irclog2html for #asterisk on 20050530

00:00.02cochiyou could also do extreme skydiving one day, climb the K2 the other, go to death sea, go to marian rift. and then die of some obscure oxygene-problems no doc ever seen
00:01.03mishehuhmm...  think I'll pass up on that idea, cochi
00:01.11Klarmishehu: That's too bad :(
00:01.18mishehuKlar: why is it too bad?
00:01.33opus__can you specify a longer column width when you do various manager api?
00:01.36cochireally? worth dying a strange death ? ;)
00:01.49opus__I get things like Name/username being truncated
00:01.56mishehucochi: better way:  death by snu-snu
00:02.11cochi;))
00:02.23cochiis there a better way of dying ;)
00:02.57mishehuonly by snu-snu
00:02.58mishehuheh
00:03.25cochisnusnu at death sea + the k2 ;)
00:03.41mishehuno no, snu snu can only happen on the planet of the amazon women
00:03.43opus__death by asterisk debugging
00:03.57mishehuas seen as in futurama's "amazon women in the mood" episode
00:03.58cochidon't they got any death sea + k2 there? boring :(
00:04.09mishehudead sea, not death sea.
00:04.14cochiflat as the netherlands eh
00:04.16cochiuhm k
00:04.21mishehuheh.
00:04.24cochideath sea'd be spooky. i admit that
00:04.28Klarmishehu: I've been to other places before and it wasn't so great
00:05.15mishehuKlar: well, without bad, there is no good, right?
00:05.16*** join/#asterisk outsidefactor (chrismarti@203-206-241-250.dyn.iinet.net.au)
00:05.41*** part/#asterisk outsidefactor (chrismarti@203-206-241-250.dyn.iinet.net.au)
00:06.42mishehubah.  need to make a copy of my reserve duty exemption card
00:07.13mishehuso much to do, no time to play with * head
00:20.34*** join/#asterisk mountie (~mountie@CPEdeaddeaddead-CM000a739acaa4.cpe.net.cable.rogers.com)
00:25.42*** join/#asterisk newbien (~e@116.242.33.65.cfl.res.rr.com)
00:26.08tzafrir_laptopmishehu, can you think of an interesting asterisk demo?
00:26.49tzafrir_laptopInteresting stuff easy to demo over a remote phone and stuff that "normal PBX-s" won't do
00:27.22tzafrir_laptopE.g.: any nice, interactive AGIs?
00:27.46cochinightynight
00:27.57*** part/#asterisk delphiuk (~Owner@host217-43-97-191.range217-43.btcentralplus.com)
00:28.06*** join/#asterisk wwalker (~wwalker@wwalker.sustaining.supporter.pdpc)
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00:28.37wwalkerIs there a way to control the rate / duration of DTMF signals sent via SendDTMF?
00:29.48Nuxitzafrir_laptop, I can play poker and blackjack with my AGI.
00:31.07tzafrir_laptopNuxi, where can I find that AGI?
00:31.23NuxiI think they're in phpagi examples
00:32.25blopanyone who's using smsq --process ? :)
00:32.36NuxiHmmm. Nope.  I can pastebin them.
00:32.57tzafrir_laptopplease do
00:35.27Nuxiok, this is deck.php:   http://pastebin.ca/12930
00:35.32*** join/#asterisk wrtchd (~dentont@c-24-0-114-212.hsd1.tx.comcast.net)
00:36.05mishehutzafrir_laptop: been meaning to write a couple agis myself, but haven't had time to fool around with any.
00:36.10*** part/#asterisk dersteer (~travis@24-231-151-119.dhcp.aldl.mi.charter.com)
00:36.14wrtchdGood Evening, it is a fine day here in Dallas.
00:36.31opus__manager api is just completely worthless
00:36.50wrtchdDoes anyone have a preference in a perdictive dialer that runs on top of Asterisk?
00:36.52opus__IAXPeers doesn't send ActionID so I need to patch my code to expect that, and its really hard
00:36.53*** part/#asterisk iheartcanada (~iheartcan@lfc.tor.istop.com)
00:37.18Nuxitzafrir_laptop, that's took much work.  Try this:  http://24.223.0.9/games.tgz
00:38.36Nuxirequires phpagi
00:39.40*** join/#asterisk wrtchd (~dentont@c-24-0-114-212.hsd1.tx.comcast.net)
00:39.44*** join/#asterisk Samoied (~Samoied@200-138-237-177.fnsce7004.dsl.brasiltelecom.net.br)
00:40.22Nuxiexample use of poker class http://pastebin.ca/12931
00:40.26Nuxiblackjack is the same.
00:42.40opus__games?
00:42.53Nuxipoker and blackjack.
00:42.57opus__cool
00:43.28NuxiI don't know why I didn't send these to be included in phpagi, must have forgotten to put them int the tgz.
00:45.36NuxiI'm sure once asterisk supports video, more games will come.  I'm waiting for pong and tetris.
00:48.27Nuxigotta go...
00:51.11*** join/#asterisk RoyK (~roy@179.80-203-29.nextgentel.com)
00:51.23RoyKhelo
00:51.32*** join/#asterisk Ariel_ (~Ariel@dsl-20-177.cofs.net)
00:51.50*** join/#asterisk iq (~iq@207-224-100-126.omah.qwest.net)
00:54.54RoyKshit
00:55.19RoyKsomething's fscked up in app_queue
00:57.21cluecon[file]big surprise there
00:58.16*** join/#asterisk bjohnson (~bjohnson@ip219-172.dsl.istop.com)
00:58.35cluecon[file]:)
00:59.04RoyKcluecon[file]: see http://bugs.digium.com/view.php?id=4318
00:59.44*** part/#asterisk Samoied (~Samoied@200-138-237-177.fnsce7004.dsl.brasiltelecom.net.br)
01:03.03opus__is it possible to invoke manager api calls from the CGI?
01:03.27opus__i'm sorry
01:03.28opus__from the CLI
01:08.37wwalkerOK, does anyone have wav or gsm files of the DTMF tones?
01:09.14*** join/#asterisk pdracevich (~bob@smtp.aucklandtax.co.nz)
01:09.36pdracevichhello all has anyone used idefisk?
01:10.03pdracevichwhen I ring someone it dials and connects but does not give me a ringing sound
01:13.27wrtchdI would just like someone to suggest a predictive dialer...so I can get to the point where I have a problem with not getting a ringing sound
01:21.43*** join/#asterisk Sedorox (brandon@sedorox.staff.smartserv)
01:27.57*** join/#asterisk Nukemizer (~Nuke@160.7.249.15)
01:30.47NuxiIt's possible to call the manager api from AGI.
01:31.44*** join/#asterisk MrBelvedr (~tt@ip68-227-218-250.dc.dc.cox.net)
01:32.05*** join/#asterisk Inv_arp (junya@adsl-3-248-220.mia.bellsouth.net)
01:33.51*** join/#asterisk WilliamK (~wkeller@c-24-0-130-177.hsd1.tx.comcast.net)
01:35.52NukemizerI hae been looking and experimenting without success and am hoping for some guidance. I want to "read" a variable and play two prompts in the "read" process. Can this be done ?
01:35.58Qwellany bug marshalls around by chance?
01:36.08cluecon[file]Qwell: maybe.
01:36.20MikeJ[Laptop]?
01:36.25Qwellbug 4403, I don't really know what to do, if I'm correct in my thinking
01:36.30Nukemizerexten => s,8,Read(Selection,greeting1,digits/3,1)    "if you want to listen again" "press3" -- all the while listening for digit three
01:36.35cluecon[file]not that one...
01:36.48cluecon[file]oh just that
01:36.56QwellI'd like to fix it, but I need help/advice...
01:36.58MikeJ[Laptop]Nukemizer, background
01:37.39NukemizerMikeJ.. thank you looking it up
01:37.50QwellI have a very good feeling it was broken before I touched anything...
01:38.14cluecon[file]I'd rather stay away from that ODBC voicemail junk
01:38.23cluecon[file]it's ugly
01:39.32MikeJ[Laptop]well.. try it without your patch and find out ;)
01:40.26QwellMikeJ[Laptop]: well, I don't even know how to set it up :p
01:40.38Qwellbut, am I correct in my thinking, about the name mismatch?
01:41.03Qwell(and the two for loops...whats up with that?)
01:41.13MikeJ[Laptop]honestly... I didn't really look <shrug>
01:41.18Qwell(unrelated, but...weird)
01:41.44cluecon[file]so very very ugly...
01:41.46cluecon[file]just like Mike!
01:43.58JuggieQwell, when i left another message
01:44.00Juggieit worked fine
01:44.04Juggieso i'm not sure whats up
01:44.20Juggiei did change voicemail to be only wav between the msg that woudnt work
01:44.22Juggieand the one that did
01:44.26Juggieso that may be why? no idea
01:47.46*** join/#asterisk valence (~valence@Quebec-HSE-ppp230300.qc.sympatico.ca)
01:47.56Qwellit probably won't make it in anyhow
01:49.47*** join/#asterisk angler_ (~angler@suid.digium.com) [NETSPLIT VICTIM]
01:50.30MikeJ[Laptop]Qwell, here's the trick on getting smthing to make it...  get it tested up the wazoo....
01:50.52MikeJ[Laptop]I will do my best to get it through code and arcitecture review
01:51.05QwellMikeJ[Laptop]: nobody really cares enough about voicemail in odbc to test it it seems, honestly...
01:51.26MikeJ[Laptop]well.. the best you can at least
01:51.38MikeJ[Laptop]and get testing comments into the bug
01:51.52QwellJuggie: How difficult was this to setup?  Mind PMing me to help me out?
01:52.17Qwellit seems pretty useful
01:52.36*** join/#asterisk NewSole (dave@i216-58-44-245.avalonworks.net)
01:52.44MikeJ[Laptop]also look at 1188
01:53.11MikeJ[Laptop]as I hope to push that one in soon too
01:53.19*** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk) [NETSPLIT VICTIM]
01:53.43Qwellwell, if it were in a db, the fs wouldn't really slow down
01:53.55MikeJ[Laptop]look at the patch
01:54.01Qwelllooking
01:54.08MikeJ[Laptop]it makes the lookup optionally on a hash
01:58.00*** join/#asterisk Druken (~dskj@CPE00119539b9cc-CM000e5cde4ca2.cpe.net.cable.rogers.com)
01:58.04Drukenhello all
01:58.39Drukenhas anyone had a problem with pap2's only ringing once? after the first ring, the call is gone ?
01:58.58MikeJ[Laptop]search the bugtracker
01:59.09MikeJ[Laptop]there are 1 or 2 pap2 issues in there
01:59.14MikeJ[Laptop]I don't recall what for
02:00.25Drukenwhere's the bug tracker again? hehe
02:02.02*** join/#asterisk bprice20 (~brandon@cpe-24-195-22-108.nycap.res.rr.com)
02:03.05Qwellbugs.digium.com
02:03.49*** join/#asterisk remmo (~rem@smack.isp.net.au)
02:13.51QwellMikeJ[Laptop]: Are you saying to use the functions in that patch in mine (and extend my patch to cover all of vm in odbc?)?
02:16.10MikeJ[Laptop]no
02:16.11*** join/#asterisk Tetraboy (~we@h9.195.40.162.ip.alltel.net)
02:16.19MikeJ[Laptop]I was saying it was interesting
02:16.33MikeJ[Laptop]and marginally related..
02:16.37MikeJ[Laptop]so take a look
02:16.48Qwellahh
02:17.50Qwellyeah, it certainly does make things nicer
02:20.16QwellI especially like removing the "eh...the directory might already exist...but we don't care, we'll just let it error" stuff.  heh
02:20.48cluecon[file]waddddddddup
02:20.50Qwell/location = ribs
02:21.09Qwellemerging away the unixodbc stuff
02:21.17cluecon[file]oh no, not emerging!
02:23.03Nuxishouldn't it be unmerging or demerging or something?
02:23.25Qwellno, just emerging
02:26.40*** join/#asterisk apardo (~apardo@80.26.160.197)
02:27.03Qwellis it normal to have a large bruise after having blood drawn, two days previously?  heh
02:27.47NukemizerMikeJ, Thanks for the direction to "Background"  Just what I needed :)
02:29.30MikeJ[Laptop]:)
02:31.10cluecon[file]to all those in the world who want everything for nothing I say, "screw you"
02:31.39MikeJ[Laptop]heh...
02:31.44MikeJ[Laptop]your in a good mood
02:31.52Qwellcluecon[file]: thank you sir, may I have another?
02:32.01cluecon[file]I read asterisk-users posts
02:32.09cluecon[file]they make me mad
02:32.21Qwellmy condolences
02:32.27TetraboyCan multiple phones be hooked into one FXS, and work like in a regular house?
02:32.53Qwell~ren
02:33.01Qwellguess not
02:33.17JuggieTetraboy, yes
02:33.18cluecon[file]Tetraboy: depends on how much power the FXS port puts out
02:33.19QwellTetraboy: You can on some cards.  More usually work if the phone has its own external power supply
02:33.21cluecon[file]but usually yes
02:35.00MikeJ[Laptop]so that's a definate maybe?
02:35.01MikeJ[Laptop]:)
02:35.08TetraboyWhat's the cheapest way to get 1 FXO, and 1 FXS?
02:35.22MikeJ[Laptop]sipura spa ... ummm
02:35.31MikeJ[Laptop]can't remember the model #
02:35.57Qwell3000?
02:36.13MikeJ[Laptop]I don't know...  look on the website
02:38.38TetraboyI know which one your talking about.
02:38.50MikeJ[Laptop]good then
02:39.59*** join/#asterisk Drew2006 (Drew2006@66-190-252-14.dhcp.mdfd.or.charter.com)
02:40.11*** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode)
02:40.53Drew2006Can anyone help me with a firmware problem i am having with a packet8 proprietary voip adaptor?
02:43.13*** join/#asterisk NewSole (dave@i216-58-44-245.avalonworks.net)
02:44.26MikeJ[Laptop]packet8
02:44.41MikeJ[Laptop]this is #asterisk not #packet8
02:44.41NivexDruken: Doubtful
02:44.47Nivexerr
02:45.01Nivexno wonder my tab complete didn't work... he left.
02:45.11MikeJ[Laptop]who?
02:45.20MikeJ[Laptop]o
02:45.35MikeJ[Laptop]he's a jackass anyways
02:45.39MikeJ[Laptop]:D
02:47.03NewSoleheeehaaa
02:47.56*** join/#asterisk TrickyPhillips (~Trickyphi@adsl-68-124-61-95.dsl.irvnca.pacbell.net)
02:48.17wrtchdHas anyone used GnuDialer?
