IRC log for #asterisk on 20090921

00:00.11drmessanoThats not built into Eyebeam.. Maybe MyspaceIM has it
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00:03.11Katty:>
00:03.12Kattyhas freshly laundered laundry
00:06.06dandate2hmm freshly laundered money mmm
00:07.17drmessanowatches the used car salesman at work
00:08.07phixKatty: :D
00:10.49dymaxionhi anyone here use Asterisk-GUI?  the other room is v quiet.
00:11.05dymaxionWoudl you recommend not using a GUI if possible?
00:11.10Kattyinfobot: gui?
00:11.10infobotgui is, like, (Graphical User Interface) Any system that uses graphics to represent the functions of a program. All Windows operating systems are GUI. or point-and-drool, double-click, drag-and-drop, ... or at http://pla-netx.com/linebackn/guis/index.html.  Of course Real Programmers use the command line interface.  See cli
00:11.59phixReal Programmers also live in their parents basement too
00:12.50dymaxion:-)   indeed.. well this is for small 10 person office.. I'm happy playing around with config files , however other users who may end up administering may not be comfortable with this...
00:13.32dymaxionI was suprised though.. the vanilla install that I was delivered with my rowetel blackfin IP04,  had a load of pre-configured asterisk files  (eg.  extensions.conf  is full of loads of seemingly irrelevant stuff.. is this normal ?
00:14.34dymaxiondemo contexts,  Context 'numberplan-international, Context 'numberplan-iaxtel' created by 'pbx_config'   etc.. is this because I installed the GUI?
00:14.52dymaxioni'm considering going back to command line, but not sure waht to delete in order to get a clean start
00:16.10TJNII~cli
00:16.11infoboti heard cli is a Command Line Interface, the best form of interface around, of course  Call-Level Interface, originally developed by SQL Access Group, but now known as Microsoft's ODBC standard. an x86 assembly instruction  Common Language Infrastructure (See mono or .net)
00:16.37dymaxionthe GUI as far as I can tell,  dynamically generates the asterisk.conf files.   I hope by installing the GUI i haven't broken everything... at the moment, SIP clients cannot register...  any ideas why?
00:17.06TJNIIThat link in ~gui is no good anymore.  Someone with bot rights should remove it.
00:17.30dymaxionhttp://pastebin.org/19153
00:17.46dymaxioni get  SIP/2.0 403 but can't figure out why
00:21.52drmessano~gui
00:21.53infobotgui is, like, (Graphical User Interface) Any system that uses graphics to represent the functions of a program. All Windows operating systems are GUI. or point-and-drool, double-click, drag-and-drop, such as http://toastytech.com/guis/miscwin1xp.png.  Of course Real Programmers use the command line interface.  See cli
00:22.07drmessanoTry that link, TJNII
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00:31.08TJNIII did.  4 Times.  Got a parking site
00:31.23TJNIIWhoops, Ignore that!
00:31.32TJNIIMuch better.
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01:01.04Kattymariah carey has such an awesome voice. it's a shame she wasted it.
01:01.14darkdrgn2k3Hey guys, is there a way to ring multiple extensions an inbound route?
01:01.30Kattydarkdrgn2k3: Dial(SIP/100&SIP/101) etc
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01:01.38darkdrgn2k3Katty: thank you
01:01.42Kattydarkdrgn2k3: mhmm
01:05.18[TK]D-Fenderdarkdrgn2k3: #freepbx <------
01:08.32darkdrgn2k3[TK]D-Fender: wasnt a freepbx question :-P pure asterix dialplan question
01:08.44darkdrgn2k3but i should hang around there to;)
01:09.09[TK]D-Fenderdarkdrgn2k3: Considering you're using FreePBX it IS a GUI question
01:09.31[TK]D-Fenderdarkdrgn2k3: You are in dialplan territory and you have sold your soul to the lowest bidder.  Now move along...
01:10.07darkdrgn2k3[TK]D-Fender: not if im treding in the "custom" conf files  :-P
01:10.26darkdrgn2k3but i guess your right
01:10.30darkdrgn2k3as always...
01:10.30[TK]D-Fenderdarkdrgn2k3: uhhh huh
01:11.05darkdrgn2k3LOL im not gonna live any of this down till i reinstall from scratch am i?
01:11.11darkdrgn2k3and probably even then :-S
01:12.17[TK]D-Fenderdarkdrgn2k3: You're like that guy who keeps trying to order Big Macs..... at BURGER KING.
01:12.55darkdrgn2k3[TK]D-Fender: i once orderd a vegie backen burger.. you should of seen the looks i got :-D
01:22.57dmzhey y'all, just heard that web-meetme has a sql injection and remote script execution
01:24.31[TK]D-Fender\o/
01:25.00dmznot sure the details, but they were sent to the developers a few hours ago
01:26.06drmessanoPeople use that?
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01:32.39TarantulafudgeI'm building an asterisk box for the company I work for, and I was just wondering whether I should go with Core2Quad or XeonQuad
01:33.06TarantulafudgeI was thinking that the Core2Quad would probably be better for transcoding
01:33.24Tarantulafudgebut I'm not sure, and I don't have any facts to back that up
01:33.25[TK]D-FenderTarantulafudge: xeon doesn't say what core tech.
01:34.06Tarantulafudgeits a comparible 64bit model
01:34.33Tarantulafudgegot it
01:35.56Tarantulafudgeeither the  Core2Quad Q8200 at 2.33Ghz (4mb) or the XeonQuad 2.4 x3200 (8mb)
01:36.56Tarantulafudgeprice difference is about $20
01:37.02Tarantulafudgethe Core2 being cheaper
01:37.13Tarantulafudge[TK]D-Fender: any idea?
01:37.57[TK]D-FenderTarantulafudge: Idea : go read the core specs and reviews if you have trouble interpreting the core specs
01:38.20[TK]D-FenderTarantulafudge: And I dunnooo... maybe BENCHMARKS
01:38.40TarantulafudgeI can't...
01:39.12darkdrgn2k3Tarantulafudge: stupid question perhaps buy Why?
01:39.20darkdrgn2k3actualy i have a feeling its a stupid answer :-P
01:39.27TarantulafudgeI don't have the hardware I have to order it
01:39.42darkdrgn2k3Tarantulafudge: reaserch.. they post benchmarks
01:39.44TarantulafudgeI was just wondering which would be better for use with asterisk
01:40.08[TK]D-Fenderwww.tomshardware.com
01:40.12Tarantulafudgebenchmarks don't tell me much about asterisk performance
01:40.13darkdrgn2k3http://www.lmgtfy.com?q=xeon+benchmark+chart
01:40.21[TK]D-FenderTarantulafudge: Reviews have BENCHMARKS.
01:42.35darkdrgn2k3wow look at that "High End CPU's"
01:42.42darkdrgn2k3Intell AND amd :-P
01:43.20Kattystretches
01:43.23Kattymmmmmmmmmmmrelaxed.
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02:35.16psilikonI'm trying to test out an asterisk setup. Can I just add an extension to extensions.conf that will playback audio?
02:36.03TJNIIYep.
02:37.21psilikonI have added extension, yet when I try them I get extension: '600' rejected because extension not found
02:37.58TJNIIDid you reload the dialplan?
02:38.36psilikonYes. When I do a 'dialplan show' I get everything from extensions.ael. Do I need to disable extensions.ael?
02:38.49psilikonDoes it have priority over extensions.conf?
02:40.50TJNII* should use both, AFAIK.
02:41.32[TK]D-Fenderpsilikon: Whatever you added either isn't properly formatted, or in the right place
02:41.40psilikonI am trying to follow Asterisk: TFOT 2nd edition. yet I cannot get Echo() to work
02:41.51TJNIIpsilikon: Pastebin what you have.
02:42.02TJNII~pb
02:42.03infobot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding 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 apt-get install pastebinit
02:42.41psilikonhttp://pastebin.com/m22fc0fb7
02:44.36TJNIIAnd what do you have context= set to in sip.conf
02:45.03psilikonUnder general?
02:45.32TJNIIIf you don't have it set on a per-phone basis, yes.
02:46.24psilikonUnder general: context=from-callcentric
02:46.55TJNIIAnd do you have a seperate context= set on the phone you are using?
02:47.08psilikonyes
02:47.09TJNIIOr, should I say, the entry for thwe phone you are using
02:47.16TJNIIAnd what is that?
02:47.24psilikoncontext=to-callcentric
02:48.09TJNIISo all calls go out to callcentric, correct?
02:48.33psilikonyes
02:48.48TJNIINow, you have programmed your test extensions in both [default] and [internal], but you have no included those contexts anywhere
02:49.51TJNIISo those extensions don't work for you, basically, because * doesn't know it is supposed to use those extensions for that peer.
02:49.56psilikonOk, you made me understand that I need those extensions under the right place in the extensions.conf.
02:50.13psilikonI am having trouble understanding the basics...
02:50.13TJNIIYep
02:50.47psilikonso it is sending everything from that peer (the ATA phone) to [to-callcentric]
02:51.02TJNIIYep
02:52.04TJNIIAlso, that patterm match looks like it will only match two digit numbers, which is probably not what you want.
02:52.36psilikonOk that is cool. I am sitting here trying to follow this Asterisk book..
02:53.07psilikonProbably not. How do I change it.
02:53.15psilikonWhat other test sounds can asterisk provide?
02:54.03TJNIITest sounds?  Whatever you want, I guess.  (Provided you have a sound file for it.)
02:55.26psilikonReally? So how do I link an mp3?
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02:56.50TJNIIAFAIK, and with the exception of MOH, you need to recode it.
02:57.41psilikonTo wav?
02:57.41[TK]D-FenderTJNII: No
02:57.43TJNIIBut, I'm sure if I'm wrong [TK]D-Fender will call me on it.  He has a way of doing that.
02:57.48TJNIISee, beat me to it!
02:57.51[TK]D-FenderTJNII: Too late :p
02:58.32[TK]D-Fenderpsilikon: PASTEBIN is your friend.
02:58.34[TK]D-Fender~pb
02:58.35infobot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding 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 apt-get install pastebinit
02:58.36[TK]D-Fender^^^^^^^^^
02:59.12[TK]D-Fenderpsilikon: Would help to see your actual dialplan.  Hopefully it isn't based of that phsycho-kludge sample from the tarball...
02:59.18TJNII[TK]D-Fender: His contexts are messed up.  He pastebinned it earlier.
02:59.48psilikonso how do i make a 7 digit number go out through the sip provider?
03:00.10[TK]D-Fenderpsilikon: like you are already doing : exten => _XX,1,Dial(SIP/${EXTEN}@callcentric)
03:00.26[TK]D-Fenderpsilikon: Only using a pattern that will match the kind of # you want to dial out
03:00.39[TK]D-Fenderpsythat matches any 2 digits
03:00.43TJNII[TK]D-Fender: Also, I didn't see mp3 in a core show file formats, so I assumed playback wouldn't play mp3s.  Is there another function, or am I missing mp3 support on my system.
03:00.58[TK]D-FenderTJNII: Asterisk-addons <-
03:01.07TJNIInods
03:02.07psilikonThis doesn't make sense to me yet: exten => _XX,1,Dial(SIP/${EXTEN}@callcentric What do I need to read so I can start make sense of the logic?
03:03.58darkdrgn2k3psilikon: you only want 2 digit phoen nubmers?
03:04.03TJNIIX Matches any single digit from 0 to 9.  So XX matches 00-99.
03:04.27darkdrgn2k3psilikon: i think you want I think you want NXXXXXX
03:04.30darkdrgn2k3oops
03:04.49darkdrgn2k3N=2-9 X=0-9  :)
03:04.49TJNIIpsilikon: http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html/asterisk-book.html#asterisk-CHP-5-SECT-3.6
03:04.58darkdrgn2k3or that
03:06.23[TK]D-Fenderpsilikon: I can also tell you that there is no way callcentrix is going to accept you dialing a 7 digit number
03:06.43[TK]D-Fenderpsilikon: Now you may consider a 7 digit number local to you because of an assumed area code
03:07.16[TK]D-Fenderpsilikon: at which point if you want to dial that # you'll want to pass callcentric the areacodealong with
03:07.26[TK]D-Fenderpsilikon: At which point I may as well hand you the answer :
03:07.53[TK]D-Fenderpsilikon: exten => _NXXXXXX,1,Dial(SIP/555${EXTEN}@callcentric)
03:08.00TJNIIOne of the handiest (imho) things I do with * is automatically add the area code to 7 digit numbers so they are past to my ITSP as 10 digits.
03:08.04[TK]D-Fenderpsilikon: replacing 555 with your preferred default area code
03:08.21drmessanoTJNII: Fuckin enabler!
03:08.21TJNIII get so annoyed with telcos that make me do 10 digit dialing for #s in the same area code....
03:08.34drmessanohisses at TJNII
03:08.45drmessano10 digits 4EVA
03:08.55TJNIIsplashes Holy Water on drmessano
03:09.00drmessanolol
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03:09.08drmessanoNoooooooo
03:09.27p3nguinpsilikon: Are you trying to match outgoing CallCentric numbers?
03:09.33drmessanobursts into flames and smashes blindly through the door into #trixbox
03:09.35[TK]D-Fenderpsilikon: Another quick lesson : ${EXTEN} hold the number that patched the first part of that EXTEN => line.,  As you can see by the action we are taking (appllication : Dial()) we are passing that # as part of the tech we are dialing and quite literally shoving digits in front
03:10.31[TK]D-Fenderpsilikon: *'s use of variables is PRECISELY as dumb as it looks.  it is plain text substitution.
03:25.07psilikonI have a hard copy of Asterisk: The Future of Telephony 2nd Edition however I guess I am getting confused when it comes to the fact that I am using a SIP provider
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03:32.26[TK]D-Fenderpsilikon: the fact they are an ITSP isn't really the issue.  its a question of knowing what numbers the tech you are calling is expecting.  Many ITSP's expect you to even dial the 1 in front
03:33.14psilikonYeah, I think callcentrics expects a 1
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03:33.37[TK]D-Fenderpsilikon: Where I am, 10 digit dialing is madatory.  This started maybe 4-5 years ago, and to accomodate old habits, I too allow 7-digit dialing and prefix it myself for what would be a relatively safe assumption of the primary area code here
03:36.01psilikonthat is what I would like to do.
03:36.38psilikonSo you guys are telling me that even though I am using an ITSP I can still get everything I need to know out of *: TFOT?
03:40.02psilikonI am trying to place a test call from my ITSP but I am getting the extension not found message.
