IRC log for #asterisk on 20090514

00:00.10drmessanoFirst off
00:00.10pauliusI'm not using a multimeter.
00:00.10drmessanoTheres this thing
00:00.11[TK]D-Fenderpaulius: Echo is more than impedence.
00:00.14drmessanoCalled an impedence bridge
00:00.20paulius[TK]D-Fender: Would you care explaining?
00:01.08[TK]D-Fenderpaulius: Not particularly.  This answer is long and should be obtained the same way as your previous question ; JFGI
00:01.18pauliusRight. I'm on it.
00:01.20pauliusI'll let you know.
00:01.29pauliusBtw, [TK]D-Fender, this Cisco phone works amazingly.
00:01.34pauliusAnd the voice quality is amazing.
00:02.37[TK]D-Fenderpaulius: Still won't beat a Polycom, even one cost half as much
00:02.38*** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm)
00:02.52paulius...
00:04.06[TK]D-Fenderpaulius: 7971 doesn't even support G722
00:04.20Gremlinis writing an application to read PCM 16-bit linear resolution sound from Madonna CDs and play it as hold music.
00:04.24[TK]D-FenderOr so many other common codecs...
00:04.32pauliushmm
00:04.33QwellCiscos use Polycom hardware anyways :p
00:04.33[TK]D-FenderSilly Crisco...
00:04.37GremlinGSM is a great codec
00:04.37pauliuswell but it's using something which works
00:05.11[TK]D-FenderGremlin: Writing an app?  * jsut DOES this already :p
00:05.35GremlinFrom CDs? Or should I rip my Madonna CDs to OGG?
00:05.47[TK]D-FenderGremlin: To any format * supports
00:06.08GremlinWill Asterisk support a SIP trunk with 16 lines over a cable Internet connection?
00:06.15[TK]D-FenderGremlin: I'd suggest doing it to the same format as used by the channels that are on hold so as not to transcode unnecessarily
00:06.28[TK]D-FenderGremlin: Depends on your cable internet conenction now doesn't it?
00:06.33GremlinRight, saving CPU cycles is good.
00:06.56Gremlin[TK]D-Fender: I know it does, but Time Warner won't tell me that.
00:07.37GremlinISPs have moved away from stating specific numbers--marketing like the vagueness of adjectives like "blazing fast".
00:07.38[TK]D-FenderGremlin: You gave me 1 set of numbers without any for the OTHER half.
00:07.55[TK]D-FenderGremlin: Well I'm not asking them, I'm asking YOU
00:07.57GremlinI know I did.
00:08.15GremlinAnd my best guess is 768 Kbps.
00:08.16[TK]D-FenderGremlin: If you can't come up with a number, don't even ask the question :)
00:08.48[TK]D-FenderGremlin: And 16 calls on that?  extremely unlikely in the best possible scenario
00:08.57GremlinI would do a test, but currently I have no Internet access there at all (I blame the gov't).
00:09.21[TK]D-Fender13 *16
00:09.32GremlinWell, 768 Kbps would be the upload speed.
00:09.42[TK]D-FenderGremlin: I know..
00:10.09GremlinTeliax told me they have a GSM codec that can do 8 Kbps.
00:10.17[TK]D-FenderGremlin: Actually, it might be doable, but only over an IAX2 trunk connection with something like GSM as a codec
00:10.37[TK]D-FenderGremlin: no, their & *'s GSM 6.10 = 13kbps
00:10.57[TK]D-FenderGremlin: + packet overhead, whih you'll need IAX2 trunking to survive
00:12.08nullable_typeD-Fender >> I set a channel variable via Set(SOURCE_NUMBER=17789602222) function but it doesn't show when i tried core show channels concise. Do you know why. Or did you mean i pull the channel variable some other way?
00:12.12[TK]D-FenderGremlin: Correction... seems you CAN survive non-trucked with GSM or G.729
00:12.15GremlinI could always do cable+ADSL
00:12.24[TK]D-Fendertrunked*
00:12.36[TK]D-Fendernullable_type: Other way for the channel var method.
00:12.51[TK]D-Fendernullable_type: Which I told you.  Then again I told yuo ANOTHER way by changing your CHANNEL line.
00:13.04[TK]D-Fendernullable_type: I think you dropped the programme on the floor.
00:13.15GremlinI' a bit confused with what you mean by trunked. I would assume that I am trunked if I set up Asterisk to use Teliax over Session Initiation Protocol.
00:13.57GremlinIf trunk means the PBX lines.
00:14.04drmessanolol
00:14.13*** part/#asterisk beek (n=klinebl@pdpc/supporter/professional/beek)
00:14.20*** join/#asterisk leif[mobile] (n=leifmads@asterisk/documenteur-extraordinaire/blitzrage)
00:14.20*** mode/#asterisk [+o leif[mobile]] by ChanServ
00:14.24GremlinDisclaimer: I'm an idiot.
00:15.05leif[mobile]bash.org +1
00:15.22GremlinYou submitted this to bash.org? :o
00:15.39leif[mobile]bash.org +2
00:18.45Pan3DGremlin: just read up a bit. if I can pick it up, you can :)
00:19.04GremlinOkay. (initiating bash.org evasion)
00:19.34Pan3Dhaha, nice evasion
00:23.16*** join/#asterisk ruben23 (n=AGENT@122.55.48.242)
00:23.44ruben23hi anyone have idea to increase my asterisk server performance..
00:24.28ruben23i already got E1 connection but when 25 calls at the same time my voice quality still are compromise..
00:25.36ruben23got a dual core 4GBddr2 ram with 500 GB Sata Drive
00:27.28*** join/#asterisk MrNaz (n=mrnaz@ppp118-208-158-191.lns10.mel4.internode.on.net)
00:32.51stopeecho type calls?
00:32.57[TK]D-Fenderruben23: That description gives no indication of being too weak
00:33.41ruben23<PROTECTED>
00:34.19[TK]D-Fenderruben23: Well you've told us nothing suspicious for your problem
00:34.26stopewhat version of * ? are you running g729?
00:35.00alrstrixbox?  music-on-hold? queues?
00:35.05ruben23yes g729 codec..i got this error log on call- http://pastebin.com/m2f564a05
00:35.20ruben23im using asterisk as backend for vicidial..
00:38.29ruben23<PROTECTED>
00:40.31alrsruben23: using hardhdlc instead of dchan in zaptel.conf helps a lot if you have a two or four-port Digium card
00:42.09alrsif you are using multiple one-port cards I predict problems
00:54.04*** join/#asterisk trnzmeta (n=bleh@secure27.lnk.telstra.net)
00:56.30*** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri)
01:01.06jayteealrs, hardhdlc with PRI instead of dchan?
01:01.14alrsyes
01:02.27jayteewhere'd you hear about that?
01:02.54alrsfound it somewhere in a commit message in Digium's version control
01:03.46jayteeah, the joy of well coordinated and consolidated OSS documentation :-)
01:03.46*** join/#asterisk styelz (n=yoohoo@m0o0.mooo.com) [NETSPLIT VICTIM]
01:03.46*** join/#asterisk viq (n=viq@unaffiliated/viq) [NETSPLIT VICTIM]
01:03.46*** join/#asterisk simond (n=simon@syria.uc.org) [NETSPLIT VICTIM]
01:03.46*** join/#asterisk Beave (n=beave@DCC.SEND.startkeylogger.000.telephreak.org) [NETSPLIT VICTIM]
01:04.49alrsjaytee: It was after spending a whole lot of time following a support guy's suggestion that voicebus would fix everything
01:04.49alrsthis was last year when that was just coming out
01:04.49alrsof course, voicebus is only for analog cards
01:04.49alrswhatevs
01:04.59*** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) [NETSPLIT VICTIM]
01:04.59*** join/#asterisk pdmmm (i=pdm@pdm.sh) [NETSPLIT VICTIM]
01:04.59*** join/#asterisk Superbartt (n=bart@ipd50a21c9.speed.planet.nl) [NETSPLIT VICTIM]
01:04.59*** join/#asterisk Katty (n=asterisk@mail.copi-rite.com) [NETSPLIT VICTIM]
01:04.59*** join/#asterisk jjshoe (n=jjshoe@h69-129-142-83.mdsnwi.tisp.static.tds.net) [NETSPLIT VICTIM]
01:05.00*** join/#asterisk chris|wk (n=JoeMoes@85.183.18.3) [NETSPLIT VICTIM]
01:05.00*** join/#asterisk seanbright (n=sean@asterisk/contributor-and-bug-marshal/seanbright) [NETSPLIT VICTIM]
01:05.01*** join/#asterisk ltd_wk (i=z@patwk.transact.net.au) [NETSPLIT VICTIM]
01:05.01*** join/#asterisk `Sauron (n=sauron@dsl001-130-033.aus1.dsl.speakeasy.net) [NETSPLIT VICTIM]
01:05.01*** join/#asterisk wdoekes (n=walter@wjd.osso.nl) [NETSPLIT VICTIM]
01:05.01*** join/#asterisk Arkaos` (n=arkaos@chimera.c0ws.biz) [NETSPLIT VICTIM]
01:05.01*** join/#asterisk rajiv (n=rajiv@gentoo/developer/rajiv) [NETSPLIT VICTIM]
01:05.11jayteewell, I've got two servers with TE212P dual port cards in them and I'm using dchan. It's rock solid so I think I'll leave well enough alone.
01:05.21alrsmakes sense to me
01:05.37*** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) [NETSPLIT VICTIM]
01:05.37*** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) [NETSPLIT VICTIM]
01:05.37*** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) [NETSPLIT VICTIM]
01:05.37*** join/#asterisk kc8pxy (n=gecko@99-182-113-98.lightspeed.clmboh.sbcglobal.net) [NETSPLIT VICTIM]
01:05.37*** join/#asterisk kjs (i=kjs@fedora/kjs) [NETSPLIT VICTIM]
01:05.37*** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) [NETSPLIT VICTIM]
01:05.37*** join/#asterisk Mw3 (i=mw3@mw3.hu) [NETSPLIT VICTIM]
01:05.37*** join/#asterisk joat (n=joat@ip70-174-79-200.hr.hr.cox.net) [NETSPLIT VICTIM]
01:05.37*** join/#asterisk _Raptor_ (i=raptorbl@andariel.informatik.uni-erlangen.de) [NETSPLIT VICTIM]
01:05.37*** join/#asterisk quintana (n=sylvain@aghnar.doowan.net) [NETSPLIT VICTIM]
01:05.37*** join/#asterisk keith4 (n=keith@unaffiliated/keith4) [NETSPLIT VICTIM]
01:05.37*** mode/#asterisk [+o putnopvut] by irc.freenode.net
01:06.26jayteeI have one of my two spans do an HDLC Abort every wednesday at 1:58AM so I suspect the telco is doing a reset of their stuff at that time. It's too precise and regular.
01:07.12*** join/#asterisk Shizuo (i=shizuo@200-171-49-211.dsl.telesp.net.br)
01:08.28drmessanojaytee
01:08.35drmessanoIts 2am, check your damn clocks
01:08.37jayteedrmessano,
01:09.07jayteewell, my clocks are synched to the domain controllers
01:09.32*** join/#asterisk saftsack (n=saftsack@p5792458A.dip.t-dialin.net)
01:09.34jayteeso yeah, maybe Time Warner is doing it at 2AM and my servers are 2 minutes fast
01:09.56jayteebut nothing's broke so I don't give a damn.
01:09.58drmessanoHmm
01:10.03drmessanoI see what you did there
01:10.36jayteelook, unless it means more money in my pocket then we aren't having this argument!
01:12.05*** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica)
01:14.53pauliusSo does anyone know how to eliminate echo on the SPA 3102?
01:15.12ShizuoSPA?
01:15.18pauliusLinksys adapter thingy.
01:15.21ShizuoIS this some kind of digium crap?
01:15.23ShizuoOh
01:15.26pauliusGives an FXS/FXO port.
01:15.29rob0Unplug it. ;)
01:15.35rob0SCNR
01:15.39pauliusYes it's crap but I didn't feel like paying tons for a Cisco system.
01:15.51ShizuoYeah, digium sucks
01:16.06rob0oh the Sipuras are not so bad, for analog stuff.
01:16.28rob0definitely THE cost effective way to get into this.
01:16.31*** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) [NETSPLIT VICTIM]
01:16.33ShizuoSure
01:16.34*** join/#asterisk mellow-yellow (n=mellow-y@exchange.norris-stevens.com) [NETSPLIT VICTIM]
01:16.39ShizuoAs long as it's not Digium (crap)
01:16.40ShizuoIt's ok
01:16.41[TK]D-FenderShizuo: Only one shitting all over Digium.. or heck even mentioning their name here... is you...
01:16.44pauliusrob0: Right. exact;y/
01:16.56pauliusrob0: But do you have any guidance on how to eliminate the echo?
01:16.58*** join/#asterisk brian (n=brian@unaffiliated/brian) [NETSPLIT VICTIM]
01:16.59pauliusIt's quite annoying
01:17.23rob0I buy Digium when appropriate. I certainly support what they've done for the community.
01:17.33[TK]D-Fenderpaulius: www.voxilla.com <- go check their forums.  Plenty of guides on how to tweak to deal with echo situations.
01:17.34ShizuoRipped off?
01:18.06*** join/#asterisk propellerhead (n=yogurt2u@host253.200-82-98.telecom.net.ar)
01:18.21rob0Digium support helped a lot when I was getting started, with a TDM card. Can't complain.
01:18.45ShizuoThey're ok
01:18.58ShizuoThe only issue I have with them is paying IRC shills
01:19.06ShizuoAnd making Asterisk baitware
01:19.16[TK]D-FenderShizuo: "baitware"?
01:19.27ShizuoYes
01:19.33[TK]D-FenderShizuo: Do explain...
01:19.40ShizuoOpen-source stuff where the real deal is a paid software
01:19.50[TK]D-FenderShizuo: And who are "IRC shills"?
01:19.52ShizuoJust like Alfresco and others
01:19.58[TK]D-FenderShizuo: What "real deal"?
01:20.04*** join/#asterisk JayTee52 (n=jforde05@unaffiliated/jaytee)
01:20.12ShizuoThe non-community version
01:20.23[TK]D-FenderShizuo: What makes that "the real deal"?
01:20.27ShizuoWell
01:20.41rob0Huh? The free software asterisk is fully functional.
01:20.41ShizuoAs lots of main developers are connected to the sponsor (jusr like Alfresco)
01:20.44JayTee52comes with a stick on decal?
01:20.52ShizuoThe community distribution is a pain in the ass (ON PURPOSE) to install
01:21.00ShizuoWhile the paid version is easy and trouble-free
01:21.16[TK]D-FenderShizuo: BS... 3 stupid compiles.  No different than any other version
01:21.22[TK]D-FenderShizuo: Software is software
01:21.26rob0Mine ... has no pain from installing asterisk. I *can* think of a PITA, however.
01:21.44[TK]D-FenderShizuo: and "on puropose"?  This is OSS SOURCE.  Its no different to compile than jsut about any other project
01:21.47Shizuo[TK]D-Fender: Only 3 stupid compiles?
01:21.54[TK]D-FenderShizuo: Yup
01:21.57Shizuo[TK]D-Fender: Why is there NO stupid compiles at the commercial version?
01:22.07[TK]D-FenderShizuo: * itself, Zaptel/DAHDI, and Libpri
01:22.22Shizuo[TK]D-Fender: Why not a community version with no stupid compiles?
01:22.25[TK]D-FenderShizuo: I dunno, maybe because it comes preconifugred on an entire ISO?
01:22.35[TK]D-FenderShizuo: And maybe you're ignrant of PACKAGED *.
01:22.36ShizuoBecause that would drain support money from Digium
01:22.42[TK]D-FenderShizuo: Which wouldn't surprise me
01:22.56ShizuoAlfresco works that way too
01:22.58[TK]D-FenderShizuo: Digium has a binary repo, as does every major distro maker.
01:23.05ShizuoLulz
01:23.09[TK]D-FenderShizuo: Your point is now wholly zoid.  Next?
01:23.10rob0The PITA I am thinking of is a person who comes into a Digium-sponsored and -run IRC channel to bash Digium.
01:23.18ShizuoWhat a paid shill
01:23.31[TK]D-FenderShizuo: Ok, Enough of your bullshit FUD
01:23.33rob0I wish I was paid.
01:23.43ShizuoShiiiil, shiil
01:23.50rob0ignored
01:23.53*** mode/#asterisk [+o [TK]D-Fender] by ChanServ
01:23.54ShizuoThis place should be called ##digium
01:23.57ShizuoNot #asterisk
01:24.04Shizuo<PROTECTED>
01:24.36[TK]D-FenderShizuo: Let me get an electron microscope so I can FIND IT FIRST
01:24.46*** mode/#asterisk [+b *!*@200-171-49-211.dsl.telesp.net.br] by [TK]D-Fender
01:24.47*** kick/#asterisk [Shizuo!n=joe@64.235.218.194] by [TK]D-Fender ([TK]D-Fender)
01:24.54JayTee52thank you!
01:25.00rob0I guess that was what he wanted.
01:25.01[TK]D-FenderSo long moron...
01:25.02MaliutaLapwas waiting on that one
01:26.01JayTee52where is br? brazil?
01:26.08rob0yes
01:26.19MaliutaLapprobably a compromised machine
01:27.04MaliutaLapone of the channels I'm on on another network is suffering from botnets trying to appear like genuine users
01:27.22JayTee52seems to me alot of cantankerous assclowns from there come in here lookin to raise a ruckus
01:27.38rob0he was/is real, replying
01:27.52rob0not a bot
01:28.04MaliutaLaprob0: yeah, doesn't mean he was the owner of the machine he was on
01:28.08JayTee52yeah, way too stupid to be a bot
01:28.14rob0haha
01:28.19MaliutaLaprob0: bots and compromised are 2 different things
01:28.32JayTee52not necessarily
01:28.38JayTee52could be both at the same time
01:29.11JayTee52a computer compromised by a bot is still compromised and has a bot, the jerkoff using it might not be compromised, just stupid
01:29.12*** join/#asterisk thehar (i=thehar@thehar.xmission.com)
01:29.26MaliutaLapand not all bots are bad
01:29.33MaliutaLap~jbot
01:29.34infobotwell, jbot is a hack!, or known to have only said one useful thing. a tool, or dating the fembots, or  [TK]D-Fender's b*tch, or suck, or a pain in the ass
01:29.44JayTee52certainly not. infobot is a standup bot
01:29.56*** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-a846428e71332e56)
01:29.59JayTee52~botsnack
01:29.59infobot:), JayTee52
01:30.08JayTee52I like feeding em
01:30.39JayTee52damn, I'm still on as a ghost with my regular account due to the power blib I had here 10 minutes ago.
01:31.09JayTee52but I'm not showing up in the userlist?
01:31.28jayteebetter
01:31.45jayteeI really need to buy a 10 Gigawatt UPS
01:31.55drmessanoebay
01:32.08jayteeof course! why didn't I think of that? :-)
01:32.31jayteehow've ya been, danny?
01:33.35KyleKI only need 1.21 gigawatts
01:33.46drmessanoMe? Pretty shitty
01:34.04jayteesorry to hear that bud!
01:34.31drmessanoIts all good
01:34.33jayteework's been pretty dull lately. back to break/fix and install crap.
01:34.53drmessanoNo high drama?
01:35.04jayteebut at least the landlord had someone fix the damn roof after my kitchen ceiling collapsed from the rain leaking in.
01:35.13drmessanoGood god
01:35.15drmessanoThat sucks
01:35.45jayteenow of course God has decided to "f" with me some more by sending a storm with high winds and heavy rain :-(
01:36.09jayteelong as my kitchen floor is dry tomorrow I'll be happy
01:36.53jayteeIf not I'll just toilet paper a local church as revenge
01:37.07drmessanoIt'll be ok.. One thing I know about god, other than that he doesnt exist, is that it rains
01:37.23jayteehahaha!!
01:37.43MaliutaLapdrmessano: that's god pee'ing on us
01:37.45jayteeor as Woody Allen once said, "Not only is there no God but try and get a plumber on a Sunday"
01:38.02drmessanoJESUS DIES IN THE 7TH BOOK, JUST LIKE DUMBLEDORE <--- SPOILER
01:38.12jayteeLOL
01:38.35[TK]D-Fenderdrmessano: :F
01:39.01drmessanoThat would be an awesome bash
01:39.39MaliutaLapdrmessano: we can bash you over the head with it?
01:39.59drmessano"I just got back from Bible study"  "Hey man, Jesus dies about halfway through" "Damnit!"
01:40.17drmessanoI love spoilers
01:40.25KyleKawww man
01:40.29drmessanoBy the way, in the sixth sense, Bruce Willis is really DEAD
01:40.32KyleKi just got started ;)
01:41.34drmessanoMeatloaf dies at the end of Bat Out Of Hell 3
01:41.39drmessanoO.o
01:42.18jayteewhenever I get cornered by a born again and they bring up the point about the bible being "the literal word of God" I like to point out that in one of gospels when Jesus goes before Pilate he's wearing a scarlet robe and in another gospel he's wearing a purple robe. "So, like He didn't proofread His own work? Dude, that's so lame for someone omniscient and omnipotent, don't ya think?"
01:42.41[TK]D-Fenderdrmessano: http://tinyurl.com/5g3sq4
01:42.45drmessanoROFL
01:43.46drmessanoOh crap
01:43.52drmessanoI havent seen 3 of those
01:43.54drmessano:(((((
01:43.57drmessanoBACKFAIL
01:45.09drmessanoKristin shot JR.. one of my favs
01:46.06drmessanoI watched every episode of Dallas on TNN at the time, one a day for like a year and a half
01:46.30drmessanoGot to watch the Who shot JR? episode.. but only had to wait til Monday to find out
01:46.31drmessanoheh
01:47.06a1faanybody noticed increase ammount of spam in asterisk logs?
01:47.20KyleKi haven't looked at my logs at all yet
01:47.28jayteespam?
01:47.30a1fa[May 13 18:11:18] NOTICE[29301]: chan_sip.c:14654 handle_request_invite: Failed to authenticate user "MeucciSolutions" <sip:MeucciSolutions@93.190.143.10>
01:47.31drmessanoYes, the world is burning
01:47.34drmessanoRun!
01:48.00a1fapeople constantly dicking with my internet facing asterisk server
01:48.26jayteefortunately my * servers don't "talk" to the outside world, just internal peers. If ya gotta get somewhere else it's PRI baby!