02:48.50*** join/#asterisk szw2001 (~vip@222.68.31.103)
02:48.54*** join/#asterisk bprice20 (~brandon@cpe-24-195-22-108.nycap.res.rr.com)
02:50.22*** join/#asterisk irv999 (~irv999@ool-457249eb.dyn.optonline.net)
02:50.53irv999lo all
02:51.02MikeJ[Laptop]high all
02:52.06irv999efveryone is so talkative tonight!
02:52.19MikeJ[Laptop]I am
02:53.01MikeJ[Laptop]why, nothng to do but hang out on IRC... fill your free time
02:53.11MikeJ[Laptop]go to the bugtracker at bugs.digium.com
02:53.15MikeJ[Laptop]and test patches
02:53.40irv999eh.. I need to get my new Dr client fixed on asterisk.. I am using this as free support.. :-D
02:54.06MikeJ[Laptop]you want free support
02:54.11MikeJ[Laptop]go test some patches..
02:54.15MikeJ[Laptop]it's give and take
02:55.04irv999mike: can't on this client.. will on my own pbx.
02:55.09irv999life and death situationo
02:55.12MikeJ[Laptop]perfect
02:55.26MocHail
02:55.29MikeJ[Laptop]test and comment on the bugtracker
02:55.35MikeJ[Laptop]frozen rain?
02:58.50wrtchdI am installing FC3 just as fast as I can.  Then to install asterisk for the first time.  Then I can test for you..
02:59.04MikeJ[Laptop]:)
02:59.10MikeJ[Laptop]make sure to look at the wiki
02:59.10wrtchdhave you used a predictive dialer yet?
02:59.29MikeJ[Laptop]I have used a predictive dialer.. a few of them
02:59.34MikeJ[Laptop]none asterisk based
02:59.36wrtchdI have been all over the place and I have three install guides..
02:59.44wrtchdah..I need on that is * based.
02:59.53irv999is there any way to see on an IP600 polycom phone all the calls that are parked?
02:59.55wrtchdI found one called GNUDIALER..looks promissing.
03:00.03MikeJ[Laptop]make sure to note the wiki details for FC3, spevifically the udev stuff
03:00.22MikeJ[Laptop]irv999, ummmm
03:00.23*** join/#asterisk JunK-Y (~junky@modemcable174.107-81-70.mc.videotron.ca)
03:00.27MikeJ[Laptop]ask moc
03:00.31wrtchdthe wiki is where?  on digium's site?
03:00.33MikeJ[Laptop]he knows those
03:00.39irv999ok..
03:00.39MikeJ[Laptop]www.voip-info.org
03:00.42JunK-Y~wikis
03:00.43jbotfrom memory, wikis is http://www.voip-info.org
03:00.46wrtchdTY
03:00.47irv999mike: which phone do you usually use?
03:00.51MikeJ[Laptop]he was just here a second ago
03:01.04MikeJ[Laptop]irv999, I have a lot of PRI connected box
03:01.06MikeJ[Laptop]es
03:01.07Qwell"You need a newer version of libpri"  libpri isn't even installed, heh
03:01.14MikeJ[Laptop]have a few cisco phones...
03:01.18MikeJ[Laptop]and some ata;s
03:01.20QwellIs that normal?  I must admit, I don't use cvs very much
03:01.22wrtchdYou have PRI?  what digium card do you use?
03:01.28MikeJ[Laptop]but don't use a lot of phones
03:01.37MikeJ[Laptop]te110p
03:01.53MikeJ[Laptop]and a new quad
03:01.54wrtchdWhat signalling do you need to order when you order the PRI?
03:02.13MikeJ[Laptop]any of the ones that are supported
03:02.24MikeJ[Laptop]I do a lot of trunking between other pbx's
03:02.29wrtchdok.
03:02.43MikeJ[Laptop]NI2, 5ess, 4ess... .. there are a few otheres
03:03.43MikeJ[Laptop]I'm using 5ess and NI2
03:03.55wrtchdI am new to the telephony.. I started in netware, and then was forced to windows.  I have FC3 down cold, and I am trying to get good with asterisk so I can get out of the windows world i am mired in.
03:04.18MikeJ[Laptop]I do a little bit of everything
03:04.30MikeJ[Laptop]use the wiki, love the wiki
03:04.34MikeJ[Laptop]~docs
03:04.35jbotDocumentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org
03:04.39wrtchdI enjoy it all..I will read the wiki
03:04.45MikeJ[Laptop]asteriskdocs are good too
03:05.10MikeJ[Laptop]plus there is an oriley book coming out soon
03:05.13cypromisnah
03:05.18cypromisthat's blitzrage stuff
03:05.20MikeJ[Laptop]<PROTECTED>
03:05.20cypromiscan't be good
03:05.21cypromis;)
03:05.25*** part/#asterisk JunK-Y (~junky@modemcable174.107-81-70.mc.videotron.ca)
03:05.26MikeJ[Laptop]:p
03:05.38wrtchdlast question..am I worried about what asterisk will support or the cards will support? or both..
03:05.39MikeJ[Laptop]cypromis, go to sleep you insomniac
03:05.48MikeJ[Laptop]??
03:05.50MikeJ[Laptop]huh
03:06.02MikeJ[Laptop]it's 5 a.m.
03:06.08cypromis5:06
03:06.13MikeJ[Laptop]hehhe
03:06.34MikeJ[Laptop]pri protocol support is in libpri, not in the cards
03:08.11wrtchdah ok.  so once I have the cards setup then asterisk should work just fine.  good.   and they want us to work for free.  Thank you MikeJ.  I will go test for you when I get something working.  don't know how much help I will be for a bit but thanks for the info anyway..
03:08.15MikeJ[Laptop]http://asterisk.org/index.php?menu=features
03:08.17*** join/#asterisk tiko_007 (~root@218.108.186.220)
03:08.32MikeJ[Laptop]np
03:11.58tiko_007hi,in asterik ,how can i change the Frame Size of Voice from 160 to larger? the Voice Frame Size
03:12.08cypromisyou can't
03:12.21cypromisit's hardcoded
03:12.23MikeJ[Laptop]crowbar, and torch
03:12.36MikeJ[Laptop]o... and sledgehammer
03:12.56MikeJ[Laptop]well, sure you can, but it will break a bunch of stuff that assumes 160
03:13.59wrtchdwhy would you want to change the frame type?
03:14.20Nuxiwrtchd, * runs in windows.
03:14.25cypromissize
03:14.27cypromisnot type
03:14.46wrtchdMikeJ ---- THANK YOU so much. you found for me in 2 minutes what I have searched for all day.
03:15.05MikeJ[Laptop]np
03:15.23tiko_007in my drive Play Mem buffer should larger than 192
03:15.27cypromiscomnpiling erlang on a via 500 is not really fun
03:15.33wrtchdNuxi, don't go calling names.  I was forced into windows.  I don't want to be there.  I have seen the light and have converted all home machines to FC3..lol
03:15.48wrtchdI am a Netware man..
03:15.48MikeJ[Laptop]hehe
03:16.08MikeJ[Laptop]wrtchd, it does not run on netware;)
03:16.16cypromishmm
03:16.21cypromisasterisk.nlm
03:16.22cypromis?
03:16.22MikeJ[Laptop]no home machine?
03:16.30cypromisheheh
03:16.31cypromisno
03:16.34cypromisalthough not true
03:16.38cypromisI have a wrt54gs
03:16.39MikeJ[Laptop]wait.. your never there
03:16.40cypromislol
03:16.48cypromistrue
03:16.48MikeJ[Laptop]run * on that
03:16.49MikeJ[Laptop]?
03:16.56MikeJ[Laptop]I need to go eat
03:16.58cypromisnah
03:17.09cypromisbut my hitachi wireless connects via that to my office *
03:17.24MikeJ[Laptop]y
03:17.30Qwellbloody hell...why does cvs think I have libpri installed?
03:17.50cypromiswhere did you get a thinking cvs ?
03:17.54cypromisI want one as well
03:18.14*** join/#asterisk Nugget (nugget@dazed.slacker.com)
03:18.19NuggetLinux is poo.
03:18.22wrtchdSorry my child was up way after bed time.. had to lay down the law...nlm..
03:18.52cypromisMikeJ[Laptop]: I am curious if I could run irix on the wrt
03:18.54Sedorox[23:15] <wrtchd> I am a Netware man..
03:18.56Sedoroxw00t!!!!
03:19.03cypromisin the and it's just a overfeatured indy
03:19.04cypromishehe
03:19.10cypromiss/a/e/
03:19.27wrtchdlol
03:19.39SedoroxLinux, Novell and Astrisk all on the same box...
03:19.39Sedoroxyum
03:20.16wrtchdthat would be fun..I could print from my phone.
03:20.24Nuggethaha
03:20.32Sedoroxlol
03:20.34wrtchddoes asterisk support NDPS?
03:20.50cypromisno
03:20.57wrtchdit was a joke..
03:21.09cypromisbut your welcome to add it
03:21.30wrtchdbut I would like to control asterisk with e-directory....I was told my novell rep that edirectory could control anything..
03:21.34*** join/#asterisk stilex (StilexIP@pc-24-151-79-248.newm2.ct.charter.com)
03:21.37wrtchdeven asterisk..
03:21.47wrtchdI have since changed Novell reps..
03:22.03tiko_007quit
03:22.07*** part/#asterisk tiko_007 (~root@218.108.186.220)
03:22.24wrtchdWell thanks for your help today.  I am out of here.
03:22.37*** part/#asterisk wrtchd (~dentont@c-24-0-114-212.hsd1.tx.comcast.net)
03:22.41*** join/#asterisk geesus (~paul@220-244-218-250-qld.tpgi.com.au)
03:29.46tzangerhttp://daynews.ru/index.php?act=show_news&id=939751%20%3C-%20lol
03:29.47tzangerhahahaha
03:31.55Qwellnice
03:32.38*** join/#asterisk irv999 (~irv999@ool-457249eb.dyn.optonline.net)
03:32.40irv999re!
03:38.35*** join/#asterisk Abizzals (drone1@d154-20-86-32.bchsia.telus.net)
03:39.01AbizzalsAnyone know how to force Asterisk to not when my FXO receives calls
03:39.13AbizzalsI tried putting in a blank context but it just skipped over that to default
03:39.32opus__<PROTECTED>
03:39.36opus__rephrase that pls
03:40.06AbizzalsTo never answer
03:40.12AbizzalsI want to make outgoing calls only
03:40.30AbizzalsYet the card is picking up any calls that come in, and depositing people at the default contexxt
03:40.51docelm0Anyone in here sell florida DID's?
03:41.02*** join/#asterisk dwC- (~dwc@S0106002078a93f9d.vs.shawcable.net)
03:41.07cypromisput only one extension in default:
03:41.11cypromisexten => s,1,hangup
03:41.26dwC-good evening
03:42.08dwC-is there any way to disable a fxo card from answering alltogether?
03:42.18geesusLOL
03:42.21docelm0unplug it.
03:42.23dwC-hah
03:42.26dwC-yeah thats the easy way
03:42.27geesusLOL!
03:42.29AbizzalsThat's not an elegant solution :P
03:42.32dwC-i am sure you guys have heard this one before
03:42.35dwC-oh shit
03:42.37geesusdwC-: we were just answering that
03:42.38dwC-ab you bitch
03:42.41docelm0Well it does the job doesnt it.
03:42.53dwC-i was asking for Abizzals and didnt see him here.. sorry
03:43.01Abizzals:P
03:43.23cypromisglasses help
03:43.26cypromissometimes at least
03:43.52dwC-big channel nick lists arent easy to look at in bX cypromis
03:44.19AbizzalsAwesome that worked, it didn't pickup at all
03:44.19Sedoroxlol
03:44.37dwC-good shit
03:45.06geesusmmm I just started using gaim for IRC today :P
03:45.06AbizzalsThanks cypromis I appreciate the assistance, and I'm sorry for dwCs intrusion.
03:45.06Sedoroxyou could also remove it from the dialplan?
03:45.26Sedoroxget xchat :p
03:45.29dwC-intrusion lol
03:55.10*** part/#asterisk Nukemizer (~Nuke@160.7.249.15)
03:58.25*** join/#asterisk TheEmperor (TheEmperor@210.19.250.123)
03:58.56*** join/#asterisk pitz (0@hsdbsk142-165-160-81.sasknet.sk.ca)
03:59.38TheEmperorhello
04:00.14*** part/#asterisk Nuxi (~nuxi@cust-sdsl-204-250-82-202.bzn-co-i1000-01.bridgeband.net)
04:01.16docelm0olleh
04:01.48docelmoThat really sucked!
04:04.27*** join/#asterisk DaLion (~DaLion@Toronto-HSE-ppp3883885.sympatico.ca)
04:04.31DaLionhi
04:04.55DaLionhow does one make cdrtool make free calls to other users... seems it still using pri's
04:05.08pitzok so i'm thinking of doing asterisk, at home, with a p133 machine
04:05.13DaLiontried to figure best routing but nothing in this software is obvious
04:05.17pitzand 4 ethernet based SIP phones
04:05.18DaLionpits dont
04:05.33DaLioncompression and codecs will killl it
04:05.44pitzDaLion well some online examples show 20-30 lines with a p2-300
04:05.44DaLionso use native formats and no music on hold
04:05.53DaLionyeah highly optimized
04:06.05DaLionand .. modded code
04:06.12DaLiontry if you wish
04:06.21DaLionjust dont expect too much out of it
04:06.29pitzits that bad, eh?
04:06.51DaLioni get probs with dual xeons 2 giger rams p4 -3.33GHZ
04:06.59DaLionimagine
04:07.00pitzdoing what though?
04:07.01doolphwho care
04:07.05DaLionlol
04:07.07DaLioni do
04:07.11TheEmperoranyone know why i can't see the webbrowser when installing fedora core?
04:07.18DaLion!jbot cdrtool
04:07.28doolphTheEmperor then install it
04:07.30DaLiondarn
04:08.14TheEmperordoolph:i installed it when installing fedora...
04:08.33TheEmperordoolph:but when it boots up all i get are the command line prompts, even if i do startx same thing :(
04:08.56doolphyou need some linux class
04:09.21doolphtry some os more easy for you, like centos or debian
04:09.34TheEmperorwhat is linux class?
04:09.41doolphi think fedore supports the command yum
04:09.46TheEmperoroh you mean back to school? :)
04:09.56pitzok so how does dual p3-450s sound?
04:10.06pitzasterisk uses threads, right?
04:10.30doolphpitz recommended are i686 class
04:10.55doolphthat means at least 1Ghz and up
04:10.56pitzdarnit i dont want to buy new hardware...  lol
04:11.08doolphbut you can still try
04:11.16doolphand use codecs like gsm
04:11.21doolphthat dont need that cpu
04:11.34pitzdoes that kill quality though?