03:44.13[TK]D-Fenderpsilikon: an ITSP is just one more thing * can talk to
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03:44.48[TK]D-Fenderpsilikon: the book teaches to basics of what different modules can interacto with.  The dialplan is the core of everything.  It is 95% of * and something you must master
03:44.51drmessanopsilikon: What would an ITSP have to do with anything?  Asterisk can speak SIP, IAX, Skype, DAHDI, Skinny, MGCP, OH323, Jingle, Bluetooth, etc
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03:45.33psilikonI hear ya. It is just that right now the dialplan is making little sense.
03:45.47drmessanoThat has nothing to do with using an ITSP
03:45.53psilikongood
03:47.09drmessanoThe dialplan could care less about what technology you're using, as a general rule.. There are protocol specific aspects, but "DO THIS, DO THAT" is protocol agnostic
03:48.05psilikonI really appreciate the help/lessons from you guys. I'm gonna get this stuff. Gotta o to bed for now tho.
03:48.08drmessanoWait til you start writing dialplan to patch around glitchy phones, asterisk bugs, or a shitty analog card
03:48.17[TK]D-Fenderpsilikon: If you configure the SIP parameters with an ITSP, you typcially receive calls from them.  these calls enter the dialplan and are processed however you tell it to
03:48.42drmessano[TK]D-Fender: Unless your name is "S"
03:48.49drmessanoThen, youre screwed
03:48.56[TK]D-Fenderpsilikon: If you call a number from a SIP phone (sort, hard, or whatever), it also enters the dialplan.  Same with a Zaptel FXO (line) interface, FXS,e tc
03:49.22[TK]D-Fenderpsilikon: * processes calls.  Setting up the device itself is petty.  what matters is what you do with calls.  that is the dialplan
03:49.33drmessanoAsterisk: The VoIP wirenut
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03:54.15joobiehey guys.. is tehre any published dialplan i can use to figure out which numbers are dialling where? currently if i get a number dialed that's not in my list of known numbers, i then manully research to find out what regex would suit the number and where it goes
03:54.18joobiegetting to be a PITA
03:58.23[TK]D-Fenderjoobie: ... huh?
04:00.13joobieTK, when users dial a number it goes through to the SIP provider and bills us at whatever.. my billing system then parses through the CDR and classifies the number to a country (using regex on the dialed number) and then bills accordingly.. at the moment each month i have like 5 new numbers taht my regex doesnt handle, so i have to manually google to try figure out a regex for it.. just wondering if there's a good source to get all the number combina
04:00.13joobietions and where they dial to
04:00.49[TK]D-Fenderjoobie: this is an abstract non-* programming thing...
04:01.30[TK]D-Fenderjoobie: If you hit an unknown pattern how would you know how to brak down a number?
04:03.35joobieif it's unknown, it barfs at the moment
04:03.45joobiethen i manually investigate, put a regex in place and it moves along
04:04.16joobieso each month, i build more and more regex into my billing.. but i'd rather than get a huge list of all countries and do all the regex combinations in one hit
04:14.52korolevis this for countries?
04:15.05korolevor for any dialing pattern?
04:16.47korolevjoobie for made up dialing patterns or even local calls that would overlap with international country codes you would need separate tables
04:17.25korolevif its only international, make a db table to store all country codes, you can get them from here: http://voipjet.com/ratescsv.php
04:18.03korolevand then use Longest Common String instead of regexes to figure out the country code out of the dialed string
04:18.14joobieit's for multiple countries
04:18.45korolevgrab the csv from voipjet, thats pretty accurate except for US area codes
04:18.56korolevbut it contains all other nanpa codes
04:19.18joobiecool exactly what i was after
04:19.18korolevand all international, broken by mobile and fixed
04:19.19joobiethanks man
04:19.22korolevnp
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04:34.28drmessanoYay, unbanned
04:35.59[TK]D-Fender?
04:38.23drmessanoNah, not really
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05:02.08VarnishedOtterHi, does anyone here have experience with app_konference? I have it working but it has extremely high latency (3-5 secs)
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06:19.23alexhqHow to configure asterisk to use radius for authentication?
06:24.43KyleKhuh les.net is adding a monthly fee for the pleasure of having an account with them
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06:32.00korolevwith the rates they have to anywhere i call, its really no pleasure
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06:49.56Guggess
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06:54.13tm1985I have a analog line and uses dahdi what is then the best exten for dialing *21*?
06:54.28Micc_I need a way to play periodic messages to people on hold.
06:54.33Micc_In a queue.
06:56.59fiddurMicc_: http://www.voip-info.org/wiki/view/Asterisk+config+queues.conf
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07:13.22maxagazis it better to use ogg or gsm to compress sound with asterisk ?
07:13.50maxagazthe problem with gsm is that there is no graphical application to read it on linux
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07:34.59tzafrir_laptopmaxagaz, 'play' plays it
07:36.50maxagaztzafrir_laptop, play is not a graphical player, it's okay for me, but not for my colleagues
07:37.20tm1985I have a analog line and uses dahdi what is then the best exten for dialing *21*?
07:37.48tzafrir_laptopmaxagaz, what do you use?
07:38.05maxagaztzafrir_laptop, play
07:38.14maxagaztzafrir_laptop, but i'd like an alternative
07:38.20tzafrir_laptoptry 'WAV' . It is wav/gsm . Same size as gsm, but a wav container
07:38.42tzafrir_laptopso players are more likely to detect it
07:39.01maxagaztzafrir_laptop, i'm also thinking about changing the compression format to ogg
07:39.09tzafrir_laptoptm1985, Dial(DAHDI/1/*21*); ?
07:39.11maxagaztzafrir_laptop, wav? better than flac ?
07:39.39tzafrir_laptopmaxagaz, that is like asking if vorbis is better than wav
07:39.53tzafrir_laptopwav and ogg are containers, not codecs
07:40.39tzafrir_laptopfurthermore, Asterisk uses 8kHz, 16 bits per sample, mono
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07:41.33tzafrir_laptopflac is compressed, but not lossy, so here's a limit to what it can compress
07:41.44tzafrir_laptopThis means that it doesn't loose information
07:42.00tzafrir_laptopBut sometimes you don't really care about information
07:42.41maxagaztzafrir_laptop, ok, then i think wav is a good solution
07:43.07tzafrir_laptopgsm, speex and others are not only lossy, but also intended to preserve human voice - "speech"
07:43.19maxagaztzafrir_laptop, but i thought flac was designed to replace wav
07:44.00tzafrir_laptopI guess you can save some disk space with flac
07:44.11tzafrir_laptopideally ~50%
07:44.26tzafrir_laptopOTOH, it takes more CPU time compressing
07:44.44maxagaztzafrir_laptop, if gsm, speex and others are containers, how can they affect the quality of sound ? isn't it the codec that affect it ?
07:44.59tzafrir_laptopgsm and speex are codecs
07:45.09tzafrir_laptopand yess, they are lossy
07:45.40tzafrir_laptopgsm has a very noticable effect. speex: less so
07:48.36maxagaztzafrir_laptop, so gsm is both a codec and a container ?
07:48.49tzafrir_laptopno. It's just a codec
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07:49.19tzafrir_laptopa .gsm file is a raw file - merely the content of a stream, with no headers
07:49.22maxagaztzafrir_laptop, you mean that my .gsm sound files have no container ?
07:52.31maxagaztzafrir_laptop, what is the interest of putting a raw file in a container if it can be played directly ?
07:52.47tm1985when I try to call an the analog channel I got this output http://pastebin.com/d5e8b6e8a
07:53.49tzafrir_laptopWhat exactly is *21* ? What exactly is 0498506822# ?
07:54.55tm1985*21* a number from our provider so that we can redirect call to a number If we do this the call will be redirect by the provider and not asterisk
07:55.35tm1985the number a the number of certain phone and it has to end with a # because the context is *21*TELNR#
08:01.14tm1985do you have any idea?
08:02.02tzafrir_laptopmaybe you need to wait a second or so after the *21* ?
08:02.23tzafrir_laptopIn that case, dial *21*ww0498506822#
08:02.44tzafrir_laptopThe 'digit' 'w' means 'wait for half a second'
08:03.57tm1985I just tried to call an normal number I does It what that also
08:06.31tm1985http://pastebin.com/d7bf15899 this is the dahdichannel!! Is maybe something wrong or missing?
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08:26.01tm1985http://pastebin.com/d7bf15899 this is the dahdichannel!! Is maybe something wrong or missing?
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08:43.59maxagaztzafrir_laptop, thanks for your explanation, i understand a lot better now
08:45.03tzafrir_laptoptm1985, what does it do, exactly?
08:45.23tm1985the *21*?
08:46.33maxagaztzafrir_laptop, wav is a codec with no comrpession at all ?
08:47.28maxagazmaxagaz, sorry, stupid question...
08:47.29tzafrir_laptopwav is not the codec
08:47.48tm1985When we dial this, the provider redirect to call to the telephone number you put into *21*TELNR# then the call doesn't comes to asterisk
08:50.29maxagaztzafrir_laptop, what i don't understand is that when i use wav/gsm, the sound is smaller than when i use gsm only
08:51.00maxagaztzafrir_laptop, but wav add a header, so it should make the sound bigger, isn't it ?
08:51.19tzafrir_laptopit shouldn't be smaller. It should be a few bytes larger
08:52.43maxagaztzafrir_laptop, it does this when converting a gsm file into wav
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09:00.12maxagazalso, what's the difference betwee gsm and speex ?
09:00.25maxagazwhy not using ogg/speex by default ?
09:01.05maxagaz(as it is all free software)
09:01.14tm1985I have a analog line and uses dahdi what is then the best exten for dialing *21*?
09:01.43troffaskymaxagaz, core show translation will tell you why :-)
09:06.15maxagaztroffasky, what does mean the table it displays ?
09:07.29maxagaztroffasky, it means that speex is much slower...?
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09:08.54s14ckhow can I do to detect when somebodys pickup the call through fxsks device?
09:11.23tzafrir_laptopit means that it takes more CPU time
09:11.58tzafrir_laptopbtw: the table as calculated at startup is no so useful. try:  core show translation recalc 1000
09:12.08tzafrir_laptopto get a better granularity
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09:18.08mbrevdaanyone here with nvfax detect installed?
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09:26.37maxagaztzafrir_laptop, i don't understand this table, if my sound is saved into speex, why should it be translated into gsm ?
09:28.28troffaskymaxagaz, read the description at the top of the table
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09:39.13L2Logicanyone know if libpri will compile 64 bit?
09:42.12troffaskyI assume so as it's installed on my 64 bit system from debian packages
09:43.27L2Logicty troffasky
09:45.01tm1985http://pastebin.com/d7bf15899 => This is my analog channel. Does anyone see what is wrong. Because I can't dail out????
09:47.16ChainsawL2Logic: Yes, both 1.2 and 1.4 will compile fine on X86_64 hardware.
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10:16.12tzafrir_laptoptm1985, have you tried dialing the same number with an analog phone?
10:19.57tm1985yes and that worked
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10:25.05trentsterhey all, whenever i put someone on hold and music on hold plays I get this error continiously logged in the CLI "RTCP SR transmission error, rtcp halted" this does not seem to effect the call or the music...any ideas what is causing this?
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10:58.51eliasphi
11:02.00eliaspwhen a call is routed forward by the followme rules, it seems the followme message is played back at the same time when the call is forwarded, so the receiver hears just the last words of the followme message... is there a way to define the execution order (parallel/serial)...
11:02.08eliasp?
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11:16.01afinkIs there any way to do a manual sip peer poke?
11:23.04trentsterhey all, whenever i put someone on hold and music on hold plays I get this error continiously logged in the CLI "RTCP SR transmission error, rtcp halted" this does not seem to effect the call or the music...any ideas what is causing this?
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11:32.54TommyBottenafink: You can use sipsak or something similar to construct your own SIP packets
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11:49.19aiksa[LV]Hello everyone
11:51.11aiksa[LV]I sometimes see a strange behaviour on  1.4.22 regarding the callerid(num). It seems that it gets overwritten by asterisk to the pattern of the extension the call is on. Anyone else have seen this?
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11:58.48TommyBottenaiksa[LV]: Could you please elaborate?. I didnt quite understand.
12:02.28aiksa[LV]TommyBotten: I have a dialplan where a call would cycle through a number of extensions
12:02.45aiksa[LV]the first priority would ring SIP/01, the second SIP/03 ... etc.
12:03.50aiksa[LV]now - for the sake of the ease of use they are not maped directly to a specific external extension, but have an "extension" pattern of "simpleQ"
12:04.45aiksa[LV]calls to specific external numbers are then sent to this through a Goto(simpleQ,1)
12:04.53aiksa[LV]is that clear so far?
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12:07.07aiksa[LV]under a circumstances I have not been able to track down so far sometimes a call arriving at one of the destinations SIP/01 or SIP/03, will not have the callerid received from PSTN, but "simpleQ" both as a name displayed by voip phone (zoiper and snom) and as a callerid key in AMI Newstate event
12:08.14aiksa[LV]what is the most strange thing - that untill now I have not been able to track what makes this difference
12:09.33aiksa[LV]I have seen a first call with a coorect callerid pass by, and then the next one with the 'simpleQ' instead. Both dialed to the same extension through the same PSTN PRI line.
12:09.48aiksa[LV]I have not seen this behaviour before, that is why I was asking.
12:10.19aiksa[LV]could that be related to that 'o' option for the Dial command, mentioned somwhere in the wikis
12:10.22aiksa[LV]?
12:13.28aiksa[LV]TommyBotten: did this make it clearer
12:13.30aiksa[LV]?
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12:37.56Grofhey guys
12:37.58Grofneed help
12:38.06Grof"Unrecognized pridialplan NPI modifier: g"
12:38.07Grof?
12:38.20Grofi just upgraded from asterisk 1.2 to 1.6
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12:40.02manxpower~answers
12:40.03infobotanswers is probably Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki:  http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt
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12:45.10TommyBottenaiksa[LV]: Probably... as o: Restore the Asterisk v1.0 CallerId behaviour (send the original caller's ID) in Asterisk v1.2 (default: send this extension's number)
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12:56.00HatrixHi guys, i have a tandberg video solution and i try to record the video in asterisk. tcpdump tells me all is fine, audio and video gets send from the tandberg to my asterisk, but the record command only records the audio? i read at voip-info and googled but all i could find is that record should save the audio and video file in a separate raw file (which i would need)  but ... it doesn't, any ideas?
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13:05.17Kattyyawns
13:05.35Naikrovekyawns
13:05.40Naikrovekit's contageous
13:05.42Kattypamples Naikrovek
13:05.56Naikroveklooks up "pamples."