01:48.33*** join/#asterisk bkw_ (n=brian@freeswitch/developer/bkw)
01:48.40a1fai would consantly have to adjust firewall rules
01:48.52a1fato block off these idiots
01:48.56a1fai need a sip smart firewall
01:49.16a1fa3 - 404s or 401s.. and you are out for a timeout
01:49.17bkw_I wouldn't trust a SIP ALG
01:49.29KyleKmake/build a firewall that limits access to USA or Canada
01:49.35drmessanoJust use decent usernames and strong passwords
01:49.40KyleKim sure a lot of people would find that handy
01:49.44a1faKyleK: i got people outside of USA
01:49.48KyleKoh
01:49.56jayteewhen someone actually invents a secure internet I might reconsider using SIP but until then it's a crapshoot and a bug farm.
01:50.01a1fai guess i just need to open for /16s
01:50.12a1faSIP is not that bad
01:50.18a1fajust like any other web protocol
01:50.23KyleKi kinda like insecure internet
01:50.29KyleKsecure yo shit peeps
01:50.29a1fainternet is fine
01:50.30drmessanoSeriously.. do you run any other daemons on the public internet?  That shit gets pounded like prom night
01:50.35a1faits the idiots on the internet
01:50.37drmessanoJust use common sense
01:50.41a1fayeah
01:50.47drmessanoZOMG THE SIP IS FALLING <-- bad response
01:50.49KyleKI run ssh on a port for my local computer
01:50.57a1fai run ssh, http, https, 5060...
01:51.02a1fathey all get pounded
01:51.03a1faconstantly
01:51.06KyleKreally?
01:51.10a1fayes
01:51.12drmessanoGreat, welcome to SIP being exposed
01:51.16KyleKI run just https and didn't seem to get pounded
01:51.18drmessanoNow secure it and move on
01:51.48jayteescrew common sense. 96% of all attempts to penetrate my security are from friggin China and the bulk of the remainder are from Eastern European shithole countries like Romania.
01:51.57a1fahaha
01:51.58a1fa:)
01:52.03a1fano internet laws in romaina
01:52.06a1fapound away boys
01:52.12jayteeI don't trust those commie bastards or former commie bastards
01:52.31drmessanoAs long as there's stupid people, they will always win.  101/101 is NOT a secure set of credentials
01:52.36drmessanoNor is 101/fluffy
01:52.40jayteehehehe
01:52.46*** join/#asterisk Kumbang (n=unknown@125.163.83.153)
01:52.55KyleKhmmm a bit more access than i expected
01:53.00a1faall my passwords are Fu@#$Ciked-dO2??\Up!
01:53.39drmessanoall my passwords are the names of Asterisk developers spelled backwards
01:53.43drmessanoNow come on, hack me
01:54.00a1fa:)
01:54.02a1faring ring
01:54.08a1fa1-900-BANANA-FONE
01:54.10MaliutaLapjaytee: romania only went bolshie because of the americans
01:54.47a1fadont make excuses
01:54.47a1faalthough
01:54.47MaliutaLapjaytee: and their smelly deal with stalin
01:54.47a1fathe only romaian I like is Sandra Romain
01:54.47a1fa:)
01:54.48rob0There are, for all practical purposes, no Internet laws anywhere. Even where(/if) smart laws exist, LEO's are not smart enough to enforce them. If they were, 419'ers everywhere would be stomped out quickly.
01:54.50MaliutaLapjaytee: trust the US to do a deal with someone who killed more people than hiler
01:54.55MaliutaLaphitler even
01:55.19a1facmon
01:55.30a1fano body commenting on Sandra Romain
01:55.31*** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home)
01:55.34a1fa!g Sandra Romain
01:55.34*** join/#asterisk joako_ (n=andrew@opensuse/member/joak0)
01:55.43a1fayou guys suck :)
01:55.57joako_Can someone recommend a VoIP route to Cuba as reliable as AT&T?
01:55.59drmessanoEveryone is so paranoid that their precious Asterisk/FreeSWITCH/OpenSIPSomethingoftheweek SIP daemon is being exposed to all these horrible bad guys doing naughty bad awful things.. Well, no shit.. I dont understand the freak out about it
01:56.38*** join/#asterisk Talkradio (i=talkradi@linuxgeneration.ca)
01:56.59KyleKwere americans not allowed to phone cuba up until now?
01:57.00a1fai am not freaking out
01:57.06a1falol
01:57.07KyleKkeep hearing about cuba :)
01:58.02drmessanoa1fa: So you're the one person in the 50 this week that's come in here screaming ZOMG MY LOGS ARE FULL OF HAXORS that isn't freaking out?
01:58.04MaliutaLapKyleK: I don't think americans are allowed to think about cuba as anything more than the "red threat" without being done for treason
01:58.05drmessanoNoted.
01:58.31a1fadrmessano: i just noted the increase in the last 24h
01:58.37drmessanoThe red threat?  What is this the 1960s?
01:58.37a1fathere must be some kind of exploit
01:58.50a1fa:)
01:59.03joako_KyleK: When I had AT&T/Bellsouth and their long distance service calling worked rather well. Using various VoIP providers I have tried at least 30 calls in the past 2 weeks and only 2 or 3 calls have even completed to the point of carrying on a conversation and there was massive delay. I gave up, bit the bullet and just paid $1.79/min on my mobile phone (AT&T) and the call connected with no satellite delay. Is there any way to get a decent service
01:59.14MaliutaLapdrmessano: you live in the backwards country, not me
01:59.16*** join/#asterisk JenniferAkemi (n=Jennifer@76-10-182-237.dsl.teksavvy.com)
01:59.17drmessanoa1fa: There is.. it's documented in RFC 2543
01:59.20joako_Mind you I have no issues with calls to 1st world countries on VoIP
01:59.32MaliutaLapdrmessano: I'm sure I'm on a list _as_ the red threat
01:59.43MaliutaLaphard to be a left wing activist and not be
02:00.01MaliutaLapespecially what the rest of the world considers left
02:00.14Pan3Dheh
02:00.32drmessanoMaliutaLap: No one under the age of 40 cares about Cuba
02:00.45drmessanoMaliutaLap: Its really a non-issue anymore
02:01.40MaliutaLapdrmessano: what's the average age of congress?
02:01.46Pan3Ddrmessano: indeed, to everyone *except* those with a vested interest in perpetuating the bad vibes.
02:01.58MaliutaLapdrmessano: I think it's still an issue in the halls of government
02:01.59drmessanoMaliutaLap: 94
02:02.06Pan3Dhahaah
02:02.08drmessanoMaliutaLap: Make that 93, one just died
02:02.10Pan3DSenator Byrde
02:02.28MaliutaLapSenator Bryde of Satan?
02:02.46MaliutaLapor has Cheney been shooting again?
02:03.09drmessanoBesides which, Congress could care less about Cuba too.. They're trying to stop internet pirates, clog the tubes, and make AT&T richer
02:03.39Pan3Dheh, I was flying in and out of Venezuela during the past 7 years. Nothing like being a walking tarket for the hate mongers :)
02:04.40joako_drmessano: Oh, so that explains it. VoIP crappy routes to cuba to boost people to switch back to AT&T
02:05.01drmessanojoako_: THATs the ticket
02:05.22jayteetold you it was a conspiracy! same people that framed Pete Rose!
02:05.33[TK]D-FenderUSA needs an enemy close by they can distract the population with.
02:05.54jayteeI nominate Canada! Free Maple Syrup for all!
02:06.04phunyguyhates conspiracy theories.
02:06.33jayteephunyguy, yes it's well documented in your NSA file :-)
02:06.39*** join/#asterisk maxagaz (n=maxagaz@soho2.i-xanadu.com)
02:07.13phunyguyhmm?
02:07.17phunyguyNSA?
02:07.19drmessanophunyguy: You also hate brussel sprouts
02:07.26jayteeNo Such Agency
02:07.36drmessanophunyguy: and you are a big fan of Tool
02:07.42drmessanophunyguy: Its all in your file
02:07.44phunyguyeveryone hates brussel sprouts
02:07.49phunyguyand everyone loves tool
02:07.52phunyguy:)
02:07.52drmessanoI dont
02:07.59phunyguyokokok one.
02:08.10drmessanoTool sounds like broken powertools having sex
02:08.11phunyguy(I mentioned Tool yesterday - so there)
02:08.11jayteenot Bob DeBenedictus. Bob loves the brussel sprouts
02:08.29*** join/#asterisk matsk (n=matkar@c-118ae253.174-6-64736c10.cust.bredbandsbolaget.se)
02:08.44drmessanoWant to spice up a party.. throw some Neil Diamond on
02:09.04*** part/#asterisk matsk (n=matkar@c-118ae253.174-6-64736c10.cust.bredbandsbolaget.se)
02:09.11drmessanoHot August Night <--- Nuff Said
02:09.49drmessanohttp://www.abc.net.au/reslib/200708/r170041_637422.jpg
02:11.15*** join/#asterisk Kobaz (n=kobaz@its.kobaz.net)
02:13.07Kobazi have a standard analog phone on a grandstream sip gateway, dtmf payload type is 97, and dtmf is rfc2883... i dial out on an iax connection... and i have a Read on the other end.... it never sees any digits comming in... even though I can chanspy either leg of the call, and hear the dtmf's go through
02:13.13jayteeNeil Diamond rocks!!!!!
02:13.26drmessanoNeil is the man
02:13.32phunyguyo_O
02:13.56drmessanoHot August Night - Red, Red Wine <--- Even more enuff said
02:14.09drmessanoGets no better
02:14.42[TK]D-Fenderdrmessano: that sounds like a threat ;)
02:15.11Kobazi'll paste up some debug in a sec
02:15.16drmessanolol
02:16.06[TK]D-FenderKobaz: And prove on the first leg that * gets the DTMF.
02:16.27drmessanoI still deeply resent he changed up the tempo in his live performances of Red Red Wine after UB40 went #1 with a reggae version
02:16.30Kobaz[TK]D-Fender: k
02:16.42drmessanoI had more respect than that
02:16.44drmessano:(
02:17.48*** join/#asterisk jsolis (n=jimmy@201.240.109.106)
02:17.55[TK]D-Fenderdrmessano: think about how Sugar Ray re-released "Fly" without the Rap guy from the first vid & single, and Shania Twain undid that duet she did with (other country guy here)
02:18.14Kobaz[TK]D-Fender: should the dtmf's show up in sip debug?
02:18.27Kobaz[TK]D-Fender: i've never debugged dtmf stuff before... not sure where to look
02:18.44[TK]D-FenderKobaz: the only proff that is acceptable is the hard kind.  Using * dialplan apps that depend on it.
02:19.10[TK]D-FenderKobaz: Prove it can navigate VoicemailMain, Read, etc
02:19.50drmessano[TK]D-Fender: The Shania thing was hysterical.. it was a duet on Country Radio and not on mainstream.. then to back up her hypocrisy, she released a second version of her album with all the alternate cuts
02:19.54drmessanoWTFFFF
02:20.13drmessanoCounty and Sellout version
02:20.19*** join/#asterisk thuddwhirr (n=wolthuis@mimezine.com)
02:20.52*** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun)
02:21.25Kobaz[TK]D-Fender: yeah, okay... good point.... no dtmf getting to voicemail
02:22.24[TK]D-FenderKobaz: Run along now :)
02:22.41Kobazno no but
02:22.45Kobazhow do i fix this :P
02:23.13thuddwhirranyone here want to help me troubleshoot a weird SIP issue?  :)
02:24.00jayteeis listening to Neil Diamond - Cherry, Cherry [3:13 (11%)]
02:24.35[TK]D-FenderKobaz: Show us something useful
02:25.16Kobaz[TK]D-Fender: well, i dont get anything in sip debug, no digits recieved on a Read()
02:25.25Kobazi can't show something i don't have
02:25.39[TK]D-FenderKobaz: How about CONFIGS from both sides <-
02:25.45drmessanojaytee: FTW
02:26.03jayteehehe
02:26.04drmessanoGuess now would be a bad time to mention I have a 4 CD box set of his
02:26.09drmessanoheh
02:26.27jayteeI've got Hot August Night and Hot August Night II
02:26.30[TK]D-Fenderdrmessano: Sounds like a good firing range back-stop ;)
02:26.34Kobazgateway: dtmf:rfc2833  payload type:97      asterisk: peer 5505:   DTMFmode     : rfc2833
02:26.55[TK]D-FenderKobaz: real pastebin please.
02:27.06Kobazcomming
02:27.39drmessanojaytee: I used to pretend Neil Diamond was my real Dad.. that maybe mom made a boo boo after a really exciting concert.. But NOOOOOO
02:27.45drmessanoThanks for nothing, mom
02:28.07drmessanostomps off in disgust
02:28.20Kobazhttp://pastebin.com/m5f984574
02:28.21Kobazthat's the phone
02:28.24Kobazer
02:28.28Kobazsip peer on *
02:29.05thuddwhirrive got my asterisk server hooked up to a sip trunk.  I have a DID with that sip provider pointing at my asterisk.  i can originate out fine, and i can receive incoming calls fine, but if I try and originate to that DID, asterisk immediatly issues a cancel when it sees the second invite, with no explination.
02:29.20thuddwhirranyone have any thoughts on how to debug that? or why that wouldnt be allowed?
02:29.24Kobaztrying to figure out how to dump the gateway config
02:29.49jayteedrmessano, regardless of whether Neil was your real father or not, if you weren't here in this world it would be a much darker, colder and less humorous place so thank god for whatever sperm donor did the deed!
02:30.05Kobazthuddwhirr: you'll need to elaborate... i'm not following
02:30.33Kobazthuddwhirr: are you're trying to call yourself?
02:30.39thuddwhirryes
02:30.51thuddwhirrlooped through my sip provider
02:31.04[TK]D-FenderKobaz: imagebin.ca
02:31.06Kobazare you sending the right digits
02:31.18[TK]D-FenderKobaz: but next I want a failed call with SIP DEBUG for your app test
02:31.24Kobazie: does your provider require/notrequire the areacode prefix, etc
02:32.20[TK]D-Fenderthuddwhirr: maybe you should PASTEBIN the call with SIP debug enabeld so we can see what's happening
02:32.20thuddwhirri belive so.  i send out the invite, get the right awk back, and then I see the invite for the incoming leg.
02:32.30[TK]D-Fender~pb
02:32.31infobot[~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://www.rafb.net/paste , or , http://bin.cakephp.org/
02:32.40thuddwhirrok
02:33.28*** join/#asterisk lanning (n=lanning@173.8.187.197)
02:37.53Kobazhttp://pastebin.com/m2ba5961a
02:38.00Kobazthere's the voicemail with no dtmf
02:38.09Kobazhttp://imagebin.ca/view/2qqHrHu.html
02:38.18Kobazthere's the bit of gateway config
02:41.23Kobazi guess i can play around with the other dtmf types
02:41.28Kobazin-audio actually doesn't help
02:41.45Kobazall the other phones use rfc2883 just fine (ie: polycoms and such)
02:41.56Kobazi can turn on sip info
02:43.55*** join/#asterisk chendy (n=chatzill@58.251.102.216)
02:44.13Kobaz[TK]D-Fender: okay so.... * is getting dtmf now that i turned on sip info
02:44.36Kobaz[TK]D-Fender: shitty gateway? that rfc2883 doesn't work?
02:45.19*** join/#asterisk metfan2007 (n=jc@189.146.141.79)
02:45.23[TK]D-FenderKobaz: Possibe, but I'm also concerned that it trys as "anonymous"
02:45.30[TK]D-FenderKobaz: And keeps failing multiple auths
02:46.13Kobazah, i see
02:48.19metfan2007hi all! I have a queue for incomming calls for 10 agents, using callbacklogin, the agents need to make outbound calls too, the problem is that when an agent is making an outbound call, asterisk does not know that he is busy, so It tries to send an incomming call
02:48.51metfan2007how do I tell asterisk that an agent is busy when making outbound calls withoud logoff agent?
02:49.29Kobazhttp://pastebin.com/m285b35e6
02:49.35Kobazwell i turned off the anonymous checkbox
02:49.42Kobazit's sending the real username now
02:50.07Kobazstill get an unauthorized bit on the first exchange
02:51.07[TK]D-Fendermetfan2007: There is no association between chan_agent using a local channel & Exten and a particular SIP device plaing a call
02:51.36[TK]D-Fendermetfan2007: One uses dialplan, the other is a hardware device (Sorry, could be other than SIP, but hopefully you get the point)
02:51.46[TK]D-Fendermetfan2007: there is no way for * to know.
02:52.02[TK]D-FenderKobaz: And still bad DTMF
02:52.05[TK]D-Fender?
02:52.17[TK]D-FenderKobaz: Checked your firmware?  newer or better out?
02:54.07Kobazpossibly
02:57.12jaytee[TK]D-Fender, have you seen this from Digg's front page tonight? http://www.acesandeighths.com/8ball_6.html
02:57.40jayteeI can think of at least 3 people they should have listed that aren't there
02:58.22[TK]D-Fenderjaytee: Yup, finished them all about 15 mins ago
02:58.33jayteehehe, figures :-)
02:58.40[TK]D-Fenderjaytee: Antoine Dufour, Eric Johnson, plewnty more
02:59.15jayteeyeah, Antoine for sure. Tuck Andress
02:59.53drmessanoSlash
02:59.59drmessanoJust sayin
03:00.04jayteeyou're the one that turned me on to Antoine
03:01.36Kobazthere's new fiemware... lets hope it doesn't break anything
03:02.08Kobazfirmware
03:06.28Kobaz[TK]D-Fender: nope... no rfc2883 dtmf after reboot... sip info dtmf still works
03:08.02jaytee[TK]D-Fender, hey in the youtube vid of Spiritual Groove that's a solo of Antoine, not the one with Tommy Gauthier, is he playing a cutaway classical or a cutaway steel string? The neck looks like a classical but the strings look like standard bronze and the treble sounds like steel not nylon.
03:09.55*** part/#asterisk jsolis (n=jimmy@201.240.109.106)
03:15.12MaliutaLapYay! off to meet Sexy Geek Girl  ... don't wait up for me :P
03:18.49joakoOMFG you can download Linksys SPC tool from cisco.com now without any bullshit
03:20.04drmessanoYes
03:20.15drmessanoThats pretty badass
03:21.24jayteeSPC tool? whazzat?
03:21.31*** join/#asterisk Kumbang (n=unknown@125.163.83.153)
03:21.49joakojaytee: the util to generate config files for Linksys SPA-phones and ATAs
03:22.03drmessanoSkeleton configs, basically
03:22.08drmessanoLets you encrypt them as well
03:22.11jayteeah!
03:22.35joakoTime to take down http://spc.pifiu.com I suppose... when I first started the site the logs showed plenty of visitors from .cisco.com lol
03:23.01drmessanoThats your site?
03:23.14joakodrmessano: Yes
03:24.03drmessanoRight on!  I got my first SPC stuff from there
03:26.21jayteenite all, time for bed
03:27.30*** join/#asterisk propellerhead (n=yogurt2u@host1.190-30-31.telecom.net.ar)
03:32.34*** join/#asterisk Failrar (n=Failrar@coffee.ipv6.kaufmann.tc)
03:34.44*** join/#asterisk Braxus (n=braxus@netblock-68-183-230-56.dslextreme.com)
03:42.46*** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net)
03:49.10*** join/#asterisk blkry (n=blkry@97.95.233.232)
04:08.36*** join/#asterisk CunningPike (n=CunningP@S01060014bf81366b.vc.shawcable.net)
04:10.50VaGoNeTaSshit, cant make cellphone calls
04:10.54VaGoNeTaSchanunavail
04:11.01VaGoNeTaSand is not being blocked by the telco
04:11.14VaGoNeTaSwhat could it be
04:12.11[TK]D-FenderVaGoNeTaS: Absolutely anything
04:12.21VaGoNeTaSwhat u mean
04:12.24[TK]D-Fenderperhaps nothing....
04:12.29[TK]D-FenderSomwhre in the middle?
04:12.29VaGoNeTaSi'm in a different place
04:12.31VaGoNeTaSwith 2 e1 lines
04:12.33[TK]D-FenderNo.. clearly to the left
04:12.47VaGoNeTaSi can receive but not make
04:13.11VaGoNeTaSim gonna pb the error so u can see
04:13.44VaGoNeTaS#
04:14.13VaGoNeTaSpastebin.ca/1422337
04:14.30VaGoNeTaSi cant make this shit working properly
04:15.03[TK]D-Fender...........
04:15.07[TK]D-Fenderwhy do I bother...
04:15.18[TK]D-FenderVaGoNeTaS: You don't even show the friggen DIAL
04:15.21[TK]D-Fenderand NO DEBUG in there
04:16.00[TK]D-FenderVaGoNeTaS: there is nothing usable in that PB
04:16.08VaGoNeTaSi did
04:16.12VaGoNeTaSpri debug span 1
04:16.18VaGoNeTaSso im gonna show you the whole message ok
04:16.24[TK]D-Fender<PROTECTED>
04:16.29[TK]D-FenderVaGoNeTaS: We don't see the DIAL!
04:16.37VaGoNeTaSwhat u want then
04:16.43[TK]D-FenderVaGoNeTaS: How the hell do we know what you're doing there?
04:16.44VaGoNeTaSwhat u want me to pb to
04:16.49VaGoNeTaShahahaha
04:16.52VaGoNeTaSim trying to reach a cellphone
04:16.57[TK]D-FenderVaGoNeTaS: show the ENTIRE &^#$ing call
04:17.01VaGoNeTaSk
04:17.05VaGoNeTaSwait a sec
04:17.06[TK]D-FenderVaGoNeTaS: Ther is no DIAL in your pastebin.
04:20.35VaGoNeTaSim dialing with a ftphone
04:20.45VaGoNeTaSim pretty much sure that the problem is the fucking dahdi
04:20.56VaGoNeTaSit doesnt well support the fucking Redfone quad box
04:20.59VaGoNeTaSpiece of shit
04:21.11VaGoNeTaSshitty 3G box
04:22.43*** join/#asterisk bbryant1 (n=brett@c-68-59-20-153.hsd1.sc.comcast.net)
04:24.05[TK]D-FenderVaGoNeTaS: It is a device I would never recommend
04:24.08[TK]D-FenderVaGoNeTaS: http://support.red-fone.com/index.php?_m=knowledgebase&_a=viewarticle&kbarticleid=20
04:24.20*** join/#asterisk justdave (n=dave@unaffiliated/justdave)
04:24.33VaGoNeTaSso you with me
04:24.38VaGoNeTaSits a fucking piece of shit
04:24.50VaGoNeTaSan expensive piece of shit
04:29.33[TK]D-FenderVaGoNeTaS: Call them for support
04:30.05VaGoNeTaSdude
04:30.14VaGoNeTaSif im gonna call redfone is just for one reason
04:30.18VaGoNeTaSisulting them
04:30.23VaGoNeTaSxD
04:30.45VaGoNeTaSi really prefer digium card
04:31.02VaGoNeTaST110 maybe would've been better
04:33.33*** join/#asterisk werdan7 (n=w7@freenode/staff/wikimedia.werdan7)
04:39.38*** join/#asterisk GeekBoy (n=kvirc@200.94.225.248)
04:40.25*** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home)
04:42.53GeekBoywhen doing "make all" for DAHDI,  It is returning, " You do not appear to have the sources for the 2.6.18-92.1.18.el5.028stab060.8 kernel installed."