04:11.41doolphnop
04:11.48doolphgsm are most used though
04:11.49pitzjust burns more bandwidth?
04:11.53doolphyes
04:12.02doolphbut acceptable
04:12.05pitzwell on my internal ethernet, thats no problem
04:12.11pitzi'm not doing VOIP outside the house yet
04:12.14doolphyeah
04:12.14doolphfine
04:12.15pitzerr won't be...
04:12.24pitzjust internal for now..
04:14.56*** join/#asterisk twisted (~twisted@twisted-professional-pdpc.developer.and.friend.of.asterisk)
04:14.56*** mode/#asterisk [+o twisted] by ChanServ
04:15.06doolphwill you use zaptel
04:16.22pitznot planning on using any PCI devices.  i'll go with ethernet-attached TA's
04:16.32pitzbefore i give up a PCI slot
04:16.53doolphthen how do you connect to outside
04:17.03pitzright now -- there is no 'outside' ;-)
04:17.15pitzeventually through an IP link
04:17.27doolphyeah
04:17.28doolphgood idea
04:27.51pitzsorry
04:30.09TheEmperorcan someone tell me what this means?
04:30.11TheEmperorMay 30 12:24:11 NOTICE[1237730496]: rtp.c:434 ast_rtp_read: Unknown RTP codec 19 received
04:33.06doolphmaybe unknow codec?
04:36.03TheEmperori see
04:38.39*** join/#asterisk baka03 (~baka_osx@203-206-48-60.dyn.iinet.net.au)
04:45.17DaLionbtw i woudnt say 1 giger is new hardware
04:45.29DaLioni would say 3+ is
04:46.03DaLioni got 4-5 just in the closet.. and probably 20+ under 1 Ghz serving as paper weigths
04:46.24io_errorha, I still have a Pentium 166 kicking around here
04:46.40DaLionhehe
04:46.43geesusI got a 6800GT on the weekend, next thing on my list is a CPU :D
04:46.47io_errorup until last week it was my mail server
04:46.51DaLion;)
04:47.00io_errornow it's FINALLY been decommissioned
04:47.11DaLioni mixed ECC with non ECC rdram and its messing up once per 36 hours
04:47.15geesus"replaced by a speedy 533!"
04:47.36DaLionyeah.. spamassasing takes up cpu ;)
04:47.48io_errorgeesus: heh, no, 4x 2.8Ghz Xeon
04:47.53DaLionany cdrtool genious around tonight ?
04:48.25DaLiontrying to figure if any docs on that stuff
04:48.28*** join/#asterisk mrkyr (bviitanen@h24-207-81-82.cst.dccnet.com)
04:48.30DaLionneed to make some routed free
04:48.35DaLionroutes even
04:48.42doolphareski?
04:48.44baka03hi guys
04:49.06baka03i need some help with setting up freshtel for dialing out
04:49.34doolphcreate a trunk then use that context
04:49.38baka03i have my config files (with username and passwords removed) at http://www.wollongongastronomy.com/asterisk/
04:50.25baka03the files i have changed are iax.conf and extensions.conf
04:51.49gordonjcpc
04:51.53TetraboyI'm upgrading my BSD server from a 300celeron to a 600celeron!
04:52.35gordonjcphow do you find asterisk on a 300 celeron?
04:58.29TetraboyI haven't installed asterisk yet.
04:58.35*** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk)
05:17.59*** join/#asterisk Dutts (~dutts@81.168.70.41)
05:23.51*** join/#asterisk lilo_ (lilo@levin-pdpc.staff.freenode)
05:27.15niZonomgwtfbbq
05:28.00niZonlin2voip aka 1089980 Alberta Inc
05:28.07niZon+k
05:28.12niZonowns a phonesex operation
05:28.24niZonhttp://www.talksugar.com/legal/member_agreement.php?SESSID=6c03381d4df3c758ecb204d717d28d67
05:28.25geesuspro!
05:28.33*** join/#asterisk moua (david@men75-2-82-66-50-159.fbx.proxad.net)
05:29.05niZonmaybe some fat whore who talks to horny bastards all day is going to activate my DID
05:29.14geesushahah has anybody come in here asking how to use asterisk as a phone sex line?
05:29.27niZonnot that I know of
05:29.55niZon"Call 858884 on freeworlddialup for FREE phone sex!"
05:30.07niZon(not real)
05:30.11orbiniZon: darn.
05:30.19niZonjust in case someone wants to call whoever that is
05:30.57niZoni don't even remember my FWD number
05:32.45louddo they cancel your account fo rnot using it ?
05:40.34*** join/#asterisk lilo_ (lilo@levin-pdpc.staff.freenode)
05:40.46niZonno
05:40.51niZoni use it sometimes
05:40.52niZonlol
05:43.57*** part/#asterisk Dislex666 (hidden-use@osiris.mundane.co.za)
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06:31.45*** part/#asterisk ronaldl79 (~ronaldl79@63.97.186.35)
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06:59.49QwellMikeJ[Laptop]: I'll have a better patch for 4403 tomorrow.  Hopefully the problem that guy is having goes away with the suggestion I posted...
07:01.17wasimwoe is me ... nobody up for that mgcp bounty? what if we up it to $5k
07:03.04*** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de)
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07:07.20gordonjcpanyone awake in the US?
07:08.21cochimorning everybody :)
07:08.28gordonjcphey cochi
07:08.33tzafrirhi cochi
07:09.07*** join/#asterisk wshs (screwy@69-168-246-252.bflony.adelphia.net)
07:09.15cochi:)
07:09.17gordonjcpcochi: you're in the US, right?
07:09.36tzafrirUS? Morning?
07:09.36cochinope. good ole germany ;)
07:09.45cochivery early morning then *g*
07:09.53cochi"oh. 3am... morning everybody" *g*
07:10.06wasimno new elftor for a year almost ...
07:10.19cochi9am is worse enough. don't intend to get up -that- early ;)
07:10.32gordonjcpcochi: east coast?
07:10.45gordonjcpcochi: ah?
07:10.49gordonjcpd'oh
07:10.50gordonjcpffs
07:11.16gordonjcp8am here in Glorious Sunny Greenock
07:12.09cochi*g*
07:12.18cochiah UK.
07:12.21cochior Ireland ;)
07:14.09gordonjcpScotland
07:14.19gordonjcpthe Greenock bit is a sort of a clue
07:14.27cochiah yeah
07:14.52cochijust learned from my UK bud that scotland is a "real" country *duck* i always thought it just was some sort of state of the UK. silly me :(
07:15.02cochibut i was always horrible at geography anyway
07:15.23io_errorcochi: that's okay, most Americans can't tell the difference
07:15.31cochiwell that error's probably a result of trying to transfer structures from Germany over to another country
07:15.37cochibut i am no American goddamn ;))
07:16.11io_errorheh, I am an American :)
07:16.26gordonjcpcochi: well, the UK is a kind of odd setup
07:16.30cochipoor io_error :)
07:16.39gordonjcpScotland is part of the UK, but a separate country from England
07:16.40io_errorcochi: oh, I manage to survive somehow :)
07:16.42cochithe people sometimes seem odd, too *g*
07:16.49cochi*g*
07:17.27tzafrirIt is part of the UK
07:17.43gordonjcpyes
07:17.48cochiand a separate country ;))
07:17.55gordonjcpagain, yes
07:18.05cochiin contrast to the EU. which's no country. but has a parliament. same strange constellation somehow ;)
07:18.17gordonjcpour money is the same, but we have a separate government, mostly
07:18.21cochiand a constitution which's valid in just some countries *headscratch*
07:18.24cochihehe yep
07:18.25gordonjcpexcept some things are still shared with westminster
07:18.35gordonjcpa completely different legal system
07:18.40cochiand to make stuff stranger: Wales is -just- a part of the UK and -no- country (yet) ;)
07:18.44gordonjcpand a completely different education system
07:18.56*** join/#asterisk dandre (~dandre@was59-3-82-236-48-30.fbx.proxad.net)
07:18.58gordonjcpcochi: well, yeah
07:19.22cochimust be related to living on an island (although big one). people get strange over there usually ;))
07:19.26cochitime for breakfast :)
07:20.02gordonjcpcochi: not forgetting that all the different bits are culturally very different
07:20.14gordonjcpin England the pubs all shut at 10pm
07:21.04cochiwell not too different of Germany
07:21.21cochiwe got our 16 states. many laws are shared, some differ. and the people are different anyway.
07:21.36cochitwo states even got their own constitution and are loosely related to Germany
07:21.48cochithe more i think about it the more strange it gets *hm*
07:22.29io_errorcochi: try living in the U.S. :)
07:22.40cochiugh. never
07:22.48cochii'll not even come over for holiday, sry ;)
07:22.51io_erroreh, it's not so bad, once you ignore the government :)
07:23.02gordonjcpah, see, that's the problem
07:23.03io_errorand stay out of the South
07:23.07cochiwell and the majority of people ;)
07:23.15io_erroreh, we're nice :)
07:23.39cochidon#t doubt that. got a friend over there at Florida
07:23.59cochialthough we weren't in contact for some time now
07:24.03gordonjcpI'd like to go to the US, but not until they get rid of a lot of the stupid restrictions on coming into the country
07:24.11cochi*g* yup
07:24.54io_errorgordonjcp: yeah, I know :(
07:25.02Romikin israel pub's shutup 2-4am
07:25.15io_error2am here
07:25.24io_errorin some states 1am, or 3am, or never
07:25.51*** join/#asterisk scanna (~scannachi@81-174-16-211.f5.ngi.it)
07:26.28Romikanybody has expirience with dialers?
07:27.07gordonjcpRomik: 2am here, in a lot of places
07:27.18gordonjcpusually at weekends though
07:27.27cochidunno if pubs have to shutdown here at all *mh*
07:27.34gordonjcpthat said the pub I was in last night shut at 11pm, unusually early
07:27.36cochithink not. know plenty who're open around the clock
07:27.56gordonjcpcochi: there are pubs open from 7am here, too
07:28.16gordonjcpbut they tend to make the Mos Eisley cantina look like a genteel sailing club bar
07:28.29cochibrr :)
07:28.51gordonjcpsome of the clubs are a bit like that
07:29.00gordonjcptwo huge bouncers on the door
07:29.20gordonjcpyou walk up, one of them says "Oi, are you carrying any kind of weapon?"
07:29.24gordonjcp"Uhm, nope"
07:29.46gordonjcp"No knives, nothing like that?"
07:29.48gordonjcp"Nope..."
07:29.48cochio.O well we got that on some, too. but mostly not
07:29.58gordonjcp"Ah, shit, right - here, borrow mine just now then"
07:30.46cochilol ;)
07:30.50io_errorgordonjcp: hahaha :)
07:30.53cochi"knives required" kind of bar ;)
07:32.09gordonjcpyeah man
07:32.24gordonjcpyou don't so much sign up for membership of some of the clubs, as weigh in
07:34.17cochi</breakfast> <burps />
07:34.53cochireminds me of doing a "how cochi's day is structured" xml file for fun :)
07:36.19*** join/#asterisk lehel (~lehel@82.79.20.17)
07:36.23lehelhello
07:41.34gordonjcphello
08:08.19orbi*cricket*
08:12.58gordonjcp<pok> HOWZAT!?
08:16.28orbierf?
08:16.50orbimultiple personalities?
08:16.54orbiCorydon!?!
08:17.25orbijeez, im just seeing all sorts of people i know on this network
08:18.54gordonjcpheh
08:25.54orbiGranted, the people i see on this network are typically the craziest people i've met elsewhere.
08:25.59orbiim not sure if thats a good thing or not
08:27.05gordonjcpheh
08:27.35orbihmm. you must be a bot that says "heh" over and over again.
08:27.36orbi;)
08:28.16gordonjcpheh
08:28.25orbii knew it.
08:28.42orbithe gordonjcp goes "heh"
08:28.45gordonjcptell me why you think I must be a bot that says "heh" over and over again.
08:29.03gordonjcp</eliza>
08:29.17wasimpk is the only country that could have lost the bloody test to the windies
08:29.30gordonjcpno, I'm just not very communicative after four hour's sleep three nights running and a 5am start
08:29.47orbii can understand that
08:29.55orbii've missed a lot of sleep the last couple nights.. bad toothache
08:30.22gordonjcphm
08:30.39wasimorbi: bite on a clove
08:30.40gordonjcpthis is just early starts at work, and a complete inability to get to bed at a sensible time
08:30.45gordonjcpwasim: yup
08:30.55orbihehe.. the darth vader voice change helmet on amazon says "This is not a protective device".   I thinkt hey should ask "Except in sexual situations.  It will protect you from ever having contact with the opposite sex."
08:31.03gordonjcpheh
08:31.12gordonjcphang on
08:31.37orbiwasim: i can just use the prescription eugenol (clove oil) mix i have, or the lidocaine, or the 20mg of oxycodone i have in me.  None of which are helping. :`(
08:31.57gordonjcphttp://ars.userfriendly.org/cartoons/?id=20050526
08:32.05gordonjcpbah, clove oil
08:32.10gordonjcpuse the real thing
08:36.03opus__word
08:38.06opus__does anybody here work with manager api?
08:38.35opus__sip.broadvoice.  147.135.0.128               255.255.255.255  5060     OK (100 ms)
08:38.50opus__thats the output to 'sip show peers', notice how 'sip.broadvoice.' is truncated
08:38.55opus__is there anyway to get around that?
08:39.01opus__without modifying the source?
08:39.08wasimsip show peers
08:40.31*** join/#asterisk twisted (~twisted@twisted-professional-pdpc.developer.and.friend.of.asterisk)
08:40.31*** mode/#asterisk [+o twisted] by ChanServ
08:42.59opus__whats the big deal about t.38
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08:57.54*** join/#asterisk bublbobl (~chatzilla@nat-80-64-224-17.e-qual.fr)
08:58.03bublboblhi all
08:59.42gordonjcphey bublbobl
09:01.09*** part/#asterisk kino (kino@c-67-190-65-108.hsd1.co.comcast.net)
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09:24.19timecopheh
09:24.48gordonjcpheh
09:24.57lehelheh
09:28.33jayk_heh
09:28.35*** join/#asterisk axscode (~1sdfgsdfg@203.177.235.188)
09:29.03axscodehi guyz.. how would i create channel so that i can use the three way conference?
09:29.12axscodeis it only threeway conference? or i can add more nodes?
09:29.22wasimaxscode: show application meetme
09:29.56axscodei did that... but when i call that. it says no channel something..