13:06.53coppiceI think she's trying to put a diaper on you
13:08.46Kattyhttp://www.youtube.com/watch?v=lZ2nhg6az64 <- Pampling.
13:10.12Kattyoh damn
13:10.20Kattymy dad just sent me a friendship request
13:10.21Kattyon facebook
13:11.31creativxKatty: bummer!
13:11.42creativxi ignored my moms friend request till she figured out how to resend it
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13:13.19Kattyi'm not ignoring my dad.
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13:39.32michelhi all
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13:40.27MWEhi all
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13:41.33Naikrovekhello
13:42.20MWEI'm bussy with a project were a caller calls a script (duhhh) but the caller had to enter a meetme room and before that there must be an outbound with PIN check. Is het possible to make an outbound which also runs an special script while the caller is in the meetme room?
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13:46.19[TK]D-FenderMWE: You can Originate a call before sending him into thet room.  This callout will be completely independant however
13:47.11MWEI tried a serveal things but I'm stuck @ it right now.. maybe somebody could help me out? http://pastebin.com/m4c17ead4
13:49.28MWEjust before the caller enters the meetme room the script had to make an outbound and the callee had to enter the room
13:49.47MWEI checked out http://www.voip-info.org/wiki/view/Asterisk+local+channels
13:50.40Kattyfacebook suggests i know allison smith
13:51.40fuxu2facebook always suggests people to me that I don't know
13:52.40fuxu2katty: www.facebook.com/iconicflux ... send me a friend request
13:52.51Kattybut i don't know you
13:52.57fuxu2you do now
13:53.03Kattydo i?
13:53.10Kattyi don't think i do.
13:53.29[TK]D-FenderMWE: that does not spawn a SEPARATE call.
13:53.48[TK]D-FenderMWE: that has nothing to do witha  separate call processing in the background while letting the caller fall through to the MeetMe
13:53.49fuxu2sure.. we're on #asterisk.. obviously that means we have at least that in common.. and facebook isn't linkedin. :)
13:54.07MWEbut how can I fix that [TK]D-Fender ?
13:54.08[TK]D-FenderMWE: ORIGINAT <----
13:54.14[TK]D-FenderMWE: ORIGINATE <----
13:54.16Kattyyeah i only have friends and family on facebook.
13:54.22[TK]D-FenderMWE: Go look it up on the WIKI
13:54.24Kattywith the exception of fender.
13:54.30[TK]D-Fender:|
13:54.30fuxu2if anyone here plays mafia wars on facebook.. you really should add me because I've got a pretty strong crew
13:54.34Kattyi have to keep my enemies close.
13:54.40Kattyi mean.
13:54.42Kattyuhh
13:54.45[TK]D-FenderKatty: I only have people I don't see IRL there :)
13:54.46Kattyhugs [TK]D-Fender
13:54.48Katty;)
13:55.11[TK]D-Fender's facebook page is EMPTY and will remain as such.
13:55.12Katty[TK]D-Fender: my dad sent me a facebook request this morning.
13:55.16[TK]D-FenderFacebook = EVIL
13:55.22[TK]D-FenderKatty: IGNORE <-
13:55.28Kattyno.
13:55.30[TK]D-FenderKatty: ESPEciALLY family
13:55.31Kattyi shan't
13:55.38Naikrovekfacebook is not a font of morality, but i wouldn't say it's evil
13:56.10[TK]D-FenderKatty: BS... I already know how Bible-thumping-crazy your lot are, and if you value your privacy you'll deny him...
13:56.37Naikroveksomeone I didn't know added me as a friend over the weekend, and i removed her after I saw her profile and didn't know who she was.  I wonder if she lurks in here...
13:56.44[TK]D-FenderNaikrovek: You're right, it isn't inherently evil, that is more like the inevitable conclusion due to mankinds own failings.
13:57.14fuxu2facebook for me isn't really about people I actually know. If I wanted to keep in touch with them I'd use IM or email.
13:57.36fuxu2so basically.. I use facebook for groups, games, etc.. and try not to put much about me or my family on it
13:57.56[TK]D-FenderFacebook is a way for someone you used to know who can't find you any other way to do so, but never put anything up there.
13:58.24[TK]D-Fenderfuxu2: Far better game resources out there including *gasp* installable softwaqre
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13:58.52Katty[TK]D-Fender: mom's crazy.
13:59.00Katty[TK]D-Fender: dad's moderately crazy, but respectful
13:59.14[TK]D-FenderKatty: And Mom will go THROUGH your Dad if she's that crazy
13:59.31Kattypats [TK]D-Fender
13:59.33fuxu2fender: true.. but I there's not an easy way to brag about how you're doing via installable software.. :)
13:59.35Kattyi appreciate your concern, dear.
13:59.36[TK]D-FenderKatty: Katty You've been warned :|
14:00.09[TK]D-Fenderfuxu2: Twitter had it right at the 4th character...
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14:02.00fuxu2wow.. people are still using twitter? :-P
14:02.02Kattyconsiders breakfast.
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14:02.19creativxim gonna start twatter
14:02.49[TK]D-Fendercreativx: Wouldn't surprise me if a porn site already reserved that one...
14:03.01creativxhehe, no doubt
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14:03.38fuxu2i really want to make a game on facebook that's all about warring religions..
14:04.02fuxu2like catholics vs protestants vs muslims vs jews.. etc.. give them all funny skills and stuff..
14:04.39fuxu2scientologists also.. they'd totally bring out their e-meters and f. people up mentally..
14:06.00creativxthat sounds meaningful and profitable..
14:06.02[TK]D-Fenderfuxu2: .... that would actually likely sell....
14:06.28[TK]D-FenderfexEngendering the very worst traits of humanity...
14:06.33[TK]D-Fenderfuxu2: Engendering the very worst traits of humanity...
14:06.49troffaskythat's facebook in a nutshell is it not?
14:07.08[TK]D-Fendertroffasky: No, once again, only an inevitable conclusion.
14:07.22fuxu2fender: yeah .. on facebook you could totally have each side's effectiveness be swayed by the number of converts to their side..
14:07.43fuxu2and let people buy more invites..
14:07.52*** join/#asterisk Faithful (n=Faithful@124.217.119.152)
14:07.57coppiceI'm puzzled why they call this stuff social networking
14:08.08creativxlack of better word
14:08.09Naikrovekthat game would end fast.  scientologists would put out a call to arms, and EVERY scientologist would join the game and win.
14:08.23[TK]D-Fendercoppice: Because its as close as many of them get to actually interacting with people...
14:08.31coppicean obvious better term would be antisocial networking
14:08.32fuxu2naikrovek: who cares as long as I'm rich off their dimes..
14:08.33fuxu2:)
14:08.34troffaskyNaikrovek, nah, that's just what they want you to think would happen
14:08.55Naikrovekif there's one thing religious folks like above all else, it's defending their own religion, and eradicating all other trains of thought.
14:09.12fuxu2naikrovek: that's my thinking about the whole game and why I want it
14:09.19fuxu2I even started working on it at one point..
14:09.43fuxu2need some graphics artists and stuff like that though..
14:10.03troffaskydon't forget vi and emacs as religions
14:10.14fuxu2ahahahaha! troffasky! ahahaha that's awesome
14:10.19Naikrovekif some of these religions had their way we'd attend church every morning and night, and you could only eat at church, could only have children when the church said so, all taxes woudl go to the church, if you were lucky enough to be employed at a church, etc.
14:10.36*** join/#asterisk wcselby (n=wcselby@216.110.88.254)
14:11.09Katty:>
14:11.12Kattyhi wcselby!
14:12.09fuxu2Anyways.. if anyone here wants to actually work on it with me, let me know.
14:12.38p3nguinWhat are we working on?
14:12.50fuxu2p3nguin: Holy Wars the game.
14:13.09p3nguindevelopment?
14:13.48fuxu2yeah.. a lot of the development I can do but I don't have any graphics skills and it always helps to have other people that want to help on the game design and balance..
14:14.24[TK]D-Fenderdraws the bestest stick-people EVAR
14:15.01*** join/#asterisk mumtazah (n=mumtazah@152.78.48.60.wmu01-home.tm.net.my)
14:15.15wcselbyo/ Katty
14:18.37obruTHello everyone, I got question regarding getting value of the variable after hangup, is it possible ? in the dialplan, after creating a channel, I'm calling one application that creates some variables, after hangup, I can display them with Noop() application, but at the end I would like to execute one AGI application that needs to fetch those variables, is it possible ? I would like to avoid passing the values via agi script cmd line options....
14:19.19MWE[TK]D-Fender, do you have a example of that originate.. I read the wiki but is it something like the hangup() just a function call?
14:19.43p3nguinobrut: I don't see why that can't be done.  What's the problem?
14:20.43p3nguinmwe: You just need to see how originate is used?
14:20.47*** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson)
14:20.47*** mode/#asterisk [+o putnopvut] by ChanServ
14:20.51MWEyeah p3nguin
14:20.59manxpowerMWE: You didn't find anything useful in the OFFICIAL docs (located in /path/to/src/asterisk/doc)?
14:21.03p3nguinexample  originate SIP/101 extension 101@wakeup
14:21.05obruTp3nguin: getting the value from the agi script after hangup returns error, like the variable is not set
14:21.25manxpowerobruT: which specific variable?
14:22.14obruTmanxpower: for example any variable that sendfax application sets
14:22.14[TK]D-FenderMWE: AMI ORIGINATE.  its in the book, its in the WIKI.  Lookup "call files" while you're at it
14:22.14MWEmanxpower, I've got a book "The future of telephony' but I didn't know what there was
14:22.44manxpowerMWE: the docs included with Asterisk are always the current ones
14:23.03obruTmanxpower: or any variable that i set in the extensions.conf, even if I set it after hangup
14:23.04*** join/#asterisk wr| (n=wr@p54BE4186.dip.t-dialin.net)
14:23.35[TK]D-FenderobruT: DeadAGI, not AGI
14:23.35manxpoweryou should not set dialplan variables after hangup.
14:23.58obruT[TK]D-Fender: DeadAGI is obsolete in 1.6
14:24.47p3nguinI have a peer that I currently use qualify= for.  His lag often increases above the threshold.  Should I consider not using qualify= at all, or increate the value even higher?
14:24.48obruTmanxpower: well, I can't get value of the variables that are set before hangup
14:25.08manxpowerobruT: pastebin an example
14:26.28manxpowerp3nguin: consider not uding qualify
14:26.38Kattyhugs p3nguin
14:26.40Kattyhugs manxpower
14:26.45p3nguinkatty: Mornin'
14:26.55Kattyhow's everything upstate today
14:27.06p3nguinkatty: Doing well so far.
14:27.30Kattyexcellent.
14:27.34*** join/#asterisk propellerhead (n=yogurt2u@host238.190-136-114.telecom.net.ar)
14:29.36obruTmanxpower: http://free-ka.t-com.hr/ib/stuff/p/asterisk.txt this is section of extensions.conf, first it calls agi script, this agi script successfully fetches variables, after hangup it displays all of the variables, but on the second agi call, all of those variables are unset
14:29.39*** join/#asterisk karleeto (n=karl@server.nashvilleproweb.com)
14:29.53karleetofor the hole in my router for an iax trunk, is it tcp or udp?
14:30.10*** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage)
14:30.10*** mode/#asterisk [+o leifmadsen] by ChanServ
14:30.16wcselbykarleeto - udp
14:31.01p3nguinI'm suddenly reminded of a jingle for a roofing company.
14:31.22*** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm)
14:31.44p3nguin"For a hole in your roof, or a whole new roof - Fredrick Roofing."
14:31.48manxpowerobruT: why do you need to call it in the HANGUP?
14:32.01*** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk)
14:32.47Kattyplots breakfast
14:33.05obruTmanxpower: when the channel hangups, execution of the first agi script stops immediately, sometimes before I update some status, now I'm trying to do it after hangup
14:33.07p3nguinkatty: Cocoa Puffs?
14:34.04obruTmanxpower: also I have to pass some variables to the status, I could do it via cmd line, but I would like one potential security hole :)
14:34.13Kattyi have chex, eggs in the fridge...
14:34.16Kattyleftover stroganoff
14:34.21Kattypossibly other stuff.
14:36.21manxpowerobruT: what status are you trying to update?  It is possible to make your script exit when the channel hangs up
14:36.26p3nguinStroganoff for breakfast?  I'll pass.
14:36.53*** join/#asterisk Superbartt (n=bart@ipd50a21c9.speed.planet.nl)
14:37.16SuperbarttQuick question: Is there a way to run a script/command when a caller leaves the queue and gets connected to the agent?
14:37.21coppiceStrogan Off sounds like a dreadful insult
14:38.13*** join/#asterisk ChkDigit (n=mike@static24-72-71-175.regina.accesscomm.ca)
14:39.03[TK]D-FenderSuperbartt: Run a daemon polling the AMI messages
14:39.47Superbartthmm ok
14:39.48Kattymmm
14:39.53Kattystroganoff for breakfast is awesome.
14:39.55Kattychows.
14:40.28obruTmanxpower: some status in my database, it depends of the value of the variables that SendFax application sets, asterisk just stops execution of the agi script after hangup, I have no controll when and where in execution will it happen... If the other side waits for me that i make hangup, then everything finishes ok, but if the other side hangups before my agi script finishes, it can be anytime, I just cannot make clean exit...
14:42.42*** join/#asterisk fskrotzki (n=fskrotzk@mail.perspectivepartners.com)
14:42.56Superbartt[TK]D-Fender does it also return events like: $from-CID was in the queue and connets to agent X
14:43.00manxpowerobruT: I wish you the best of luck.
14:43.04Kattyi watched a film about corn last night. it was pretty sad.
14:43.12[TK]D-FendersubrGo read the manual on this
14:43.15Kattywell the flim wasn't.
14:43.18[TK]D-FenderSuperbartt: Go read the manual on this
14:43.21Kattythe corn production is.
14:43.56Superbarttsorry [TK]D-Fender, there's an company busting my balls and they want to know if i can make it in 15 minutes ;P
14:44.19[TK]D-FenderSuperbartt: Go invest in Tylenol
14:44.25Kattyif it's that important, i'd suggest hiring a consultant.
14:44.26Superbarttsay what? :P
14:44.49[TK]D-FenderKatty: Sounds liek he is one... and underqualified
14:44.57Katty[TK]D-Fender: hush, dear.
14:45.01SuperbarttKatty I am the "consultant" :p
14:45.02Katty[TK]D-Fender: there's no need for that.