04:47.29*** join/#asterisk [T]ank (n=ckwall@c-71-199-25-239.hsd1.ut.comcast.net)
04:49.24[T]ankgetting an error on echo cancelation after upgrade to 1.6. Here are the details, can anyone help? http://pastebin.ca/1422357
04:51.38[TK]D-Fender[T]ank: your system.conf did not set the ec to use for that channel
04:51.43[TK]D-Fender[T]ank: go read the samples
04:59.54*** join/#asterisk Gopaul (n=Miranda@61.17.185.118)
05:00.49[T]ankHow do i know what echo canceller I have? mg2, kb1, sec2, and sec.?
05:06.43*** join/#asterisk Defraz (n=tim@24-117-236-174.cpe.cableone.net)
05:14.04*** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun)
05:15.36[T]ank[TK]D-Fender: reading about ec says that the dahdi-linux compiles mg2, kb1, sec2, and sec cancellation types. I just need to specify which one to use. I also see in the samples echocancel=yes. I have tried all of the above on chan 1 and then restart asterisk. am I even close to where I should be? I am getting the same results with everything I try
05:16.06[TK]D-Fender[T]ank: You set the EC i your system.conf PER CHANNEL.. go read the samples a few more times.
05:16.22[TK]D-Fender[T]ank: * compiles them ALL, and loads them dynamically as assigned
05:16.31[TK]D-Fender[T]ank: No assign = none used.
05:16.40[T]ankright... got that
05:17.02[TK]D-Fendercheckout time, later all
05:17.05[T]ankits a fxoks port so my understanding is that there is only 1 chan. am i correct in that?
05:17.14[TK]D-Fender[T]ank: yes
05:18.24[T]ankso I have tried echocanceller=<whatever>,1 with each of the echocancel types tried in the whatever field.
05:18.40[T]ankis that not correct?
05:19.03[T]ankeach type gives me the same result of chan_dahdi.c:2010 dahdi_enable_ec: Unable to enable echo cancellation on channel 1 (No such device)
05:20.49*** join/#asterisk ectospasm (i=ectospas@i.hate.your.vhosts.shellium.org)
05:20.55[T]ankso if mg2, kb1, sec2, and sec are all installed, then any one of them should have worked, right?
05:31.05*** join/#asterisk bijit (n=benji@200.122.188.156)
05:31.09[T]ankanyone else here know anything about echo cancellation in /etc/dahdi/system.conf
05:31.20Qwell[T]ank: you are correct
05:31.57[T]ankQwell: I am not sure I am doing any of this right...
05:32.09QwellWhich version of 1.6 are you using?
05:32.14[T]ankIs there any chance I do not have echo cancellation installed?
05:32.31[T]ank1.6.0.9
05:32.46Qwellare your channels working otherwise?
05:33.06[T]ankyep
05:33.16[T]ankjust getting echo on the outside callers side of the call
05:33.25[T]ankthey hear them selves echo when they speak.
05:33.29Qwellwhere are you putting the echocanceller line?
05:33.36[T]ankthe asterisk side of the call sounds just fine
05:33.47[T]anklet me repaste the config...
05:33.53Qwellecho is handled on the far end.  if they hear echo, they need to fix it
05:34.20[T]ankhttp://pastebin.ca/1422375
05:35.06Qwell2.1.0.4?
05:35.38[T]ankits every single person that calls that gets the echo
05:35.42Qwellerr, 2.1.0.2 for tools
05:35.52[T]ankoh.. let me check the tools version.
05:36.09[T]ankyes, 2.1.0.2
05:36.32[T]ankI also get the error i mentioned above in the cli regarding echo
05:36.46[T]ank[May 13 23:20:27] WARNING[12419]: chan_dahdi.c:2010 dahdi_enable_ec: Unable to enable echo cancellation on channel 1 (No such device)
05:37.23[T]ankin case it is relevant: Wildcard TDM400P REV I Board 5
05:37.31Qwellfrom what I can tell, your config is right..
05:38.09Qwelldoes `modinfo dahdi_echocan_mg2` show anything?
05:38.51[T]ankat the linux command line, right?
05:38.57Qwellyeah
05:39.22[T]ankdrops the cursor to the next line like something should happen. Never completes.
05:39.31[T]ankwait... just copleted.
05:39.33[T]ankpasting info
05:39.55[T]ankhttp://pastebin.ca/1422379
05:40.41Qwellit would yell at you if it didn't exist
05:41.09Qwellyeah...
05:42.28QwellDo you have channels=1 anywhere?
05:42.59Qwelland I assume you've run ztcfg?  (I don't know if that's needed..)
05:44.02[T]ankyou mean the line that would read channel => 1?
05:44.11[T]ankthat is in my /etc/asterisk/chan_dahdi.conf
05:44.13Qwellno
05:44.23Qwelldunno, the sample system.conf has channels=X
05:44.36[T]anklol... dahdi_cfg -vv
05:44.43[T]ankChannel 01: FXO Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 01)
05:44.45[T]anktesting
05:45.04[T]ankduh
05:45.06[T]ankthat worked
05:45.10Qwell:D
05:45.12[T]anki have been restarting asterisk
05:45.18[T]ankforgot to run the dahdi_cfg
05:45.45[T]ankif the school of hard knocks gave deplomas, I would have a doctorate.
05:46.29*** join/#asterisk j_kroon (n=jkroon@dsl-240-132-169.telkomadsl.co.za)
05:47.32Qwellif you were a wireless AP, what would your IP be?
05:47.34[T]ankecho is gone also
05:47.42Qwellwoot
05:48.00*** join/#asterisk HeMan (n=jimmy@ssh.southpole.se)
05:48.16[T]ankQwell: thank you, good night
05:48.16Qwellwonders if his AP even has an accessible IP...
05:50.40Qwellah hah!  found it.
05:50.54*** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-c0f5ba2e2485ca53)
05:51.15HeManHi! Where can I find information about the current revission "system" with 1.6.0.9, 1.6.1.0, 1.6.2.0-beta and 1.4?
05:57.37HeManis 1.6.0.9 considered stable?
06:02.03Qwellhmm
06:03.18Qwellandroidvnc is...rather awesome.
06:04.42[T]anklearns new trick from Qwell
06:04.45[T]ank:-D
06:04.50[T]anknow I can die happy
06:07.27*** join/#asterisk Winkie (n=urmom@ur.fa.gs)
06:15.37*** join/#asterisk gego (n=rick@b238085.customer.hansenet.de)
06:22.18*** join/#asterisk Kumbang (n=unknown@125.163.83.153)
06:26.53*** join/#asterisk grEvenX (n=even@apb9hb.ip.ssc.net)
06:29.20*** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1)
06:30.17*** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1)
06:34.00*** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk)
06:39.25*** join/#asterisk xrmx__ (n=rm@host119-200-dynamic.180-80-r.retail.telecomitalia.it)
06:55.41*** join/#asterisk botox93 (n=botox93@213.221.82.242)
06:59.33*** join/#asterisk miloux (n=KVIrc@milu.rit.se)
07:00.58*** join/#asterisk ck_28 (n=CK@212.98.141.199)
07:02.40ck_28good morning all
07:02.51ck_28i followed up digium free fax instalion
07:03.05ck_28i am trying to send a file to fax
07:06.34ck_28kindly find my config and debug at http://pastebin.com/d60a8686a
07:08.25ck_28any one can help ?
07:22.00ck_28Dear Admins any help
07:31.49*** join/#asterisk Subdolus (n=subby@subby.afraid.org)
07:32.45*** join/#asterisk sulex (n=sulex@pdpc/supporter/professional/sulex)
07:37.52*** join/#asterisk Corydon76-dig (i=twelve@pdpc/supporter/bronze/Corydon76-home)
07:37.52*** mode/#asterisk [+o Corydon76-dig] by ChanServ
07:51.32ck_28@russellb can you help me
07:52.21tzafrir_laptopck_28, don't ask specific people to help you
07:52.39tzafrir_laptopthe guy you asked has other things on his mind, as he hinted you.
07:52.59tzafrir_laptopNot to mention that it's in the middle of the night at where he is right now
07:54.07ck_28:)
07:54.42*** join/#asterisk Perun (n=perun@2001:6f8:1316:1234:216:3eff:fe07:3160)
07:54.44Perunhi all
07:54.56*** join/#asterisk war9407 (i=war@liquidswords.org)
07:55.13PerunI want to use usb hfc isdn cards for my asterisk... what should I use, zaptel module or misdn?
07:56.31tzafrir_laptopwhat version of asterisk do you use?
08:03.36tzafrir_laptopprefers zaptel/dahdi , but is biased
08:08.05ck_28tzafrir_laptop asterisk version question is for me ?
08:08.27tzafrir_laptopno. For Perun
08:08.38Peruntzafrir_laptop: :1.4.21.2~dfsg-3
08:08.45Perundebian pkg
08:08.45tzafrir_laptopI can't support non-free software. Others may be able to
08:09.31tzafrir_laptopPerun, in that case the answer is simple: mISDN (1.x) has been vetoed out of Unstable Debian by its maintained (Simon Richter)
08:09.43tzafrir_laptopzaphfc is included in the Debian zaptel package
08:09.50tzafrir_laptopm-a a-i zaptel
08:10.33Perunso, I should use zaptel?
08:10.50tzafrir_laptopI would say that it's the path of least resistance
08:11.07tzafrir_laptopIf you want to use mISDN you need to invest much more work
08:13.10Perunbut what is better for the future? I have seen misdn is the new isdn stack in vanilla kernel?
08:14.08ck_28Channel 'SIP/add-084f3c00' fax session '25' is complete, result: 'FAILED' (FAX_NO_FAX), error: 'T1_TIMEOUT', pages: 0, resolution: 'unknown', transfer rate: '2400', remoteSID: ''
08:14.21ck_28what is the resolution
08:16.59*** join/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/zeeek)
08:18.04ck_28[44423.655] T38(0/0/0) Tx CC_FAX_IND
08:18.05ck_28675     <Call   100>    : FaxIndicated from(64)
08:18.05ck_28676     <SIP    0>      : No authentication information available
08:18.05ck_28677     <SIP    100>    : Send INVITE Request
08:18.05ck_28678     <SIP    100>    : Receive 100 Trying
08:18.05ck_28679     <SIP    100>    : Transaction (72 INVITE) proceeding
08:18.07ck_28[44424.695] T38(0/0/0) discard T30D from VoPP by no sess
08:20.45tzafrir_laptopPerun, mISDN2 is not mISDN
08:21.03tzafrir_laptopthe one in the kernel is mISDN2
08:21.40tzafrir_laptopAdding chan_lcr support to the Debian package is work in progress, and will likly happen in the squeeze cycle
08:23.37ck_28tzafrir_laptop in which  i can post my question ?
08:23.44ck_28channnel*
08:31.00Peruntzafrir_laptop: aa and can I use misdn2 with asterisk and hfc usb cards? (want the NT mode to)
08:31.22tzafrir_laptophfc-usb is indeed not supported by zaptel
08:31.49tzafrir_laptopI suggest you ask on #debian-voip in irc.oftc.net
08:32.56*** join/#asterisk qdk (n=qdk@81.7.168.130)
08:35.18KyleKhmm can germany dial a north american toll free number? i know the caller would pay for it, but can a person over there call it?
08:35.46ZeeekKyleK: I know we can in France
08:36.11Zeeeka message is read saying "you are calling a tollfree USA number, the bad news is it ain't free for you.
08:36.30KyleKk
08:36.34Zeeekso it says if you stay on the line, you will be charged
08:37.33KyleKah, I might get a random USA did as well for that then, setting up like 1800grandma kinda thing
08:37.40*** join/#asterisk kamanashisroy (n=kamanash@119.30.36.19)
08:48.26*** join/#asterisk QaDeS (n=mklaus@drms-590d5bc3.pool.einsundeins.de)
08:50.03*** join/#asterisk ck_28 (n=CK@212.98.141.199)
08:55.37*** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-6a95046b5b831224)
09:05.56*** join/#asterisk kaptengu (n=kaptengu@unaffiliated/kaptengu)
09:16.36*** join/#asterisk Get_The_Fish (n=IceChat7@173-14-4-113-Colorado.hfc.comcastbusiness.net)
09:17.11Get_The_Fishhello all... can anyone point me to a number that will work as a DTMF tester, like an IVR that will read back the digits that you entered?
09:17.34Get_The_FishI have users that complain about DTMF not working intermittently
09:18.51ck_28any one can help me to solve the fax t38 capability --T38(0/0/0) discard T30D from VoPP by no previous V21_Flag
09:20.12ZeeekGet_The_Fish:  someone published a dialplan that does that. Possible voicepulse connect or onsip.com or teliax ?
09:21.01*** join/#asterisk Ast001 (n=uros@cable-89-216-155-28.dynamic.sbb.rs)
09:21.26Get_The_Fishwell, I was really looking for a toll free or other number that does that- in other words, I use the provider that I have to connect to the pstn.
09:21.32Ast001Hi is there any method for turning off remote unix connection in asterisk cli ?
09:22.02tzafrir_laptopGet_The_Fish, exten => _X.,1,SayDigits(${EXTEN}); or some variation
09:22.06Get_The_FishI have found some that will let me register with them and test, but that doesnt test my configuration with my provider.  Am I making sense?
09:22.44Ast001I just don't want to see remote unix connection every time myscript connects with asterisk with asterisk -rx
09:23.18Get_The_FishI have that, and it's helpful, but I want to know what the "other end" of the call is receiving.  Because asterisk is saying everything is peachy, but my users are saying otherwise.  Suspecting my provider.
09:23.54tzafrir_laptopAst001, this message is a verbose message at verbosity level 3
09:24.12Ast001so I need higher level ?
09:24.17Get_The_Fishlower level
09:24.22Ast001or lower ? ok thanks
09:24.23tzafrir_laptopYou asked Asterisk to give you plenty of noisy messages. Why do you complain when it does so?
09:24.32Get_The_Fishtry "core set verbose 2"
09:24.34Get_The_Fishor lower
09:24.41tzafrir_laptopcore set verbose 0
09:24.43Ast001ok thank you.
09:24.58tzafrir_laptopUnless you actually want to troubleshoot / trace something
09:25.31Ast001it works thank you tzafrir
09:25.40Get_The_Fishtroubleshooting is overrated.  Just keep restarting until it works or windows update tells you they have a a fix.... oh wait...
09:26.45tzafrir_laptopsvn update
09:26.48tzafrir_laptopmake install
09:26.52Get_The_Fishlol
09:40.34*** join/#asterisk troubled (n=troubled@unaffiliated/troubled)
09:42.37*** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw)
09:43.50ck_28tzafrir_laptop you are the only active person in this channel :)
09:43.59ck_28tzafrir_laptop can you guide me or give me a key
09:44.40tzafrir_laptopI have already told you that I don't know this digium fax thing
09:44.54ck_28thanks
09:45.17ck_28what is the gigium channel name ?
09:51.48*** join/#asterisk jetlagmk2 (i=jetlag@pool-70-106-82-2.hag.east.verizon.net)
09:53.57*** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net)
10:17.39*** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano)
10:36.39Get_The_Fishtzafrir, you still up and around in here?  Any ideas on why a SIP trunk's register packet would have the receiving host's IP in the from: header coming from asterisk?
10:44.04*** join/#asterisk bmg505 (n=leon@196-209-78-114-rndf-esr-5.dynamic.isadsl.co.za)
10:46.39Get_The_Fishdoesnt look like the fromdomain setting in the peer details is working properly, the from: field in the SIP message header is showing the address of the host that is receiving the invite.  Any help on this?
10:54.23kaldemarGet_The_Fish: is it a register or an invite or a reply to one of those? show something concrete.
10:55.01Get_The_Fishregister
10:55.09Get_The_Fishwhat do you want to see
10:56.22Get_The_FishFrom:  <sip:john@68.164.111.XX>;tag=as3fcc5278
10:56.34Get_The_FishTo:  <sip:john@68.164.111.XX>
10:57.16kaldemarthe whole message dialog.
10:57.36Get_The_Fishone sec, lemme pastebin it
10:58.13kaldemarand relevant sip configurations from both ends, secrets masked.
10:58.22*** join/#asterisk ceeriael (n=ceeriael@93.167.108.90)
10:59.16Get_The_Fishk one sec
11:00.13milouxif i do sip show channels, i have one thats: x.x.x.x   3172        3c26701a8220-q7  0x0 (nothing)    No       Tx: NOTIFY - And that sip dev. is not recieving any calls. How do i clear it? its not listed under soft hangup
11:00.36ck_28hi any one can help me to solve the fax t38 capability --T38(0/0/0) discard T30D from VoPP by no previous V21_Flag
11:03.28Get_The_Fishkaldemar: http://www.pastebin.ca/1422536
11:03.52Get_The_Fishceeriael is in here, we've been working together on this.
11:04.00ceeriael=)
11:04.43ceeriaelit seems the nomater what is put in "fromdomain" it uses the target IP in the FROM field.
11:05.12Get_The_FishI found some guy that submitted a bug to asterisk on this, then said that he solved it without mentioning how.  This is a configuration issue somewhere, just not sure where
11:06.08*** join/#asterisk ThoMe (i=tm@tm.muc.de)
11:06.10ThoMehello.
11:06.16ThoMehow i can use/install the codec 729a?
11:06.28kaldemarnothing wrong with that single message. do you have a problem of some kind?
11:06.34Get_The_FishThoMe, you need to purchase it from digium.
11:06.41ThoMeGet_The_Fish: oh ok
11:06.49ThoMeGet_The_Fish: and a alternate?
11:06.58kaldemarand peer definitions don't affect register statements you have in the sending end.
11:07.19Get_The_FishThoMe: http://store.digium.com/productview.php?category_id=5&product_code=8G729CODEC
11:07.33Get_The_FishThoMe: no alternates will be discussed by me.
11:08.00Get_The_Fishkaldemar, ahhhhh good to know
11:08.11ck_28no one have an idea using digium fax?
11:08.35Get_The_Fishso kaldemar, no matter what you have in the peer details for that host, it doesnt really matter in a registration attempt, is that correct?
11:09.21kaldemarin the sending end it doesn't. register messages are build based on register statements only (register => ...).
11:09.29*** join/#asterisk saftsack (n=saftsack@p5792458A.dip.t-dialin.net)
11:09.42Get_The_Fishok, well, that explains quite a bit then.
11:09.50kaldemar-build
11:10.04Get_The_Fish1.4.22
11:10.34kaldemari meant i had an extra word in my sentence. :) so do you have a real issue with that?
11:11.40*** join/#asterisk sergee (n=serg@voip1.west-call.com)
11:12.03Get_The_Fishoh, gotcha.  Well, I think that the register statement isnt configured correctly is what it is.  Now that I understand the way that the registration packet is done it makes more sense, will look at the registration string.
11:16.25Get_The_Fishkaldemar, from my reading, the register string is correct here.... john:password@68.164.111.XX/john
11:16.50Get_The_FishI cant figure out why it would use 68.164.111.xx as the from: field as well.
11:21.49*** join/#asterisk Phurl (n=mdupont@82.114.88.30)
11:22.07kaldemarGet_The_Fish: see rfc 3261 section 10.2
11:22.16*** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at)
11:22.20*** join/#asterisk timeshell__ (n=chatzill@206.248.136.108)
11:23.03Phurlhi all, i am an experienced hacker learning asterisk and vicidial for the first time. Check out my work twitter http://twitter.com/VcDlAstrsk . Currently looking for some troubleshooting help on the vicidial
11:23.37phunyguy1337 h@x0r!
11:23.39kaldemarGet_The_Fish: it is not a misbehavior.
11:24.56Phurlphr3ak, w0r7
11:25.34phunyguy3xcus3 m3?  w@t j00 ca11 m3?
11:26.22Phurlhahah
11:26.57*** join/#asterisk DarkRift (n=dark@65.92.171.162)
11:27.29Get_The_Fishdamn, you are correct....thanks kaldemar, I missed that....
11:28.35kaldemarno problem.
11:47.03*** join/#asterisk Great_Anta_Baka (n=tensai@196.33.159.83)
11:56.48*** join/#asterisk ingenius (n=alektro@netsolution.com.ar)
12:05.23*** join/#asterisk bmg505 (n=leon@196-209-78-114-rndf-esr-5.dynamic.isadsl.co.za)
12:06.51*** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com)
12:08.16*** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net)
12:10.44ck_28<PROTECTED>
12:11.08*** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net)
12:21.42Great_Anta_Bakahow do you check how long a call has been going on for from the * CLI?
12:22.43*** join/#asterisk stope (n=nobody@chelmsford-cable-69-60-242-213.unitz.ca)
12:24.55*** join/#asterisk esaym (n=user@cpe-24-174-186-34.satx.res.rr.com)
12:25.34ck_28any one tried to install opt
12:27.22tzafrir_laptopck_28, what is 'opt'?
12:28.03tzafrir_laptopGreat_Anta_Baka, for starters 'channel show NAME_OF_CHANNEL'
12:28.18Great_Anta_Bakaah ty
12:28.20tzafrir_laptopwhich would be the specific channel and not the call, but would be good enough, I guess
12:28.33*** join/#asterisk [netman] (n=netman@175.Red-79-145-182.dynamicIP.rima-tde.net)
12:28.36*** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163)
12:32.06*** join/#asterisk ariel_ (i=3fd6eca9@gateway/web/ajax/mibbit.com/x-5346077f9fd4ff6f)
12:32.52ck_28sorry opal
12:33.12ck_28tzafrir_laptop http://www.voip-info.org/wiki/view/T38modem+configuration+with+Asterisk
12:33.31ck_28requested to enable t38 modem
12:33.43tzafrir_laptopck_28, let's start with the simple things: what version of Asterisk do you use?
12:34.04ck_281.6.0.1
12:34.08ck_281.6.1.0
12:35.49ck_28tzafrir_laptop i installed 1.6.0.9 first  then update it to 1.6.1.0
12:38.59*** join/#asterisk tobias (n=tobias@24.225.71.33)
12:39.04*** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk)
12:39.46*** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk)
12:40.14*** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk)
12:45.51*** join/#asterisk thomasrr (n=scroogey@195-240-213-212.ip.telfort.nl)
12:45.53thomasrrhello
12:48.55thomasrrdoes anyone here have experience with voipbuaster voip-in numbers?