09:30.06axscodebut i did edit the meetme.conf
09:30.17*** join/#asterisk ramtha (~tk@gw.01063telecom.de)
09:30.19axscodei added 30000 channel
09:30.53axscodei added on: exten => 300,1,Meetme(30000)
09:31.37axscodebut it says.. an auto response voice.. channel dont exist.. or not initialized
09:31.49axscodeit says not initialized..
09:32.17gordonjcpaxscode: got zaptel timer?
09:32.53*** join/#asterisk iheartcanada (~iheartcan@lfc.tor.istop.com)
09:33.07ramthahi, some one got astcc working?
09:33.10axscodegordon.. i dont have a zaptel.. its all SIP
09:33.17ramthaif i dial into the callingcard app
09:33.22ramthai get AGI Rx << STREAM FILE astcc-down ""
09:33.35*** join/#asterisk Martohtar (Martohtar@82.196.218.80)
09:33.42ramthasound "is currantly unavailable"
09:33.50gordonjcpaxscode: you need some sort of zaptel driver to provide a timer
09:33.55gordonjcpotherwise meetme will not work
09:34.06wasimaxscode: yep, like even ztdummy
09:34.44gordonjcpwasim: indeed
09:34.55gordonjcpztdummy was a faff to install in netbsd
09:36.34*** join/#asterisk axscode (~1sdfgsdfg@203.177.235.188)
09:36.55axscodedo i need a zaptel device? or just the driveR? but i bet i have it installed coz my installation is from a CVS.
09:37.01gordonjcpyou can use ztdummy
09:39.17axscodewhats a ztdummy?
09:40.12postelits a dummy device you can compile if you dont have zaptel hardware installed
09:40.32axscodehmm ok ok.. so then i can use a conference feature?
09:40.41axscodeso upto how many user can join the conference?
09:41.24postelhow many friends you got?
09:42.43axscodethat depends...
09:43.01axscodewhat do u mean.. true or somewhat.. close..
09:43.07postelall of them
09:44.11axscodenice..
09:44.17axscodeso its like a party conference...
09:44.35axscodewow.. i can sense phone s*x orgy
09:44.42axscodeehehhe.. kidding..
09:44.53postelu dirty boy you, no more kleenex tissue for you
09:45.07axscodehaha.. kidding man..
09:45.17axscodedo u happen to know where can i download that ztdummy?
09:45.24axscodefor freeBSD?
09:46.01lehelhow do i have to configure Comedian Mail.. to send me a mail.. when i get a voicemail message?
09:46.04postelits source dude, if you got gcc on fbsd you're good to go
09:46.28axscodewow nice.. so... where can i get it?
09:46.45axscodeim quite excited about this conference thing..
09:47.02axscodeand it took me 2 weeks to learn asterisk... too bad dont u think!?
09:48.07postelaxscode: its on the cvs tree
09:49.19*** join/#asterisk cds (~chris@nfluid.plus.com)
09:49.29*** join/#asterisk _omer (dfsdf@202.147.167.213)
09:49.40gordonjcpaxscode: I built ztdummy from source in netbsd, it was fiddly but eventually it worked
09:49.43axscodei dont have internet at home
09:49.47axscodeso i have to download it..
09:49.48_omerCan I use Asterisk for Local Loop ?
09:49.51axscodeim using freeBSD..
09:49.53gordonjcpif you get stuck I may be able to point you in the right direction
09:50.01gordonjcp_omer: what do you mean?
09:50.17axscodegordonjcp.. i like this channel.. very friendly..
09:50.21axscodedont u think gordonjcp?
09:50.49_omergordonjcp: I want to be exactly like Freeworlddial.com . . .
09:51.08gordonjcpaxscode: get it right up ye
09:51.24*** join/#asterisk RichiH (richih@richih.staff.freenode)
09:51.32gordonjcp_omer: you mean letting people call each other from sip phones?
09:51.46axscodegords... would u like to point me to the direction now? i have to DL that ztdummy
09:51.51cdsHey chaps.  trying to get at CVS.... logging in with pwd anoncvs gives connection refused.... has anon access been disabled for the time being?
09:52.11RichiHare there any suggestions on video-chat capable sip clients for mac os x?
09:52.12_omergordonjcp: yes ....call each other and some other TELCO as well.....In simple words...a Local Loop Carrier..
09:52.22RichiHalso, a sip client for linux
09:52.28wasim_omer: yes you can
09:52.47axscodegords.. question... what do i need to use a video phone? is there anything i should turn-on on asterisk1?
09:53.18RichiHaxscode: no, you only need a video capable client
09:53.30axscodewow.. cool..
09:53.30RichiHproblem is, it seems all clients for mac cost money
09:53.41axscodedo u happen to know any softphone that i can use?
09:53.48_omerwasim: I know I can, but I think freeworlddial.com is developed by SER (SIP Express Router).
09:54.04wasim_omer: its not developed by SER, they use SER, yes
09:54.10gordonjcp_omer: yes, probably
09:54.21gordonjcpaxscode: uhm, I used pkgsrc
09:54.40_omerthen? which one is good to use? asterisk or SER? what's the difference?
09:54.58axscodehmm.. gordonjcp: this time.. softphone on windows base.
09:55.04wasim_omer: one is a registrar, * also does registry services, but its primarily a switch, you'd use both in their own place
09:55.23axscodeis there an opensource softphone?
09:55.43wasim_omer: one implementation is to use ser to handle the registries, and * on the backend to handle the switching
09:56.33_omeralright...
09:57.30_omerthanks wasim.
09:58.32wasim_omer: http://www.voip-info.org/wiki-Asterisk+at+large
09:59.29cds... any ideas on non-annon cvs access?
09:59.44postelyou want to commit?
10:00.10_omerwasim: thanks...let me read that...
10:00.25cdsNah - just trying to get the lastest src from cvs.
10:00.35tzafrircds, bugs.digium.com ?
10:00.49cdsFriend asked me to look into it for him, so the latest src would be good.
10:01.04cdsFollowed the download directions as usual, but connection refused.
10:01.20cdsWondered if annon cvs was unavailable deliberately.
10:01.28tzafrircds, maybe the connection to port 2401 is blocked?
10:01.37cdsDuh!
10:02.14cdsWait.. hang on, my cvs access everywhere else is ok.
10:04.39*** join/#asterisk Jedirl (~fdsafasdf@213.162.200.226)
10:04.40JedirlHello
10:05.31JedirlI'm going to buy a simple 1-BRI HFC ISDN PCi device for testing with asterisk. Which driver should I use? chan_modem_i4l? chan_capi? zaphfc?
10:06.25Jedirlanyone experienced with asterisk + BRI ?
10:06.37*** join/#asterisk axscode (~1sdfgsdfg@203.177.235.188)
10:07.27*** join/#asterisk Maksim (~max@213.142.207.20)
10:07.40wasimcds: cvs seems to be broken
10:09.44cdswasim: My firewall was blocking port 2401... sa tzafrir suggested. I usually use CVS over ssh so hadn't noticed this
10:10.03cdstzafrir: wasim: loggin in works now :)
10:10.15*** join/#asterisk Dislex666 (hidden-use@osiris.mundane.co.za)
10:10.17axscodeguys any link to ztdummy source for freebsD?
10:10.20Dislex666howzit all!
10:10.56Dislex666can someone puleeease give me some information regarding h323 static routing
10:11.32axscodequestion guyz.. if i got the timer? what will i do with the timer?
10:12.12*** join/#asterisk RoyK (~roy@80.239.107.80)
10:12.17Jedirlso noone here uses bri? :(
10:12.30postelaxscode: compiel it and install it, its a kernel modulew
10:12.39postels/modulew/module
10:12.56RoyKkapjod is the bri master
10:12.57axscodethen.. after that.. im good to go.. or any  configuration?
10:13.02RoyKnot here, though
10:13.11RoyKJedirl: what hardware?
10:13.54JedirlRoyK: HFC PCI 1-BRI
10:14.20JedirlI don't know what driver should I use
10:14.29Jedirlchan_modem_i4l? chan_capi? zaphfc?
10:14.36RoyKbristuff
10:14.40RoyKonly bristuff
10:14.44RoyKthat is, zaphfc
10:14.49Jedirlbristuff tries to compile asterisk itself
10:14.49tzafrirbristuf/zaphfc?
10:14.50RoyKi4l sucks
10:14.51Dislex666does anyone know h323 without a GK?
10:15.03RoyKcapi needs a capi board
10:15.16RoyK~chan_h323
10:15.19Jedirlthe HFC HiSax driver doesn't provide capi?
10:15.24RoyKno
10:15.29RoyKthat's a completely different thing
10:15.35RoyKcapi is capi.
10:15.38Jedirlok
10:15.50RoyKbut get bristuff from junghanns.net
10:16.05JedirlRoyK: I got bristuff, but I already have a working asterisk, and it tries to compile itself another one
10:16.14*** join/#asterisk SeSe (~sese@pmoran.demon.co.uk)
10:16.39tzafrirJedirl, the debian packages have bristuff in them. Sadly enough, bristuff is indeed an intrusive patch
10:16.50JedirlI'm using centos here
10:16.52Jedirl:(
10:17.12Jedirlthen, I *do* need to compile another asterisk, it's not just a module, right?
10:17.16RoyKtzafrir: intrusive?
10:17.43tzafriryou need to patch asterisk, libpri and zaptel, and build an extra zaptel module
10:18.01JedirlI'll let "install.sh" do its job then
10:18.05tzafrirand you end up with versions that are binary-incompatible with the older ones
10:18.50axscodehow do i check for asterisk timer?
10:18.51tzafrirso sharing modules between them can be a pain
10:18.56JedirlI'm going to buy a *really* cheap ISDN BRI card just for testing... in production I'll use a TE405
10:19.00RoyKtzafrir: too bad fscking Digium can't do asterisk GPL for real
10:19.09RoyKtzanger: that's the root to all such problems
10:19.21JedirlRoyK: why? asterisk is not really GPL'ed?
10:19.27tzafrirRoyK, from what I understand it's not the license but the copyrights
10:19.38Jedirlthey want copyright assignements like mysql and that, right?
10:19.47RoyKJedirl: yes
10:19.52tzafrirI mean: some people don't want to assign copyrights to diguim for their work
10:20.09RoyKtzafrir: which is absolutely normal....
10:20.29RoyKtzafrir: spending time to make open source better is one thing. working for free for a californian company is another
10:20.36axscodewhat else can asterisk can do? aside from voicemail, sms, conference, video phones..?
10:20.51RoyKaxscode: what else do you want?
10:20.58JedirlI agree
10:20.58RoyKit can do a bunch of stuff......
10:21.00tzafrirZapATeler?
10:21.06tzafrirPlay Poker?
10:21.08cochii'm pretty sure * can also make your coffee ;)
10:21.18RoyKchan_htcpcp doesn't exist yet
10:21.18axscodei really dont know.. what else it can do?
10:21.20Jedirlother projects don't do copyright assignement but they grant a very permissive redistribution license
10:21.29Jedirlto the company that develops the project
10:22.03tzafriraxscode, it can be a cause of a headache, for once
10:22.16Jedirlgreat
10:22.27Jedirlbristuff doesn't compile on my machine
10:22.28Jedirl:(
10:22.46leheltzafrir: how to make asterisk (AMP) to send mail when i get a new voicemail?
10:22.51axscodehehhe.. not that much though.. but it tooks me two weeks to figure it out without manuals.
10:23.15tessier_lehel: Asterisk usually does that by default
10:23.25tzafrirlehel, sorry, not sure. are emails configured in /etc/asterisk/voicemail.conf ?
10:23.39tzafririt is being rewritten, but this is the end result
10:24.04lehelthey are
10:24.25tzafrirnext step: is a message being sent?
10:24.37tzafrirmaybe a message is sent and there are mail config issues?
10:25.45lehelno message sent.. i don't think there are mail config problems..
10:26.42leheldoes "context" matters?.. or i don't know what could be the problem
10:26.48Dislex666do any of you guys know a h323 guru on the channel?
10:26.48Sato1damn, i upgraded from rh9 to fc3 and now my phonejack does not work with it
10:27.07JedirlDislex666: I use chan_h323 but with a gatekeeper
10:27.14JedirlDislex666: never tried without it
10:27.15Sato1anyone experienced with ixj compiling in 2.6.x?
10:27.34Dislex666Jedirl: what do you use for a GK?
10:27.39JedirlDislex666: GnuGK :)
10:28.01Dislex666ok
10:28.38Dislex666Jedirl: so when you assign extensions you just put it like this: exten => 27215551234,1,Dial(H323/${EXTEN})
10:29.16JedirlI don't remember the exact syntax
10:29.37axscodequestion everyone... is xtent is an OpenSource Software?
10:29.51Jedirlbut it was somewhat like that, yes
10:30.08Dislex666thanks anyway :)
10:30.16Jedirlanyway I don't create extens which dial to H323, I just create dial strings from an AGI string
10:30.55Dislex666agi?
10:31.08*** join/#asterisk Aze` (~aze@host229-162.pool80105.interbusiness.it)
10:31.42Jedirlan external script
10:31.49Jedirlwhich talks to asterisk
10:31.54Jedirland tells it what to do
10:32.55Dislex666aah i see
10:33.11axscodeany web admin for asterisk for apache?
10:34.11Aze`anyone use cisco 7912 ?
10:37.48postelaxscode: http://amp.coalescentsystems.ca/
10:41.08tzangerRoyK: actually Digium is from Tennissee (probably spelled wrong)
10:41.24tzangerer no, Alabama
10:47.22cypromisOh moon of alabama we now must say goodbye ...
10:47.22cypromislol
10:51.50tessier_hmm...I know that song...
10:52.05*** join/#asterisk mAsH` (~mAsH@ppp-217-133-150-46.cust-adsl.tiscali.it)
10:52.06tessier_Anyone know Vonage's billing increment? 6s? 1min?
10:52.11mAsH`hi all
10:52.37InfraRed9 hours
10:53.33tessier_The Doors recorded that song.
10:53.45*** join/#asterisk biafra23 (~biafra@off-96.gate5.de)
10:53.45*** join/#asterisk pikeymike (~Mike@80-192-3-190.cable.ubr04.glen.blueyonder.co.uk)
10:54.07pikeymikeCan someone help me with options for load balancing an Asterisk setup ?
10:54.19*** join/#asterisk DrFrancky (~chaos@pirus.securax.be)
10:54.28pikeymikeI know about round robing DNS for SRV records, but what other options are there?
10:56.57mAsH`anyone never  use attended transfer with at-320 ?
10:58.22*** join/#asterisk axscode (~1sdfgsdfg@203.177.235.188)
11:00.56Jedirlok
11:01.00Jedirlfinally compiled bristuff
11:01.05Jedirlwhat a hell of patching :D!