14:45.10Superbarttand true [TK]D-Fender ;)
14:45.14[TK]D-FenderKatty: I stand validated :p
14:45.16Superbarttit's the truth :p but I can do it
14:45.27Kattythat's the spirit!! (=
14:45.35[TK]D-FenderSuperbartt: then get your hands off your nuts and SEIZE THE DAY!
14:45.45Superbarttlol :p
14:45.52Superbarttif you just say yes i believe you :P
14:46.26*** join/#asterisk Faithful (n=Faithful@124.217.119.151)
14:48.23*** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler)
14:50.23Naikrovekpoor [TK]D-Fender - knows more than all of us and is unemployed, while we all have jobs (at least I think he said he was unemployed.)
14:50.56troffaskythat's cos only unemployed people have time to RTFM ;-)
14:51.19*** join/#asterisk cusco (n=trilili@2001:0:53aa:64c:1c5a:126a:b2c9:8871)
14:51.31cuscohi
14:52.09cuscoiff a application in make menuselect says "Depends on xxx (M) or (E)"
14:52.15cuscowhat is (M) and (E)
14:52.28p3nguinMonkeys and Elephants
14:52.33cuscoI would say (E) means external?
14:52.46cusco...
14:53.17p3nguinOkay, maybe not.
14:53.19*** join/#asterisk |Cybex| (n=John@80.100.126.176)
14:53.33cuscoand where canI find app res_jabber
14:55.03Kattyhmm. maybe stroganoff for breakfast wasn't such a great idea afterall. feeling kinda naseous now :/
14:55.13p3nguinI tried to warn you.
14:55.22wcselbyKatty - lol
14:55.26Kattyyes, yes you did
14:57.04coppiceKellogg's Stroganoff probably isn't a good idea, then
14:57.12*** part/#asterisk gego (n=rick@b238085.customer.hansenet.de)
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15:00.12[TK]D-FenderNaikrovek: Nope, rock solid day-job...
15:00.23Naikrovek[TK]D-Fender: oh, cool.
15:00.31Naikrovekwonders where he got that idea then.
15:00.39[TK]D-FenderNaikrovek: And I believe that anyone who depends on * as their livelyhood had better RTFM...
15:00.46Naikrovekyeah
15:00.56Naikrovekyour job an * job?
15:01.00[TK]D-FenderNaikrovek: Probably manxpower's on/off status
15:01.13[TK]D-FenderNaikrovek: General IT including running our * box.
15:01.24Naikrovekcool.  same here.
15:01.26[TK]D-FenderNaikrovek: I was using it long before and consult on the side
15:02.12[TK]D-FenderNaikrovek: I burned a week with #$&^ing Dell PCIe interrupt losses and have lost some face on out last upgrade attempt
15:02.28[TK]D-FenderNaikrovek: and lost 2 weekends.  I'm completely fried, in a Kentuky kinda way
15:02.37Naikrovektake a day off
15:02.48[TK]D-FenderNaikrovek: I'm trying to take my normal WEEKENDS off
15:03.31wcselby[TK]D-Fender - ouch
15:03.31cuscochange hardware
15:03.52[TK]D-FenderI don't even have the energy to kill people that really deserve it.
15:04.21wcselby[TK]D-Fender - do you have a PFY running around with you somewhere?  You sound like you could be a real BoFH
15:04.29wcselbyand I mean that in the best possible way
15:04.33wcselby:-)
15:04.39[TK]D-Fenderwcselby: PFY?
15:05.46wcselby[TK]D-Fender - have you read any of Simon Travaglia's BoFH series?  http://www.theregister.co.uk/odds/bofh/
15:05.53[TK]D-Fenderwcselby: And I am anti-BOFH.  I do not subscribe to the closet-troll power-tripping mentality.  Technology is there to serve people, not the other way around.
15:06.00[TK]D-Fenderwcselby: Nope
15:06.38*** join/#asterisk maour (n=gnu@unaffiliated/maour)
15:06.47wcselbyi enjoy it as good comedy, however I don't act that way towards clients / users
15:06.56Naikrovektoo many do
15:07.04Naikrovekfortunately they don't last long
15:07.06[TK]D-Fenderwcselby: And I make solutions that work, cheaper faster and "freer"
15:07.23Naikrovekthen they complain on facebook "I did the work of 10 men" and BS like that
15:07.52wcselbyNaikrovek - lol yeah too true.  I've seen that mentality a lot...unfortunately
15:08.33wcselby[TK]D-Fender - so is your upgrade complete or are you still dealing with hardware issues?
15:08.47Naikroveki ALWAYS reply to those guys when I can: "yes, all smart businessmen fire the one man who can do the work of ten men.  right?  perhaps it's more likely that you are an ass."
15:08.49*** join/#asterisk Faithful (n=Faithful@124.217.119.131)
15:09.13wcselbyNaikrovek - lol
15:09.20[TK]D-Fenderwcselby: Plagued.  It was a hardware/software full upgrade and I still had the old server.  Back to square one operationally speaking..
15:09.30Naikrovekeek
15:09.46[TK]D-Fenderwcselby: And I'm going to bludgeon the McFuck out of the hardware vendor until they can make their products cooperate
15:09.46wcselby[TK]D-Fender - UGH, I hate that.  At least you were able to roll back to an operational status.
15:09.50p3nguinIs there a variable that essentially says "all available SIP channels"?  A usage of it might be to ring all SIP phones by the variable as opposed to using Dial(SIP/1001&SIP/1002&SIP/1003&SIP/you-get-the-idea)
15:09.59Naikroveklol McFuck awesome
15:10.07[TK]D-Fenderp3nguin: No.
15:10.08wcselby[TK]D-Fender - lol
15:10.29wcselbyp3nguin - I guess you could define one in the [globals] context of your extensions.conf file though
15:10.31Naikrovekp3nguin: assign people to groups, then assign all groups into an ALL alias
15:10.47wcselbyp3nguin - would be a pain in the ass to maintain though
15:10.48p3nguinwcselby: That was precisely my reason for asking.
15:10.53Naikrovekyeah it would be a pain
15:11.08p3nguinI figured if something already existed, there was no reason to make a new var for it.
15:11.19[TK]D-FenderMaintaining that constant would also be a pain, but if its usageoccurs in multiple places, at least LESS of a pain.
15:11.42[TK]D-Fenderp3nguin: Feel free to write an external script to do this...
15:12.02*** part/#asterisk manxpower (n=EWieling@133.sub-70-214-74.myvzw.com)
15:12.11[TK]D-Fenderp3nguin: Shouldn't be too hard
15:12.12Naikroveki do it like i used to do mailing lists on sendmail.  1st floor was tech, producers, tech mgmt, producer mgmt.  2nd floor was execs, sales, marketing.  all was 1st & 2nd
15:15.46*** join/#asterisk ArchGT (n=ArchGT@unaffiliated/archgt)
15:16.17*** join/#asterisk jcape (n=jcape@209.120.251.81)
15:16.28p3nguinnaikrovek: When you said assign people to groups, did you have in mind to use callgroup for each peer in sip.conf?
15:17.25p3nguin"group" turns out to be a rather ambiguous term.
15:18.43Naikrovekwell i just assign people to logical groups.  so mgmt has like 4 people in it, they each have their own name -> extension mapping as well
15:19.29p3nguinI assume you didn't mean callgroup, now that I read what it does/doesn't do.  "Callgroups are not intended to call a group of phones"
15:19.57Naikrovekah, no
15:20.19p3nguinIn which file do you contstruct said "group"?
15:20.58*** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com)
15:21.09p3nguinSeems like it would be easiest by using a variable in extension.conf's globals.
15:21.20Naikrovekp3nguin: yes i use globals
15:21.37Naikrovekbut i have a freepbx system so i had to hide them away somewhere
15:21.48[TK]D-Fenderp3nguin: there is no grouping or function for what you are looking to do.
15:22.32[TK]D-Fenderp3nguin: Any "select all' that you want for dials, page, etc that you want to do will require scripting on your part.
15:29.42*** join/#asterisk moy (n=moy@74.12.131.104)
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15:30.24*** part/#asterisk Hatrix (n=Hatrix@213.201.24.127.static.user.ono.com)
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15:32.18KattyARGH
15:32.22Kattymy nose itches
15:33.29*** join/#asterisk spck (n=spck@unioncab.com)
15:35.32wcselbycd /usr/src/asterisk-1.4.26/doc
15:35.35wcselbybleh
15:35.39wcselbywrong window
15:35.43Kattyi was gonna say
15:38.33*** join/#asterisk Carlos_Tico (n=grillo_v@c-98-201-162-34.hsd1.tx.comcast.net)
15:38.38Carlos_Ticohello...
15:39.13Carlos_Ticoare you there [TK]D-Fender
15:39.28wcselby~ask
15:39.29infobotQuestions in the channel should be specific, informative, complete, concise, and on-topic.  Don't ask if you can ask a question first.  Don't ask if a person is there; just ask what you intended to ask them.  Better questions more frequently yield better answers.  We are all here voluntarily or against our will.
15:39.58*** join/#asterisk Whitor (n=Whitor@rrcs-24-97-4-146.nys.biz.rr.com)
15:40.00Carlos_Ticogot a problem with DISA
15:40.17jayteecd /home/dickcheney/chickenporn
15:40.22jayteeooops!
15:40.26jayteecd ..
15:40.39[TK]D-FendercarGo ahead and show it
15:40.45[TK]D-FenderCarlos_Tico: Go ahead and show it
15:41.26[TK]D-FenderCarlos_Tico: And you should ask the channel in general, not target individuals.
15:41.37p3nguinDoes asterisk use .ttml files?
15:41.53Carlos_Ticohttp://pastebin.com/mde4872f
15:41.59Carlos_Ticook
15:42.09Carlos_Ticoi get the dial tone after dialing Disa
15:42.16Carlos_Ticobut then i cannot dial nothing
15:43.14Carlos_Tico[Sep 21 10:41:07] WARNING[10508]: app_disa.c:246 disa_exec: DISA password file not found on chan Zap/1-1
15:45.36Carlos_Ticoany ideas ?
15:46.59[TK]D-Fenderp3nguin: No.
15:47.28[TK]D-FenderCarlos_Tico: Error means what it says.  It didn't find a match for whatever PIN was entered
15:47.37p3nguinI didn't think so, but someone asked and I didn't know for certain.  I wasn't familiar with seeing those file extensions on asterisk.
15:47.49[TK]D-FenderCarlos_Tico: "DISA password file not found" you specify a password file, and it can't find it
15:48.04Carlos_Ticobut i got the same error even with Disa without a password
15:48.32[TK]D-FenderCarlos_Tico: Show a real pastebin of the complete failure and your configs
15:49.56Carlos_TicoExecuting [s@voicemenu-custom-1:2] DISA("Zap/1-1", "|DLPN_DialPlan1") in new stack
15:49.56Carlos_Ticothen
15:50.03Carlos_Ticoi dial and nothing... just bussy tone
15:50.14Carlos_Ticono other msg
15:50.53*** join/#asterisk raden_work (n=jon@69-179-99-17.stat.centurytel.net)
15:51.00[TK]D-FenderCarlos_Tico: PAStebIN
15:51.13[TK]D-FenderCarlos_Tico: I said the complete failure.
15:51.13raden_workis there a way when i forward calls to forward caller ID ?
15:51.16[TK]D-Fender~pb
15:51.17infobot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding 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 apt-get install pastebinit
15:51.38[TK]D-Fenderraden_work: "forward" how?
15:52.33raden_workwe forward our extension when we out of office I would like the actual calling number to be deisplayed instead of our company number
15:52.45[TK]D-Fenderraden_work: "forward" how? <-
15:53.01raden_workastdb
15:53.12[TK]D-Fenderraden_work: AstDB has nothing to do with forwarding...
15:53.23raden_workwell wtf do you mean forward how ?>
15:53.54[TK]D-Fenderraden_work: What commands/functions are actually used in this "forward".
15:54.07[TK]D-Fenderraden_work: Clarify your use of this term
15:54.18Carlos_Ticohttp://pastebin.com/d45c339df
15:54.26[TK]D-Fenderraden_work: It is currently rather vague including where the call is going in the call process
15:55.18[TK]D-FenderCarlos_Tico: how many digits do you get to enter before it hangs up?
15:55.20p3nguinraden_work: Do you mean transfer them?  Do you mean when they use Followme?  Clarify.
15:55.22raden_workwhen we dial *72, # we forward to our cell phones normally, I want the caller id information to come through to the cell phones
15:55.47[TK]D-Fenderraden_work: what is *72?  this isn't some magic number for *...
15:56.29raden_workall im asking is can it be done ?
15:56.29Carlos_Ticoi can enter what ever number of digits... then a little silence and bussy tone
15:57.18Carlos_Ticonot other msg in CLI
15:57.37raden_workI set everything is astdb
15:59.20*** join/#asterisk LtScarr (i=benno@palm.hoeg.nl)
15:59.26LtScarrhey guys
15:59.38LtScarri have a security related question
16:00.21LtScarris it posible to dail extensions through sip without registering?
16:00.42LtScarrand is that default behavior?
16:01.27*** join/#asterisk wildzero-cw (n=chatzill@p50997436.dip0.t-ipconnect.de)
16:01.58*** join/#asterisk raden_work (n=tanning@69.179.99.17)
16:02.32raden_work[TK]D-Fender, http://pastebin.com/m21512a2e
16:03.29*** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose)
16:03.30raden_workthats how i have it working at the moment id like to take the inbound caller ID and send it to the forwarded number
16:04.12*** join/#asterisk Defraz (n=Defraz@corp.fuzecore.com)
16:04.22raden_workmaybe even with a number in front of it soo i know its forwarded
16:05.14*** join/#asterisk ltd (n=z@pat.transact.net.au)
16:09.13Carlos_Ticohow can i fix this error ?? - ast_get_srv: SRV lookup for '_sip._udp.proxy01.sipphone.com' mapped to host proxy01.sipphone.com, port 5060
16:10.19QwellCarlos_Tico: Who says it's an error?
16:10.53Carlos_Ticoyesterday TK D-fender told me
16:10.59Carlos_Ticothat it was a NAT issue
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16:12.36wildzero-cwhello, 1.4.27-rc1 is available and im testing it (with no problems), how much time it it use from rc1 to final release on average?
16:13.39Qwellwildzero-cw: when it's ready, basically
16:13.41Naikrovekwildzero-cw: few weeks i'm guessing.  why do you need THAT version
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16:16.15wildzero-cwNaikrovek: because i just build a 1.4.27-rc1 deb (debian) for my system, and want to wait to distribute it until it's final. iam using 1.4.20 and it's much to old
16:16.28Qwellwildzero-cw: well, you could use 1.4.26.2 for now..