12:51.38ck_28tzafrir_laptop should i continue or you are busy ?
12:56.19*** join/#asterisk martyn-job (i=be18869a@gateway/web/ajax/mibbit.com/x-4504d6a8c96f778c)
12:56.25martyn-jobHi
12:56.25martyn-jobevery1 :D
12:57.49martyn-jobI need to know if you know some application or way to do an IVR with connection to authorizing like ( VIsta, Master Card) to pay some products or service that my IVR companny want to sale across my IVR ..
12:57.53martyn-jobWhat do you know about it ?
12:58.20thomasrrif you are getting a "number is not in service" that's a problem on voipbuster side right (got the number frmo them [voip-in])
12:58.46thomasrrits not lke a config issue on my side in asterisk when i am calling from my mobile phone
12:58.51*** part/#asterisk gego (n=rick@b238085.customer.hansenet.de)
12:59.00*** join/#asterisk Superbartt (n=bart@ipd50a21c9.speed.planet.nl)
12:59.33PhurlUnable to request channel SIP/1234 what to look for? please one tip.
12:59.51thomasrrare you sure the extension exists?
12:59.54thomasrrin extensions.* ?
13:01.54SuPrSluGPhurl: what's the output of sip show peers @ the cli?
13:02.15Phurl1003/1003                  192.168.111.67   D          5061     OK (3 ms)
13:02.19Phurli can call them,?
13:02.34PhurlSubdolus, so the peer has to be seen?
13:03.07SuPrSluGso you don't have and extension made for 1234. you need to make one
13:03.07SuPrSluGs/and/an
13:04.09Phurlthats it, great. that is what i wanted to know.
13:04.13Phurlthanks
13:04.29thomasrrPhurl: do you know something about voupbuster voip-in numbers?
13:04.49*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)
13:04.54*** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net)
13:05.11Phurlthomasrr, no, but I can ask someone. what do you what to know? I am in kosovo atm
13:05.42thomasrrif the message "This number is not in service" is a problem on voipbuster end
13:05.43thomasrror mine :)
13:05.55Phurlnopw
13:05.55thomasrri think voipbuster because nothing in the log on my side
13:06.03Phurlcannot help with that. sorry dude.
13:06.14Phurli assumed you googled?
13:06.19thomasrryes
13:06.45*** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net)
13:08.07Get_The_Fishthomasrr, make sure that the sip invite is reaching your * box.  go to the asterisk CLI and type "sip set debug"
13:08.16Get_The_Fishsee if you see an invite from them
13:08.20*** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net)
13:10.59thomasrrcool our sat is getting launched now :)
13:11.03thomasrrgo europe!
13:12.22thomasrrno such command 'sip set debug'
13:12.58plundraWhen you attach to the asterisk cli, can you enable the pretty colors you get while running asterisk in the forground?
13:13.47*** join/#asterisk Gremlin (i=826f3cf2@gateway/web/ajax/mibbit.com/x-05ff357c04f676f2)
13:13.47plundraThe opposite of -n would be the obvious option but I don't see anything like that.
13:14.17Get_The_Fishthomasrr, you are in the asterisk cli are you not?
13:14.20PhurlSubdolus, Unable to request channel SIP/cc10 /chan_sip.c: No such host: cc101
13:14.22[TK]D-Fenderthomasrr: "sip set debug on"
13:14.30Phurlso i need to setup the extension cc101
13:14.37Phurlthat would be in extensions.conf?
13:15.01Get_The_Fishis that a 1.6 thing? sip set debug worked for me in 1.4.23.1
13:15.09*** join/#asterisk HenrikBe (n=zapphir@h204n4fls32o954.telia.com)
13:15.29ck_28any one can help me in my asterisk fax t38 compatability
13:15.49GremlinDoes Asterisk support DNIS reliably?
13:15.56plundraGet_The_Fish: I believe it warned about it being deprecated in 1.4, did it not?
13:16.01thomasrrnow i am getting a different error hehe
13:16.24HenrikBeis there any tutorial or examples on using ajam (ajax asterisk manager) except from the short text on voip-info?
13:16.47Get_The_Fishit didnt for me
13:16.59Get_The_FishGremlin- yes it does
13:17.48Get_The_FishHenrikBe, yeah there is.... google around for it, I remember seeing a several pages on it
13:17.55[TK]D-FenderPhurl: Stop with these little 1-line wonders and PASTEBIN the entire failed call.
13:17.57[TK]D-Fender~pb
13:17.57infobot[~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://www.rafb.net/paste , or , http://bin.cakephp.org/
13:17.58[TK]D-Fender^^^^
13:18.17Phurl[TK]D-Fender, thanks i know pb np
13:18.50Get_The_Fishhenrikbe: http://www.the-asterisk-book.com/unstable/manager-interface-ajam.html
13:19.27thomasrrhmm no notify
13:19.27HenrikBegtf: thanks!
13:19.38Get_The_Fishnp
13:19.41thomasrrmaybe its the bloody hipath pbx who steals it :)
13:20.23GremlinIs it possible to get Asterisk to "patch" the call through to a certain line/whatever based on the number dialed?
13:20.31Get_The_Fishhenrikbe, here's the doxygen from source on it: http://www.asterisk.org/doxygen/1.4/AstHTTP.html
13:21.04[TK]D-FenderGremlin: You can do whatever the hell you want.  Its your dialplan <-
13:21.15Get_The_FishGremlin, yes, it is- Freepbx will allow you to do this from the gui.  No idea on the dialplan code, but you can use theirs as an example.
13:21.56Phurl[TK]D-Fender http://www.pastebin.ca/1422632
13:22.03GremlinOkay.
13:23.54stopeI have an SLA problem, http://pastebin.ca/1422633   the phones both ring and the line lights light up and flash but the phones don't actually ring
13:23.59stopeam I missing something?
13:24.14stopethe caller can hear the ringing...
13:24.31GremlinAny hope of running Asterisk on a 486 or Pentium 75 MHz, or should I look at something decent like Pentium 166 or higher?
13:25.14GremlinI'll have up to ten simultaneous "lines" coming in from SIP with a fairly CPU intensive codec.
13:25.25stopeget a more powerful machine
13:25.26*** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee)
13:25.29[TK]D-FenderPhurl: I take it you're trying to call out via AMI Originate, correct?
13:25.43[TK]D-FenderGremlin: NO CHANCE
13:25.47Get_The_FishGremlin, no
13:25.54Phurli think so, I am using the vicidialer, let me see the command to call.
13:25.55GremlinCeleron 1.3GHz?
13:26.09[TK]D-FenderGremlin: Your details are still vague
13:26.18Get_The_Fishdamn man, you have a 75 MHz pentium that still runs?  That thing should be in a museum :)
13:26.21GremlinYeah, they are.
13:26.35Get_The_Fishceleron 1.3 should work.
13:26.38[TK]D-FenderPhurl: It is trying to call a DEVICE, not an "extension".
13:26.42GremlinOkay, cool.
13:26.58Phurlgood, so the call command is wrong.
13:27.06Phurllet me see what it is sending
13:27.07GremlinNow I just need 10 USB to RJ11 ATAs all running off of two USB ports.
13:27.11*** join/#asterisk Psychobilly (n=moi@adsl72-48.kln.forthnet.gr)
13:27.12[TK]D-FenderPhurl: so taht would be "SIP/1003", etc
13:27.48Psychobillyhello, how do i convert this diaplan line in ael 2 syntax:   exten => 402,hint,SIP/402
13:30.22*** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman)
13:32.05*** join/#asterisk nauticalthinker (n=mratliff@c-76-122-200-95.hsd1.tn.comcast.net)
13:32.45nauticalanyone have much experience with doing an Active Directory integration for unified communications?
13:33.21*** join/#asterisk axisys (n=axisys@155.70.141.45)
13:33.43*** join/#asterisk chandoo (n=chandoo@ool-4353b978.dyn.optonline.net)
13:35.01[TK]D-Fendernautical: I'm sorry, could you be a little more vague please...
13:35.49Get_The_Fishnautical, yes.  Be prepared to fight with Microsucks kludge of a LDAP implementation
13:36.23Get_The_Fishmight want to try likewise as a client for AD, and go from there.
13:36.44Psychobillyany hints about hint syntax in ael? :>
13:37.34[TK]D-FenderPsychobilly: http://www.voip-info.org/wiki/view/Asterisk+AEL2
13:37.47*** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm)
13:38.33SuPrSluG<PROTECTED>
13:39.06Phurl[TK]D-Fender, thanks it looks like a configuration problem in the vicidial, the dial manger table contains 1 Channel: SIP/cc100
13:39.27[TK]D-FenderPhurl: GUI's are not our problem.
13:39.34jayteeI have over 200 phones on my * system. My boss wants me to determine if it's feasible to have something setup where we can dial all the phones at once and broadcast a message. I don't see how without using MeetMe and that many simultaneous calls into a conference bridge on the server I think would cripple it or at least not work as hoped. Am I wrong?
13:39.40Phurlof course. thanks for your help.
13:40.13PsychobillySubdolus i ve tried this, aelpasereturns error
13:40.15Get_The_FishPhurl which version of asterisk did you say that you were using?
13:40.17Phurl[TK]D-Fender, thanks
13:40.32PsychobillySuPrSluG i mean
13:40.39mort_gibjaytee: Sending text to the phones is not an option??
13:40.40[TK]D-Fenderjaytee: that's kinda rough.  What I might do is add an extran conferencing server into the mix and split them up and bridge the meetmes
13:41.01PhurlGet_The_Fish, i think ubunut 1.4
13:41.18jayteemort_gib, nope not all of them are Polycoms, some are analog off of Linksys ATAs.
13:41.20Get_The_Fishok, nevermind then.
13:41.46Get_The_Fishjaytee, you can do this out of the box with Freepbx, so you can look at their code to get an idea of what you need to do to get that working
13:41.47jaytee[TK]D-Fender, that's a thought. I'd use Page() anyways since it's a one-way audio setup and all the callees are muted.
13:42.03PhurlGet_The_Fish, Asterisk 1.4.21.2~dfsg-3ubuntu2
13:42.03*** join/#asterisk elred (i=sauron@fucksheep.org)
13:42.04mort_gibjaytee: Then TK's advice might be better
13:42.15jayteemort_gib, thanks for the suggestion though!
13:42.23ck_28any one can check this http://pastebin.com/m79a3c265
13:42.42Get_The_FishPhurl, a buddy of mine tried this with 1.6 and was getting something similar is why I was asking
13:42.46[TK]D-Fenderjaytee: master spawn process looping out to a page-pool server.  1 real issue you'll run into is call setup lag.
13:43.10*** join/#asterisk tobias (n=tobias@cpe-071-070-219-040.nc.res.rr.com)
13:43.10*** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) [NETSPLIT VICTIM]
13:43.13[TK]D-Fenderjaytee: remembering that you'r ack-ing a LOT of calls.  Sometimes can kill a few sec's to get everyone in which makes the speaker feel awkward
13:43.38[TK]D-Fenderjaytee: (beep) .. uneasy pause ... "start speaking"
13:44.13jayteewe had a major power failure and right now the network engineer is in the boss's office getting reamed out because the UPS systems failed to keep up with the load and the Microsoft Operation Manager monitoring failed to send pages for the power outage
13:44.38*** join/#asterisk Jacke (i=jacke@85.128.98.27)
13:44.38*** join/#asterisk DaveCanoe (n=Dave@66.96.16.50) [NETSPLIT VICTIM]
13:44.38*** join/#asterisk carrar (i=tim@198.136.194.10) [NETSPLIT VICTIM]
13:44.41jayteeboth my IVR server and my primary * server "bounced" but came right back up online.
13:45.23Get_The_FishMOM?  Seriously????
13:45.24ck_28checking for gcc option to accept ANSI C... none needed which package is missing
13:45.30mort_gibSo MoM didn't work... I'm shanek
13:45.40Get_The_Fishshocked
13:45.49mort_gibs/shanek/shaken
13:46.31PhurlGet_The_Fish, here is a similar problem. no answer. http://code.google.com/p/outcall/issues/detail?id=17
13:46.38mort_gibI have trusted MS with my important data for so may year, and with only a lot of problems
13:46.44jayteeI had Nagios working great for stuff like that but all the linux bigots that run the show here decided to axe it in favor of MOM. It took me about a month to get Nagios working for all the server monitoring and stuff while doing other tasks. The Network Nitwit has taken over 6 months and still can't get it right.
13:46.54[TK]D-FenderPhurl: No... the channel you chose is not VALID
13:47.33[TK]D-FenderPhurl: Phurl there is no peer named "cc101" in your SIP SHOW PEERS
13:47.38Phurl[TK]D-Fender, yes, here is an example of a valid call that I was able to place :http://pastebin.ca/1422645
13:47.40mort_gibjaytee: I had two "Pro IT guys" recommending Juniper Firewalls
13:47.58jayteelol
13:48.00Phurlthe problem is with the vicidial call generated
13:48.06Phurli am now looking into it, thanks
13:48.08mort_gibjaytee: they came up with a £50.000 project to "secure" the infrastructure
13:48.20[TK]D-FenderPhurl: That code means nothing to me.
13:48.27mort_gibjaytee: I left 8 months later, they were still at it
13:48.33jayteemort_gib, hahahaa
13:48.34[TK]D-FenderPhurl: Sho an actual failed call from *'s point of view.
13:48.54mort_gibjaytee: I did a similar setup for another client using openBSD (CARP rulez) in two weeks
13:49.02[TK]D-FenderPhurl: We do not care how broken your dialer is, only enough to prove thats it is broken
13:49.16Get_The_Fishmort and jaytee... same here, cant tell you how many times I've seen that.
13:49.39jaytee[TK]D-Fender, on the idea of a page-pool, is there some kind of wildcard for dialing all the sip peers? or would I have to code something to pull the list of registered peers and loop through them in a macro?
13:50.30[TK]D-Fenderjaytee: No wildcard... you'll have to script something up or hard-code
13:51.08[TK]D-Fenderjaytee: And you'll get hit with the dialplan line length limit as well.  You'll probably ahve to cascade this a few levels deep.  Hence the nasty setup delay
13:51.27jaytee[TK]D-Fender, that's what I thought. I'm going to look into doing a script that will pull the sip peers.
13:52.01mort_gibGet_The_Fish: In this case client wanted to aggregate ADSL lines, and have redundant Firewalls doing so, along with IPSec 4 sites infrastructure
13:52.20mort_gibGet_The_Fish: I had issues with online banking, that was it, eh sort of
13:53.09Get_The_Fishjaytee, this might help you: http://www.pastebin.ca/1422658  this is the freepbx implementation of intercom
13:53.54Get_The_Fishmort, did you use a distro for pf like pfsense or anything?  I had issues with CARP, especially with SIP
13:54.18[TK]D-FenderGet_The_Fish: LOL... NO chance :)
13:54.31Get_The_FishTK, how so?
13:54.35[TK]D-FenderGet_The_Fish: jaytee knows what he's doing, and FreePBX wasn't made to scal for this
13:54.53HeManI'm trying to do call pickup and I've got it to work when I call internally but not when I have an incomming call
13:54.54Get_The_Fishtk, I can understand that, but it's a start :)
13:54.59mort_gibGet_The_Fish: No, I used OpenBSD vanilla, but in all honesty they don't run any "realtime" protos
13:55.04Phurl[TK]D-Fender, thank you for your advice. I think I found the problem. in the vicidial has that c101 in the "phone extension" field. I am going to change it to a know peer.
13:55.05Get_The_Fishgotcha
13:58.35SuPrSluGPhurl: are you using vicidialnow?
13:58.43PhurlSubdolus, yes
13:59.18rue_mohrI have a good one, the people at the office dont want to attend their transfers, and they want the calls tey transfer off to come back to them if nobody answers... suppose I can set a call source variable
13:59.51*** join/#asterisk spck (n=spck@unioncab.com)
14:00.04*** part/#asterisk spck (n=spck@unioncab.com)
14:00.11*** join/#asterisk spck (n=spck@unioncab.com)
14:00.19[TK]D-Fenderrue_mohr: Go read the CHANNELVARIABLES doc that came with *
14:00.20*** join/#asterisk axisys (n=axisys@155.70.141.45)
14:00.33rue_mohrhmmm
14:02.20*** join/#asterisk jmacz (n=jmacz@190.144.75.22)
14:04.39*** join/#asterisk youngproguru (n=quassel@74.10.229.45)
14:06.06youngproguruneato
14:06.25Phurl[TK]D-Fender, thanks it is working now
14:07.03Phurli changed the Phone extension: to a known peer and the astguiclient.php can now dial. twinkle gets the call.
14:07.05Phurl!!!
14:07.06Phurlthanks
14:08.21*** join/#asterisk coppice (n=chatzill@119.82.66.88)
14:12.41Subdolustwitches
14:16.39*** join/#asterisk theHub (n=theHub@69.177.93.21)
14:17.43HeManI got call pickup to work with incomming calls now but transfered calls are strange to pickup
14:24.03HeManthe call has to be picked up with original destination
14:24.04*** join/#asterisk trentcreek (n=kvirc@200.94.227.117)
14:24.10HeMancan that be changed in any way?
14:29.52trentcreekI am attempting to compile DAHDI, It is wanting the kernel source for  2.6.18-92.1.18.el5.028stab060.8, but the Kernel on the system is 2.6.18-128.1.10.el5-i686
14:30.04trentcreekHow can I get around this?
14:32.05tzafrir_laptopare you sure?  uname -r
14:32.15pmhaddad-workdoes anyone have a good e911 tutorial or dialplan example? I'm still stuck on this :/
14:33.43trentcreektzafrir_laptop: the version it needs came up
14:36.35*** join/#asterisk machoman48 (n=machoman@89.203.164.69)
14:39.18*** join/#asterisk my007ms (i=master@botmaster.x86.be)
14:39.19trentcreektzafrir_laptop: but, yum list kernel\* does not return the correct sources listed
14:39.34my007mswhat is windows software i can use to convert sound file to gsm files
14:39.50[TK]D-Fendermy007ms: Audacity
14:40.19my007mshttp://audacity.sourceforge.net/ ?
14:40.40my007msthanks :) [TK]D-Fender :)
14:41.03[TK]D-Fendermy007ms: How many do you have to convert?
14:41.23my007ms4 file in wav format need them to be gsm
14:41.40[TK]D-Fendermy007ms: Just use * CLI for this
14:41.46my007msi used to use sox but i am in windows and have no access to any linux box
14:41.47tzafrir_laptoptrentcreek, which version came up where? What about uname -r ?
14:42.08trentcreekusing yum list kernel\*
14:42.11[TK]D-Fendermy007ms: You're in * and you say you have no access to any *NIX box?
14:42.15[TK]D-Fender#asterisk
14:42.20trentcreekuname -r returns correct
14:42.25tzafrir_laptopinstall Linux under virtualbox
14:42.37my007msyes i mean ssh access :
14:42.44my007mswith root to install sox :)
14:42.50tzafrir_laptopproblem solved
14:43.51[TK]D-Fendermy007ms: Just use * to convert if * supports the base format at all
14:44.37*** join/#asterisk GlobeTrotter (n=GlobeTro@201.218.90.155)
14:45.51*** join/#asterisk jasonwoot (n=some@bookit-dev.com)
14:47.02trentcreek[TK]D-Fender: uname -r returns the vesion it wants
14:47.27*** join/#asterisk BobPierce (n=BobPierc@216.36.132.162)
14:47.49*** join/#asterisk Black_L (n=chatzill@wsip-98-175-64-147.ga.at.cox.net)
14:47.52Black_LHEllo?
14:48.48Black_LAllo?
14:49.35*** join/#asterisk kamanashisroy (n=kamanash@119.30.36.20)
14:49.49*** join/#asterisk n3hxs (n=HAMming@static-162-84-42-161.slsbmd.east.verizon.net)
14:50.42jaytee[TK]D-Fender, my servers restarted this morning due to a power failure. Since the restart the Polycom 550 callerid display shows the phone number and an IP address like SIP: 5146@XXX.XX.XXX.XXX. The Polycom 330's aren't experiencing this issue. I did a sip reload but that didn't change anything and rebooting the 550's doesn't either.
14:51.12jayteeI'm thinking I need to reload chan_sip.so
14:51.19[TK]D-Fenderjaytee: Usually you see an IP when a call is sent from a different IP than the one its reg'd to
14:51.52[TK]D-Fenderjaytee: Actually.. jsut on a different subnet <-
14:52.07*** join/#asterisk |Cybex| (n=John@80.100.126.176)
14:52.37jaytee[TK]D-Fender, you mean the phone has a different IP now than what is in the sip registry for that device? I'll check that, thanks.
14:52.38[TK]D-Fenderjaytee: So if the phone is 192.168.10.123 and *'s interface is 192.168.11.1 then I believe you end up seeing the IP as well
14:57.40*** join/#asterisk davood (n=Davood@86.109.41.134)
14:58.16Kattyhai!
14:58.28*** join/#asterisk LT (n=lt@unaffiliated/lt)
14:58.43[TK]D-FenderKatty: Mew.
14:58.47[TK]D-FenderKatty: OH HAI
14:58.48Kattyhugs on [TK]D-Fender
15:00.52*** part/#asterisk JenniferAkemi (n=Jennifer@76-10-182-237.dsl.teksavvy.com)
15:01.20*** join/#asterisk mmlj4 (n=jkelly@70.171.94.246)
15:01.30*** join/#asterisk propellerhead (n=yogurt2u@host105.190-136-230.telecom.net.ar)
15:01.45jayteehugs Katty
15:02.03eppigy^________________________^
15:03.06Katty:>>>>
15:03.09Kattyhuggles jaytee
15:03.11Kattypamples eppigy
15:03.50jaytee[TK]D-Fender, figured it out. You were correct. My server has dual nics and the secondary interface is in the same VLAN as the phones but not the interface the phones register to. When the server rebooted after the power failure the eth1 interface came up automatically. I've since disabled it from coming up at startup.
15:04.05[TK]D-Fenderjaytee: You're welcome
15:04.25jaytee[TK]D-Fender, oh, yeah, thank you!!! :-)
15:04.40*** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home)
15:05.46eppigy:D
15:07.21Black_LHey guys
15:07.32Black_LHow do i configure a dial plan to run as soon as a phone is picked up?
15:07.47Black_LI need to pick up the phone, and have it automatically dial 2 more numbers into a conference with it.
15:08.29Black_LAt the same time
15:08.41Black_LAnd to not stop ringing those numbers for any reason other then them picking up.
15:09.06sulexI'd like to limit to 10 mins the amount of time an agent can spend with a caller entered in a queue. not the time a user spends in a queue waiting for an agent to pickup the call, but the amount of time spent talking to the agent himself... ideas?
