefan27 | stracing 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 |
17:06.15 | boatmn810 | correction - with newt |
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17:08.31 | Stefan27 | looks like some thread is stuck in an infinite while loop on line 1706 in res_rtp_asterisk.c? |
17:09.14 | Stefan27 | but what puzzles me is that that function is surrounded by a block #ifdef HAVE_PJPROJECT ... #endif |
17:09.37 | Penguin | ~spam |
17:09.37 | infobot | methinks 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. |
17:09.54 | Penguin | useless |
17:10.12 | Stefan27 | i didn't even load any asterisk module with the name "pj" in it |
17:10.16 | Stefan27 | i use chan_sip |
17:10.29 | boatmn810 | do you have it all turned off ? |
17:10.48 | Penguin | It has to be related to pjsip. |
17:10.58 | Stefan27 | I did compile asterisk with pjprojects |
17:11.20 | Stefan27 | but i decided only to load chan_sip and make no reference to pjsip stuff from dialplan |
17:11.20 | WIMPy | Maybe you should try without. |
17:11.58 | Stefan27 | I will - but it took me a long time to produce this bug |
17:12.12 | Stefan27 | more than 10000 calls |
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17:12.26 | boatmn810 | thats about 1 1/2 days for me |
17:13.01 | boatmn810 | so box has been up awhile ? |
17:13.03 | boatmn810 | how logn / |
17:13.07 | boatmn810 | how long ? |
17:13.08 | Stefan27 | debug-fd-leak option requires special compilation flags? |
17:14.15 | WIMPy | It's in menuselect |
17:15.41 | Stefan27 | can i from the OS force a close of these FDs without restarting the asterisk process? |
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17:23.05 | Stefan27 | Yeah I closed them, so if they were not actual leaks asterisk will likely seg fault later |
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17:48.53 | jeev | is asterisk on switchvox any different than asterisk opensource? |
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17:52.35 | R1ppa | had 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-Fender | R1ppa: Exactly what we've already told you hours ago |
17:55.42 | R1ppa | [TK]D-Fender, "or some networking issue" is what I am working on |
17:56.26 | [TK]D-Fender | Have you dumped traffic to see if it's looking? |
17:56.34 | [TK]D-Fender | Because this isn't an Asterisk issue... |
17:56.59 | R1ppa | [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-Fender | We 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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18:15.54 | 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 |
18:16.51 | Penguin | Is the time some hours and some minutes wrong, or just hours? |
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18:17.28 | R1ppa | Penguin, everything lol Dec 31st |
18:17.43 | Penguin | IP phones? |
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18:18.23 | R1ppa | Penguin, yarrr Polycom soundpoint 430 |
18:18.26 | Penguin | They 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 |
18:18.39 | R1ppa | Penguin, yes, calls inbound and out are fine |
18:20.47 | R1ppa | [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-Fender | R1ppa: have you tried changing the resync time on it? Confirmed where it's actually applying settings from? |
18:34.16 | [TK]D-Fender | R1ppa: Polycoms can grab from multiple places but may use just the one specified in its configs |
18:36.59 | Penguin | Can'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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18:38.19 | R1ppa | Penguin, 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 |
18:38.59 | Penguin | Have you restarted any phone to see if it gets time when it starts up? |
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19:19.47 | WIMPy | When 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-Fender | that's whenever the channel dies |
19:20.23 | [TK]D-Fender | both are reasons |
19:21.15 | WIMPy | It doesn't seem consistant to me. |
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19:46.39 | R1ppa | Penguin, 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-Fender | Change the resync delay |
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21:22.16 | Simon-- | is there a manager command that (intentionally or not) emits a log entry? just for debugging if something is working.. |
21:23.01 | Simon-- | I found QueueLog(), but that isn't inband with what needs debugging |
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21:27.25 | ipengineer | Does anyone know what would cause these channels to hang in the âRingâ state like this? This is v13: https://gist.github.com/zconkle/06cbab4eff3aba9e28e7 |
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21:42.16 | Simon-- | command agi exec verbose..sounds disgusting :) |
21:43.03 | WIMPy | And probably has side-effects. |
21:52.54 | Simon-- | scanning source for ast_log().. |
21:54.44 | Simon-- | [Oct 23 14:54:38] NOTICE[4240] manager.c: Invalid 'Cause: log me please' in manager action Hangup |
21:54.45 | Simon-- | yay |
21:54.56 | Simon-- | that was a bitch |
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