16:16.37p3nguinUse something newer, but not all the way into the rc.
16:17.47wildzero-cwyes, but then is need to test 1.4.26.2 and 1.4.27 and distribute them twice, so ei just wait until it's final
16:18.22p3nguinWhat's the big deal?
16:18.46p3nguinUse 1.4.26.2 now.  Distribute it.  Use it.  Like it.
16:18.53p3nguinWhat's the problem?
16:19.37p3nguinLike you were already asked, why do you need THAT version?  Why not use 1.4.26.2 for the next few years?
16:19.48p3nguinIt's certainly stable enough.
16:21.06*** join/#asterisk geneticx (n=chatzill@host-208-88-126-198.biznesshosting.net)
16:21.55geneticxHello everyone, has anyone implemented or has any experience with skype for asterisk?
16:23.02Carlos_Ticoanyone here managed to setup magic jack as a trunk on asterisk ?
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16:23.33Superbarttgeneticx it kinda sux imho :P
16:23.59p3nguinWhere is the app_dial timeout set for followme?
16:24.19geneticxSuperbartt: interesting, how come? any bad experiences you might want to share?
16:24.23Naikrovekmagic jack?  don't cheap out
16:24.44Superbarttgeneticx pain in the ass to setup, still an actual skype running
16:25.39geneticxSuperbatt: ah I knew it.. I hate when
16:26.34geneticxSuperbatt: when they want to throw untested stuff like this into production before even trying it
16:27.02geneticxSuperbatt: you probably saved me a lot of headaches then..
16:27.47geneticxthey were ready to have about 8 people on skype for asterisk..you should've seen my face..
16:32.32[TK]D-Fenderraden_work: exten => 103,n(forward),Dial(LOCAL/${DB(CFIM/${EXTEN})}@to-callcentric,18) <-- this is not forwarding
16:32.53[TK]D-Fenderraden_work: forwarding is an action taken by the device like on a SIP phone.
16:33.03[TK]D-Fenderraden_work: You are simply choosing to dial out callcentric
16:33.14[TK]D-Fenderraden_work: So if they let you rig your Caller ID, then go do it
16:34.41QwellSuperbartt: umm, no.
16:34.59[TK]D-FenderLtScarr: Without who registereing?  Who is placing the call?
16:35.17QwellSuperbartt: You are talking about a completely different product.  SFA does not require any hacks that the other ones do.
16:35.21[TK]D-FenderLtAnd no, registration is normally not needed to place calls from a device as long as it know where to call.
16:35.59Qwellgeneticx: get yourself 1 license and try it out.  if it works, get the rest.
16:36.25Qwellit's rather trivial to setup/use.  and it's...actually supported.
16:36.37SuperbarttQwell really? Last time I checked for that (a while ago) it required an running skype etc... Was mostly a dirty hack
16:36.45Qwell~skypeforasterisk
16:36.45infoboti heard skypeforasterisk is a Digium-made channel driver that allows Asterisk to connect to the Skype network using a Skype-supported connection method. Unlike all other solutions it does not require X11, kernel modules, or a Windows machine. See http://www.digium.com/skype for details
16:36.58Naikrovekskypeforasterisk is relatively new i believe, yes?
16:37.03Qwellyes
16:37.08Naikrovekcool
16:37.11Qwellreleased at the beginning of the month
16:37.28wcselbywow
16:37.48wcselbyjust made a custom background for my 7961, and when it's displayed on the phone it looks compeltely different
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16:37.58Naikrovekwcselby: explain completely different
16:38.03Qwellwcselby: welcome to Cisco
16:38.07wcselbys/compeltely/completely/
16:38.09Naikrovekchange colors?
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16:38.13Superbarttnice Qwell :) Excuse me for my wrong opinion
16:38.19Naikrovek7961 color screen?
16:38.25wcselbyyeah, what I gave as a clear background I got a black background
16:38.28[TK]D-FenderNaikrovek: No
16:38.42wcselbyNaikrovek - no, greyscale
16:38.43Naikrovekah transparency is always iffy when changing devices
16:38.44Qwellwcselby: clear requires transparency
16:38.46[TK]D-FenderNaikrovek: higher res, 802.3af, etc
16:38.54Naikrovek[TK]D-Fender: ah okay thanks
16:39.17Qwellthere's no such thing as transparency in grayscale, heh
16:39.35wcselbywell, yeah, transparent background I guess.
16:39.45Naikrovekwell you can assign a color to be the same color as the native BG color, probably white
16:39.49wcselbyheh, okay.  need to relook at the logo
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16:45.12geneticxQwell: Ok, ill take that into consideration. Thanks..
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16:53.42raden_work[TK]D-Fender, is there a better way todo what im doing and basically your saying all i have todo is set my own CID and send  id have to switch to vitelity when forwarding but itd work i guess
16:54.19[TK]D-Fenderraden_work: Go set your callerID.  And this is not "forwarding" and this has absolutely no relationship to that.
16:54.25Qwellraden_work: punctuation
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17:08.45wcselbyhmmmmm
17:08.48wcselbywhat to eat for lunch
17:08.53wcselbydecisions decisions
17:09.57Pan3Dfood is always a good choice
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17:33.26Kattywcselby: something low on the glycemic index.
17:34.12Qwellwheat germ
17:35.43Kattyi was actually going to suggest cheese.
17:36.25Katty1 cup of milk = the same glycemic load as a piece of toast.
17:37.17Kattyaka load of 9. 1 cup of cheese is a load of 3
17:37.42Kattya cup of watermelon is 3, cup of grapes is 5
17:38.00Kattysomehow, i've always found watermelon sweeter than grapes tho.
17:38.07Kattyand watermelon definately sweeter than milk.
17:38.14Kattybut milk has a high glycemic level.
17:38.19p3nguinIf ${EXTEN:1} will = strip the first character off the extension, what do we use to strip the last character off of it?
17:43.41henkoegema${EXTEN:-1}
17:44.04Qwellhenkoegema: that will get only the last
17:44.31Qwell${EXTEN:0:-1}
17:46.40henkoegema:)
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18:14.16wcselbyQwell - is there logic to say if I only want the last 10 digits of a number using the ${EXTEN:x:y} logic?
18:14.31[TK]D-Fenderwcselby: "core show function LEN"
18:15.26wcselbyLEN
18:15.29wcselbyokay thanks [TK]D-Fender
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18:22.55raden_workwhat would this be considered then if its not forwarding redirection ?
18:23.13raden_workwhy do all the telcos call the same procedure forwarding ?
18:23.17raden_workim a lil confussed
18:23.31lirakis_raden_work, what are you talking about
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18:24.44Pan3Dheh
18:26.16[TK]D-Fenderraden_work: You are doing a dumb dial yourself in the dialplan.  the fact you bothered to SHOOSE between to possible action is irrelevent.  the CID from the channel is the same as when it started
18:26.44[TK]D-Fenderraden_work: When you call a SIP phone and IT decides to bounce you, THAT is a forward (SIP 302 Redirect)
18:26.58[TK]D-Fenderraden_work: THAT can have implications on who the call is coming from
18:27.11[TK]D-Fenderraden_work: Here you are just shoosing between door #1, and door #2 in the dialpla
18:27.17raden_workis there a way to set a redirect in asterisk ?
18:27.32[TK]D-Fenderraden_work: which is a conceptual "forwarding", but the term must be avoided in the programming sense
18:27.41[TK]D-Fenderraden_work: this ISN"T a REDIRECT
18:27.44*** part/#asterisk fuxu2 (i=iconicfl@www.kevinlynn.com)
18:27.46[TK]D-Fenderraden_work: just stop calling it that
18:28.03[TK]D-Fenderraden_work: Its a dumb dial just like you use in your typical PSTN numbered dialplan patterns
18:28.07*** join/#asterisk noRTFM (i=iconicfl@www.kevinlynn.com)
18:28.13[TK]D-Fenderraden_work: So go set the CALLERID() already...
18:28.24lirakis_[TK]D-Fender, i think he got it... he just wants to know if he can do a redirect instead of a door 1 or door 2 option
18:28.40[TK]D-FenderlirakisNo, I'm sure it'll take a few dozen more swings still...
18:28.45lirakis_heh heh
18:28.53raden_worklirakis, exactly
18:28.55Pan3Draden_work: so, you want to call into an * box and have that call redirected to another number without any user intervention?
18:28.56[TK]D-Fenderputs some rusty nails onto his ClueBat (tm)
18:29.03Pan3Dlol
18:29.04Kattytakes cluebat away from [TK]D-Fender. Again.
18:29.11[TK]D-FenderPan3D: Don't... for the love of god, don't...
18:29.17lirakis_lol
18:29.25Pan3Dthis is why we can't have nice things
18:29.39LtScarr[TK]D-Fender: to answer your question earlier:
18:29.39LtScarr18:34 < [TK]D-Fender> LtScarr: Without who registereing?  Who is placing the call?
18:30.03[TK]D-FenderPan3D: Yes, you'd break them.  Now hand them back gently
18:30.24Pan3Dlol
18:30.52LtScarri got sip id's like SIP/59823 using my dialplan without registering
18:31.03LtScarrand 59823 isn't a valid account
18:31.23fuxu2dang.. ya know.. I totally was tryin to make up a funny question to ask..
18:31.32fuxu2and I just couldn't think of anything..
18:31.43fuxu2someone took away my creativity.. :(
18:31.56raden_workPan3D, sometimes yes
18:31.57LtScarri solved it by setting the default context to nothing
18:32.00Pan3Dlol
18:32.13[TK]D-FenderLtScarr: O RLY?
18:32.17[TK]D-FenderLtDo share...
18:32.21Pan3Draden_work: OK, I think [TK]D-Fender has a point. You should step back a bit and read teh book.
18:33.13Pan3Dthere's a chapter on telecom in general which is useful for lingo, and some specific examples of how dial plans work (including what it is you want to do... sometimes)
18:33.13[TK]D-Fenderraden_work: Set(CALLERID(num)=1800GETCLUE)
18:33.16lirakis_shameless plug for redirect based routing http://routengn.com/
18:33.40LtScarri'm running 1.4.21.2
18:33.44Pan3D[TK]D-Fender: lol, be nice
18:33.56[TK]D-FenderPan3D: i AM, i EVEN GAVE HIM THE CORRECT SYNTAX
18:34.03[TK]D-FenderCaps fail..
18:34.35[TK]D-FenderLtScarr: Well if you allwed unauth'd guests access to a dangerous context like that... YGWYPF :)
18:35.05garymcYGWYPF.... wowo what the hell is that? :)
18:35.08LtScarr[TK]D-Fender: i don't want any unauth'd guests :-)
18:35.13[TK]D-Fender~ygwypf
18:35.14infobotygwypf is probably You Get What You Pay For.  If the sole factor in your decision to purchase a product or service is that it's cheaper than everything else out there, don't be surprised if it's also worse in every other respect than everything else out there.
18:35.25garymcAhhhh
18:35.32garymcgood one
18:35.47[TK]D-Fendergarymc: Polite version of "Suck it biatch :p"
18:35.50Pan3Dthat description should be updated to include bad decisions that are non-financial
18:35.53garymc:P
18:35.55garymclol
18:35.57Pan3Di.e. open contexts
18:36.16LtScarr[TK]D-Fender: well i ran into this by coincidence
18:36.17garymcis glad he bought the good stuff :P
18:36.46LtScarrno real harm done at all
18:38.00LtScarrjust curious about if it's default behavior
18:38.01[TK]D-FenderLtScarr: "allowguest=no" <-
18:38.42LtScarris that sip specific?
18:38.47Pan3Dman, I am so glad I read the book before doing anything with *. It seems 70% of the questions in here are basic stuff covered in the book.
18:39.21[TK]D-FenderPan3D: another 515 once you include the docs in the tarball
18:39.23hescoAt the *CLI>, I see: "Auto fallthrough, channel 'SIP/hugh_desk-08368fb0' status is 'UNKNOWN'", the NoOp()'s that are and are not presented at the *CLI> tell me the dialplan is failing on this priority: "exten => 993,n,AGI(smartdialer.agi,GET_LIST_REPORT,${SID},${AGENT_CHANNEL})", the .agi script passes the perl -wc test; and if I bypass this logic, other priorities have no problem calling this agi script using different functions.  At the moment,
18:39.23hescothis function of the agi script only prints a debug statement, although the module->method called by it and presently commented out does pass basic regression tests.  So what is this error, what does it mean?
18:39.24[TK]D-Fender15%
18:39.35LtScarrthe wiki page about security doesn't cover it...
18:39.48[TK]D-FenderWIKI = CRAP
18:40.15LtScarralright good point :-)
18:40.25[TK]D-Fenderrandom ancient shit written by well-meaning yet still all-too-often wrong people
18:41.03[TK]D-FenderLtScarr: Should have seen yesterday's scuffle over the giant misconception that is the "s" Asterisk Standard Extension.
18:41.46LtScarr:)
18:41.46wcselbyooooh
18:41.48wcselbysounds like fun
18:42.07hesco~paste
18:42.07infoboti guess paste is http://rafb.net/paste/, or see also pb, or http://bin.cakephp.org/
18:42.14[TK]D-Fenderwcselby: Diet Fun (tm).  Just like Real Fun (tm), only half the fun.
18:42.19wcselbylol
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18:42.48wcselbyahhh
18:42.58wcselbyI think what I want is ${EXTEN:-10}
18:43.06p3nguinIs ${CALLERIDNUM} no longer valid?
18:43.12LtScarri wonder how many servers don't have allowguest=no...
18:43.18wcselbythat will give me the last 10 digits of the variable
18:43.24ian6LtScarr: most, I would think. The default allows it.
18:43.28wcselbyp3nguin - not since 1.0
18:43.42wcselbyp3nguin - that is, depricated in 1.2, not used in 1.4
18:43.43p3nguinNo wonder my changes weren't working out.
18:43.51wcselby${CALLERID(num}
18:43.58wcselby${CALLERID(num)}
18:43.58p3nguinThought so.
18:44.03p3nguinJust needed to confirm.
18:44.11Pan3DLtScarr: probably a lot -- particularly when folks probably use that instead of troubleshooting auth problems
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18:49.07wcselbywow
18:49.18wcselbyi just got a sales call on a phoneline that I haven't given the number out to, ever
18:49.26p3nguinsucks
18:49.29wcselbyin fact, I just set the DID up less than two weeks ago
18:49.34wcselbymaybe just over two weeks
18:49.38wcselbyfunny
18:49.48hescoOK, I am retrying my question this time supported by a paste ( http://bin.cakephp.org/view/193968548 ) which I trust will give some context.  Any pointers would be appreciated.