15:09.11*** join/#asterisk spck (n=spck@unioncab.com)
15:09.42Black_LIf you can just program in a timer that sets off a user-event to end the call after 10 minutes.
15:09.59[TK]D-Fendersulex: "core show application dial" <-
15:10.44*** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net)
15:10.57*** part/#asterisk davood (n=Davood@86.109.41.134)
15:11.37sulex[TK]D-Fender: I'm not using dial, I'm using queue
15:11.40mmlj4Black_L: Answer()
15:11.59Black_Lmmlj4: That doesn't tell me anything...
15:12.06[TK]D-Fendersulex: You might very well be using dial, go lookat precisely what calls your "agent"
15:12.15Kattydinged.
15:13.11sulex[TK]D-Fender: sorry but I think I did not get what you mean, can you expand it?
15:13.32[TK]D-Fendersulex: I mean go look at how the queue calls your agents
15:13.57*** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri)
15:14.08*** join/#asterisk moy (n=moy@74.12.124.89)
15:14.29mmlj4that's the first part of your project... the line will be answered immediately
15:14.40*** join/#asterisk Deeewayne (n=dwayne@75.76.254.162)
15:14.40*** mode/#asterisk [+o Deeewayne] by ChanServ
15:15.00kaldemarBlack_L: what kind of a phone?
15:15.27Black_LJust a typical phone that plugs into the back of my x86 switch box
15:15.34*** join/#asterisk bmg505 (n=leon@196-209-78-114-rndf-esr-5.dynamic.isadsl.co.za)
15:15.35sulex[TK]D-Fender: and again... the queue command is calling the agent at the moment... not a dial application. the only timeout I can set on Queue() is the time the user will be let waiting for an agent to pickup... are you suggesting me to leave the Queue() usage for Dial() ?
15:15.51kaldemarBlack_L: was it you asking the same thing yesterday too?
15:15.56Black_LYes
15:15.59Black_LNever got it working
15:16.09kaldemarwell, did you define the channel as immediate?
15:16.31Black_LYes
15:16.42kaldemarhow did it not work?
15:16.53Black_LI have no idea how to program a dial plan
15:17.05Black_LAnd i can't figure out where the other coder put the conf file now...
15:17.42Black_LI think i dial numbers using ext => priority, number, dial() right?
15:17.52kaldemarwhat exactly do you want the phone to do when someone picks it up? just join a conference?
15:18.36kaldemar"ext => priority, number, dial()" is a no.
15:18.37Black_LCall two other numbers and link them into a 3 way
15:18.44Black_LDo not stop dialing for any reason
15:18.54*** join/#asterisk jcape (n=jcape@209.120.251.81)
15:18.55Black_LRing until someone picks up, DO NOT STOP
15:19.23Black_LEven if one of the other line picks up don't stop ringing on the one that hasn't until they pick up
15:20.52kaldemaryou can't do it with a simple dial then.
15:21.16kaldemari've done that using a meetme conference and callfiles.
15:21.32Black_LOk then
15:21.35Black_LHow do i do that?
15:21.47[TK]D-Fendersulex: Again you are not looking at wht the queue is DOING.
15:21.47kaldemarmake the exten put a call file for each callee to the spool directory and then put the caller in a conference.
15:22.06[TK]D-Fendersulex: pastebin a call going into the queue and look at PRECISELY how this "agent" is getting called
15:22.08[TK]D-Fender~pb
15:22.09infobot[~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://www.rafb.net/paste , or , http://bin.cakephp.org/
15:22.11[TK]D-Fender^^^^^^^
15:22.18kaldemarmake the callfiles connect the callees to the same conference upon answer.
15:22.34[TK]D-Fenderkaldemar: Yup
15:22.38Black_Lkaldemar: Ok so.... any idea how i would do that?
15:23.09[TK]D-FenderBlack_L: [11:21]<kaldemar>make the exten put a call file for each callee to the spool directory and then put the caller in a conference. <--- he just told you
15:23.25Black_LYou don't get it
15:23.31Black_LI know NOTHING about this system...
15:23.37sulex[TK]D-Fender: I think I got you now... eureka! the point is, the queue command is using dial to call the agent... maybe if I set an absolute time out on the agent extension I get what I need? :)
15:23.50Black_LI'm assigned jobs and i'm new so i'm expected to figure it out on the go
15:23.53kaldemarBlack_L: you have all the keywords now. do you have a copy of the book?
15:23.55Black_LWhich is a pain in the ass
15:24.06Black_LI have Trixbox made easy now
15:24.11[TK]D-FenderBlack_L: Go lookup "call files" on the WIKI, and go read the instructions for MeetMe
15:24.25Black_LAye
15:24.27[TK]D-FenderBlack_L: Thats a book I take it..
15:24.27*** join/#asterisk outtolunc (n=me@c-71-198-197-201.hsd1.ca.comcast.net)
15:24.37kaldemarif you read up on the dialplan on the simplest level, callfiles and meetme, you should be able to figure it out pretty fast.
15:24.41Black_LYes
15:24.43Qwell[TK]D-Fender: a book by...Kerry
15:24.48mort_gibBlack_L: Hey, do you need cleenex??
15:24.56[TK]D-FenderBlack_L: Your GUI is not going to do ANYTHING for you, this is entirely custom.
15:24.56Black_LWhy would i need cleenex?
15:25.03Black_LYeah i know...
15:25.07Black_LI have Putty open
15:25.09mort_gibBlack_L: To wipe your eyes
15:25.12Black_LDoing it through Putty, not the GUI
15:25.38Black_LWhat's the address for the Asterisk wiki?
15:25.49[TK]D-FenderBlack_L: If you understand nothing about * configuration itself, you're in for plenty of extra pain just fighting your way around your GUI's crap
15:25.53[TK]D-Fender~wikis
15:25.54infobot[~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE.  http://www.voip-info.org (c) Arte Marketing Inc / CommPartners
15:26.10[TK]D-FenderQwell: LOL!
15:26.23[TK]D-FenderQwell: the blank leading the blind!
15:26.50Black_LD-Fender: Don't need to tell me man
15:27.05Black_LBeen working on this all yesterday. Not getting anything else done so i'm getting a little agitated.
15:28.38QwellBlack_L: why not reinstall from scratch and ignore any of the GUI stuff?  Why bother working around it?
15:29.22tzafrir_laptopor rather, "freepbx", and not "gui in general. FreePBX makes a very complicated dialplan
15:29.25kaldemarBlack_L: familiarize yourself with the things mentioned earlier and you'll get it. unless if you've shot yourself in the leg with a GUI, maybe.
15:31.18[TK]D-FenderBlack_L: You'll need to make a custom extension/application, in there you'll need to drop in 2 call files (or issue AMI/CLI Originate commands) to have 8 call out to the 2 other parties.  You'll then need dialplan to dump them into a MeetMe room upon some sort of confirmation that they accept.  You'll then have to dump that call (yours) into the same room and wait for the others to join.
15:32.06[TK]D-FenderBlack_L: **AND** all of the depends on *'s ability to track call-progress, etc.  More work required to auto-kick people from the conference when you're done so you don't accidentally hang channels, etc.
15:32.17[TK]D-Fenderfun fun fun...
15:33.12Black_LSorry
15:33.13Black_LHad to pee
15:33.17kaldemarlet him start simple, that's just plain depressing. :)
15:33.45Black_LMy boss is determined to use Asterisk
15:33.48Black_LI mean Trixbox
15:33.50Black_LI have no idea why
15:33.59Black_LSo i can't ditch it
15:34.07*** join/#asterisk MikeJ (n=MikeJ@freeswitch/developer/mikej)
15:35.02[TK]D-FenderBlack_L: Well you've got plenty of * to learn for this, or go hire a consultant
15:35.30[TK]D-FenderBlack_L: So why is it he needs a miracle bat-phone to initiate a 3way forced call?
15:35.40Black_LLocal airport
15:35.47Black_LIf a plane crashes, the bat phone shall be maned
15:36.03Black_LAnd god so help us if the fire and emergency departments don't pick up
15:36.06[TK]D-FenderBlack_L: Holy shit, you're putting LIVES on the line with Trixbox?
15:36.17[TK]D-FenderSILLY RABBIT TRIXBOX IS FOR KIDS!
15:36.18kaldemarskims 3 seconds off of coffee latency.
15:36.28Black_LIt's a backup
15:36.31Black_LIt's not the main line.
15:36.48Qwellwait, what?
15:36.51*** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net)
15:37.07Black_LANYWAYS
15:37.11Black_LNeed to get this working
15:37.15Black_LAnd flawlessly obviously
15:37.20QwellNot going to happen.
15:37.23Black_LTime to burn up the search box
15:37.36mort_gibBlack_L: I would opt for * rather than TrixBox
15:37.46mort_gibOr FreePBX or, or
15:37.52[TK]D-FenderBlack_L: Well I jsut told you what you'll have to do and account for
15:37.55Black_LI don't know why he wants to use Trixbox so badly
15:38.03Black_LBut he's the boss
15:38.03QwellBlack_L: because he's dumb.  tell him no.
15:38.08Black_LSo we are using Trixbox
15:38.09[TK]D-Fendertrixbox isn't really so much of a problem.
15:38.15[TK]D-Fenderthis is jsut costom code ont he side
15:38.21[TK]D-FenderFor a single task
15:38.23Black_LI can code
15:38.27mort_gibBlack_L: Many bosses are like that, ask Katty
15:38.29kaldemarmake your point well and you can ditch trixbox in no time.
15:38.30Black_LI just don't know how to use this stuff.
15:38.39[TK]D-FenderBlack_L: You've been told the 3-4 pieces required.  get to work
15:38.44[TK]D-FenderTRABAJO
15:38.46Black_LAlright alright
15:38.52[TK]D-Fender(c) eppigy
15:38.56Black_LI'm already searching documentation
15:42.30*** join/#asterisk Vec (n=Vec@host-87-74-7-50.dslgb.com)
15:44.46*** join/#asterisk BlargMaN00 (n=blarg@12.234.16.130)
15:45.28*** join/#asterisk grantm (n=grant@68.142.138.4)
15:46.03*** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net)
15:47.03mort_gibIs there any plugins available for monitoring asterisk from Nagios??
15:49.14*** part/#asterisk bkw_ (n=brian@freeswitch/developer/bkw)
15:49.23fiddurhttp://www.voip-info.org/wiki/view/Asterisk+monitoring
15:49.24spcktshark -w output.cap "port 5060 or port 5061"
15:49.29*** join/#asterisk bmoraca (n=chatzill@66.242.174.254)
15:49.30spckerps
15:56.33Kattyyes.
15:56.35Kattybosses dumb.
15:56.38Kattyvery, very dumb.
15:57.48Kattyalso!
15:57.49Kattylunch.
15:57.50Kattyafks
15:58.19spcki ignore my phb's
15:59.04*** join/#asterisk marv[work] (n=timr@router.asteriasgi.com)
16:10.03VecFilipe
16:10.04Vecu there
16:10.13*** join/#asterisk CunningPike (n=CunningP@204.239.10.119)
16:11.06*** join/#asterisk shareenergy (n=go@host-87-74-7-50.dslgb.com)
16:11.17spckhas anyone been to the opensips bootcamp?
16:11.49*** join/#asterisk spck (n=spck@unioncab.com)
16:11.53spcksrry wrong rom
16:14.01therealcircuthey all
16:16.03*** join/#asterisk grantm (n=grant@68.142.138.4)
16:18.11*** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri)
16:18.33*** join/#asterisk coppice (n=chatzill@119.82.66.88)
16:18.49*** join/#asterisk astassistant (n=skip@65-126-63-1.dia.static.qwest.net)
16:19.40therealcircutdead today
16:19.49therealcircut[TK]D-Fender must have meetings
16:22.02*** join/#asterisk d00gster (n=doughant@bas1-cooksville01-1279552044.dsl.bell.ca)
16:27.50therealcircutis it possible to playback a recording over a page without using call parking?
16:28.40bmoracatherealcircut: er...huh?  what are you trying to accomplish?
16:29.22[TK]D-Fendertherealcircut: sure. make sure the channel that is calling the page is doing a playback.  guess what kind of channel this would be
16:29.50therealcircutbasically i dial extension 779, it records a message, then it pages my phone on the desk and plays back that message
16:30.22[TK]D-Fendertherealcircut: Easily doable so far
16:30.24therealcircutthen it will start ringing the phone the original caller to the extension that was dialed.
16:30.41therealcircutsorry that was jibberish
16:31.46therealcircutbasically, call comes in. receptionist answers and finds out who it is. Then she dials 7+persons extension which makes a recording
16:32.13therealcircutthen asterisk should call that persons extension, playback the recording over a page
16:32.26therealcircutand then send the original caller to that persons extension.
16:32.33therealcircutallowing it to ring
16:33.13therealcircutand going to vm if theres no answer
16:33.16therealcircutsounds doable right?
16:34.29[TK]D-Fendertherealcircut: So the callee's phone gets 1 call as a Page" over the speaker.  When that ends the intended actual call starts to ring in?
16:34.43therealcircutyup
16:35.15[TK]D-Fendertherealcircut: Tricky but possible
16:36.00therealcircutdo i need to setup another extension for it somehow
16:36.02[TK]D-Fendertherealcircut: What is the real point of this?
16:36.13therealcircutannounced callerid?
16:36.17therealcircuti dunno the sheep want it
16:36.34[TK]D-Fendertherealcircut: Have they considered looking at the CID on the stupid phone?
16:37.14therealcircutand risk wiplash for having to turn their heads??
16:37.48therealcircutright now it does the playback fine, parks the call
16:38.03therealcircutbut they dont like having to press the parked call button to pickup the call
16:38.06[TK]D-Fendertherealcircut: When you say "recording", you mean the receptionist says "Joe from Acme Inc" for instance?
16:38.07Kattypokes head in
16:38.14therealcircutyea
16:38.40stopewhen the system is up and running, if we're using BLF and SLA on a hosted pbx for many clients, is that kept in memory or is it constantly read from the disk when required?
16:38.50*** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri)
16:39.48[TK]D-Fendertherealcircut: Make your exten do teh record, then launch a call file or AMI originate to do the page with a local channel  as the "Channel:" to do the playback" and end this call.  On the exten that triggered the script, allow X seconds to pass (guesstimate) and then process the rest of the normal dialplan.
16:40.11[TK]D-Fenderlunch BBIAB
16:41.23therealcircutdcc me some burritos
16:41.26therealcircutpls kthnx
16:42.40Kattyhas been watching star trek episodes on youtube
16:55.15*** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim)
16:55.54*** join/#asterisk Octothorpe (n=Octothor@pdpc/supporter/professional/octothorpe)
16:57.22rbdhi guys, I have an ast 1.4 call acting as a B2BUA for G729/G711 calls....I do a loadtest where I shove 50 calls through it (1 call starting every other second, and after all 50 are established, lasting about an hour)... the CPU goes quite high (>100%) when the calls are starting, but then during the test it is at about 20% ... is this normal?
16:57.32*** join/#asterisk Great_Anta_Baka (n=tensai@196.33.159.83)
16:57.50rbdthe dial plan has only a simple Dial() command in it to initiate the bridging
16:59.44*** join/#asterisk |Cybex| (n=John@80.100.126.176)
17:18.05*** join/#asterisk Great_Anta_Baka (n=tensai@196.33.159.83)
17:18.54*** join/#asterisk lanning (n=lanning@nat/yahoo/x-a530d3523d5a8ecc)
17:23.45*** join/#asterisk Flyser (n=Flyser@unaffiliated/flyser)
17:30.44stoperbd: I would assume that its the call setup and teardown that causes the spike in cpu usage
17:33.22*** join/#asterisk Defraz (n=T0tal@fw-poky.fuzecore.com)
17:33.36*** join/#asterisk seidren (n=chatzill@38.111.96.113)
17:33.53seidrenhi everyone
17:34.31Kattyhi
17:36.14seidrennoob here!!! could someone please enlighten me on how an outbound call center works. I need to know how many phone lines are needed to setup
17:36.33jasonwoot42
17:36.42seidrenha ha
17:36.49seidrenfor real though
17:37.04seidrenis one enough.. i feel stupid to ask
17:37.23FlyserIs it correct, that I can remove all Hangups from the end of the priority chain in my extensions.conf if autofallthrough is enabled?
17:37.54FlyserOr is there a slight difference in the behaviour of these options
17:38.03Kattyi think a better question is how many epople are ina  call center, and will they all require a channel at the same time.
17:39.26seidrensay i have 10 people making outbound calls simultaneously. do i need 10 phone numbers to make this happen or
17:39.43seidreni was rephrasing that...
17:39.56Kattystop thinking about physical lines for a moment
17:39.59[TK]D-Fenderseidren: How many lines depends on your needs.
17:40.15[TK]D-Fenderseidren: And separate NUMBERS from LINES
17:40.35seidrenok good..
17:40.35Katty10 people on the phone at the same time, means 10 channels
17:40.39[TK]D-Fenderseidren: If you want 10 agents calling out, thats 10 channels right there.
17:40.52seidrenok 10 channels
17:41.36Kattyhow many people calling in?
17:42.13seidrencalling in is unknown
17:42.20seidrenbut the main purpose is to call out
17:42.23Kattythen you need to find that out
17:42.30[TK]D-Fenderseidren: Come up with an answer for how many channels you will need
17:42.32seidrensay 2 coming in
17:42.38Katty12 channels.
17:42.41seidren2 channels coming in
17:42.46seidrenso ya total 12
17:42.47Katty2 phone numbers?
17:42.49Kattyor 1 phone number?
17:42.58seidren1 is good
17:43.02Kattywell there you go
17:43.10Kattythat's what you need
17:44.04[TK]D-Fenderseidren: Call up your telco and have them quote you a T1 PRI (full 23 channel, or partial for 13
17:44.04seidrenok.. so now to make this happen.. do i call my local telephone company and subscribe for 1 phone line ? and asterisk takes care of the rest ?
17:44.13[TK]D-Fenderseidren: No
17:44.17seidrenah ya
17:44.24Kattysip 'trunks' might be an option as well
17:44.27[TK]D-Fenderseidren: Asterisk does not let some boring analog line carry 500 calls at once
17:44.56[TK]D-Fenderseidren: * lets you use a variety of ocnnectivity options you already posess
17:45.15seidrenok so I need a T1 PRI
17:46.00[TK]D-Fenderseidren: That is one option.  Using an ITSP if you have the bandwidth is another
17:46.02[TK]D-Fender~itsp
17:46.03infobot[~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs
17:46.06[TK]D-Fender~pri
17:46.07infobot[pri] [~pri] Primary Rate Interface, often called T1 or E1 (European Standard). E1 offers 30 ISDN B-Channels a 64kBit/s + 1 D-Channel with 64kBit/s. The T1 has 23 B-Channels + 1 D-Channel. Cards to use with *: T100P, E100P, TE410P, R1T1,R2T1,R4T1, etc.
17:46.09*** join/#asterisk michaely (n=Mike@207.114.199.107)
17:46.54seidrenok good.. thank you so much
17:47.18*** part/#asterisk machoman48 (n=machoman@89.203.164.69)
17:47.22seidreni think i understand the scene.
17:47.38Kattycheers
17:47.41Kattytime for margaritas then
17:47.42jasonwootFender is in league with the POTS providers
17:48.01jasonwootSIP trunks or death
17:48.12Kattylittle red cookbook, little red cookbook
17:48.24Qwelljasonwoot: SIP "trunks" cause deaths.
17:48.30coppiceFender is in league with the devil.... oh, sorry, you said that
17:48.32seidrenthe T1 option seems like it needs special cards..
17:48.44jasonwootSIP trunks cured my lyme disease
17:48.46Kattyeverything needs cards.
17:48.50Kattyobviously
17:48.51seidrenITSP only needs as much bandwidth as possible
17:49.36seidrenbecause I should be able to connect computers to the asterisk box via ethernet and call through
17:49.53*** part/#asterisk michaely (n=Mike@207.114.199.107)
17:50.05seidrenam i correct ?
17:50.44Qwellseidren: you are.
17:50.52Qwellbut then what happens when your DSL line goes down?
17:51.04Kattylooks like you're not calling the police department
17:51.10Kattyor the fire department
17:51.14Qwellor the ISP
17:51.25jasonwootanswer= dont get DSL
17:51.26Kattyor me!
17:51.47seidrenT1 wont fail then ? right ?
17:51.56jasonwootyou can have a local provider pull fiber for the cost of a T1 install/lease
17:51.59Qwellseidren: It will fail.
17:52.07QwellBut there are SLAs on them.
17:52.12Kattyeverything fails, at some point or another
17:52.13Kattyeven you.
17:52.22QwellI don't fail.
17:52.25Qwellever.
17:52.32jasonwootT1's will bounce the signaling and drop the calls, with SIP trunks, there's a good chance it will recover
17:52.51[TK]D-FenderAsterisk : When failure is NOT an option (it comes bundled with the software)
17:52.51*** join/#asterisk nephfl (n=none@wsip-98-175-64-147.ga.at.cox.net)
17:52.57nephflhello, i need some dialplan help
17:53.03Qwell[TK]D-Fender: ...
17:53.04Kattyi need a therapist
17:53.12Kattyand maybe a visit to the spa
17:53.35nephflI have 2 zap extenions set up for immediate=yes and they dial to a meetme conference
17:53.40[TK]D-Fenderseidren: T1's are monitored circuits where the telco offers SLA for their service.  It is probably the most reliable phone link you're likely to find
17:53.48[TK]D-FenderQwell: <3
17:53.53nephflbut i need the other extensions to ring in order to know when someone has joined the conference
17:53.53jayteedid you ever notice that the word therapist contains two separate words? the AND rapist
17:54.21jasonwootI will venture a guess: who is Jaleel White?
17:54.31seidrenthanks a lot people...
17:55.04seidreni'll probably come back here when i have more questions.. but for the moment.. i guess i have my options layed out...
17:55.15nephflhow do i get the remote extension to ring?
17:55.20seidrenmoney will decide the rest
17:55.22seidrenbye
17:55.30nephflextensions
17:56.20jasonwootmo money, mo problems
17:56.20jayteehums "money changes everything..."
17:56.37nephflcould i have the first extension dial the other two and transfer them to meetme? or is there a simply command to do it?
17:58.19seidren~itsp
17:58.20infobot[~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs
17:58.34seidren~itsplist-ca
17:58.35infobot[~itsplist-ca] Here are some popular ITSPs (Canada) starting with the more respected ones : http://www.unlimitel.ca , http://www.les.net , http://www.babytel.ca
17:58.41nephflwhat i have is an emergency phone system with 2 zap extensions and an analog connection to pots... i need it so that if i pick up either extension the other rings and dials out to 911 on the pots line
17:58.56*** join/#asterisk IBC_Jkenney (n=Jkenney@99.23.50.73)
17:59.30IBC_JkenneyHello I have a problem i need for asterisk to write partial CDR record to postgres then complete record after call hangup
18:00.01[TK]D-Fendernephfl: What does the other do BESIDES jsut ringing?