11:03.41RoyKJedirl: heh. there's a script that does it for you
11:04.49bublboblI would like [from-sip-external] context to be sent to digital receptionnist, my syntax is probably wrong, how would you declare the extension ?
11:09.40*** join/#asterisk newbien (~e@116.242.33.65.cfl.res.rr.com)
11:09.47*** join/#asterisk sigterm (sigterm@devious.info)
11:15.55_omercan I call at any External SIP Address from Asterisk CLI?
11:17.40*** join/#asterisk Mc_Tr (~Mc_Tr@bacterio.knet.es)
11:17.48Mc_Trhello!
11:19.02*** join/#asterisk mrtwister (~mrtwister@cable-1-32.cgates.lt)
11:23.13*** part/#asterisk cds (~chris@nfluid.plus.com)
11:23.42tzafrir_omer, what I usuall do is drop a call file to generate a call from my echo-test extension (using Local/) to any extension.
11:24.03tzafrirI have a script for that, and using ! I can also do that from the asterisk CLI
11:24.32*** join/#asterisk Dutts (~dutts@81.168.70.41)
11:28.58JedirlRoyK: the script "works"
11:29.12JedirlRoyK: but it makes some assumptions that shouldn't be there I guess
11:29.19*** join/#asterisk nifter (~nifter@190-141.SPEEDe.golden.net)
11:29.23JedirlRoyK: out-of-the-box it won't compile on any RHES-based linux
11:29.40RoyKThou Shalt Not Use Distro Kernels With Aserisk
11:29.57Jedirlwhy?
11:30.18mrtwistertest. one asteris and cisco 53xx in same notwork oh323, calling DID number (caller -> cisco -> * ) g729. DTMF working nice. second test. same. 729. asterisk other country. delay is stable, 200 ms. calling cisco -> *. dtmf doubling. like 123456 on first test, 1122344556 on second test. any ideas how to solve? it is echo problem? what is hell there? :)
11:30.22RoyKbecause they include all sorts of shite
11:30.28RoyKuse a kernel from kernel.org
11:30.29RoyKperiod
11:30.43Jedirlhehehe
11:30.49Jedirlwell now it works :D
11:33.33gordonjcpxheliox: use adjustkernel, it's good
11:34.00sigtermbah make lint =)
11:34.42gordonjcplint: not found
11:34.50sigtermhaha
11:35.13xhelioxgordonjcp: ?
11:35.38gordonjcpxheliox: netbsd tool for setting up kernels
11:35.46xhelioxuh.
11:35.51gordonjcpgives you a sensible baseline config, based on what's in your system
11:35.52xhelioxokay? I don't use netbsd...
11:36.10gordonjcpah well
11:36.36gordonjcpcontinue to struggle on with an inferior hobbyist OS then
11:36.55sigtermoh boy... here we go
11:37.05gordonjcp:-)
11:37.42gordonjcpwhich makes me stand out a bit in the office, I can tell you
11:38.04sigtermi bet
11:41.18*** join/#asterisk gambolputty (~gambolput@cblmdm72-240-241-108.buckeyecom.net)
11:43.38bublbobl:'( I have an incoming sip call (From: <sip:549494810@192.168.2.213> To: <sip:0196@192.168.2.211>), I can't drop it on Digital recep^tionnist, any help ?
11:43.48*** part/#asterisk gordonjcp (~gordonjcp@cpc3-broo2-5-0-cust232.renf.cable.ntl.com)
11:44.07Inv_arpsigterm:   sigterm puts his troll outfit on     just read it  hahaha
11:47.14*** join/#asterisk meppl (mephisto@p54AABE54.dip.t-dialin.net)
11:48.50*** join/#asterisk tessier_ (~treed@203.210.212.138)
11:49.37*** join/#asterisk ceegee (~christian@p508A4152.dip.t-dialin.net)
11:49.43ceegeehello
11:53.23*** join/#asterisk Lloydio (duph@81-86-196-70.dsl.pipex.com)
11:56.21tzafrirRoyK, what's so wrong with distro kernels?
11:56.40tzafrirzaptel is generally a module that builds out-of-tree
11:56.55RoyKnonstandard everything....
11:57.08RoyKyou can't build anything out-of-tree in 2.6 iirc
11:58.11tzafrirRoyK, make -c path/to/kernel/dir SUBDIRS=$PWD modules
11:58.15tzafriror something similar
11:58.25tzafrirLook at the zaptel makefile
11:58.38RoyKI stick with kernel.org kernels anyway
11:58.49RoyKI don't want all that shite most distros put into them
11:59.06tzafrirMost problems with distro kernels is that by being different many patches won't apply to them
11:59.28tzafrirBut that's irrelevant to zaptel, which is a nice set of external modules
12:01.08RoyKall i know is i have had problems building zaptel for distro kernels
12:01.13RoyKsome time back
12:01.17RoyKso i don't use them anymore
12:02.01RoyKi really don't see the point of using them in the first place, when the kernel.org kernels have less fancy stuff and then is prolly more stable than the ones with animated framebuffers and whatnot
12:02.10Ahrimanesanyone using app_prepaid_auth_cid.so? have just installed it, but could really use some sample data
12:02.31ceegeedoes anyone of you use AMP?
12:03.38postelceegee: no, you're the only one, that 99% popularity in sourceforge is false stats
12:04.10ceegeepostel: :-)
12:06.28xheliox"prolly more stable" so you've obviously done extensive research and testing with good record keeping to back your blovating up, right? Because 30 minutes ago, you were stating no one should use them and now you're stating that it's "prolly more stable" -- which is it? And did you ever stop to think that the patches the vendor provides might be a requirement for software the end user is using and they're not savvy enough to realize it?
12:06.28xhelioxGod, people like you make me insane. So black and white, never mind the fact that 2.6 is basically the testing branch of the kernel, considering there's no other testing branch. And we all know that there's never instability with patches committed directly to the 2.6 branch. </rant>
12:06.46*** join/#asterisk shammash (~shammash@130.136.31.125)
12:06.52ceegeepostel: you use it?
12:06.59*** part/#asterisk shammash (~shammash@130.136.31.125)
12:07.08postelceegee: no, im the 1%
12:07.26newlerm..even numbers are stable, not testing.
12:07.49Ahrimanesxheliox: good to get off your chest, eh? :D
12:08.09xhelioxAhrimanes: Yeah, I feel a little better, thanks. :)
12:09.44xhelioxDidn't Linus himself say that the final testing and tweaking should be done by the distributions, not the people from kernel.org? I know I'm paraphrasing a great deal.. but I know kerneltrap has something similar to that from several months ago.
12:10.09Ahrimanesxheliox: you're mentioning many of the reasons why i shifted to FreeBSD :)
12:10.19sigtermlol
12:10.20Jedirluntil early 2.4 I agree with using kernel.org
12:10.22tzafrirxheliox, 2.6 is the current kernel.
12:10.37tzafrir2.6.11.11 is the current stable kernel :-p
12:10.47Ahrimaneseven mumbers are "stable" and uneven are development
12:10.59tzafrirAhrimanes, that is not correct anymore
12:11.03xhelioxThe latest stable version of the Linux kernel is:  2.6.11.11
12:11.05JedirlAhrimanes: not now :D
12:11.06Ahrimanestzafrir: oh ok
12:11.09Ahrimanesjeez
12:11.18xhelioxtzafrir: I know that, but Ii
12:11.31Ahrimaneseven with 2.4.x it seemed ok to change vm system from 2.4.10 to 2.4.15 or something.. christ
12:11.41xhelioxI'm suggesting that "stable" might not be all it once was from kernel.org :)
12:11.45tzafrirCurrently the "stable" branch is actually Linus's tree, whereas Andrew Morton's tree is the "unstable" version
12:11.56xhelioxI mean, what was the scsi deal from 2.6.10 to 2.6.11?
12:11.58xhelioxThat's stable?
12:12.36tzafrirxheliox, heck, I'm stuck with Debian's 2.6.8
12:12.41xhelioxI'm not saying there's anything wrong with using kernel.org's release, but he was so insistent on how crap distro kernels are and I simply don't see it that way.
12:13.40Ahrimanesbesides i dont like the whole having kernel and base userland developed asynchronously
12:14.19tzafrirAhrimanes, What do you mean "userland"? As if ther was one userland
12:14.37tzafrirglibc and gcc are separate, for instance
12:15.09Ahrimanestzafrir: exactly my point
12:15.22tzafrirBut strangely enough, it works
12:15.27Ahrimanesthe kernel lives seperately from userland.. i prefer them to live together
12:16.16tzafrirAhrimanes, if this is what you want: http://web.yl.is.s.u-tokyo.ac.jp/~tosh/kml/
12:16.26sigtermpeace and harmony everafter.
12:16.46tzafrirtill sigterm sets them apart
12:16.53Ahrimanestzafrir: hehe nah, am sticking to FreeBSD where kernel and userland are developed in sync
12:17.19tzafrirAhrimanes, the problem is that such a model doesn't scale well enough
12:17.30Ahrimanestzafrir: in which way?
12:17.31sigtermer
12:17.37Ahrimanessigterm: hehe
12:18.28*** join/#asterisk makhtar (~ageller@mail.bulletinnews.com)
12:18.59tzafrirFrankly, you pointed at problems with the kernel developemnt process, and not with the interface between it and libc/gcc
12:19.20Ahrimanesuhm yes
12:20.01tzafrirkernel.org is generally not meant for production. distro kernels are meant for production.
12:20.25tzafrirThat is: kernel.org kernels are meant to be used by proffesionsls and by testers.
12:20.44tzafrirTherefore it is, well, OK, if they have some problems.
12:20.48Ahrimanestzafrir: i know.. but the whole philosophy in linux is that it's really only the kernel, everything else is up to distros
12:21.10tzafrirBut it helps expose problems
12:21.39tzafrirYou shouldn't give such a central place to the kernel. It is just one component that distros use
12:21.53Ahrimanesi know, but that's the thing i dont like
12:22.00xhelioxYou know what..
12:22.07xhelioxCan't we just all come to the same agreement that we always do..
12:22.11xhelioxIt's good to have these choices.
12:22.12Ahrimaneshehe
12:22.14Ahrimanesyep
12:22.15xhelioxAnd to eaceh their own. :)
12:22.25posteltzafrir: what on earth you're talking about, the kernel.org kernels ARE the kernels used in production, distro kernels have custom patches, with a kernel.org kernel you can roll your own distro, or LFS, with a custom FOO distro-custom-patched-kernel you cant
12:22.43Ahrimanesflamewar comming up.. hehe
12:22.44*** join/#asterisk Fraeggl (~Fraeggl@rkom.r-kom.de)
12:23.04postelno, im not gonna defend it, its SO obvious i wont even troll over it
12:23.20Ahrimanes:)
12:23.54*** part/#asterisk welby (~welby@tollcross.edihost.co.uk)
12:24.01*** join/#asterisk speakman (~speak@t1o30p68.telia.com)
12:24.17speakmanhello people! :)
12:24.17Fraegglhi all
12:24.18tzafrirpostel, I sure can. In fact, I already have
12:24.20speakmananyone awake?
12:24.38Fraegglis there a way to access the rpid from a sip-message in a dialplan ?
12:24.48sigtermspeakman: no were all typing in our sleep
12:24.53sigterm;)
12:24.55Fraeggleg via an predefined variable?
12:24.58speakmanlol ok!
12:25.06posteltzafrir: " You shouldn't give such a central place to the kernel. It is just one component that distros use", its a MONOLITHIC os for crying out loud, and the kernel is "one component distros use" for you?
12:25.19postel...
12:25.44*** join/#asterisk Aze` (~aze@host229-162.pool80105.interbusiness.it)
12:25.56speakmanI'm desperatly trying to make Asterisk connect with my ISP with a 64k channel over E1. But the pppd tries to end Config-Requests until it times out.
12:26.06tzafrirpostel, for a distro, the kernel is basically a package. BTW: this applies for hurd as well
12:26.10postel....
12:26.14Aze`Anyone can hel me with cisco 7912/7905 ?
12:26.15speakmanIt seems like Asterisk isn't making data-calls, but regular voice-calls.
12:26.18Aze`Anyone can help me with cisco 7912/7905 ?
12:26.20Ahrimanesback to work.. anyone using: http://www.voip-info.org/wiki-CallingCard+Applications ?
12:26.26tzafrirBut then again, I see some actually on-topic transport in the channel
12:26.29speakmanHow to tell Asterisk to make a *data*-call ?
12:27.11sigtermspeakman: your extensions conf setup wrong possibly?
12:27.13*** join/#asterisk IsMe (me@211.24.146.11)
12:27.51IsMehi all, i got 2 g729 license, how do i use them in sip.conf ?
12:27.58speakmansigterm: it has: exten => 1234,1,Answer      exten => 1234,2,PPPD(/usr/sbin/pppd dial myisp)
12:28.37*** join/#asterisk welby (~welby@tollcross.edihost.co.uk)
12:28.49Fraegglhow does asterisk "process" Remote-Party-ID from a sip invite, does someone know ?
12:28.56Fraegglcouldnt find anything in the web about it
12:29.14speakmanAnd then I move a .call-file into /var/spool/asterisk/outgoing to make the call...
12:29.27speakmanCan I specify in the .call-file that the call is a data-call?
12:29.47sigtermspeakman: im not sure on that one, maybe someone else can help you there
12:30.10speakmansigterm: ok, but it seems to me most idleing people in here.. :)
12:30.41sigtermthere alive, just hiding =)
12:30.43speakmanNoone is even commenting people's questions...
12:30.49speakmansigterm: seems so.. ;)
12:31.13sigtermthey are recovering from sore fingers from battling out a kernel war
12:31.14sigterm;)
12:31.34xhelioxIt's a holiday in the UK and US, maybe elsewhere. I imagine a lot of people are out doing something fun. :)
12:31.41sigtermpatience, someone will know , i hope =)
12:32.05speakmanlol newly war? :)
12:32.20sigtermon that note, i think im going to bed so i can do something today
12:32.46xhelioxlol
12:32.52xhelioxwhat? spend the day playing with Asterisk, again?
12:33.38speakmanIt's time for a break.. back in a while!
12:33.40sigtermpossibly =)
12:33.55xhelioxGo sleep. Any word back from Teliax?
12:41.52Fraegglcan someone held me set the Remote-Party-ID (rpid) in extensions.conf for a SIP -> PSTN Gateway ?
12:41.59Fraeggls/held/help
12:45.05*** join/#asterisk Juggie (~agony@CPE00c049d9f271-CM00137186c8d8.cpe.net.cable.rogers.com)
12:47.18IsMehi all, i got 2 g729 license, how do i use them in sip.conf ?
12:48.12tzafrirspk|idle, what is a "data call"?