18:51.07wcselbyhesco - show us the CLI output of the failure
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18:51.28[TK]D-Fenderhesco: Now show us a new pastebin where we actually see EXECutION
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18:52.01hescowcselby: coming your way.
18:53.12hescohttp://bin.cakephp.org/view/654130842
18:53.49hescoI get a fast busy and hang up, leading to the 'Call completed' line
18:55.17wcselbywhat's your verbosity level set to?
18:55.22hesco3
18:55.36wcselbypop it to 30 and redo the CLI output please
18:55.43hescocoming your way
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18:58.19[TK]D-Fenderhesco: 993 != 991
18:58.23[TK]D-Fenderhesco: MATH FAIL
18:58.25hescohttp://bin.cakephp.org/view/1134546063
18:58.55bpgoldsbI just upgraded to 1.6.1, and it appears both chan_console and chan_alsa were compiled.  However, when I start asterisk, chan_console loads first, and when it tries to load chan_alsa, I get 'Already have a handler for type 'Console'' + 'Unable to register channel class 'Console''
18:58.59bpgoldsbAny ideas?
18:59.26hescothanks [TK]D-Fender: copy-n-paste error does it to me again
18:59.26wcselby[TK]D-Fender - lol totally missed that
18:59.35Qwellbpgoldsb: don't load one
18:59.39Qwellbpgoldsb: see modules.conf
19:00.28bpgoldsbSure, I know how to not load one.  I figured maybe one was deprecated in a newer version or something.
19:03.36p3nguinIn 1.4, is Set(Global(OldCID)=${CALLERID(num)}) the correct way to set a global variable OldCID to the value of the current CALLERID(num)?
19:04.09[TK]D-Fenderp3nguin: No, GLOBAL is a functions... and like all others is case-sensitive <-
19:04.14[TK]D-Fenderp3nguin: Also, globals = evil
19:04.28[TK]D-Fenderp3nguin: And in most implementations, the wrong solutions
19:04.38p3nguinI'm trying to get a variable to be carried through several contexts.
19:04.51[TK]D-Fenderp3nguin: there is no variable scope within a call
19:05.00[TK]D-Fenderp3nguin: Misconception #2
19:06.24*** join/#asterisk voipmonk (n=voipmonk@69.172.93.45)
19:06.33p3nguinI tried to set the variable using Set(OldCID=${CALLERID(num)}) in one context and then read ${OldCID} in another context.  It was null.
19:06.55[TK]D-Fenderp3nguin: And I don't see your code and the attempt./..
19:07.20[TK]D-Fender~pb
19:07.21infobot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding 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 apt-get install pastebinit
19:07.29p3nguinIf that's the wrong way, why not just tell me the right way?
19:08.12carrarbelligerent about needing help
19:08.26bpgoldsbp3nguin, Are you setting it before you bridge with another channel?
19:08.29bpgoldsbi.e. from a dial
19:08.46[TK]D-Fenderp3nguin: Yes, that looks fine.  I'm also sure you're doing something else wrong
19:09.10bpgoldsbIf so you might need to look at variable inheritance, as [TK]D-Fender pointed out to me.
19:09.12[TK]D-Fenderp3nguin: So show me the evidence and I'll show you where you went wrong.  Odds are its in the part you're not telling me
19:09.43[TK]D-Fenderbpgoldsb: there is no inheritance.  he jsut said one context to another
19:09.53[TK]D-Fenderbpgoldsb: So yeh, I want PROOF
19:10.07carrarYou can't handle the PROOF!!!
19:10.26[TK]D-Fendercalls a Code Red on carrar
19:10.30carrarheh
19:11.46p3nguinCould have been a mere typo.  I changed back from trying to set it globally, and it seems to be carried into the next context.
19:12.31[TK]D-Fenderz0mg a miracle!
19:12.37p3nguinNot really.
19:13.06*** join/#asterisk hardwire (n=hardwire@216-67-99-228.static.acsalaska.net)
19:13.16hardwireanybody have to restart iaxmodem/hylafax often?
19:13.32hardwireI can't quite figure out which process stops performing correctly.
19:13.56Kattyissues an Amber Alert on eppigy
19:16.19Naikroveki wonder if those amber alerts actually work
19:16.40hardwiresteal a child and see.
19:16.47Kattyyes, they do.
19:16.54Naikroveknah i have two kids already.  don't want a third
19:16.59Kattymost turn out to be kidnappings by parents.
19:17.04Naikrovekyes
19:17.09Kattyor other relatives.
19:17.15hardwireKatty: did you see the castle episode about that?
19:17.18Naikroveklike 95% of kidnappings and sexual abuse on children is family
19:17.50Naikrovekbut they can't automatically make the family suspects because that's profiling
19:19.07Naikrovekwhich sucks because profiling effing works
19:19.28russellbhow about that Asterisk!
19:19.36Naikroveklol, yeah
19:19.46Naikrovekman i gotta watch my off topic BS
19:19.50Naikroveki'm bad at that...
19:19.56Kattyyou think you're bad?
19:20.04Kattywhen was the last time i talked about asterisk? ;P
19:20.13Naikrovekjust now you did
19:20.29[TK]D-FenderKatty: Yeah... you KeySystem sellout :p
19:20.41Kattygrumps.
19:20.53*** join/#asterisk TimToady_ (n=moi@adsl148-239.kln.forthnet.gr)
19:20.55jayteeDK Strata FTW!!!
19:21.18Kattyhugs on jaytee
19:21.44Kattyjaytee: i am toshiba certified.
19:21.45jayteeI wonder if Avaya's gonna get saddled with all of Nortel's Norstar key system business crap
19:21.47[TK]D-FenderjayOne of my earlies clients was a Toshiba reseller getting into * & VoIP :)
19:21.55[TK]D-Fenderjaytee: One of my earliest clients was a Toshiba reseller getting into * & VoIP :)
19:21.58jayteeKatty, I'm certifiable!
19:22.06[TK]D-Fendercommits jaytee
19:22.12Kattyvisits jaytee
19:22.15jaytee900 million
19:22.17*** join/#asterisk scalex000 (n=chatzill@190.166.149.204)
19:22.30jayteewhat a fall from grace that one was.
19:22.31Kattyi can't help it.
19:22.37Kattymy company wants to sell what Sells.
19:22.50Kattyand what Sells around here, is Toshiba and Samsung POS el cheapo systems.
19:23.02scalex000good afternoon
19:23.11Kattyhowdy.
19:24.48Kattymisses eppigy
19:24.51jayteeKatty, in this period with this economy, you keep whatever job you have and sell whatever shit you have to sell if you've got at least a lick of common sense. Let the retards stand on loyalty and principle. When they start whining about the fact they're starving tell them you'll make them a loyalty and principle sandwich.
19:25.00jayteeyeah, where is Dave today?
19:25.22Kattyidk, maybe he went to take a nap
19:25.27Kattyhe told me he was sleepy earlier.
19:25.50[TK]D-Fenderjaytee: Ferengi Rule of Acquisition #109: Dignity and an empty sack are still worth an empty sack
19:26.22jaytee[TK]D-Fender, if I ever get venture capital I'm tagging you for my CTO
19:26.35Naikroveki love those ferengi rules
19:26.56Naikrovekack, offtopic again
19:26.59Naikrovekshyte
19:27.07[TK]D-Fenderjaytee: Ferengi Rule of Acquisition #1: Once you have their money, you never give it back
19:27.29wonderworldi need to write a bill right now and can't decide if i'd charge a bit too much or not. i am another moral-victim.
19:27.34Kattygoes hunting for eppigy
19:28.04*** join/#asterisk eppigy (n=Dave@216-139-245-58.aus.us.siteprotect.com)
19:28.06eppigyhello
19:28.07jayteepeople who know all of the Ferengi Rules of Acquisition are still only have as nerdy and weird as people who speak fluent Klingon
19:28.08eppigyi am dave
19:28.12jayteeDAVE!!!!!!!
19:28.20jayteewe were just wondering about you
19:28.27eppigyHI
19:28.36eppigymy shell box got rebooted
19:28.44Kattypsh. shell box.
19:28.45eppigyand it takes me a while to rejoin everything
19:28.48Kattyi can still find you.
19:28.51eppigyyesh
19:28.53KattyYOU CAN"T HIDE FROM ME
19:28.54eppigyyou know where i live
19:28.59[TK]D-Fender's shell box has a conch in it
19:29.08Katty[TK]D-Fender: you're a conch.
19:29.13jayteemmmm, conch chowdah!
19:29.30[TK]D-Fendereppigy: nothing says "I love you" quite like a restraining order...
19:29.57jayteewonders which version of 1.6.x to download to test sip tcp to Exchange UM
19:30.00Kattyjaytee: curried :>
19:30.18Kattywait, what?
19:30.28*** join/#asterisk oej (n=olle@ns.webway.se)
19:30.40[TK]D-Fenderjaytee: 1.6.0.15
19:30.46Kattysip tcp to exchange?
19:30.53Kattylinkylink?
19:30.58jaytee[TK]D-Fender, thanks. 1.6.1.x still to new?
19:30.59Kattyinfobot: exchange
19:31.12Qwellinfobot: exchange Katty with a banana
19:31.13Kattyinfobot: well of course it doesn't. it's exchange.
19:31.43KattyQwell: do you have a receipt for that transaction?
19:31.50jayteeKatty, we use Exchange UM with * 1.4 but use sipX as a udp/tcp transform proxy to UM since * 1.4 only speaks udp and Exchange is Microsoft and they ALWAYS HAVE TO BE DIFFERENT
19:32.08Kattywhat is 'UM'?
19:32.10Qwelljaytee: well, technically both violate the RFCs
19:32.15jayteeUnified Messaging
19:32.28jayteeQwell, both?
19:32.30Kattyohisee.
19:32.34Qwellboth are required I do believe
19:32.35eppigyunified messing
19:32.46Qwellboth = Asterisk and Exchange
19:32.53Kattywho comes up with these Big Words
19:32.59Qwelland both = TCP and UDP
19:33.00KattyUnified Messaging
19:33.02Naikroveksince when does anythign not violate the RFCs it claims to imiplement
19:33.05QwellKatty: Cisco
19:33.12Kattywell cisco can go...
19:33.14Kattyeat a sandwich
19:33.15jayteeah, so 1.4 violatest the RFC and Exchange 2007 UM does too. but then that means 1.6 doesn't violate it!!!
19:33.23Qwelljaytee: something like that
19:33.28Naikrovekjaytee: safe to assume that everything does, really
19:33.35[TK]D-Fenderjaytee: relatively... it is a full release so if you want to play around it shouldn't amtter too much
19:33.43Naikrovekbut as i understand it * 1.6 and Exchange 2007 can talk just fine
19:33.43jayteeCisco Kid weren't no friend of mine.
19:34.00Kattyi really like cisco, actually.
19:34.03Kattythey have some amazing products.
19:34.09Kattywith an amazing price tag.
19:34.16Naikrovekoh yes
19:34.21Kattyeven their VP's halo has an amazing price tag
19:34.23Naikrovekbut really good products
19:34.25Kattyi think he forgot to snip it off.
19:34.40Kattywe won't go there tho.
19:34.50Kattyi'm particularly fond of the their blackberry app
19:34.58jayteeI think drmessano said that 1.6.0.15 worked well. I'll start there and if it's stable that'll be my 1.6 upgrade path to do away with sipX as an in between.
19:35.04Kattyit's a little status/presense thingy
19:35.11Kattywith COLORFUL DOTS
19:35.15Kattymore dots! more dots!
19:35.39jayteeDots are ok but I prefer Crows being a big licorice fan
19:35.52KattyQwell: you did get the reference, right?
19:35.56KattyQwell: more dots! more dots!
19:36.27Kattydoesn't have any dots :<
19:36.35Kattyi do have a couple Hots, tho.
19:36.36Naikrovekhas freckles
19:38.55wcselbyI have a user saying that when one other user transfers a call from a queue to her, all she hears is hold music. Me being the call being transfered, I can hear the person who says all she hears is hold music.  I do not hear the hold music.  Here's the CLI from when it happens: http://pastebin.com/d1255631c
19:39.14Naikrovekthat sounds weird
19:39.17wcselbyany thoughts?
19:39.49Kattyi think something is faschnuckadid up
19:40.11Kattylooks at post
19:41.48*** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm)
19:44.44*** join/#asterisk lesouvage (n=lesouvag@82.73.69.76)
19:47.53Naikrovekjust spoke with comcast about maybe getting a new internet connection.  tried to explain to her that I spoke with the business ethernet guys (fiber) and she didn't seem to know the difference between coaxial cable and fiber optic
19:48.21Naikrovektold her that they quoted me $1800/month for 20mbps each direction because they had to build infrastructure to my business
19:48.35Naikrovekeh nevermind
19:48.41Naikrovekthey're just so stupid
19:48.48wcselbylol
19:48.56wcselbyyou spoke with the wrong person, obviously
19:49.06Naikrovekshe was an accoutn manager
19:49.54Naikrovekbut yes wrong person
19:50.49*** join/#asterisk jtodd (i=jc3v8cag@ns.fox-den.com)
19:50.49*** mode/#asterisk [+o jtodd] by ChanServ
19:53.44*** join/#asterisk zpertee (n=chatzill@68.142.169.62)
19:53.53*** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim)
19:56.11zperteeDoes anyone know if Intel 537E cards will work with asterisk?
19:56.55*** join/#asterisk IBC_jkenney (n=jkenney@ip65-44-169-66.z169-44-65.customer.algx.net)
19:57.43IBC_jkenneyi need a bit of assistance i am using chanspy and what i want to do is when you enter a specific passcode you are listening to a specific $SPYGOUP
19:57.48IBC_jkenneyhas aonyone done this
19:59.58Qwellzpertee: That is a modem.  No it will not.
20:00.06IBC_jkenneyi need a bit of assistance i am using chanspy and what i want to do is when you enter a specific passcode you are listening to a specific $SPYGOUP
20:00.08IBC_jkenneyhas aonyone done this
20:00.31wcselbytry asking one more time Iamnacho
20:00.32wcselbyerm
20:00.35wcselbyIBC_jkenney
20:01.14zperteeQwell: ok.  I had heard reports of at least one intel card being nothing more than a clone of a digium card
20:01.19IBC_jkenneyCould someone please assist me with this problem i am having i would greatly appreciate it
20:01.27*** join/#asterisk el_critter (n=critter@200.8.97.41)
20:02.17wcselbyIBC_jkenney - use the Read() command to set a variable then run a check against that variable to determine which $SPYGROUP to run chanspy on
20:02.34*** join/#asterisk joako (n=joako@opensuse/member/joak0)
20:02.58IBC_jkenneyI want to use the function to read the pw and channel variable from a file
20:03.08IBC_jkenneybut i think i have the syntax wrong or something
20:03.19IBC_jkenneyi'l like to get this working b.o.b today
20:03.24wcselbyIBC_jkenney - okay.