18:00.11*** join/#asterisk funxion (n=nunya@63.214.236.169)
18:00.17bmoracanephfl: you should probably investigate just using a POTS line directly.  asterisk + emergency = too many pieces in the link and too many points of failure
18:00.27nephflwhen you pick up all 3 extensions must be in conference
18:00.59nephflif i pick up line one 2 and pots should dial then both need to ring until someone answers and all 3 are on conference
18:01.09funxionanyone know of a reason that the asterisk app would lock up and drop its sip peers becuase of trunk deadlocks?
18:01.45*** join/#asterisk Justnulling2 (n=Justnull@ool-4b7fd02a.static.optonline.net)
18:02.46*** join/#asterisk ruben23 (n=AGENT@124.107.3.178)
18:04.20nephflany ideas
18:04.47*** part/#asterisk seidren (n=chatzill@38.111.96.113)
18:08.49jayteepony rides on commercial airplane trips
18:10.36jasonwootfunxion: using ZAP too?
18:10.45funxionyup
18:10.48funxionE1 pri
18:10.51jasonwootPCI card?
18:11.38*** join/#asterisk mweichert (n=mweicher@216.13.154.21)
18:12.18jayteejust got in two new Dell Mini laptops. we're going to use them with bar code readers at points of entry to scan tickets and membership cards. I'm gonna name them Frodo and Bilbo
18:13.23mweicherthello, I have trixbox installed, setup some extensions, and would like to customize one of the extensions by playing back an mp3. To do this, I think I would add some dialplan rules in extensions_custom.conf, correct, under the [ext-local-custom]
18:13.27mweichertcontext
18:14.01jayteemweichert, wrong channel. try #trixbox
18:15.14eppigy8[]
18:15.24eppigyTRABAJO
18:15.26mweichertjaytee, okay, thanks
18:15.27eppigyNO NECESSITO
18:16.09jaytee¡usted ahora trabaja!
18:16.54[TK]D-Fendernephfl: You'll need to script 2 call files to have your outbound calls get dumped into  a mettme with you
18:18.16jplankfor a polycom, does this look off to anyone? 9,011x.t
18:18.32jplank(digit map for a polycom)
18:22.23*** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net)
18:25.32Get_The_Fishcan anyone suggest a good way to see what channels are transcoding? I am using more g729 than I think that I should, and I need to find a good way to troubleshoot this
18:26.07Get_The_FishI'm grepping the logs for recordings, playing audio, and out of license warnings
18:27.24trentcreekWhat can be used to stress test the combination of the box and trunks?
18:27.45trentcreekFor example I want to simulate 100 calls
18:28.23Get_The_Fishtrent, look into sipp and see if that will work
18:28.32*** join/#asterisk BobPierce (n=BobPierc@216.36.132.162)
18:31.46*** join/#asterisk ayeso (n=chatzill@ext-52.sagetelecom.net)
18:31.53trentcreekokay, thanks
18:32.07Get_The_Fishjplank, this is my polycom digit map, it's the default if it helps: "[2-9]11|0T|011xxx.T|[0-1][2-9]xxxxxxxxx|[2-9]xxxxxxxxx|[2-9]xxxT"
18:32.52ayesoIf I have modified the source code for app_voicmail.c, (this will sound stupid im sure) how can i compile it and install it into a current working asterisk platform that already has voicemail running?
18:33.41[TK]D-Fenderjplank: Dial 9 prefix... how 1980
18:34.21[TK]D-Fenderayeso: Just recompile & install everything, then connect to * and specifically reload that module
18:35.47Get_The_FishI hate to ask again, but I could use help on this...can anyone suggest a good way to see what channels are transcoding? I am using more g729 than I think that I should, and I need to find a good way to troubleshoot this
18:39.03*** join/#asterisk matsk (n=matkar@c-198ae253.174-6-64736c10.cust.bredbandsbolaget.se)
18:39.15[TK]D-FenderGet_The_Fish: "sip show channels"
18:39.31[TK]D-FenderGet_The_Fish: Map that to "show channels concise"
18:40.28Get_The_Fishsweet TK, thanks man...
18:41.03*** join/#asterisk DaveCanoe (n=Dave@strike.dclg.ca)
18:41.18*** join/#asterisk jcape (n=jcape@209.120.251.81)
18:41.53*** join/#asterisk theHub (n=theHub@69.177.93.21)
18:44.33*** join/#asterisk mmlj4 (n=jkelly@70.171.94.246)
18:44.33*** join/#asterisk carrar (i=tim@198.136.194.10)
18:50.24Get_The_Fishok, so would it makes sense that when a call is sent from the queue it would be using a transcoder?
18:50.59*** join/#asterisk ingenius (n=alektro@netsolution.com.ar)
18:51.01[TK]D-FenderGet_The_Fish: Depends if all the sounds are in G729 or not
18:51.07[TK]D-FenderGet_The_Fish: Or if you're recording.
18:52.13Get_The_Fishall the sounds (that I know of) are in g729, as is the only codec accepted on the trunk, and same with all the ua's
18:52.33Get_The_Fishnot recording all calls.
18:53.26ruben23hi how am i going to setup my asterisk server to be fault tolerant or for failover..
18:54.32*** join/#asterisk mnicholson (n=mnichols@nat/digium/x-79303cae626cd56f)
18:55.01*** join/#asterisk stupidnic (n=foo@cpe-70-94-229-122.sw.res.rr.com)
18:56.25stupidnicCan somebody tell me how to fix "Firmware version 0 not supported by this driver contact Voicetronix to have it updated"? The card is an older TDM400 in a server that I had to replace the OS on. I used Debian Lenny, should I revert back to Etch?
18:56.58*** join/#asterisk neurosys (n=vinix@173-9-159-182-miami.txt.hfc.comcastbusiness.net)
18:58.28alrsstupidnic: I have never used anything Voicetronix.  Perhaps you could just get rid of the libvpb0 and vpb-utils packages?
18:59.13stupidnichmm
18:59.25stupidnicif I try and remove libvpb0 it tries to remove asterisk as well
19:00.26alrsstupidnic: Does that voicetronix error actually keep anything from working?
19:00.35stupidnicNot sure
19:00.58*** join/#asterisk ttl- (n=patrick@d5153AE82.access.telenet.be)
19:01.06stupidnicI do have a problem that has only occured after I installed the new system (HD failed earleri today on an otherwise perfectly working setup)
19:01.36stupidnicgranted I installed asterisk from apt-get, but the main version is the same as what I was running
19:01.55alrsI'm running Lenny-packaged Asterisk on production machines and loving it
19:02.03stupidnic1.4.18 in etch versus 1.4.22 in lenny
19:02.13stupidnicYeah, I am not having trouble with VoIP stuff
19:02.19stupidnicjust TDM/Zaptel
19:02.23[TK]D-FenderWhat on earth does Voicetronix have to do with a TDM400 ?
19:02.24alrshave you built the zaptel drivers?
19:02.33alrsin module-assistant?
19:02.38stupidnicalrs: yes m-a etc
19:03.00stupidnicI can see the Wildcard detecting the channels properly in dmesg
19:03.27stupidnicbut for some reason when I ring the FXO line, asterisk never sees it as ringing
19:03.47stupidnic[TK]D-Fender: it was an error in the dmesg that I saw right before the wildcard loading
19:03.53stupidnicso I don't really know...
19:04.04alrscard shows up OK in zttool?
19:04.25*** join/#asterisk SkywaIker (n=pirch@58.147.17.166)
19:04.27stupidnicYes
19:04.30stupidnicI can see it there
19:04.31keith4my asterisk box is NAT'd. I have nat=yes for all external peers, and have never had a problem with them. currently, there is one client, also NAT'd, that can hear me fine, but I can't hear her
19:05.01keith4sip debug shows: <--- SIP read from 195.113.65.8:9664 --->, followed by two blank lines, and <------------------->
19:07.09[TK]D-Fenderkeith4: PASTEBIN
19:07.12[TK]D-Fender^^
19:09.29keith4k, hold on
19:10.40*** join/#asterisk thomasrr (n=scroogey@195.240.213.212)
19:10.51thomasrrhello
19:11.06thomasrrdoes anyone here have experience with using vopibuster voup-in numbers together with asterisk?
19:14.22[TK]D-Fenderthomasrr: Is specific experience required for your actual question?
19:14.57jameswfno free lunch today :(
19:15.06hardwireanybody know of a place to send really old and outdated phones for old phone systems.
19:15.44keith4[TK]D-Fender: http://pastebin.com/d38130527
19:16.17*** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net)
19:17.24keith4[TK]D-Fender: I'm calling from 6107583228 to 6107171796, which goes Voicepulse->asterisk->sabrina (105)
19:18.58[TK]D-Fenderkeith4: PB your sip.conf
19:19.17eppigyhi
19:19.24*** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net)
19:19.47[TK]D-Fendereppigy: YOU ARE DAVE!
19:20.30Kattyoh
19:20.30Kattydave
19:20.31eppigyYES
19:20.32Kattyhi dave
19:20.35eppigyhi Katty
19:20.38eppigyhi [TK]D-Fender
19:20.41Kattyi'mw atching star trek
19:20.44Kattyseason 1
19:20.46eppigythe new one?
19:20.47Kattyon episode 5 now
19:20.47eppigyo
19:20.49eppigyhaha
19:20.50KattyTOS
19:21.01eppigyI am at work :[
19:21.05KattyTHE ENEMY WITHIN
19:21.06Kattyso am i
19:21.09eppigy:D
19:21.15eppigyI AM THE ENEMY
19:21.25Kattyi don't even what to know what you're within
19:21.28Kattythat's just.. eww.
19:21.55eppigyo I did not think of that
19:21.57eppigyhaha
19:22.08keith4[TK]D-Fender: http://pastebin.com/d5436f254
19:22.12eppigyit is an older electro song
19:22.44keith4I just had someone off-site/NAT'd set up 104 for a test call, and there were no audio problems.
19:23.53*** join/#asterisk jeffgus (n=jeffgus@green.zimage.com)
19:24.24nephflmy call file is just sitting there and not processing
19:26.57KattyOH NOES the captain beamed up TWICE!
19:27.43Pan3DKatty: that's a great episode
19:27.57Pan3Dthe use of shadows is great
19:28.09Pan3Dmakes him look eeeeeeviiiil
19:28.54*** join/#asterisk saftsack (n=saftsack@p5792476A.dip.t-dialin.net)
19:43.30Kattyi like the use of eyeliner
19:47.57*** join/#asterisk mweichert (n=mweicher@216.13.154.21)
19:48.13Kattythe use of stunt doubles is awful
19:48.39mweichertis it possible to specific a SOCKS proxy in sip.conf for sip channels?
19:48.50mweichert*specify
19:50.48*** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net)
19:59.12*** join/#asterisk telnettech (n=telnette@cpe-71-74-91-116.insight.res.rr.com)
20:02.02*** join/#asterisk voxter (n=voxter@190.241.16.138)
20:06.13stupidnicOn the followme app if I just provide the context that followme should use (with no |a s or n) it is still prompting me to record my name
20:06.43mweichertin a sip channel, can you specify which rtp port to use?
20:06.58Qwellstupidnic: how are you specifying the context?
20:07.39stupidnicQwell just followme( 2001 ), but I have also tried followme( 2001| ) with the same results (this use to work on my previous version)
20:08.00Qwellthat didn't answer my question
20:08.15*** join/#asterisk goupil (n=goupil@2a01:e35:2f3d:7900:240:63ff:fedc:10e)
20:08.17stupidnicthen your question was vague
20:08.28Qwellyou said you're providing the context
20:08.29Qwellhow?
20:08.35*** join/#asterisk ziram19 (n=chatzill@41.226.184.105)
20:08.47stupidnicin the extensions.conf I have it specified
20:09.02stupidnic[followme]     5. Followme(2001|)
20:09.30stupidnic2001 matches a context specifed in the followme.conf
20:09.34thomasrri am trying to get the voipbuster voip-in number working
20:09.34*** join/#asterisk Micho123 (n=mcho123@77.42.186.182)
20:09.45[TK]D-Fenderkeith4: Your REGISTERS break the rest of [general].
20:09.49thomasrronly keep getting a message like "this user is currently not online try it later"
20:09.51*** part/#asterisk spck (n=spck@unioncab.com)
20:09.56*** join/#asterisk spck (n=spck@unioncab.com)
20:09.58[TK]D-Fenderkeithey have to come AFTER everything else
20:10.01spckwish i'd quit closing the channel
20:10.03thomasrronly outgoing calls work just fine the voipbuster trunk
20:10.08thomasrrwhat could i be doing wrong?
20:10.13Micho123Hi all, Can a peer in sip.conf be a part of several contexts
20:10.13Micho123?
20:10.22[TK]D-FenderMicho123: No
20:10.33spckyou could include the context of that peer in other contexts
20:10.41spcki think
20:10.52ziram19i have a voip provider connected to asterisk 1.4 that works fine with an ivr
20:11.16[TK]D-Fenderspck: Wrong scope, but I'm sure you are thinking the right goal
20:11.47ziram19when i migrate to asterisk 1.6.1 dtmf semms not understand correctly the extension that someone entred
20:11.47spckahh k
20:11.49Micho123[TK]D-Fender, I'm using asteris realtime...I need to change context based on dialed number...Could this be done with goto?
20:12.05[TK]D-FenderMicho123: HUH!?
20:12.22KobazMicho123: you answered your own question
20:12.28[TK]D-FenderMicho123: Your context is FIXED.  It does not change based on what you dial.  What you dial goes to THAT context period.
20:13.01KobazMicho123: you need to use goto, to jump to the desired context
20:13.11*** join/#asterisk mmlj4 (n=jkelly@ip70-171-94-246.no.no.cox.net)
20:13.21[TK]D-FenderKobaz: I wouldn't go saying that outright
20:13.33Kobazokay well, you don't need... but you can use
20:13.39Micho123Kobaz, and the CDRs will be saved on cdr table with the new context or with the peer context?
20:13.55KobazMicho123: the last context will be saved in the cdr
20:14.06Micho123Kobaz, let's see then
20:14.10Micho123Kobaz, thanks
20:14.22Qwellstupidnic: show me the call trace when it happens
20:14.25keith4[TK]D-Fender: oh, crap
20:14.26Qwell~pb
20:14.27infobot[~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://www.rafb.net/paste , or , http://bin.cakephp.org/
20:15.23ziram19there is someone that can response me?
20:15.27stupidnicQwell: http://pastebin.com/m63b65539
20:15.42stupidnicthat is a short version but it illustrates the issue I am having
20:15.48jplankno one else is seeing that IP attempting to make calls through your PBX that I was talking about yesterday? Between yesterday and today, it has hit every one of my clients at least 5 times
20:16.00stupidnicI hung up when it prompted me to record my name
20:16.11keith4[TK]D-Fender: after *everything* else? or after everything else in the [general] section?
20:16.16[TK]D-Fenderziram19: You haven't shown anything and we have no reason to suspect some generalized problem against 2 non-specific * versions
20:16.21jcapejplank: What IP?
20:16.36Qwellstupidnic: what version of Asterisk?
20:16.50stupidnicAsterisk 1.4.21.2~dfsg-3
20:16.51[TK]D-Fenderkeith4: REGISTER's always after everything else under [general] and before any peer entry
20:16.59stupidnicQwell: debian lenny
20:17.02keith4[TK]D-Fender: okay, fixing
20:17.10jplank93.190.143.10
20:17.19jplankcheck out http://trixbox.org/forums/trixbox-forums/open-discussion/sip-external-hack
20:17.22jplankI'm not the only one
20:17.24stupidnicI tried finding something in the changelog
20:17.30stupidnicbut nothing has jumped out at me yet
20:17.34ziram19fender: what can i show?
20:18.15jcapeDoes it try to login as sip user 123?
20:18.16[TK]D-Fenderjplank: That post is nonesense
20:18.28jplankmore or less right
20:18.34jplankbut I'm seeing the same thing
20:18.40jplank15 different PBXs
20:18.49ziram19when i tape 100 for example * understand XX and when i reconnect my astersik 1.4 all works fine
20:19.03jplanksame IP, same originating DID, same terminating number
20:19.05[TK]D-Fenderjplank: zirmProvide CONFIGS and call debug.
20:19.19[TK]D-Fenderziram19: Provide CONFIGS and call debug.
20:20.20[TK]D-Fenderjplank: there is no backup provided by that post and it proves nothing.  a dialplan context doesn't let people in.  Its just a place calls can land if sip.conf lets them in and was configured to point there
20:20.28keith4[TK]D-Fender: like so? http://pastebin.com/d3b6263bd
20:20.35[TK]D-Fenderjplank: A dialplan error has nothing to do with that.
20:20.47jplankno, I'm not saying hes hacking anything, if you look at the sip traces he provided, they guy isn't getting through
20:20.47*** join/#asterisk iratik (n=itariki@209.248.216.146.nw.nuvox.net)
20:20.51[TK]D-Fenderkeith4: Yes
20:20.54iratikIs there a way to clear the originate queue for AMI ?
20:21.16jplankI'm just finding it interesting that this same person is getting around the Internet so fast
20:21.18jplankfrom the same IP
20:21.23[TK]D-Fenderjplank: [May 12 05:30:51] VERBOSE[30399] logger.c: -- Executing [4312297134@from-sip-external:1] NoOp("SIP/93.190.143.10-09635110", "Received incoming SIP connection from unknown pee
20:21.25[TK]D-Fenderr to 4312297134") in new stack
20:21.43[TK]D-Fenderjplank: this seems to say he IS allowing anonymous calls and let it slide right on through.
20:21.57jplanknah, thats how trixbox works
20:22.07jplankwhen its gets rejected, thats the first line in the context
20:22.23jplankvery stupid, I agree, shouldn't setup the call
20:22.27[TK]D-Fenderjplank: "says he doesn't allow anonymous calls", "show an anonymous call coming in" = user error
20:22.39ziram19fender do you want an ssh access?
20:22.41jplanknot user error, trixbox error
20:23.05jplankI'm just saying crazy how I seen the same attempt from on 15 different of my pbxs, from the same IP
20:23.15keith4[TK]D-Fender: thanks, i'll have to wait to test it until tomorrow, though. that peer is 6 hours ahead of me, and probably in bed by now ;-)
20:23.22Qwellstupidnic: report a bug.  bugs.digium.com
20:23.23[TK]D-Fenderjplank: Which if its a bug in their interface for not doing what it was conceieved for falls flatly under the realm of "fuck the fucking GUI" :)
20:23.30thomasrr[TK]D-Fender: I am using voipbuster and requested a voip-in number
20:23.31jplanklol
20:23.35jplankagreed
20:23.40[TK]D-Fenderkeith4: Yes, better
20:23.43jplankthats why I stopped using trixbox a while ago
20:23.49jplankeverything was ass backwards
20:23.52thomasrrbut somehow i can call out with the account but not receive calls (nos ip notify msgs) from the voip-in number
20:24.20jplankI must say though, a lot of it was trixbox trying to take over freepbx (which is kind of backwards in its own right)
20:24.33stupidnicQwell: Okay. I will build from source just in case (unless you think it has been addressed)
20:25.14keith4[TK]D-Fender: I just noticed that I have listed "localnet=192.168.0.0/255.255.0.0" explicitly. Is this necessary? If so, should I list the other private IP ranges? (like... what if that client is NAT'd 10.x.x.x?)
20:25.22Micho123Kobaz, Getting the following warning ...http://pastebin.com/d1dac3ea9
20:25.27Micho123can you check plz?
20:25.46ziram19fender :you wan't an ssh access?
20:25.51[TK]D-Fenderkeith4: these are YOUR ranges, not THEIRS
20:25.52Kobazyou're sending the call to an extension/context that doesn't exist
20:25.59jplankhaaaaa last two posts on that thread is great, those people allowed anonymous SIP calls
20:26.05Kobazziram19: no he doesn't want ssh access
20:26.10jplankone person logged 2000 calls in over an hour
20:26.18keith4[TK]D-Fender: okay... so what if a NAT'd client is in that same range? is that a problem?
20:26.32[TK]D-Fenderkeith4: completely unrelated
20:26.39keith4ok, just making sure
20:27.14[TK]D-FenderMicho123: Telling you exactly what context & exten its looking for, what more is there to say?
20:27.43Micho123[TK]D-Fender, I think i have a priotity issue...1 sec
20:28.28iratik[TK]D-Fender: ?
20:29.12[TK]D-Fenderiratik: ?
20:29.45iratikIs there a way to view the AMI originate queue? or clear it?
20:30.18iratikWhen you issue an Originate command, you get back "succesfully queued" .  How do i manage, look at that queue?
20:30.57[TK]D-Fenderiratik: No clue, and if I did I might have answered you
20:31.08iratikThis must be a tough problem
20:31.12iratikthanks for your consideration though
20:31.20[TK]D-Fenderiratik: Please don't hold out on the assumption I'm going to answer everyone's questions
20:31.20*** join/#asterisk haryv (n=lanny@S010600a0c93f6f7e.vs.shawcable.net)
20:31.22Qwellstupidnic: no, it's a bug
20:31.48[TK]D-Fendercheckout time, later all
20:31.53iratik[TK]D-Fender: I do assume you are the god of asterisk ...
20:32.07Qwelliratik: Gods are supposed to be omnipresent.
20:32.11Qwelliratik: He leaves occasionally.
20:32.21iratikYou are good too i think
20:32.49Kobazheh
20:33.12iratikAny clue why my AJAM integration seems to be launch thousands of originate requests ... e.g. , i execute a single originate command , then the next time i issue originate, its like they are doubled up ... until 256 originate commands are being launched.
20:33.30iratikI'm not launching thousands of originates, but its like asterisk is receiving them
20:34.27*** join/#asterisk paulius (n=paulius@unaffiliated/paulius)
20:35.05Micho123Can someone take a look to my extensions table and see why the GoTo is not working well?  http://pastebin.com/d62a48b3b
20:35.07haryvHiya everyone. Got a new phone well, not new but anyway, having a heck of a time for it to except the aastra.cfg image file, or its UI changes on the 9133i. Getting a display message of Network disconnected with a default time of jan1 12:00. Cli says the phone is registered as soon as it logges in then, this display info pops up. I have tried it with dhcp and static ip address. I can log into the phones UI, make the changes to the networ
20:35.53haryvNot sure if these phones have a log output that can be put on pastebin.ca
20:36.00Micho123I'm getting Channel 'SIP/gw-in.ergatel.net-b7814c98' sent into invalid extension '028945551' in context 'On-net', but no invalid handler
20:36.00Micho123Scheduling destruction of SIP dialog '1817550057304200033721@80.169.210.180' in 32000 ms
20:38.00iratikI figured it out... turns out firebug had reached its max log limit
20:38.53*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
20:41.29iratikI didn't see the 1,000s of originate requests from my client side interface
20:41.51iratikturns out i was rebinding the same element over and over again , exponentially increasing the number of requests
20:42.34Kattyeppigy: episode 6!!!