12:48.40tzafrira call file can generate a channel. The contents of that channel is basically up to you
12:50.43RoyKIsMe: try show g729 to see if they're active
12:50.53RoyKIsMe: then just disallow=all, allow=g729
12:51.07*** join/#asterisk webograph (~webograph@213.235.241.194)
12:52.03speakmanback!
12:52.43speakmantzafrir: data call is for non-voice conversation, basicly :)
12:52.45webographhi! i have different sip providers which i could use for local calls to different countries. how do i make asterisk route a call through a specific server?
12:53.22mrtwisterwebograph, many ways
12:53.43webographmrtwister: that's good news ... but one would be enought :-)
12:53.44IsMeRoyK: show g729 say i got 2 licensed channels but, i got "codec_g729.c:196 g729tolin_framein: Invalid data (4 bytes at the end)
12:53.56speakmantzafrir: Regarding my ISP: if a Voice-call is coming on their channel, they answer with V.90. But if it's a data-call, they open up for 64k ISDN traffic.
12:54.11mrtwisterwebograph, if you have one provider per destination or few, with almost same rate, use dial command
12:55.20webographie in my home-extensions, i put something like "exten => 0043,1,Dial(SIP/$MyISP)"?
12:55.31mrtwisterwebograph, exten _00880X.,1,Dial(SIP/${EXTEN}@bd-all1&OH323/${EXTEN:2}@1.2.3.4:1720&SIP........,60)
12:56.06webographehm, what's the delimiter in that string? @?
12:56.19speakmanwebograph: Are you talking about my problem now?
12:56.26mrtwisterwebograph, this way you will make few routes to bangladesh on one exten, diring call, asterisk will might to setup few calls
12:57.53webographin this case, what do i have to dial to call, let's say, my grandmother in bangladesh? 00880123456789 directly in my sip-phone?
12:58.03*** join/#asterisk fantomax1 (~fanto@81.208.114.250)
12:58.07fantomax1hi all
12:58.12webographspeakman: sorry, i don't know what your problem is
12:58.37fantomax1i have an * with 500 sip channels up, double xeon, 2GB ram
12:58.54fantomax1i have problems with socket and files open
12:58.58fantomax1too many
12:59.02fantomax1any suggestion ?
12:59.07speakmanwebograph: ok, just saw your $ISP-string :P
13:00.20*** join/#asterisk jazza005 (~jazz@220-253-85-31.QLD.netspace.net.au)
13:02.23Fraegglcananyone help me with a remote-party-id problem ? esp how can i use this information in a dialplan ?
13:02.53mrtwisterwebograph, tell what do you need. call flow scheme. from where to where. etf
13:03.50_omerI'm trying to MONITOR a channel....but during recording I get this..
13:03.52_omerMay 30 06:10:10 NOTICE[3591]: frame.c:135 __ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end
13:04.36_omerI get the sound like talking to an under water operator :)
13:05.00speakmanDo anyone here uses ISDN to connect to their ISP? Or uses ZapRAS/PPPD in other ways??
13:05.05_omeranyone plz?
13:07.21mrtwisteromer, do you using monitor command?
13:07.47_omeryes...
13:08.07mrtwistermoment i'll give you other example
13:08.30_omerokk
13:09.11_omerMonitor(gsm,asterisk-recorind,m) <---here is what I use....same problem with WAV..
13:09.18*** join/#asterisk Corydon76-home (orange@pcp08665860pcs.500ash01.tn.comcast.net)
13:16.40blopanyone using a voip2gsm or umts gateway ? :)
13:16.57*** join/#asterisk puppet (puppet@1-1-3-3b.ox.mlm.bostream.se)
13:17.00*** part/#asterisk puppet (puppet@1-1-3-3b.ox.mlm.bostream.se)
13:17.07*** join/#asterisk puppet (puppet@1-1-3-3b.ox.mlm.bostream.se)
13:19.11*** join/#asterisk josehap (~jose@129.210-201-80.adsl.skynet.be)
13:19.33josehapi have problems with TDM400 in be, can anyone help me ?
13:19.49konradsWASABI
13:19.56josehapthere are two 400P for a total of 5 fxs and 3 fxo
13:20.08*** part/#asterisk konrads (~konrads@193.68.74.130)
13:20.25josehapand unexpected deconnections occur frequently
13:21.14blopjosehap what error u get ?
13:21.38josehapno error, the console just telling that the line hungup
13:22.38*** part/#asterisk ceegee (~christian@p508A4152.dip.t-dialin.net)
13:22.51josehaphears like "microcuts", as the tone i get just after is the dial tone
13:23.07mrtwisteris anyone tested config: sangoma <--- E1 --> channel bank? i want to use it for gsm termination.
13:30.13Poincarejosehap: what line are you connected to?
13:30.43josehappstn in belgium
13:31.04Poincarewhich operator?
13:31.38josehapbelgacom
13:32.07Poincareyou have the problems on the fxs of fxo lines?
13:32.18josehapfxo
13:32.29Poincareall 3?
13:32.34josehapi'm sure because the sip phones get the same problems
13:32.40josehapyes
13:33.22Poincarewhat signalling are you using?
13:33.26josehapi use kewl start and when i watch the ztmonitor it seems ok as i see some activity after hangup
13:34.13speakmanAnyone arrived that are using RAS over ISDN?
13:34.16webographmrtwister: i just want * to route all calls starting with 0043 (austria, where i live) via $MyISP (using login-name and password) and all calls to germany (0049) to go via another sip-gw
13:35.33Poincarejosehap: I would say a lousy line then :-)
13:36.26josehapno solution then ? :)
13:36.35josehapmaybe the cabling is too old ?
13:36.45Poincareah
13:36.53Poincarewhat's the situation?
13:37.01Poincareand where are you located?
13:37.50josehapold house in center town of Liege and belgacom lines arriving at groud floor,  then patched on VVT to second fllor where the server is located
13:38.25Poincaredid you try to change signalling to loopstart?
13:38.39josehapno; what will it change ?
13:39.21Poincarethat it doens't try to find out for itself what signalling you are using
13:39.48josehapok
13:39.58josehapi'll try at empty call volume
13:40.15josehapalso, did you succeed in using the flash button on "belgian" phones
13:40.20josehapi mean belgacom ones
13:40.35josehapasterisk seems to never catch the event
13:40.58Poincareprobably timing
13:40.58blopgreat :) other belgian users :D
13:41.14Poincarebut I don't use analog phones on *
13:41.28josehapok
13:41.35*** join/#asterisk DarkDream (~Dreams@ip-213-49-161-150.dsl.scarlet.be)
13:41.43josehapmaybe  blop succeeded :)
13:42.00PoincareManxPower: not far from where I am
13:42.03blopthe flash thing ? :)
13:42.17josehapyes
13:42.41blopnever tried, coz i got a older pbx behind asterisk which catch the flash itself :p
13:42.41*** join/#asterisk makhtar (~ageller@mail.bulletinnews.com)
13:42.51josehap:)
13:43.01blopshould it work over an IAXy ?
13:43.15josehapok, i'll try again
13:43.19blopi may try that
13:43.20PoincareManxPower: I wasn't looking.... Can you wave again?
13:43.30josehapmaybe  we'll maje it working on iax in a few weeks
13:44.02bloplol
13:44.07*** join/#asterisk Pete_Largo (~Pete_Larg@adsl-65-71-225-121.dsl.rcsntx.swbell.net)
13:44.59cochimh any German users here, too? ;)
13:45.21speakmanAnyone got PPP working over PRI lines?
13:45.27blopwhich provider are u using for the belgian mobile phones? :)
13:45.52josehapbelgacom at the moment but i guess you have netter offer ;)
13:46.09blophum
13:46.09josehapif it's  the case i listen to you :)
13:46.11blopi cant find better rates
13:46.13*** join/#asterisk Samoied (~Samoied@popeye.opens.com.br)
13:46.14josehapbetter i meant
13:46.35josehapare you a dispatcher, some kind of centralization ?
13:46.49bublboblSoory all, one more newbie question : is it possible to tell asterisk that calls coming from a certain SIP gateway are PSTN ? :-$
13:46.55blop? :) nope i use asterisk @home
13:47.35mrtwisterbublbobl, why?
13:47.52Pete_Largoyeah, why?
13:47.56mrtwisterbublbobl, for asterisk is no matter form where call coming. do you using asterisk@home?
13:48.35bublboblI have on a same network voipPhones and a voip gateway. I don't know how to declare the gateway (not as an extension I guess) to be able to receive call (the gateway is ISDNIP gateway).
13:48.57bublboblmrtwister>  no, cvs*/debiansarge
13:49.16ManxPowertype=user sends calls to asterisk, type=peer receives calls from asterisk, type=peer sends and receives calls
13:49.40mrtwisterbublbobl, in my config i moved all dials to Local/
13:49.47mrtwisterit is nice for me
13:49.51*** join/#asterisk Cadu20 (~Cadu83@200-215-114-219.fnsce701.e.brasiltelecom.net.br)
13:50.02ManxPowerwhy do you need to declare the device as a "gateway".  SIP doesn't REALLY have the concept of "gateway"  all devices are "gateways", even if it just handles a single number.
13:50.16mrtwisterdial(Local/${EXTEN}@dial-exten)
13:50.36*** join/#asterisk Donuil (~fla@217.9.64.232)
13:50.38Cadu20Hi there, which SIP RFC does Asterisk support?
13:50.42mrtwisterregarding incoming, just route sip and h323 to somewhere
13:51.19bublboblmrtwister>  yes, that is for incoming, it works for outgoing
13:51.29Cadu20I´m having trouble trying to register a Phoneway FXO gateway. The support guy told me it can be the SIP RFC.
13:51.32Donuilhi to all... a simple question.. according to you can asterisk stream a wave file from URL?
13:52.56ManxPowerDonuil, try "show application playback" in the Asterisk CLI
13:55.37*** join/#asterisk Nuxi (~nuxi@cust-sdsl-204-250-82-202.bzn-co-i1000-01.bridgeband.net)
13:56.56mrtwisterDonuil, yes it can
13:57.03DonuilManxPower yes I've seen it but i don't understand if I can stream a file on my localhost or can stream a file from internet URL for example... can u suggest me anything?
13:57.30*** part/#asterisk Maksim (~max@213.142.207.20)
13:59.07bublboblDonuil>  I don't know if it helps maybe with mpg123 (it can reads streaming URLs and asterisk works with it)
14:00.23MikeJ[Laptop]good morning all
14:00.29ManxPowerDonuil, if it doesn't say you can, then you can't
14:00.54*** join/#asterisk hypa7ia (~leigh@1bb669ef10b16a71.session.tor)
14:00.55MikeJ[Laptop]morning ManxPower, you back from europe yet?
14:01.16Juggiehes gone for a montrh
14:01.17ManxPowerMikeJ[Laptop], no.  It's cold and rainy outside and my feet hurt, so I'm staying indoors for a little while.
14:01.34Nuggethaving a good time overall, though?  :)
14:01.41Nugget(yay europe)
14:01.48JuggieManxPower is ~eric@32.199-78-194.adsl-fix.skynet.be = no :)
14:02.08NuggetManxPower is een maffe koe.
14:02.13ManxPowerNugget, yes.  Stockholm has a great public transit system, but I didn't fall in love with the city.
14:02.40ManxPowerAntwerp has an OK public transit system.  The streets are so confusing that even the LOCALS often get lost.
14:02.54*** join/#asterisk MikeJ_ (~ircatjerr@pcp02795302pcs.roylok01.mi.comcast.net)
14:03.01ManxPowerAnd the bars here in Antwerp are...um...awesomely good on the weekends
14:03.16Nuggetinderdaad
14:04.10*** join/#asterisk tyra12 (~tyra12@203.131.137.77)
14:05.28ManxPowerI must admit that I'm looking forward to being in Amsterdam as well.
14:06.35dandreHello,
14:06.37dandreI have my asterisk box working mostly correctly but when I place a call, instead of hearing back the ring tone, I hear the music on hold. Both Zap and Sip channels are affected. Any Idea?
14:06.53*** join/#asterisk apardo (~apardo@80.26.164.145)
14:07.13tyra12hi guys....were knew to asterisk and planning to create later a SIP client for it....any advise which SIP library we could use?
14:07.27Cadu20Is Asterisk compliant with ALL SIP RFCs?
14:07.58ManxPowerCadu20, HAHAHAHAHA!!!!!  No.
14:08.16ManxPowerbut it works with the real world usage for the most common SIP RFCs in use.
14:08.18Cadu20Right, then, which RFC Asterisk support?
14:08.40Cadu20I think thats why I can´t register a PhoneWay FXO gateway...
14:08.50hypa7iaCadu20, afaik it supports a subset of the full sip functionality
14:08.52ManxPowerCadu20, It's never been an issue for me.  It Just Works.
14:09.10ManxPowerI've never had a SIP device that would not work with Asterisk, nor a SIP provider.
14:09.27Cadu20Can you use a Phoneway product in SIP PROXY MODE? I can´t... only in DIRECT CALL MODE.
14:10.19ManxPowerCadu20, Check out the asterisk instructions for using it with SER on the Wiki.  See if there's anything therer that will help you.
14:10.22Cadu20And the support guy told me that the Phoneway product is compliant with a specific RFC, and doesn´t work with other specifications.
14:10.32ManxPowerAlso, what's wrong with using it in direct call mode??
14:10.40*** join/#asterisk sudhir492 (~sudhir@4.7.59.136)
14:10.49ManxPowerCadu20, grep the asterisk source for that RFC
14:10.51sudhir492anyone there on this long weekend in US
14:11.02Cadu20I can´t use the gateway at a dynamic IP site.
14:11.10ManxPowersudhir492, Um, the long weekend was this past weekend
14:11.21Cadu20Thats why I need the register, so the gateway inform Asterisk of its IP.
14:11.42ManxPowerCadu20, register => userid:password@gatewayip doesn't work?
14:11.55sudhir492today is also part of the weekend practically
14:12.15Cadu20ManxPower, but I need register gateway => asterisk, not asterisk => gateway.
14:12.33ManxPowerCadu20, the "sip debug" is your friend.
14:12.51Cadu20NOO it isn´t! :( It is ugly... LOL :D
14:12.58sudhir492ManxPower: Do you know how to generate XML file for PAP2-NA configuration
14:13.01wwalkerAnyone know how to extend the tone duration for Dial and SendDTMF?  I have an agi that uses SendDTMF and sometimes the recieving system can get the tones, often not.  I would like to widen the tones to see if the dsp can pick it up better.
14:13.02ManxPowernot all friends are pretty.
14:13.10Cadu20Hehehe.
14:13.19ManxPowersudhir492, No.