20:03.42TJNII~questions
20:03.43infobotremember, there are no stupid questions, just stupid people. <http://www.catb.org/~esr/faqs/smart-questions.html>
20:03.44wcselbyIBC_jkenney - be sure you don't show us anything, because the more we see the less we can help you...
20:03.52TJNIIIBC_jkenney: Read that link
20:05.22*** join/#asterisk Tim_Toady (n=moi@77.49.156.2)
20:05.28el_critterhi
20:06.07Pan3Del hello
20:09.00*** join/#asterisk Slezhuk (n=mammon@95-25-249-49.broadband.corbina.ru)
20:09.46*** join/#asterisk Micc (n=Micc@c-98-225-59-171.hsd1.wa.comcast.net)
20:18.51bmoracahow well does DUNDI scale?  can I have hundreds of peers or am I limited to several?
20:19.16[TK]D-Fenderbmoraca: What is your goal?
20:19.24wcselbyis ultramonkey still the preferred HA solution for linux?
20:19.48bmoracause DUNDI to direct calls to my hosted PBXes from my media gateway servers as opposed to static call routes
20:20.01IBC_jkenneyMy Question is i would like to use the chanspy application for managers of different departments to monitor their employee's.  I am looking at storing the password and group name in a textfile and then have the channel name read from the file what is the best way to go about it.  Not every manager should have the ability to listen to calls in all area's
20:20.13*** part/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com)
20:21.14[TK]D-FenderIBC_jkenney: Go make your own AGI and auth it yourself
20:21.17TJNIIProbably an AGI script, I would say.
20:21.41IBC_jkenney<==== is looking for simple
20:21.53bmoracawhat you're asking isn't simple
20:22.05wcselbyIBC_jkenney - I gave you simple, you're looking for something more than simple
20:22.45IBC_jkenneyaccording to chanspy it can do what i'm asking i think i am just doing it wrong
20:23.19IBC_jkenney<PROTECTED>
20:23.19IBC_jkenney<PROTECTED>
20:23.19IBC_jkenney<PROTECTED>
20:23.19IBC_jkenney<PROTECTED>
20:24.01*** join/#asterisk captiancrash (n=jonmoore@70.159.118.70)
20:24.06Kattyhai IBC_jkenney
20:24.54bmoraca[TK]D-Fender: the idea is this:  instead of statically assigning my call routes from my media gateway servers, I want to instead potentially have those routes dynamically generated (this is the point of DUNDI, isn't it?).  Consider OSPF stub areas as opposed to static routing in a large IP network.
20:25.11IBC_jkenneyi wanted to do it with authenticate
20:25.11IBC_jkenneythen pass it to chanspy
20:25.11IBC_jkenneysorry
20:25.23*** join/#asterisk denon (i=denon@sassinak.net)
20:25.23*** mode/#asterisk [+o denon] by ChanServ
20:25.25[TK]D-FenderIBC_jkenney: that is Authenticate.  it won't create a mapping to channels its allowed to spy on.
20:25.40[TK]D-FenderIBC_jkenney: What you want is custom so yes, you're going to have to code.
20:25.58IBC_jkenneyhas no idea how to code
20:26.18wcselbyIBC_jkenney - use the Read() command to set a variable then run a check against that variable to determine which $SPYGROUP to run chanspy on
20:26.34*** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at)
20:26.40[TK]D-FenderIBC_jkenney: http://tinyurl.com/sulfm
20:26.49[TK]D-Fender:D
20:26.52wcselbybut whatever, it's time to head out
20:26.56wcselbylol @ [TK]D-Fender
20:27.03wcselbyheroes is on tonight!  \o/
20:27.30[TK]D-Fendercheckout time.. BBIAB
20:28.06IBC_jkenneywow
20:28.12IBC_jkenneysorry to have bothered you all
20:33.09*** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman)
20:33.21dustybinhas anyone here ever been guilty of using ChanSpy() ?
20:33.35Pan3DIBC_jkenney: heh, not a question of bothering anyone... but if you want to administer this stuff, you gotta dive in. It's not like Fisher-Price VoIP server. It requires some preparation.
20:33.56NaikrovekMr. Potato VoIP
20:33.59Pan3Dlol
20:34.11Naikrovekhe's right tho
20:34.20*** join/#asterisk hugorebelo (n=hugorebe@200-171-132-124.completo.com.br)
20:34.39Naikrovekto be fair it IS a lot to take on at once
20:34.50Naikrovekasterisk is not simple to the untrained eye
20:35.06Naikrovekbut, as [tk]d-fender always says, it's all in the dialplan
20:35.15Naikrovekget that figured out and you're 95% of the way there
20:35.21Naikrovekpeace.
20:36.14KattyIBC_jkenney: are you okay?
20:37.12KattyIBC_jkenney: i wouldn't let them get ya down--everything worth doing takes time (=
20:42.22zperteeIBC_jkenney: Katty is right.  My first asterisk job that I did I had planned on finishing it in a couple of weeks.  A few months later I was still tinkering... just roll up your sleeves and get dirty :-)
20:42.52Kattyis still tinkering after 4 years.
20:42.56*** join/#asterisk brezular (n=brezular@adsl-dyn232.78-99-68.t-com.sk)
20:43.12Kattyphone system is a never ending project.
20:43.22Kattylike a company website
20:43.31zperteecouldn't agree more
20:44.41IBC_jkenneyOh o'm ok
20:44.46IBC_jkenneyjust flustered
20:44.53IBC_jkenneyand i won't let them get me down
20:45.40IBC_jkenneyi mean if they really don't know how to do it thats all the had to say no need to get on the defensive and try to make someone else look stupid to heal their ego
20:45.41IBC_jkenney:)
20:45.44IBC_jkenneyi'm fine
20:46.00IBC_jkenneyi just prefer not to have a pissing match
20:46.02IBC_jkenneythats all
20:47.04IBC_jkenneyThis is not my first deployment this is my first complicated deployment and when its a company you work for as an employee its a different then when your a contractor
20:47.09zperteebeen in your shoes. its a real art to get the info you need without a pissing match :-)
20:47.30IBC_jkenneynot really its a matter of finding mature people
20:47.39Kattywell count me out.
20:47.42IBC_jkenneyinstead of the Kiddies
20:47.43IBC_jkenneylol
20:47.47Kattyi'm still a kid
20:48.00IBC_jkenneyi didn't mean you katty
20:48.04IBC_jkenneyat least you try
20:48.08IBC_jkenneyand are nice about it
20:48.23IBC_jkenneyi'll figure it out
20:49.26Kattydon't tell. i have a reputation to keep up around here.
20:49.33zpertee:-)
20:49.49IBC_jkenneyoh sorry
20:49.50IBC_jkenneylol
20:51.11bpgoldsbIs there a more user-friendly way to view the changes between 1.6.0 and 1.6.1 than the changelog from downloads.digium.com?
20:51.43drmessanoYou want a better changelog than the changelog?
20:52.50bpgoldsbI want an overview of big changes.
20:53.30captiancrashdrmessano, i see a yodawg of that coming soon.
20:53.49drmessanolol
20:54.12drmessanobpgoldsb: Those can all be big changes, depending on your definition
20:55.38bpgoldsbYa, this (http://svn.digium.com/svn/asterisk/tags/1.6.1.0/CHANGES) works nicely compared to the changelog
20:57.03*** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194)
20:58.07*** join/#asterisk _brent_ (n=_brent_@orem.jiveip.net)
20:59.20*** join/#asterisk t_ (i=tom@freenode/staff/tomaw)
21:02.00Kattygoshdangitanyhow
21:02.04Kattywhy am i /always/ friggin hungry
21:02.05KattyALWAYS
21:02.08Kattythrows things
21:02.21_brent_is anybody here using cdr_custom? i'm getting empty values, like this:
21:02.22_brent_,"","","","","","","","","","","","","","","","",""
21:02.39Kattylooks like a train.
21:02.42Kattywith maybe, snacks.
21:02.53_brent_the third car has nachos, if you're interested
21:03.25_brent_cdr_custom.conf has
21:03.25_brent_Master.csv => "${CDR(clid)}","${CDR(src)}",etc.
21:03.27LinuturkI feel silly asking this, but I'm trying to use Playback to play back a particular ulaw file on my system. I've got the full path to the file, but asterisk says it can't find the file :(
21:03.53_brent_Linuturk: make sure you're not including the .ulaw part of the filename in your command
21:03.58Kattyi would make sure your Playback() does not have an extension
21:04.11Kattyso Playback(/path/to/file.gsm) wouldn't work
21:05.31KattyLinuturk: if not that you could be looking at permissions and encoding problems
21:10.50_brent_with my cdr_custom problems, i can get values like ${EPOCH} to show up, but none of the ${CDR()} variables, any ideas?
21:10.58Linuturkthanks _brent_ that was the issue
21:11.01Linuturksilly me
21:11.15Linuturkand Katty ^^ :)
21:11.53Katty_brent_: i just use the postgres connection. don't do anything custom
21:12.30_brent_i've been using cdr_pgsql for a few years. just trying to measure performance vs flat files, but i've got some custom fields i'm writing currently
21:13.08_brent_but i can't get any of the CDR fields to show up in cdr_custom, let alone my custom ones
21:13.21*** join/#asterisk el_critter (n=critter@200.8.97.41)
21:15.21el_critterHi, sometimes when I call from mi SIP phone via PSTN trunk and I hang up, it seems like asterisk hangs de SIP device part but the line keeps open until something (haven't been able to determine what) resets it.
21:25.25[TK]D-Fender_brent_: pastebin your configs and the output of * starting up
21:26.33dustybin[TK]D-Fender: do you ever feel like focusing on another open-source technology? maybe mail servers?
21:27.15[TK]D-Fenderdustybin: Nope.
21:27.23dustybinwhy not?
21:27.42Qwellonly real rock stars get to work with mail servers.
21:27.43[TK]D-Fenderdustybin: I have marginal requirements on other bits, but nothing I felt compelled to go into depth with
21:27.52*** join/#asterisk MikeJ (n=MikeJ@freeswitch/developer/mikej)
21:27.57Qwelllike MikeJ here
21:28.00dustybinin freenode, every channel has there in-house purists :D
21:28.50scalex000TK: Hello
21:28.53drmessanoI could care less about telephony
21:29.03Kattyhi mike
21:29.08*** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm)
21:29.08Kattyhugs MikeJ
21:29.11Kattyhugs anthm
21:29.12drmessanoAsterisk is my fav video game.. the fact that I can make calls with it is "ok" I guess
21:29.23MikeJhmm
21:29.25MikeJruns
21:29.28Katty:<
21:29.33drmessanoSwine flu?
21:29.38Kattymakes a note to switch deoderants.
21:30.02scalex000Tk: which function I need to use to hang up a call when I close the call
21:30.38[TK]D-Fenderscalex000: Show me one failing.  And don't just ask blindly without providing backup, and definitely stop targeting individuals for support
21:30.46hardwiresniffs Katty
21:31.22[TK]D-Fendersnuffs hardwire
21:31.34[TK]D-Fendergoes to dig another ditch
21:31.41hardwiresnubbs [TK]D-Fender
21:31.51scalex000ok
21:31.52hardwirewhy is it every time I want to talk to TK I start with \[
21:31.57hardwirethen hit tab a few times and get mad.
21:31.58hardwire:P
21:34.08scalex000TK: when someone call, and hang up before i pickup the phone the call continue ringing
21:35.54Kattyhardwire: do you mind.
21:36.15*** part/#asterisk MikeJ (n=MikeJ@freeswitch/developer/mikej)
21:36.16*** join/#asterisk umay (n=chris@174-16-21-58.hlrn.qwest.net)
21:36.19Kattyhardwire: i am not a Hot Blonde in a Red Dress
21:37.07leifmadsenKatty: brunette and blue dress
21:37.19*** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq)
21:37.20outtoluncwas gonna guess green dress
21:37.38leifmadsenoh wait, I thought we were talking about me
21:37.55_brent_[TK]D-Fender: http://pastebin.com/d15b55b3e
21:38.07Kattywell.
21:38.12Kattyi am wearing blue today.
21:38.24Kattychecks for cameras in her office.
21:38.44leifmadsenKatty: mwahahaha
21:38.48leifmadsenKatty: is there one in your laptop?
21:38.53*** join/#asterisk okaratas (n=netadmin@fsf/member/okaratas)
21:38.54leifmadsenbecause it may or may not be hacked
21:38.56hardwirethis is going strange places.
21:39.06hardwireshrodingers hack.
21:39.20leifmadsenstrange? #asterisk? wow, you must be new :)
21:39.26Kattyno.
21:39.47[TK]D-Fender_bAnd what file are you looking in for your CDRs?
21:40.00_brent_/var/log/asterisk/cdr-custom/
21:40.10_brent_it's getting appended to, but it's all blank columns
21:40.16_brent_"","","",etc
21:40.32anonymouz666there's a killer bug in app_queue that report the member as "IN USE" when you hit the DND button of softphone or something like that
21:40.39Kattyreplaces [TK]D-Fender's tab button with a tab button! let's see if he notices!
21:40.45anonymouz6661.4.21.2
21:40.51anonymouz666going to update quickly
21:40.54_brent_values like ${EPOCH} show up, but none of the ${CDR(foo)} values
21:41.26hardwireanonymouz666: soooo you want a DND phone to ring?
21:42.06anonymouz666no it's a way to simulate the issue
21:42.23hardwirewhat do you want it to say?
21:42.28hardwireAVAILABLE?
21:42.40anonymouz666you turn on and off......
21:42.52*** join/#asterisk obnauticus (n=l@about/windows/regular/obnauticus)
21:43.09anonymouz666and you keep "in use" forever until you restart asterisk
21:43.24anonymouz666there is no way to delivery a call to this member again
21:43.26*** join/#asterisk puzzled_ (n=foobar@puzzled.xs4all.nl)
21:43.49hardwireah
21:44.18hardwireI've seen something similar but never found a fix.
21:44.26[TK]D-Fender_brent_: OK, I don't see it...
21:45.32_brent_don't see the pastebin or don't see ...?