20:42.46Kattyeppigy: 'Mudd's Women'
20:43.52eppigyMUDDS
20:44.33eppigyI am doing bitch work :[
20:44.53Katty:<
20:45.41eppigyyesh
20:45.57eppigythats why I am getting out of the pbx game
20:46.03eppigytoo much end user hand holding
20:46.49Katty:<
20:47.23iratikBy the way guys... not sure how relevant this is.. it has to do with guitar effects and asterisk
20:47.41Kobazhah
20:47.47Kobazthat sounds pretty relevant
20:47.50iratikYou can an amazing delay effect by doing 255 chanspy requests on a channel with a guitar and conferencing the result
20:48.02iratikit sounds like hell delay distortion
20:48.59iratikSomething like  for (i=0, i<255) Originate( Local/23646, 61388316250 ..) where 23646 is the conference you are listening in on, and 6138... stuff is the channel to monitor thet softphone which has guitar patched through the mic
20:48.59mmlj4have you tried adding an effects pedal to the mix?
20:49.14*** join/#asterisk jrhunt (n=jrhunt@69-223-16-247.ded.ameritech.net)
20:49.49iratikno.. i just ran across it accidentally when an ajax interface rebinded the same element on each request leading to an exponential number of bindings.. .which subsequently caused an exponential number of chanspy channels to be created.. maxing out the conference
20:51.16*** join/#asterisk ks3 (n=ks3@74.203.195.1)
20:51.57FlyserHi, I have a normal consumer ISDN connection with 3 MSNs. Is it possible to set up a mailbox at ${MSN}01 and something else at ${MSN}02? If yes, can you please point me to some documentation?
20:52.15ks3Is there a way to force ReceiveFAX to send a T.38 re-invite? Our provider is T.38 capable, but they expect us to reinvite with T.38 when fax tones are detected, and ReceiveFAX doesn't seem to be doing this.
20:52.18mmlj4make menuslelect... what's "module embedding"?
20:52.37FlyserI think the right word for this is direct dialing in
20:52.46Qwellmmlj4: it embeds the modules into the asterisk binary, rather than making them shared objects
20:53.05Qwellmmlj4: it's not useful to most people
20:53.33mmlj4fair enough, thanks
20:54.16Kattyeppigy: cargo safely aboard.
20:58.12eppigyKatty: :>~
21:03.00Deeewayneks3, enable faxdetect
21:04.33*** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw)
21:05.29Kattyeppigy: the definition of attractive has changed a bit, me thinks.
21:06.36ks3Deeewayne: I see reference to that in zaptel.conf for Zap channels... can it be used for SIP as well? Our only connection to PSTN is through a SIP provider.
21:06.38*** join/#asterisk sikanrong (n=sikanron@20.104.219.87.dynamic.jazztel.es)
21:09.38sikanronghey all, I've been reading the asterisk book and playing around with dialplans on my local net, so I'm familiar with a lot of these concepts, but I have a question:
21:09.43sikanrong<PROTECTED>
21:09.45eppigyKatty: how so?
21:09.48sikanrongexten => 1,1,Answer()
21:09.49sikanrongexten => 1,n,Dial(SIP/0034xxxxxxxxx)
21:09.56Kattyeppigy: they're not very attractive, to me.
21:10.09sikanronghow do you use the ITSP for routing calls to the PSTN?
21:10.30sikanrongi guess that's my question, or do you actually use zap and route stuff to a physical FXO?
21:10.36sikanrongnot sure how this stuff works quite yet
21:11.30haryvwas astralink sold to another company?
21:12.06haryvgot a message from my bank that another company aquired astralink andit was trying to use my credit card :)
21:12.46*** part/#asterisk sikanrong (n=sikanron@20.104.219.87.dynamic.jazztel.es)
21:12.52*** join/#asterisk sikanrong (n=sikanron@20.104.219.87.dynamic.jazztel.es)
21:14.12Qwellastralink?
21:14.26Qwellharyv: you mean Asterlink?
21:14.31Qwellyes, whois them.
21:15.26haryvI did
21:16.31haryvAnother one bites the dust. I cancelled my acount with this new company. Now need to find another company with a 1800 service.
21:17.08pauliusSo I've asked this millions of times, but what providers do experts like YOU use?
21:18.08stupidnicharyv: I use voicepulse and like them, not affiliated with them in any way
21:18.14*** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194)
21:18.36haryvvoicepulse has a pay as you go 1800 service?
21:18.47mmlj4haryv: teliax has that
21:19.05stupidnicharyv: not sure what exactly qualifies as pay as you go in your eyes though
21:19.26stupidnicthey charge you for the number monthly and then any incoming calls are just billed to your account
21:19.43stupidnicalthough they make you keep a balance on hand to deduct from
21:19.50stupidniclike an escrow account
21:20.06haryvI really liked the idea of a 1800 service. Dialing into my asterisk  box from a pay phone then dialing out really saved me a bunch of coin.
21:20.55stupidnicI will say this about voicepulse though... I was a bit annoyed when they yanked my local DIDs out from under me a few months back
21:21.22[TK]D-FenderToll-free origination often costs a lot more than termination does.
21:21.37haryvseems asterlinks site is still up. the sales and support extentions are not ringing.
21:21.42stupidnicapparenttly the CLEC that was providing them the numbers abruptly terminated the agreement they had so all DIDs in my area were lost
21:23.07haryvwho owns asterlink again?
21:24.10*** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no)
21:25.23bmoracaharyv: probably more expensive than it's worth.  lots of providers will charge you a surcharge for toll-free calls placed from payphones
21:25.29haryvkinda funny, the registered service is moved to hollywood california. Perhaps they layed off the techsupport and sales staff ;)
21:25.44haryvprobebly true.
21:26.48bmoracaPacWest, for instance, charges $0.80 for each payphone call, in addition to their normal rates...not really worth it
21:27.00haryvwow
21:27.11eppigyKatty: who is not? I am so lost :<
21:27.42haryvbetter to have a prepaid cell phone
21:28.03*** join/#asterisk HeXiLeD (n=H3X@unaffiliated/hexiled)
21:28.17Kattyeppigy: mudd's women.
21:28.31haryvI need to knock out this aastra 9133i phone no service issue before going on to something else.
21:33.07*** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net)
21:33.42Kattyeppigy: they're blonde, and stuff
21:33.56Kattyeppigy: too many sequins
21:36.01haryvkatty, are you a admin?
21:36.29*** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee)
21:36.52haryvhi jaytee, been a while ;)
21:37.13jayteehi haryv
21:38.03Kattyharyv: buwha?
21:38.05Kattyhugs jaytee
21:38.13jayteeman, those new Dell Mini netbooks are so sweet. they're like Hobbit laptops
21:38.21jayteehugs Katty
21:38.34Kattysweet like chocolate?
21:38.47jayteesweet like real small and real light
21:38.50haryvur still working at the zoo right ?
21:38.54jayteeyep
21:39.32haryvokay
21:39.38haryv:)
21:39.40Kattyharyv: you do not parse. please try again.
21:39.42jayteejust setup one of the Minis as an admissions scanning station with a USB bar code reader and our POS (Point of Sale, Piece of Shit, same difference) software
21:39.49*** join/#asterisk jdblack (n=jblack@pool-71-181-243-204.sctnpa.east.verizon.net)
21:40.26haryvI have put some of my IT work on hold. Market sucks :)
21:40.32haryvsome of it at least
21:41.31haryvjaytee, so what has been happening?
21:41.53jayteethe big money maker in today's IT market is server consolidation and "virtualization" ( a word I hold in contempt due to it's constant misuse and abuse)
21:42.05bmoracaharyv: aastra phones are garbage.  avoid at all costs
21:42.06jayteeharyv, same ole, same ole
21:42.12Kattyjaytee: i'll virtualize you in a minute
21:42.14drmessanojaytee: Im working on virtualizing at home
21:42.33drmessanojaytee: I moved my girlfriend in, and she sleeps next to my wife now.. in our slightly larger bed
21:42.44drmessanoIve saved 30% on energy costs
21:42.49jayteelol
21:43.00stupidnicheh
21:43.08haryvohh brother
21:43.34bmoracawe've actually noticed an increase in sales over the last two months
21:44.05Kattywoah
21:44.05Kattyhey now
21:44.44haryvbmoraca, sales in what?
21:45.00Kattyapparently larger beds.
21:45.07bmoracaharyv: general IT...computers, monitors, servers
21:45.11*** join/#asterisk goupil (n=goupil@2a01:e35:2f3d:7900:240:63ff:fedc:10e) [NETSPLIT VICTIM]
21:45.11*** join/#asterisk jasonwoot (n=some@bookit-dev.com) [NETSPLIT VICTIM]
21:45.11*** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com)
21:45.11*** join/#asterisk miloux (n=KVIrc@milu.rit.se) [NETSPLIT VICTIM]
21:45.11*** join/#asterisk Failrar (n=Failrar@coffee.ipv6.kaufmann.tc) [NETSPLIT VICTIM]
21:45.11*** join/#asterisk thuddwhirr (n=wolthuis@mimezine.com) [NETSPLIT VICTIM]
21:45.11*** join/#asterisk Talkradio (i=talkradi@linuxgeneration.ca) [NETSPLIT VICTIM]
21:45.11*** join/#asterisk styelz (n=yoohoo@m0o0.mooo.com) [NETSPLIT VICTIM]
21:45.11*** join/#asterisk viq (n=viq@unaffiliated/viq) [NETSPLIT VICTIM]
21:45.11*** join/#asterisk simond (n=simon@syria.uc.org) [NETSPLIT VICTIM]
21:45.11*** join/#asterisk Beave (n=beave@DCC.SEND.startkeylogger.000.telephreak.org) [NETSPLIT VICTIM]
21:45.30bmoracaharyv: for about 6 months, everyone was in repair mode...but we're seeing a lot more sales the last two months
21:46.12*** join/#asterisk Ziaeon (n=ziaeon@75-149-177-2-Miami.hfc.comcastbusiness.net)
21:46.42ZiaeonIs there a way to limit what extensions an incoming call can dial from an ivr?
21:46.57Kattyjust don't include it
21:47.34ZiaeonMy IVR has direct dialing enabled, you dial the extension you want, as long as that extension exists on the server, it dials it. Perhaps I am going about this the wrong way?
21:47.34bmoracaZiaeon: don't have a matching extension and use 'i' to inform user of such...like Katty said
21:47.50bmoracaZiaeon: what is it you don't want them to dial?
21:48.07ZiaeonLets say I only want them to reach the 1000 series extensions from the IVR direct dial, not the 2000 series.
21:48.12bmoracaZiaeon: if you only want them to be able to dial specific extensions, you'll probably need more specific pattern matching
21:48.28bmoracaZiaeon: then your pattern should be _1XXX instead of _XXXX
21:48.31ZiaeonSo instead of direct dial, I need strict pattern matching
21:48.43ZiaeonRemember im talking about incoming calls through an ivr
21:48.46ZiaeonFrom, say, my cell phone.
21:49.07Kattywonders what part of don't include them isn't sinking in ;)
21:49.12jayteeZiaeon, are you having them dial the digits at a prompt using Read() or WaitExten()?
21:49.24bmoracaZiaeon: let's start by having you pastebin your IVR context...
21:49.25Kattyapplies asterisk book to Ziaeon's noggin in hopes of osmosis
21:49.27bmoraca~pb
21:49.28infobot[~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://www.rafb.net/paste , or , http://bin.cakephp.org/
21:49.32bmoraca^^^^^
21:49.58ZiaeonI'm inferring that this is more of a problem with the way FreePBX generates IVRs.
21:50.04jayteeyeah, pastebin that sucker and let's see what's what
21:50.16jayteeFreePBX? OMG!!!
21:50.20ZiaeonDont whip me!
21:50.35jayteeI won't whip you, I'll just ignore you :-)
21:50.50bmoracaZiaeon: if you're going to use a GUI, you can't complain when said GUI doesn't let you out of the box it creates
21:50.59ZiaeonRoger
21:51.07ZiaeonOk, time to port this IVR to my strpped asterisk
21:51.11bmoracaZiaeon: nor can you complain when you can't find your own way out of that box :)
21:51.12Ziaeonthanks guys :|
21:51.22VaGoNeTaSbuddys
21:51.29VaGoNeTaSi got this error during the zaptel "make"
21:51.30mmlj4can I concatenate variables in extensions.conf?
21:51.31VaGoNeTaShttp://pastebin.ca/1423097
21:51.41VaGoNeTaSdoes anybody knows why i got that error?
21:51.49VaGoNeTaSits ubuntu 9.04 server
21:52.03jayteemmlj4, yes
21:52.55VaGoNeTaSi've installed the linux-headers-2.6.28-11-server which is my kernel version
21:53.03mmlj4I don't see what I want in the book :-/
21:53.23jayteemmlj4, what are you trying to do?
21:53.50[TK]D-Fendermmlj4: Set <-
21:53.55*** part/#asterisk Ziaeon (n=ziaeon@75-149-177-2-Miami.hfc.comcastbusiness.net)
21:53.59VaGoNeTaSthat's the version that supports this shitty Redfone quad
21:54.04VaGoNeTaS(zaptel)
21:54.21mmlj4I want to put several numbers into a string, separated by colons
21:54.42VaGoNeTaShttp://pastebin.ca/1423097
21:54.50mmlj4say, "$extension:$callerid"
21:54.57[TK]D-Fendermmlj4: Set <-
21:55.12mmlj4because AGI is choking on multiple variables
21:55.12[TK]D-Fendermmlj4: just DO IT
21:55.25mmlj4lemme
21:55.39VaGoNeTaStk any suggestion?
21:56.48VaGoNeTaSls
21:56.59Qwell. .. goats/
21:57.12jayteelol
21:57.22jayteepygmy goats
21:57.42[TK]D-FenderHoney Bunches of Goats
21:57.44[TK]D-FenderYUM!
21:59.55jayteeperfect example of where product testing and development suffers a communcations breakdown in a large company: the new Tums Smoothies are awesome as an antacid. The packaging they come in sucks
21:59.56*** join/#asterisk phl4kx (n=supervis@webmailserver.nisira.com.pe)
22:00.43VaGoNeTaSwell, THANK YOU SO MUCH guys
22:00.44VaGoNeTaSdamn
22:01.14jayteeVaGoNeTaS, sorry dude but my espanol is a bit rusty for reading all that make output in your pastebin
22:01.24VaGoNeTaScrap
22:01.26VaGoNeTaSdamn
22:01.36VaGoNeTaSim gonna format this shit
22:01.53*** join/#asterisk egypcio (n=egypcio@unaffiliated/egypcio)
22:01.55Qwellhow is that going to help?
22:03.03jayteeVaGoNeTaS, running such a new kernel is begging for compile issues. Jackass Jackoffalope is too damn new. Try Hardy Heron, at least it's an LTS release. Running bleeding edge distros just gives you roids, man.
22:05.01VaGoNeTaSthat's the kernel tha cames with the distro
22:05.17VaGoNeTaSso your suggestion is install an older version like 8.04-server'
22:05.20jayteeno shit Sherlock but why are you running something that was released LAST MONTH?
22:05.25Qwellor install DAHDI
22:05.48jayteehas Redfone updated their crap for DAHDI yet?
22:05.51VaGoNeTaSok im installing 8.04 server
22:05.59VaGoNeTaSQwell
22:06.04eppigyVIRTUALIZATION
22:06.09VaGoNeTaSdahdi 2.0.0 its the only version that supports Redfone
22:06.24Qwelland?
22:06.27VaGoNeTaSthe dahdi_dynamic_ethmf module
22:06.42[TK]D-FenderVaGoNeTaS: Use a more stable distro
22:06.53VaGoNeTaSi belive that is not supported by my kernel version so im installing ubuntu 8.04 server right now
22:08.24jayteeI don't care if Mark Shuttleworth went into space or not. I'm still not running a production system on anything other than RHEL or CentOS.
22:10.08eppigyYEAH SON
22:10.17jayteeDad? is that you?
22:10.21Qwelljaytee: Has Bob Young ever been in space?
22:10.23QwellI didn't think so.
22:10.39jayteeBob Young? nope, don't think so.
22:10.39eppigyWhen he takes a good dmt hit maybe
22:10.47[TK]D-FenderVaGoNeTaS: And be caeful because Ubuntu updates kernels and tons of shit behind your back which tends to break zaptel/dahdi.
22:10.48eppigyYOU KNOW WHAT I MEAN
22:10.56[TK]D-FenderVaGoNeTaS: You are ASKING for trouble using it....
22:11.00eppigyyes
22:11.01VaGoNeTaShmmm
22:11.03jayteeUbuntu, the "kernel of the week" distro
22:11.06VaGoNeTaSyou say ubuntu sucks?
22:11.08eppigyuse yum and exclude kernel updates
22:11.19bmoracaubuntu's fine as a desktop OS for tinkering around
22:11.21jayteeUbuntu uses apt not yum
22:11.28eppigyOH RLY?
22:11.30jayteebmoraca, agreed
22:11.32[TK]D-FenderVaGoNeTaS: I'm saying having your kernel change behind your back every week will leave you fucked every week.
22:11.40bmoracagentoo ftw
22:11.40eppigyim sayin use centos
22:11.44eppigywith yum
22:11.56jayteeeppigy, ah, that would be yum then
22:12.00[TK]D-FenderOk, martial arts time...
22:12.00eppigydont you dare crontradict me son
22:12.01[TK]D-FenderBBL
22:12.04[TK]D-Fender(very)
22:12.06jayteehave fun
22:12.32bmoracai fuckin hate linux...i wonder if asterisk would compile and run under cygwin...
22:12.40eppigywow dog
22:12.44eppigyhave you lost your mind
22:12.48*** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage)
22:12.48*** mode/#asterisk [+o leifmadsen] by ChanServ
22:12.54eppigylinux can be fun
22:12.59eppigyif you dont mind learnign something
22:13.06bmoracaonly if you're a masochist
22:13.07leifmadsenAsterisk 1.4.25 Release Candidate 1 (1.4.25-rc1) is now available for testing!
22:13.11eppigynegative
22:13.12mmlj4ok... Set(ZEXTEN={$XEXTEN}:{$YEXTEN}) seems to fail, or at least my AGI isn't getting it
22:13.16eppigyonce you actually know what youre doing
22:13.21eppigyit involves little pain
22:13.28leifmadsenmmlj4: you're using variables incorrectly
22:13.37mmlj4leifmadsen: enlighten me?
22:13.41leifmadsen${VARIABLE}
22:13.46mmlj4hrm
22:13.47leifmadsennot {$VARIABLE}
22:13.47jayteebmoraca, yeah but the Windows ntoskrnl.exe in W2K3 or earlier is such a latency pig it wouldn't be worth the pain
22:14.17bmoracajaytee: i wasn't being serious (about cygwin)
22:14.24jayteebesides, the asteriskwin32 port was originally an April Fools joke that someone got carried away with and it won't die a proper death.
22:14.40bmoracalol
22:14.56mmlj4that works better... but AGI still ain't being nice to me
22:14.58mmlj4:-(
22:15.30bmoracai learned a long time ago that timing sensitive applications + windows = fail...maybe with 2k8 Core mode (no GUI) would be acceptible
22:15.56bmoracai mean, if I can run asterisk well enough in ESXi, it should work on top of windoe
22:15.58bmoracaze
22:16.07jayteebmoraca, it might, haven't played with it myself. too busy
22:16.55bmoracasame here...who has time to test things out anymore anyway?
22:17.49VaGoNeTaSits about to finish
22:17.50jayteeon the other hand, I have no issues with linux, it's a fine OS when properly configured regardless of what distro you choose. I just prefer the RHEL or CentOS releases for server applications to Debian or Ubuntu.
22:17.56VaGoNeTaS(ubuntu 8.04server)
22:18.05VaGoNeTaSits the company requirement
22:18.09haryvim4centos
22:19.07eppigythe only thing that intrigues me about ubuntu currently
22:19.15eppigyis its desktop release
22:19.24eppigyand that is comes with ext4
22:19.33eppigy*it
22:20.05haryvIh8flakey customers
22:20.20*** join/#asterisk h3x (n=Hex@64.192.116.17)
22:20.20jayteeCanonical does a good job on pushing the envelope on the desktop. They've managed to move the desktop forward quite a bit over the last 4 years or so.
22:20.36jayteeGnome is still a major resource pig though.
22:20.50eppigyhow adre you
22:20.51eppigydare
22:21.09haryvguy sets appointment, then calls back a few hours says he cannot make it for that time period, sets it up for the next day, then he calls back and says he cannot make it saying he is going on a road trip.
22:21.33eppigytell him you will be in his living room until he gets back
22:21.40eppigymaking yourself at home
22:21.53haryvI am going to start asking for credit card deposits of customers do that to me.
22:22.11haryvIts a loss of several hours income of no work.
22:22.49haryvI call, leave a message..and another one does not call back. Whats the point?
22:22.49stupidnicQwell: latest trunk for 1.4 fixed the issue just in case you were wondering
22:23.16jayteehe probably wasn't, he's got enough to keep himself occupied
22:23.19tzafrir_laptophmm... that "solution" from the list didn't work well here. they forgot to escape the !
22:23.40haryvjaytee, ever work for your self before?
22:24.35jayteeharyv, yes I did, once upon a time but I like to eat so I had to choose between working for someone else and having food and money or working for myself and getting stiffed by customers.
22:25.28haryvthats true. Im wondering it it makes more sence just to work for a established company ;)
22:25.55h3xestablished companies are overrated
22:26.07haryvA well known vancouver communications company wants me to so sales for them. Will see what the meeting this saturday will unfold.
22:26.17h3xtelus? lol
22:26.22haryvnooooo
22:26.35jayteewell, I'm looking at doing it again soon as I'd rather work for myself than some of the idiots I've had to work for or with lately and I've gained business experience and learned more about making a contract that a customer can't weasel their way out of.
22:27.41*** join/#asterisk SaiSoma (n=SaiSoma@74.167.136.30)
22:28.03h3xFLACs are awesome.   I'm going to start all over with my music collection.  screw mp3
22:28.10jayteelol
22:28.30h3xi wish i could find a car amp that has S/PDIF inputs
22:28.53jayteeor a portable music device that plays FLAC files :-)
22:28.59h3xI think my archos does
22:29.13h3xbut, its pissing me off.  I'm assuming I will build a carpc
22:29.22jayteepretty sure my SanDisk doesn't
22:29.25h3xhahahah
22:29.28haryvh3x, a telus contractor wanted me to sign a 13 page contract. and, work on sunday. I said no.