14:13.24*** join/#asterisk onkeltimm (~chatzilla@213-84-102-203.adsl.xs4all.nl)
14:13.39Cadu20ManxPower, All right. Thanks a lot, again.
14:13.50sudhir492Anyone here knows how to generate XML file for PAP2-NA configuration
14:13.55Cadu20ManxPower, I´ll try the SER thing... i can´t wait anymore... HAVE to learn SER!
14:16.11sudhir492who uses Port 47 ?
14:17.34Nuxini-ftp           47/tcp    NI FTP
14:17.34Nuxini-ftp           47/udp    NI FTP
14:18.03*** join/#asterisk stilex (~occ@pcp01850015pcs.danbry01.ct.comcast.net)
14:20.55PatrickDKsudhir, you sure you don't mean protocol 47?
14:20.56wwalkeris there a pdf (or a make option) that makes printable vim manuals on US Letter (prefer double sided with enough space on the spine side for 3 holed paper....?
14:20.57PatrickDKgre?
14:21.07wwalkeroops, wrong window...
14:24.14*** part/#asterisk _omer (dfsdf@202.147.167.213)
14:27.33*** join/#asterisk mutilator (~animenodv@65.111.201.79)
14:28.25*** join/#asterisk vandien (~stephan@p50903D3C.dip.t-dialin.net)
14:32.09*** join/#asterisk sivana (~sivana@mixdown.ca)
14:32.16stilexhey anyone using nufone with a toll free DID? Wondering what hostname your iax.conf is set to register=> to
14:32.40stilexi tried all the ones i could find and they all get refused from their server
14:33.48sudhir492I am trying to hack PAP2 locked for vonage. I have upgraded the firmware to known PAP2-NA firmware. When the device comes up, it queries for ls.tftp.vonage.net to load cfg file. My servers resolves that and gives IP address itself. The device then tries to read spaMAC.xml file. I dont know how to create config file in xml format. Anyway I tried to compile a cfg file with spc and pass that to the device thinking that the downgraded firmware probably wont c
14:33.48sudhir492are for the extension. I see the spaMAC.xml file going over the wire. After that I see an ftp request on udp 47 from the device to the server. Since I am running vsftpd, which uses port 20, hence a message from sever to the device : icmp: udp port ftp unreachable [tos 0xc0]
14:33.51*** join/#asterisk Wi_Fi (~OUT@c-24-127-12-85.hsd1.ca.comcast.net)
14:34.10*** join/#asterisk jsolares (~jsolares@200.30.141.86)
14:35.29mishehueven though the pap2's were nice, I haven't bothered with them in a long time.
14:35.41mishehuthey kept changing the behavior of them.
14:36.27sudhir492mishehu: Thery are darn good (at least with the firmware I have) and very cost effective
14:37.18sudhir492Unfortunately, Cisco folks have increased the price of PAP2-NA and there is about $10 price differential between PAP2-NA and PAP2
14:37.35*** join/#asterisk QuaG|NaR (~quag1nar@adsl-0-173-39.mob.bellsouth.net)
14:38.03mishehuI thought you couldn't even get the pap2-na unless you were a distributor.
14:38.30ManxPowerI just buy SIPuras and figure the little bit of extra money is worth the reduced issues
14:38.53*** join/#asterisk pravz (~temp@219.95.247.22)
14:39.13mishehuyeah, I'd agree with ManxPower.
14:39.41pravzhi guys ...newbie in the house here... whats goin on
14:39.46mishehua few bucks more in return for less headache.
14:40.01sudhir492ManxPower: If you buy sipuras, then you are probably better off with PAP2-NA. They are cheaper than Sipuras, and they are Sipuras
14:40.11mishehupravz: for many of us, it's too early in the morning to have anything going on...
14:40.39pravzmishehu : cool dude...just curious
14:40.46sudhir492PAP2 is same but even cheaper thats why I a trying to open it. If I am successful, I am going to post the instructions on wiki
14:41.21mishehusudhir492: I wouldn't even bother posting instructions.  they will just change the behavior once again.
14:41.46mishehuplus, do you really want to attract the attention of vonage?
14:43.26QuaG|NaRwould anyone mind pointing me in the right direction getting asterisk to answer a pots line?
14:43.52QuaG|NaRim having some difficulties
14:44.05pravzguys... i've got asterisk1.0.7 downloaded, running on centOS4. i'm planning to connect my POTS line to my server and also connect analog phones to it...can someone give me a good link where I can get proper information/configuration guide for something like this...like i said newbie...really lost and dunno where to start
14:44.21mishehuQuaG|NaR: well, it would be a help if you told us what you have and what you've been trying to do.
14:44.41mishehupravz: http://www.voip-info.org
14:45.00mishehuit's a great wiki.  in my opinion, only second to wikipedia
14:46.11pravzthanks mishehu..will give it a go. will come to you if I have difficulties..thanks dude
14:46.13Nuxiyay. phpagi works on asteriskwin32.
14:46.41Ahrimaneseek, php AND win32, hehe
14:46.58mishehuasterisk and win32 is already scary enough
14:47.16Ahrimaneswin32 is scary enough..
14:47.43cochilinux has scary parts, too. lots of
14:47.44NuxiAlways good to do some portability testing tho.
14:48.36Nuxieagi support is only known to work in linux and freebsd with phpagi.
14:48.38Ahrimanesyes but at least linux has a native port.. win32 one uses cygwin and such.. too many bad things can happen
14:48.51QuaG|NaRive got asterisk1.0.7 on running with a generic x100p card.  im just trying to get it to answer the pots line at this point. the generic wildcard is listed in /var/log/messages and with ztcfg -vv.  ive got it set as channel 1 in zaptel.conf.  but i dont see it when i type 'show channels' while asterisk is running
14:48.53Nuxiwon't work in windows (probably ever)
14:49.18QuaG|NaRive got minimal answer, play in the exentions.conf
14:49.30mishehuQuaG|NaR: show channels only shows active channels.
14:49.35PatrickDKquag, you setup zapata.conf?
14:49.37NuxiI had to compile php in cygwin.  the native php port caused * to crash when it called it.
14:49.42QuaG|NaRso it will only show if the channel is open
14:50.06SuPrSluGQuaG|NaR:zap show channels
14:50.10QuaG|NaRi think my zapta.conf is setup.  ive got fxsks=1 loadzone-us defaultzone-us
14:50.47QuaG|NaRzap show channels doenst show anything either, of course none are in use at the moment
14:50.53SuPrSluGQuaG|NaR:in zapata.conf the context must = that used in extensions.conf
14:50.55NuxiI guess it's on to res_php for *.
14:51.18QuaG|NaRSuPrSluG, i bet that is my problem
14:51.35QuaG|NaRill go check, thanks very much
14:51.37PatrickDKquag, you need a channel=>1 in there at the end somewhere
14:51.56QuaG|NaRin zapata.conf?
14:52.00PatrickDKya
14:52.02SuPrSluGQuaG|NaR:it should show under zap show channels
14:52.48QuaG|NaRit doenst show any channels
14:52.55QuaG|NaRso i must have my configs wrong.
14:53.23PatrickDKya, you should probably see two channels
14:53.33PatrickDKpseudo and 1
14:53.35jsolaresit's signalling=fxs_ks not fxsks in zapata.conf
14:53.45QuaG|NaRit shows pseudo
14:54.01QuaG|NaRi do have signalling=fxs_ks
14:54.06jsolaresand of course as PatrickDK said, channel=>1 to assign all of those configurations to that channel
14:54.15QuaG|NaRchannel => 1
14:54.44QuaG|NaRbut in extensions.conf i need to have the same context name for it to pickup?
14:55.02QuaG|NaRmaybe im ahead of myself
14:55.07PatrickDKya
14:55.19PatrickDKcontext=(something valid in extentions.conf)
14:55.21jsolaresin zapata.conf you have to set the context that you will use for incoming calls in your dialplan (extensions.conf)
14:55.25PatrickDKbut that isn't your problem yet
14:55.25*** join/#asterisk MikeJ[Laptop] (~ircatjerr@pcp02795302pcs.roylok01.mi.comcast.net)
14:55.38QuaG|NaRok
14:55.58jsolaresyeah, it has to show with a zap show channels
14:56.26QuaG|NaRok, zap show channels shows - pseudo - no channel 1 or anything
14:56.37SuPrSluGQuaG|NaR:it should look like signalling=fxs_ks
14:56.38SuPrSluGcontext=inbound
14:56.40SuPrSluGchannel=>1
14:56.48*** join/#asterisk MikeJ[Laptop] (~ircatjerr@pcp02795302pcs.roylok01.mi.comcast.net)
14:57.18QuaG|NaRSuPrSluG, you mean in zapata.conf?  ive got that in there
14:57.22PatrickDKhmm, he was saying fxsks=1, is he editing zaptel.conf or zapata.conf?
14:57.39dandreHello,
14:57.41dandreI have my asterisk box working mostly correctly but when I place a call, instead of hearing back the ring tone, I hear the music on hold. Both Zap and Sip channels are affected. Any Idea?
14:57.42SuPrSluGQuaG|NaR:yes. and don't forget to restart * when done
14:57.53QuaG|NaRzapata.conf has signalling=fxs_ks as well as channel => 1
14:57.53jsolaresQuaG|NaR, what SuPrSluG said should go in zaptel.conf
14:58.10PatrickDKjsolares, zapata.conf you mean
14:58.10jsolaresno wait now you're confusing me -_-;;
14:58.11*** join/#asterisk L|NUX (linux@202.5.131.101)
14:58.47SuPrSluGQuaG|NaR:no that's zapata.conf
14:59.01jsolaresindeed, zaptel.conf should have: fxsks=1
14:59.18jsolaresand a loadzone=us
14:59.34Pete_Largo~dialplan
14:59.35jboti guess dialplan is the thing configured in extensions.conf
14:59.43QuaG|NaRok, my zaptel.conf also has defaultzone=us and those jsolares listed
15:00.10*** part/#asterisk wwalker (~wwalker@wwalker.sustaining.supporter.pdpc)
15:00.13cochigerman users here experiencing voicemail probs?
15:00.16jsolaresQuaG|NaR, run zttool, it should tell you if it's configured
15:00.40PatrickDKhmm, my zaptel.conf has, defaultzone=us, loadzone=us, fxsks=1
15:00.48*** join/#asterisk Mc_Tr (~Mc_Tr@bacterio.knet.es)
15:00.49QuaG|NaRthats what mine has
15:00.51jsolaresand ztcfg -vv should also say what's configuring
15:01.00Pete_Largo~extensions.conf
15:01.01jboti guess extensions.conf is at http://voip-info.org/wiki-Asterisk+config+extensions.conf, or know as dialplan, or known as extensions, or known as exten
15:01.03QuaG|NaRzttool shows generic clone board one
15:01.22jsolaresalarms? is it configured?
15:01.25QuaG|NaRztcfg -vv shows 1 channels configured
15:01.37QuaG|NaRno alarms
15:01.56jsolareshave you restarted your asterisk?
15:01.58PatrickDKzapata.conf has, [channels], language=en, context=pstn-in, signalling=fxs_ks, channel -> 1
15:02.28QuaG|NaRrestart will work right?
15:02.34SuPrSluGQuaG|NaR:now restart asterisk and zap show channels should show your card
15:02.37*** join/#asterisk nowork (~jfu2808@216.254.141.97)
15:03.03QuaG|NaRzap show channels still only shows the pseudo
15:03.10Pete_Largodandre check here for ringing  ...  http://voip-info.org/wiki-Asterisk+-+documentation+of+application+commands
15:03.28PatrickDKquag, you kill asterisk, and started it again?
15:03.35*** join/#asterisk MikeJ_ (~ircatjerr@pcp02795302pcs.roylok01.mi.comcast.net)
15:03.40QuaG|NaRi killed asterisk and started it with asterisk -vvvvc
15:03.58Mc_Trhas anybody use codec g726?
15:04.04PatrickDKquag, can you paste your zapata.conf to pastebin?
15:04.20QuaG|NaRpastbin?
15:04.33Mc_Tri configure a Linksys PAP with G.726
15:04.35SuPrSluG~pastebin
15:04.36jbotpastebin is, like, a place to paste your stuff without flooding the channel - try http://pastebin.ca
15:04.42Mc_Trbut asterisk not recognize the codec
15:04.42*** join/#asterisk MikeJ[Laptop] (~ircatjerr@pcp02795302pcs.roylok01.mi.comcast.net)
15:05.02QuaG|NaRyea ill get it pasted in just a sec
15:05.48noworkhi, so many hacker/cracker's trying my asterisk/linux passwd, I want to know the setting for my iptables;should i go to linux channel? but, i don't know any linux channel,anyone can give me one?i am using Fedora..thx;
15:06.57Nuxinowork, #fedora
15:07.09noworkthx
15:07.28*** join/#asterisk MikeJ[Laptop] (~ircatjerr@pcp02795302pcs.roylok01.mi.comcast.net)
15:07.48dandrePete_Largo: I just use Dial command in my dialplan and I don't see why MOH is played instead of ring tone.
15:08.06PatrickDKdandre, what is your dial command?
15:09.06SuPrSluGdandre: do you have a lower case m in the options?
15:09.19QuaG|NaRPatrickDK, my zapata.conf file is located at http://pastebin.ca/12962 there is other stuff in the file but i commented out everything except these lines
15:09.43dandrePatrickDK: exten => _00.,1,Macro(dial-out,${out_isdnft},${EXTEN:1}) where
15:09.44dandre[macro-dial-out]
15:09.46dandreexten => s,1,Dial,${ARG1}${ARG2}|30|m
15:10.07PatrickDKhmm, change m to r
15:10.13PatrickDKor maybe just adding r will fix it
15:10.16Pete_Largom = music on hold
15:10.33Pete_Largor = ringing
15:10.36Pete_Largoso change m to r
15:11.36*** join/#asterisk MikeJ_ (~ircatjerr@pcp02795302pcs.roylok01.mi.comcast.net)
15:11.51dandrePatrickDK: Pete_Largo  Thanks for your answers and sorry for such a stupid question!
15:12.22Pete_Largono stupid questions, only stupid answers :)
15:14.03Mc_TrPlease, anoyone have a sipura to work with codec g726 with asterisk?
15:14.29PatrickDKQuaG|NaR, the only real thing I have that you don't is, switchtype=national
15:14.42QuaG|NaRill put that in....
15:14.43PatrickDKbut I don't think that matters on pots
15:14.47QuaG|NaRlol
15:15.05Mc_Try have asterisk configure with: disallow=all, allow=g726
15:15.23Mc_Trand sipura with g726-24
15:15.32PatrickDKquag, oh, heh, you have a typo though
15:15.39PatrickDKusercallerid should be usecallerid
15:15.40QuaG|NaRtransfer?