21:47.37anonymouz666hardwire: I didn't search for a fix for that in SVN since my version
21:47.48anonymouz666I'll update anyway to latest version (due security issues)
21:48.00anonymouz666and then report as a bug if persists
21:50.27*** join/#asterisk _bugz_ (n=bugz@adsl-99-129-215-159.dsl.lsan03.sbcglobal.net)
21:54.06*** join/#asterisk smps (n=maher@193.170.53.51)
21:57.54_brent_is puzzled
22:00.24bpgoldsbif soft hangup <channel> isn't working, is there a way to do a hard kill of a channel from the cli?
22:02.05leifmadsenbpgoldsb: no, only a restart of the system, or patience will do it -- typically that happens from a dead lock. If you've compiled with MALLOC_DEBUG you should be able to do 'core show locks'
22:02.48bpgoldsbHmm.  Thats interesting because I'm the only one using this machine
22:03.05bpgoldsbAnd I just started asterisk, made 1 call, and an immediate deadlock
22:03.37hardwire"asterisk" is a reserved syslog word.
22:03.38hardwire:)
22:03.59bpgoldsbleifmadsen, do you know if MALLOC_DEBUG has reasons to not be on all the time?
22:04.08bpgoldsbI imagine performance, but I don't know for sure
22:04.38leifmadsenbpgoldsb: no reason I can think of. I always enable DONT_OPTIMIZE and MALLOC_DEBUG so that when/if an issue comes up, I have the information available to me to produce a useful bug report
22:05.06bpgoldsbleifmadsen, MALLOC_DEBUG but not DEBUG_CHANNEL_LOCKS?
22:05.40leifmadsenI think it's MALLOC_DEBUG to enable 'core show locks'
22:05.58bpgoldsbWell, it's a testing box, so I'll throw em both on and see what happens.
22:06.09leifmadsenheh
22:06.41*** join/#asterisk mbrevda (n=mbrevda@unaffiliated/mbrevda)
22:06.51hardwireit feels like a friday
22:06.52hardwireyou know
22:07.00leifmadsenit so does
22:07.08mbrevdaanyone know of a decent set of english uk promts for asterisk with a voice talent avlible for further work?
22:07.23hardwireoi!
22:07.34hardwireI'll do eet!
22:09.57*** join/#asterisk voxter (n=voxter@76.77.95.2)
22:11.24hardwirehi voxter
22:11.30voxtersup hw
22:12.04hardwiresame ooooold same oooold
22:12.14hardwirestill got fancy glasses?
22:12.20hardwirethought they were handsome and fancy.
22:13.09hardwirenot sure how to combine those words.. handsomeancy or maybe fansomey.
22:13.45*** join/#asterisk andres833 (n=andres83@190.156.152.225)
22:13.47_brent_fandsom
22:14.23dustybinmbrevda: http://www.enicomms.com/cutglassivr/
22:14.38mbrevdadustybin: have you used them befor?
22:14.46dustybini use now
22:15.02mbrevdahave you done custom promts?
22:15.05dustybinnot yet
22:15.10dustybini might do one day
22:15.15dustybini use the asterisk replacements
22:15.25dustybinshe has a nice voice
22:15.55dustybinanyway, time for bed
22:15.56dustybinnn
22:16.14mbrevdathnkx
22:23.25bpgoldsbleifmadsen, What should I be passing to 'core show locks'?
22:24.04leifmadsenbpgoldsb: nothing -- it just shows the currently held locks -- it can be useful to a developer when debugging issues with asterisk. Typically if you can't 'soft hangup' a channel it is because there is a lock being held
22:24.23leifmadsenwhich is not a problem in itself -- it is when it is a deadlock and it is not released that it becomes an issue
22:24.33bpgoldsbOh, I ask because core show locks wasn't giving me anything but a usage statement
22:24.46bpgoldsbactually, no, it's giving me no locks
22:24.52bpgoldsbbut soft hangup still doesn't work
22:24.54leifmadsenthen there are no locks being held -- that is normal
22:25.10leifmadsenrun it after you've done 'soft hangup' and the channel does not go away
22:25.16bpgoldsbI did.
22:25.21bpgoldsbStill no locks.
22:25.24bpgoldsbChannel still exists.
22:25.44leifmadsenthen it isn't a deadlock
22:25.54bpgoldsbYay.
22:26.37bpgoldsbMayhaps I should just throw festival integration off my todo.
22:27.01leifmadsenya, cepstral tends to be a lot better
22:27.12leifmadsenI don't even remember the last time I tried festival integration... like... years ago :)
22:27.20bpgoldsbcepstral?
22:27.50Nuggetthe only time I played with festival I found it to be about as high quality as wiring a speak 'n spell toy into the phones.
22:27.54bpgoldsbOh, paid for software.
22:28.12Qwellspeak 'n spell <3
22:28.16bpgoldsbHah, well, this is just a wake-up-call system for employees.
22:28.25bpgoldsbSo quality is not important
22:28.29Qwellconsidering the hardware is ran on...man, was that impressive tech
22:28.34Qwells/is/it/
22:34.50*** part/#asterisk mbrevda (n=mbrevda@unaffiliated/mbrevda)
22:39.26Kattyhas beef stew and garlic bread!
22:39.40*** join/#asterisk kink0 (n=xchat@86.Red-212-170-176.staticIP.rima-tde.net)
22:39.43kink0hello
22:40.19kink0quick question, why Dialtones() sounds so bad quality while Background any file sounds fine ? ( H323 channel + g729 )
22:41.55kink0sorry Playtones()
22:45.18*** join/#asterisk rossand (n=rossand@CPE000c413a19a3-CM00159a025ad4.cpe.net.cable.rogers.com)
22:45.32*** join/#asterisk kerchunk (n=kerchunk@pool-173-49-10-152.phlapa.fios.verizon.net)
22:50.08*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
22:52.22obnauticuszrjyer0su6j]ga-y9rzjyrjhyrz9hyjr9pyjzsr
22:54.06obnauticuserrr
22:54.26[TK]D-FenderH4X
22:54.41obnauticuslololololololol
22:56.17*** join/#asterisk Iamnach0 (i=Iamnacho@ip98-186-180-143.ks.ks.cox.net)
22:56.31kink0anyone has some idea about playtones sounds so choped and distortioned ? ( playing any other audio file sounds ok )
22:57.53_brent_one last try...is anyone out there using cdr_custom that can confirm whether the Master.csv => "${CDR(clid)}","${CDR(src)}",etc. template works?
23:01.40_brent_anthm: have you worked on cdr_custom?
23:02.13*** join/#asterisk Naikrovek (n=jjohnson@98.214.112.102)
23:12.43*** join/#asterisk psilikon (n=psilikon@140-1.35-65.tampabay.res.rr.com)
23:13.47psilikonSo I am following CH 5. of the book *: TFOT 2nd Ed. but I can't get any calls from SIP to my asterisk box
23:14.01psilikonmy ITSP is Callcentric
23:16.50psilikonI am using an ATA with a SIP provider and everytime I place a call to my sip number I watch asterisk give me:  Call from '' to extension '1777XXXXXXX' rejected because extension not found.
23:19.07p3nguinAh, good ol' CallCentric.
23:19.34p3nguinThis is a common problem with them.  Just match your number in the dialplan.
23:20.05psilikonp3nguin, I can't seem to pull that off for some reason
23:20.27p3nguinpsilikon: Can you show me your context where you get calls inbound from CC?
23:20.55p3nguinI'm willing to help you fix this, but you'll have to cooperate.
23:21.27psilikonhttp://pastebin.com/m794e25be
23:24.47p3nguinpsilikon: What SIP number do you want to ring when someone calls your CC number?
23:25.09psilikonSIP number as in sip extension?
23:25.26p3nguinsure
23:25.37p3nguina SIP peer that will take the call.
23:25.37psilikonI have only one phone right now. I call it extension 1000 in my sip.conf
23:27.11p3nguinhttp://pastebin.com/d11964568
23:27.33p3nguinUse that context for you inbound callcentric calls to SIP/1000.
23:28.16p3nguinThen go over to sip.conf and make sure that your callcentric context says context=from-callcentric
23:29.32p3nguinI've made the assumption that your music on hold will play while your phone is ringing, and also that you have your voicemail set up as 1000 in the default context.
23:29.38*** join/#asterisk Defraz (n=Defraz@corp.fuzecore.com)
23:30.43p3nguinAny troubles taking care of those things?
23:31.18*** join/#asterisk manxpower (n=EWieling@24.42.221.26)
23:32.55psilikonNice. It is telling me nonone is available to take the call... but I can probably debug it from here
23:33.25p3nguinMaybe SIP 1000 isn't registered?
23:33.44psilikonUnable to create channel of type 'SIP' (cause 20 - Unknown)
23:33.44psilikon<PROTECTED>
23:33.50p3nguinerm
23:34.04p3nguinYou reloaded sip and dialplan, right?
23:34.55psilikonYep
23:35.21psilikonMy sipura isn't registered.. wtf. I assumed it was by the dialtone
23:35.23p3nguinI don't know what would cause that message if your phone wasn't actually busy or congested.
23:35.30p3nguinAh.
23:35.51p3nguinAsterisk doesn't provide the dial tone on a SIP device.
23:36.34psilikonk, now I am reg'd and gonna give it another go
23:38.10psilikonp3nguin, big help man BIG help!
23:38.19p3nguinWorking?
23:38.23psilikonyep!
23:38.44p3nguinDo you get the music as a ringback on the calling phone?
23:38.49psilikonyes.
23:39.01p3nguinShould say "connecting your call" then play music until answered or going to vm.
23:39.30p3nguinYou can tweak it from there if you want to change that behavior.
23:39.47p3nguinor I can help you if you aren't sure how to change something.
23:40.03psilikonI don't see where the music playing comes in?
23:40.18p3nguin,m is music
23:40.30psilikonoh
23:40.34p3nguinin the Dial() command.
23:40.35psilikon30?
23:40.40psilikonms pause?
23:40.47p3nguin30 seconds ring timeout
23:40.50psilikonoh
23:41.17p3nguinChange it to 3 if you want to make it a challenge to answer the phone before it goes to voicemail.  :)
23:42.42*** part/#asterisk _brent_ (n=_brent_@orem.jiveip.net)
23:43.15*** join/#asterisk jcape (n=jcape@adsl-75-21-76-59.dsl.chcgil.sbcglobal.net)
23:44.04p3nguinKeep in mind that I used the silence/1 and vm-dialout to make it a pleasant experience when calling your number.  You can alter the dialplan just about any way you want, as long as it will match your phone number and dial to SIP/1000.
23:45.07psilikonnice. "m" is just generic music or can you change it?
23:45.42*** join/#asterisk Hadding (n=kyle@207.177.231.9)
23:46.09p3nguinYou can configure your music on hold via the musiconhold.conf file.
23:46.38p3nguinYou could also delete the default music and put in your own if you really wanted to.
23:46.54psilikonSo if I added another xtension under [from-callcentric] that started with "s" it would always play when a call was answered?
23:47.10p3nguinnope
23:47.38p3nguinThe 's' extension doesn't seem to work with SIP for a technical reason.
23:47.56p3nguinWhat are you trying to do?
23:48.02p3nguinRing another phone?
23:49.15psilikonI was just trying to follow the Asterisk book... but actually thanks to you I am past that part anyway.
23:50.05Haddinghttp://pastebin.ca/1574601      my extensions.conf
23:50.32Haddinghttp://pastebin.ca/1574602 sip.conf
23:50.52manxpowerThe "s" extension is matched when Asterisk has NOT RECEIVED the dialed digits.
23:51.14Haddinghelp, wher do i put this
23:51.18Hadding<PROTECTED>
23:51.20Haddingexten => 33,1,Answer exten => 33,2,AGI(cidspoof.agi)
23:51.21manxpowerWhen dialing from a SIP phone Asterisk almost always knows the dialed digits (or destination number)
23:51.32manxpowerHadding: don't flood the channel
23:51.35manxpower~pastebin
23:51.35infobot[~pastebin] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding 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://www.rafb.net/paste
23:52.03manxpowerHadding: I think you should spend some time with the Asterisk Book
23:52.04manxpower~book
23:52.05infobot[~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook
23:52.20psilikonThat is cool as sh*t. Now I need to come up with something that will allow the caller to press a button and ring my cell
23:53.20manxpower"s" is used 1) when immediate=yes is set on a zap channel 2) Macros and 3) IVRs
23:53.23p3nguinhadding: Are you really that huge of a retard?  I've told you at least three times and I know that several other people have also told you.
23:53.43manxpowersince all of those situations the concept of "dialed number" is silly
23:54.07manxpowerp3nguin: I think you are starting to understand [TK]D-Fender. 8-)
23:54.43p3nguinpsilikon: Just use followme and let it go automatically.
23:55.20Haddingp3nguin I dont know were i put it, can you edit the pastebin thing so i know exactly
23:55.55HaddingSomebody said to put it on pastebin so i did that
23:55.56p3nguinpsilikon: http://pastebin.com/d63850a0e  There's one line change I made there to send the call into followme.
23:55.58manxpowerHadding: download.  the.  book.  These are such basic questions that nobody really wants to answer questions such basic questions..
23:56.19Hadding~book
23:56.20infobot[~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook
23:56.34p3nguinIt's even written out in PLAIN ENGLISH in "2."
23:56.44p3nguin"Add the following lines to your extension config file in the same context as your SIP phone."
23:56.52*** join/#asterisk supers (i=supers@bigmatix.com)
23:56.53p3nguinHow much more CLEAR could it be?
23:57.03HaddingOkay so what context is sip phone in
23:57.12manxpowerp3nguin: some people cannot be helped
23:57.20p3nguinindeed
23:57.52p3nguinpsilikon: You'll still need to configure followme, though.  Let me know when you're ready to go at it.
23:58.05psilikonLet's do it
23:58.15HaddingMan I downloaded asterisk vmware and ubuntu for this and i cant get no help
23:58.25p3nguinpsilikon: Open up your followme.conf
23:58.49psilikonp3nguin, do I still get the cool "please wait while I connect your call + the music"?
23:58.51p3nguinpsilikon: Do you have termination service with CallCentric already?
23:58.57supershi there, i'm trying to strip off the first 4 digits before placing a call, would cut be the best function for this?
23:59.04p3nguinpsilikon: It'll work out.  You'll see.
23:59.15psilikonp3nguin, probably not since I don't know what termination service is
23:59.37psilikonk I am in followme.conf
23:59.56p3nguinsupers: Dial(SIP/${EXTEN:4}@youritspcontext)

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