22:29.30stupidnicI seem to recall that there was a patch to the ipod replacement software that enabled FLAC play back
22:29.31h3xyou should be lucky your sandisk will support mp3
22:29.35stupidnicI forget the name of it though
22:29.54h3xharyv, telus is a work of the devil
22:30.00h3xok fine its better than bell but
22:30.10h3xthey make devil kind of money
22:30.12stupidnicyeah... RockBox supports FLAC
22:30.31h3xwoah
22:31.23h3xahah
22:31.29h3xwell i have a archos 604 i think
22:31.35*** join/#asterisk Alborracho (n=chatzill@190.25.135.1)
22:31.43stupidnicYeah not supported by RockBox I don't think
22:32.05h3xargh
22:32.16Alborrachohi everyone, can someone help me with a SIP issue?
22:32.21jayteedrmessano, you still around?
22:33.13h3xMFKR....  Now they got a 3.5G archos
22:33.25h3xdamn it
22:33.28h3xthats it
22:33.48h3xyou know what, im not going to build a carpc.  first im going to set up a openwrt router in my car
22:34.04h3xso i can go wifi from the archos and bluetooth from my tomtom, to hsdpa
22:34.09drmessanoI am
22:34.11h3xits sad that i have to do it that way
22:34.11stupidnichacks into h3x's car
22:34.14jayteedon't forget the Flux Capacitor
22:34.16h3xhahaha!
22:34.23h3xits a murano it came with a flux capacitor
22:34.30stupidnichaha
22:34.40h3xits in the shop right now
22:34.43h3xapparently the fluxing died
22:34.46jayteedrmessano, know of any mini-itx boards that are using the Intel Atom?
22:34.53h3xran out of the 1.2 Gigawatts
22:35.03drmessanoOff the top of my head, no
22:35.16bmoracajaytee:  intel makes one
22:35.18stupidnicjaytee: there aren't any in that form factor to my knowledge
22:35.20jayteeI just messed with a Dell Mini today and I was impresed
22:35.29h3xthe computer died and was not reporting check engine issues
22:35.34h3xso my CVT tranny went to hell
22:35.35stupidnicplenty of micro-atx
22:35.43h3xamong other things
22:35.52jayteemicro-atx is too big for what I want
22:35.57Alborrachodoes someone know how to modify the "From" header in sip? i need to send it like this "From:<sip:c8oqz84zk7z@privacy.org>;tag=hyh8"
22:36.07h3xalbor: Edit the source code.
22:36.14*** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net)
22:36.21h3xwelcome to asterisk. hahaha.
22:36.30Alborrachoany function in special?
22:36.35h3xactually isnt there some append sip header thing
22:36.39jayteeI want to build an Asterisk Appliance and use the Atom processor. I'm going to put it in a shiny lavender colored case and put Trixbox out of business. :-)
22:36.48stupidnichmmm I wonder... it isn't clear from this bug report if this is in the 1.4 trunk or not
22:37.11h3xjaytee: screen print some strippers on the lid
22:37.16Alborrachoh3x: ok thanks
22:38.44bmoracajaytee: http://www.newegg.com/Product/Product.aspx?Item=N82E16813121383
22:38.53drmessanojaytee: I want to come up with a PBX in a brown case... call it the "Shitty PBX"... so when the boss says "Not spending that kind of money, just get some shitty PBX"
22:39.00drmessano"YAY, they asked for it by name!"
22:39.08jayteebmoraca, thanks dude!
22:39.08leifmadsendrmessano: lol
22:39.09h3xdrmessano: Now you sound like nortel's marketing with the DMS
22:39.13*** join/#asterisk lanning (n=lanning@nat/yahoo/x-5cc7e106559fd860)
22:39.16h3xShittySwitch
22:39.21jayteeLOL
22:39.28stupidnicbmoraca: thanks for that... I wasn't even aware of that
22:39.50bmoracathere's a fair number of them: http://www.newegg.com/Product/ProductList.aspx?Submit=ENE&N=+50001157&QksAutoSuggestion=&Configurator=&Subcategory=-1&description=intel+atom&Ntk=&CFG=&SpeTabStoreType=&srchInDesc=
22:39.55tzafrir_laptopjaytee, why not use a Via board?
22:40.00VaGoNeTaSyou guys were right i guess
22:40.06bmoracamore to come when nVidia's ION gets around, but those will be much more expensive
22:40.07VaGoNeTaScoz now is compiling as it should
22:40.09VaGoNeTaSthe zaptel
22:40.11jayteeit's got a PCI slot so it'll take a Digium or Sangoma card!
22:40.30VaGoNeTaSi've installed libpri, libfb, fonulator
22:40.34VaGoNeTaSand now fonulator
22:40.39VaGoNeTaSi mean zaptel
22:40.57bmoracajaytee: yeah, but the trick is finding a case that'll fit one and a PCI riser card that works properly and points the right direction.  Sangoma cards are smaller than Digium cards, so I'd recommend them, personally
22:41.09drmessanojaytee: or come out with a line of cheap routers and PBXs based on open source and call them the "So, ho?" series.. because they answer the question "Aint that a cheap ass piece of crap?"
22:41.12drmessano"So, ho?
22:41.45jayteeVaGoNeTaS, glad it's compiling for ya now.
22:41.56VaGoNeTaSyeah, actually it just finished
22:41.59VaGoNeTaSnow im doing the make install
22:42.01VaGoNeTaSim just testing
22:42.13VaGoNeTaScoz with dahdi-linux-2.0.0
22:42.14jayteebmoraca, I haven't read much on nVidia's ION.
22:42.22VaGoNeTaSi wasnt able to make calls to cellphones
22:42.24stupidnicpatches the source to include a feature I he wants
22:42.46bmoracajaytee: it uses Atom, but has a more powerful GPU component...marketed toward HTPC crowd, but way too expensive
22:43.11stupidnicYeah I was going to say it still uses the Atom
22:43.25stupidnicwe use Atom's for low power servers in the datacenter
22:43.33stupidnicdamn things are miserly on power
22:43.47stupidnic.3 amps @ 120V full load
22:44.07stupidnicI have some dual quad cores that pull 2+amps @120V full load
22:44.10bmoracastupidnic: what kind of servers?  Atom is an in-order only processor...would suck dick on pretty much everything...
22:44.54stupidnicbmoraca: we mainly use it for low impact things... tftp servers, dhcp servers, dns caching servers, etc
22:45.05stupidnicthings like that
22:45.19drmessanoI think moving to the Google Cloud is the best thing we can do to conserve power and save the planet...   Since if we screw this planet up any worse, Google has enough money to buy a new one
22:45.20jayteethe Dell Mini with an Atom runs Win XP pretty fast.
22:45.30bmoracaESXi virtualize that shit
22:45.41stupidnicdo you have any idea how much power google's DCs use?
22:45.48stupidnicI drive past one daily
22:46.01jayteeas fast as a 2.8ghz P4 just by rough "feel"
22:46.14stupidnicthere are 65 1Megawatt what generators surrounding the place
22:46.31stupidnicthat gives some idea of how much power they are using in the place
22:46.46drmessano"what" generators?  Must be what powers the search indexes
22:46.58stupidnicheh
22:47.23stupidnicthat is one DC alone sadly, there is another one across the street with even more generators
22:47.26drmessanoYahoo still uses "huh" generators
22:47.46pmhaddadlol
22:47.50jayteeand MS uses "WTF?" generators
22:48.10stupidnicgoogle tries to hide how much their DC use resources wise by puttting them in shell companies so they don't have to list them on their SEC filings
22:48.18drmessanohttp://search.yahoo.com/search?p=search&fr=yfp-t-501&toggle=1&cop=mss&ei=UTF-8
22:48.39drmessanoResult #1 for the term "search" is yahoo.. 2nd is google
22:48.40drmessanoLies
22:49.06bmoracawhich is interesting, because Yahoo uses google's engine now :P
22:49.12stupidnicgoogle is ranked the number one most visted site in the world
22:49.35stupidnicI wish I had their page rank :)
22:49.40*** join/#asterisk qdk (n=qdk@0x573d8dd1.bynqu1.dynamic.dsl.tele.dk)
22:49.54drmessanoI wonder if Google IT sets google.com as everyones home page
22:50.00bmoracastupidnic: and Al Gore has the single largest electricity bill of any one person in the US (or pretty close to it).  hypocrisy ftw!
22:50.14stupidnicbmoraca: hah
22:50.19pmhaddaddrmessano, google doesnt have internal it :P
22:50.26KavanSdrmessano, not sure...pretty sure they maintain their own equipment
22:50.37KavanSthey have a repair department, but I heard it was pretty loose
22:50.59jayteeI still love Google. If I search for a recipe for some kind of dish, say gazpacho or something I'll get like 4000 hits that are usually relevant. If I search on MSN I get, "Here's your 3000 porn sites!" "Um, I didn't want porn! I wanted a recipe!" "Sure you do! Everyone wants porn!" "No!!!!" "Connection timed out"
22:51.13drmessanoHow would google be any different than any other large corporation.. Sure they have an IT department..
22:51.42KavanSdrmessano, I don't think that they have standard limitations of access placed on the local machines
22:52.01KavanSbut I could be wrong...
22:52.06drmessanoSo they don't need basic desktop support?
22:52.12drmessanoGimme a break
22:52.23pmhaddaddrmessano, i was jokin
22:52.23stupidnicthey use linux on the desktop
22:52.49stupidnicand no.. that wasn't a joke
22:52.51stupidnic;)
22:52.52pmhaddaddrmessano, i did have a friend that worked there last summer, and i dont think he ever mentioned IT
22:53.05KavanSyeah, I don't think they have the "standard" it dept
22:53.08stupidnicthat's because nobody that works there actually does any work :)
22:53.08pmhaddadyeah
22:53.10drmessanoROFL
22:53.12pmhaddadhahaha
22:53.13KavanShaha
22:53.13KavanSyeah
22:53.21KavanSthey make money I guess
22:53.22pmhaddadjust 20 percent project
22:53.40KavanSyeah, the inherent problem will come down to...
22:53.43KavanS"everyone is not a genius"
22:53.48stupidnica buddy of mine visited the campus a year or so ago... and was like "do you people ever do anything work related?" :)
22:53.57KavanSsooner or later it will happen...then you will have this excess flow of people fucking off, who really don't produce results
22:54.16drmessanoSo no one maintains antivirus on desktops, patching, switches, routers, cabling, basic desktop issues?
22:54.20drmessanoCome on now, people
22:54.40stupidnicdrmessano: they use llinux... they don't get viruses! :)
22:54.43KavanSdrmessano, how many "unsavvy" people do you think google is really hiring?
22:54.53rob327i'm sure they have network guys
22:54.54KavanSI mean you get the accountants and paper pushers...
22:55.00drmessanoKavanS: They dont hire people to work on computers all day, #1
22:55.15KavanSwho knows lol
22:55.26KavanSI'll ask the guy I went to highschool with....I guess he does some shit for the gtalk dept.
22:55.27stupidnicmy cousin is interning there this summer... I will find out and let you all know :)
22:55.38drmessanowow
22:56.28drmessanoThe thought that a company even 1/100th of their size could get by without basic IT support is even mind boggling.. Not sure why this doesnt make sense to you guys
22:56.38drmessanoOf course Google has an IT dept
22:56.46drmessanoAs does Microsoft, and yahoo, and even Digg
22:56.49KavanSbecause IT support is like "check out" staff at the supermarket
22:57.00KavanSit's going to be less in demand as time goes on
22:57.21KavanSand I work in IT support
22:57.23drmessanoJust because google is a tech company doesnt mean everyone works on their own fucking computers
22:57.29stupidnicKavanS: you poor bastard
22:57.30bmoracaKavanS: just because someone is a programmer does not mean they can maintain a global IP network
22:57.40drmessanobmoraca: Exactly
22:57.43KavanSbmoraca, indeed, I'm not saying they entirely maintain it
22:57.53KavanSbut it's not like you are going to need the same size of IT support as you'd need 10 years ago
22:58.03KavanSat least that's my opinion
22:58.04bmoracaKavanS: in addition, the vast majority of Google's employees are sales staff and other operational people.  not programmers.
22:58.43KavanSright
22:58.54KavanSI just doubt it is as large as you'd expect
22:58.56bmoracaKavanS: Google is the same as any other large company.  they have a dedicated inhouse department for IT and probably another one for IS.  too many hands in the pot, and all that
22:59.00KavanSI'm not saying it doesn't exist
22:59.11stupidnicI remember a job posting for google from many years ago... it specifically stated that support staff needed a solid background in desktop linux, as well as openoffice
22:59.11drmessanoKavanS: I doubt its any smaller than any other company their size would have
22:59.45KavanSyeah I saw some documentary on g4tv and it showed people using their choice of OS
23:00.22drmessanoAgain, they have the same IT needs as everyone else and the non-IT staff, which includes the programmers and guys that make the "product"
23:00.49stupidnicman
23:00.53KavanSlol I just don't think it's the same size
23:00.57stupidnicthat pico-itx ion board is TINY
23:01.08KavanSbut I'll ask and find out....who knows I could be wrong
23:01.23drmessanoWhy are they gonna pay $75 an hour for a programmer who could be writing code, to work on his own PC when a $20 an hour Workstation specialist can be doing it?
23:01.28drmessanoThat makes ZERO sense
23:01.44stupidnicdrmessano: we have conceeded the point... you won... move on :)
23:01.54KavanSI don't think he's 100% correct...
23:01.57KavanSlol
23:01.58bmoracai want a job maintaining Google's network...i think that'd be all kind of fun
23:01.59KavanSsorry.
23:01.59stupidnicshhh
23:02.06stupidnicdon't feed him any more :)
23:02.09drmessanoKavanS: I dont think youve worked in a large company before
23:02.17drmessanoKavanS: So move on
23:02.30KavanSdrmessano, right...what is your definition of large?
23:02.41stupidnicbmoraca: they make their own servers and switches
23:02.54drmessanoKavanS: Greater than 35,000 employees?
23:03.15KavanSnope
23:03.34drmessanoWhat about 10,000?
23:03.51bmoracastupidnic: they use commodity parts for their servers, but I HIGHLY doubt they make their own switches and routers.
23:04.05stupidnicbmoraca: then you would be incorrect
23:04.09KavanSdrmessano, yep
23:04.26bmoracastupidnic: source.  doesn't make sense for them to rewrite what other companies have already largely perfected.
23:04.32stupidniccost
23:04.37stupidnic10Ge is stupid expensive
23:04.48stupidniceven from comapnies like forece10 and cisco
23:05.23drmessanobmoraca: They make their own LARGE switches.. Datacenter size stuff.. Because in the quantity they need, and demands they have, they can do that.. But not smaller stuff it would make more sense to just buy
23:05.40drmessanoI dont think its even component level
23:05.45KavanSdrmessano, what large firms do you work at?
23:05.57alrsGoogle designs their own motherboards and has Gigabyte build them
23:06.09drmessanoI worked for 10 years doing IT a company with about 50,000 employees
23:06.13stupidnictheir servers are CRAZY smart
23:06.14drmessanoin a*
23:06.38KavanS50,000 at an IT company?
23:06.43KavanSor IT dept. at a non-IT focused org?
23:06.44stupidnicthey incorporate a 12V battery backup on the server itself with the 5v pull down built into the server's motherboard
23:06.44drmessanoNo
23:06.48KavanSright...
23:06.52stupidnicits brilliant
23:06.53drmessanoI didnt say "AN IT COMPANY"
23:07.03alrsBITCH YOU AIN'T NO NERD
23:07.14stupidnicespecially when you consider how much it would cost to install a Powerware UPS large enough to support the load of a typical DC
23:07.35bmoracaKavanS: you seem to think that Google is an IT company...it's not.  It's an IS company...very, very, very different.
23:08.01KavanSlol
23:08.15stupidnicbmoraca: not "concrete" proof but pretty close even if the info is dated
23:08.17stupidnichttp://gigaom.com/2007/11/18/google-making-its-own-10gig-switches/
23:09.04*** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri)
23:10.21bmoracastupidnic: that's a little different.  they're not building their own switches...they're buying the components and assembling them themselves.  "building" implies designing the ASIC and switching engine
23:10.33stupidnicbmoraca: symantics then
23:10.43bmoracastupidnic: i like semantics :)
23:11.01stupidnicI don't buy components to connect my servers together... I buy Cisco :)
23:11.06stupidnicsee my point? :)
23:11.08drmessanoYou poor bastard
23:11.25stupidnicmy two 6509s disagree
23:11.53drmessanoSo?
23:12.05bmoracaif it wasn't so damn loud, i'd use my 7204 as my DSL router at home...
23:12.38stupidnicI could have easily have said Black Diamonds, its the same as the OS jihad, or any other comparison between differing factions
23:12.49stupidnicpeople like what they know
23:13.00stupidnicI prefer Cisco because I know IOS well
23:14.31drmessanoSure, but making a comment to the effect of "Look at me, I have two 6509s" doesn't offer anything to the conversation
23:14.45bmoracai've been wanting to get my hands on some NetScreens in to see how they compare with the Cisco ASA, but haven't had time, really
23:15.00stupidnicNeither did your "you poor bastard" comment which was out of context and really pointless
23:15.00drmessanoLike if you told me Linksys phone sucked and I said "My 100 SPA-941s are blinky"
23:15.51drmessanostupidnic: It wasnt out of context.. maybe you dislikes it, or didnt agree with it, but I urge you to look at the context of something being "out of context" before offering that
23:15.58drmessanodisliked*
23:16.06stupidnicunless I misconctrude the intent or direction of your comment
23:16.59mmlj4os mispselled it
23:17.22stupidnicOr I could have wasted more bandwidth with the grammer/spelling police
23:17.27jayteeI've misconstrued before but never miscontruded. I'll have to try that sometime
23:17.31drmessanogrammar?
23:18.00mmlj4stupidnic: just joshin' ya
23:18.15stupidnicmmlj4: must have missed the smiley
23:18.35bmoracai saw no smiley
23:18.50stupidnicsarcasm is a dying art apparently
23:19.00bmoracasarcasm doesn't work over the interwebs
23:19.03bmoracait's incompatible
23:19.12stupidnicbut its a series of tubes! it must
23:20.17*** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman)
23:21.18KavanSdrmessano, my only point is that I think programmers (in my humble opinion) and IT personnel in general...take better care of their equipment than someone who is not as familiar with the inner workings
23:22.21*** join/#asterisk tanner (n=tanner@unaffiliated/tanner)
23:22.45drmessanoKavanS: Google is sales and programmers for the most part.. and programmers are NOT IT people and no, they don't take better care of their PCs.. Any "IT personnel" google would have are support personnel.. Their business is not IT..
23:23.04*** join/#asterisk smeegs (n=smeegs@2a01:348:153:17:211:43ff:fea3:521d)
23:23.13drmessanoUnless you're 75 years old and think that all "computer people" are the same
23:23.18KavanSlol
23:23.22KavanSprogrammers do take better care...
23:23.27KavanSand if not, they know better
23:23.32KavanSat least the programmers I've worked with
23:23.36KavanSmaybe you've spent some time at MS though...
23:23.37drmessanoI can tell you for a fact they dont
23:23.43drmessanoNope
23:25.04alrsI'm with drmessano
23:25.16KavanSok, I'm the exception then...lol
23:25.24KavanScheers :-)
23:25.44tannerdoes asterisk support sccp?
23:25.46alrs"ok, to ssh to the server we're going to use putty."
23:25.51Alborrachohow can i compile just one module? i dont want to recompile all asterisk
23:26.07alrsoh wait, no it was scp
23:26.26alrsmoral of the story was that I had to tell the programmer five times, "no you can't double-click that, it is a commandline application"
23:26.34KavanSwtf?
23:26.41KavanSjesus christ...
23:26.47KavanSI must be the only one who works with intelligent people
23:26.55KavanSor I'm a complete idiot myself...
23:26.56drmessanoThey not stupid
23:26.58KavanSmust be the latter
23:27.00drmessanoGRRR
23:27.03drmessanoThey're not stupid
23:27.12drmessanoThey're very intelligent
23:27.16drmessanoWrite good code
23:27.38drmessanoBut CAN'T fix workstation issues, or program a switch, or a router, or make a patch cable, or trace a drop
23:27.43drmessanoetc etc
23:27.57KavanSI don't think that's absolute by any means
23:28.27alrsa lot of programmers are tool-users
23:28.41alrsthey melt down when you take away their IDE or GUI db client of choice
23:29.20tannercat + redirection FTW
23:29.32stupidnicAlborracho: did you compile from source?
23:30.38Alborrachoi compiled and installed asterisk from sources
23:30.48Alborrachobut i made a change in cha_sip.c
23:30.49pauliusBad programmers are tool users.
23:30.55pauliusAnd some programmers are just plain tools.
23:30.55Alborrachoi dont want to recompile everything
23:30.56stupidnicOkay... then just run make again
23:31.11stupidnicas long as you don't do a make clean it will only recompile that one app
23:31.13Alborrachojust the module and the do a cp
23:31.21Alborrachoohh
23:31.23Alborrachonice
23:31.26Alborrachothanks
23:31.33stupidnicI just did this 5 minutes ago
23:31.55stupidnicI did do a make install though because my module as compiled into asterisk (followme)
23:32.04stupidnics/as/is
23:32.24stupidniclooks at mmlj4
23:44.17*** join/#asterisk propellerhead (n=yogurt2u@host1.190-30-31.telecom.net.ar)
23:48.28mmlj4?
23:48.58*** join/#asterisk aces1up (n=signup@ip70-173-52-152.lv.lv.cox.net)
23:49.04mmlj4stupidnic: joke over, time to get some work done
23:49.08aces1upwhat a good brand name for ata's?
23:49.26mmlj4depends
23:49.34mmlj4you want to provide dialtone, or accept it?
23:49.58stupidnicsweet... the patch works
23:50.10aces1upjust need something that gets good voice :)
23:50.14aces1upjust using for a home net.
23:50.26aces1upbut want it to be reliable, want to plug a cordless phone into it.
23:50.40aces1upunless you know any cordless voip phones that are good for less than 100.00
23:50.50mmlj4aces1up: are you plugging phones into the ATA? or are you pluggin the ATA into the wall to get AT&T's phone service?
23:51.01aces1uppluggin phones into ata.
23:51.02mmlj4ok, fine... linksys
23:51.19aces1uplinksys?
23:51.31aces1uphrmm. ok but haven't had much luck with their routers :)

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