15:15.42Mc_Trbut in /var/log/asterisk/full y see codec error ;(
15:15.44QuaG|NaRahhh
15:15.57PatrickDKoh y, and the transfer thing :)
15:16.34QuaG|NaRis there anywhere else i should look as to why asterisk dont show the card?
15:17.21QuaG|NaRi started with minimal stuff in the zapata.conf but started adding things as i was reading more 'how to get it working'
15:18.15dandreAnother one:
15:18.17dandreI have an analog phone say PhoneA connected to a tdm400. When a phone call is established between PhoneA and another party, I want to place this call on hold and dial another one
15:18.18dandreSo I press flash and dials: this works well BUT I can't recover the other and filp flop on the two conversations.
15:18.20dandreIf I press Flash I enter conference mode between the three and another flash hangs up the last call.
15:18.21dandreAny Idea?
15:18.35seanI resolved my IAX over NAT problem if anyone cares.. or anyone remembers me asking about it..
15:20.31PatrickDKheh, that is odd, iax just works over nat, witohut problems
15:20.43PatrickDKas long as your portfwd it for incoming connections
15:20.52PatrickDKand your nat handles udp correctly
15:21.24*** join/#asterisk Ariel_ (~Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net)
15:21.40Ariel_hello everyone
15:22.23QuaG|NaRok, ive got zap show channels showing channel 1
15:22.28seanPatrickDK: well, UDP, being connectionless, works over NAT, _usually_. (-:
15:22.55seanthe problem in my case is that my * box had more than one IP, so data came in on one IP and was sourced on another (because UDP is connectionless).
15:23.18Juggiesounds like a misconfiguration :)
15:23.18QuaG|NaROMG!!  hey thanks for the help guyz
15:23.24seanThe NAT box in between saw that client was sending to server1, but server 2 was sending packets back to the firewall, and the firewall didn't know where to deliver them.
15:23.25QuaG|NaRyou helped me find where to look
15:23.32QuaG|NaRi really appriciate it, asterisk is answering now
15:23.33QuaG|NaRthanks a lot
15:23.50seanso, I fixed my sourcing issue, and NAT started passing packets as expected
15:24.12seanJuggie: of course it was misconfiguration.. but it was extemely non-obvious
15:24.30seanif I get time, I'll drop it on the wiki..
15:24.41drumkillaI don't think that should be a problem in cvs head
15:25.11MikeJ_you don't think.. you don't think...
15:25.20drumkillaha
15:25.27MikeJ_what is all this thinking about?
15:25.32drumkillawell, so many things get changed, it's hard to remember!
15:25.40MikeJ[Laptop]:)
15:25.53drumkillabut I think part of the work on ast_netsock was to address that exact issue
15:26.02Juggiedrumkilla, what are we looking at for 1.0.8?
15:26.04drumkillaand keep track of where stuff arrived, so we know where to send it from
15:26.11drumkillaJuggie: good question
15:26.15*** join/#asterisk cmk (~cmk_@p54A3D13C.dip.t-dialin.net)
15:26.28Juggiei havnt updated my cvs-stable in sometime
15:26.33drumkillaJuggie: I guess I should just start the RC cycle now
15:26.57Juggieis there a changelog from 1.0.7 to now?
15:27.12drumkillayeah
15:27.17drumkillafor the bigger stuff, at least
15:27.25drumkillait's in CVS
15:27.45seandrumkilla: you mean * is smart enough to source on the same IP it received a packet, now?
15:28.01seanah.. I should've kept reading.. (-:
15:28.11drumkillasean: in cvs head, at least
15:28.17drumkillabut don't shoot me if I made that up
15:28.38Juggiedrum, see *-stable
15:28.46seancool, thanks.. that's a much better solution than playing with my route metrics and re-initializing interfaces until it happens to work.
15:29.02*** join/#asterisk af_ (~af@ip-130-170.sn2.eutelia.it)
15:29.06af_hi to all
15:29.32af_it is possible to integrate loquendo in asterisk?
15:31.04Jedirlloquendo!
15:31.18Jedirlif you can afford loquendo, you should be able to afford loquendo's support
15:31.22Jedirlto do it :D
15:31.56Jedirlaf_: there's an app_cepstral that integrates cepstral swift TTS in asterisk... you can take it as a starting point to do your own implementation
15:32.16af_Jedirl: where could I find it?
15:32.25Jedirlin google, searching for app_cepstral.c
15:32.26Jedirl:)
15:32.30af_I am reading something in the ml list.
15:32.32af_oh
15:32.46af_loquendo is so expensive? just see a demo, it's great
15:32.55Jedirlloquendo is ***EXPENSIVE***
15:34.18*** join/#asterisk ezh_newbie (~ezh888@222.124.83.54)
15:34.57*** join/#asterisk cjk (~cjk@80.92.75.120)
15:34.59bewestwhat is loquendo?
15:35.24cjkhi, did anyon got oh323 and g729 wokring with cisco callmanager? i only get it working with g711
15:37.03Jedirlbewest: the best Text-to-speech in the world
15:37.06Jedirlbewest: by far
15:37.34*** join/#asterisk ezh_newbie (~ezh888@222.124.83.54)
15:38.31bewestdoesn't happen to be opensource does it?
15:39.02Jedirlhehehehe
15:39.02bewestfestival's not too great
15:39.08Jedirlno, it isn't opensource
15:39.16bewestwe are trying to integrate TTS in our apps
15:39.20Jedirltry cepstral
15:39.23Jedirlit's cheap
15:39.26Jedirlnot opensource but cheap
15:39.28bewestso far onl looked at festival
15:39.40Jedirland its english is moderately good
15:39.52Juggiehave you tried AT&T TTS?
15:40.08Juggietry, hey,
15:40.08Juggiei had a pretty busy weekend...it was a good one, despite the crappy weather. tennis was fun, then we played with the aerobie, sunday we had a movie marathon and we made a fancy dinner. so that was fun. got home pretty late...but all in all, it was good. now i'm at work being a busy bee.
15:40.08Juggiehope you had a good weekend!
15:40.10JuggieKrysia
15:40.12Juggieack...
15:40.14Juggiebad paste :)
15:40.17Juggiewops.
15:40.18JedirlHEHEHE
15:40.25Juggienothing like pasting emails
15:40.38*** join/#asterisk inspired (mikael@213.197.167.61)
15:40.41Juggielets try this again
15:40.50bewesthaven't tried AT&T's
15:40.54Juggiehttp://www.research.att.com/projects/tts/demo.html
15:41.05*** join/#asterisk darkskiez (~darkskiez@usergc137.dsl.pipex.com)
15:41.05Juggiepasting between remote desktops sometimes does funny things
15:42.23bewestaren't some components of AT&T and festival the same?
15:42.38Juggieno idea
15:42.44Juggiei've only tried it via this www demo page
15:42.48Juggiebut the voices are quite good
15:43.45QwellJuggie: I added a note to that bug last night.  Wanna take a look?
15:44.43Juggiei saw it this morning before i went to work
15:44.47Juggieand when i got here i got busy
15:44.48Juggielets see
15:46.04Juggiek i see it
15:49.20bewestanyone have success playing with festival's settings and raising the quality a bit?
15:49.22*** join/#asterisk WilliamK (~wkeller@c-24-0-130-177.hsd1.tx.comcast.net)
15:50.38*** join/#asterisk smurfix (~smurf@smurfix.developer.debian)
15:51.49*** join/#asterisk funxion (~chatzilla@69-165-200-17.miamfl.adelphia.net)
15:52.38funxionhello
15:52.40dandreUsing Flash on Chan Zap:
15:52.41funxionanyone here
15:52.41dandreI have an analog phone say PhoneA connected to a tdm400. When a phone call is established between PhoneA and another party, I want to place this call on hold and dial another one
15:52.43dandreSo I press flash and dials: this works well BUT I can't recover the other and filp flop on the two conversations.
15:52.44dandreIf I press Flash I enter conference mode between the three and another flash hangs up the last call.
15:52.46dandreAny Idea?
15:54.03*** join/#asterisk Sedorox (brandon@sedorox.staff.smartserv)
16:01.33*** join/#asterisk coppice (~chatzilla@14.198.17.210.dyn.pacific.net.hk)
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16:05.21*** join/#asterisk izo (~izo@izo.warpl.ipxxi.pl)
16:07.35*** join/#asterisk [TK]D-Fender (~joe@4.67.252.216.dsl1.colba.net)
16:09.34*** join/#asterisk cpatry (~grepmoo@65.39.228.5)
16:10.41*** join/#asterisk Patrick^ (~patrickm@birch4.mountaincable.net)
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16:20.45*** join/#asterisk otmar (lendl@arachne.bofh.priv.at)
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16:24.45*** join/#asterisk sudhir492 (~sudhir@4.7.59.136)
16:25.35Jedirlanyone could help me ? i have a HFC-S BRI card which zaphfc detects (zaphfc: 1 hfc-pci card(s) in this box)
16:25.44Jedirlbut zap show channels doesn't show me any channel
16:25.49mutilatorwhat happened to all the pricing on digiums site for their cards
16:26.26mutilatorwhats a TE410P go for
16:28.04darkskiez1499$ ish i think
16:28.25*** join/#asterisk newmedian (~crowlther@Quebec-HSE-ppp230300.qc.sympatico.ca)
16:29.57Corydon76-home$1495
16:30.16Corydon76-homeDitto for TE405P
16:30.43Corydon76-homeSomehow, I think they're selling far more of the 405's than the 410's
16:31.58funxionthe diff is just pci voltage no
16:32.06Sedoroxyes
16:32.37noworkdon't kick me out, one quick question u guys must expert on it:
16:32.57noworkit's on my asterisk linux box
16:32.57nowork<PROTECTED>
16:32.57nowork<nowork> any idea? how to modify lines under  "Chain
16:33.17JedirlI'm having problems setting up my zaptel interface... May 30 18:31:52 WARNING[3106]: chan_zap.c:925 zt_open: Unable to specify channel 1: No such device or address
16:33.17nowork"Chain RH-Firewall-1-INPUT (2 references)"
16:33.25*** part/#asterisk PandemiK (~PandemiK@62.2.255.42)
16:33.30Jedirlbut zaphfc in dmesg says I have a card configured
16:33.44Jedirlzaphfc: 1 hfc-pci card(s) in this box.
16:33.46otmarJedirl: what does ztcfg -v report
16:33.50otmarand zttool?
16:33.56SedoroxJedirl: doesn't mean you have all the config files setup
16:33.58Jedirlcan I paste?
16:34.00Jedirl[root@ccard01 zaphfc]# ztcfg -v
16:34.00JedirlZaptel Configuration
16:34.00Jedirl======================
16:34.00JedirlSPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1)
16:34.00Jedirl3 channels configured.
16:34.15JedirlI don't have zttool
16:34.46otmarzttool is in the zapata package, but only compiles if you have some curses libs
16:34.55Jedirlwhop
16:35.04JedirlI have ncurses-devel installed
16:35.19otmarit's not curses, some other thing
16:35.24otmarcheck the makefile
16:35.29Jedirlok
16:36.04otmarbut your ztcfg output looks fine to me
16:36.11Jedirlheh, incredible
16:36.16Jedirlasterisk now works
16:36.27Jedirlmaybe I was doing something wrong ... I don't understand this
16:36.47otmaryou need to run ztcfg after you load the kernel module
16:37.05bewestwhat if you don't?
16:37.43otmarusually modprobe takes care of this, but if you have a problem in you modules config then you need to do it manually
16:38.04Jedirloh
16:38.30bewestI'm up for trying anything to improve meetme results with ztdummy
16:38.39bewestit's unuseable as is (2.6 kernel)
16:39.21*** part/#asterisk darkskiez (~darkskiez@usergc137.dsl.pipex.com)
16:40.05dandreUsing Flash on Chan Zap:
16:40.06dandreI have an analog phone say PhoneA connected to a tdm400. When a phone call is established between PhoneA and another party, I want to place this call on hold and dial another one
16:40.08dandreSo I press flash and dials: this works well BUT I can't recover the other and filp flop on the two conversations.
16:40.09dandreIf I press Flash I enter conference mode between the three and another flash hangs up the last call.
16:42.24otmaranyway, I need AGI help: I'm using Asterisk::AGI and I'm trying to do somethig like $AGI->exec("SayAlpha","sip:bla"); but the SayAlpha application interprets the : as delimiter and skips the "bla" bit
16:43.53bewesttry parentheses?
16:44.20otmaronly tried various " \" and \\\"
16:44.21otmarwill do.
16:44.33bewest"SayAlpha", "(sip:bla)"
16:44.36bewestsometimes that works for me
16:44.41bewestmakes everything group together
16:45.17*** join/#asterisk mamcinty (~mamcinty@adsl-068-209-174-113.sip.int.bellsouth.net)
16:45.21bewestwhat does it show it's doing in CLI?
16:45.27bewestis it only getting the sip part?
16:45.40otmaryes. just S I P
16:45.50Jedirlhow can I configure in zaptel interfaces if calls are signaled as national or international?
16:46.06otmarzapata.conf
16:46.45otmaron a PRI it's pridialplan=...
16:46.56Jedirlon a BRI i guess it's the same
16:47.01otmarno idea about BRI
16:47.04vandienhi! i think some one told me sth about a asteriskAThome like ISO based on debian... does anybody know? i forgot the URL... :(
16:47.15JedirlI hate the margination BRI suffers on asterisk :(
16:47.19vandienlol, got it, thx ;)
16:47.19newmedian~aah
16:47.21jbotwell, aah is Asterisk@Home. The Asterisk@Home support forum is here: http://sourceforge.net/forum/forum.php?forum_id=420324
16:48.04vandieni meant xorcom.com/rapid :)
16:49.25*** join/#asterisk gambolputty (~gambolput@cblmdm72-240-241-108.buckeyecom.net)
16:49.31funxionvandien xorcom is based on debian
16:49.36funxionits not that great though
16:49.43funxioneasier to jsut build it yourself
16:49.53newmedian~distrochooser
16:49.54jbotmethinks distrochooser is a web page wizard that helps you choose which Linux Distro is most suited to your needs, at http://www.zegeniestudios.net/ldc/index.php
16:50.18Jedirlhow can I specify multiple "nationalprefix = "
16:50.20funxionisnt there a linux dist that is made specific for asterisk
16:50.20Jedirlin zapata.conf?
16:50.27Jedirlfunxion: xorcom rapid
16:50.38funxionno xorcom is based on debian
16:50.46vandienfunxion: well, i just want to have a look :)
16:50.57funxionvandien its not bad
16:51.02Jedirlit's a asterisk-specific debian
16:51.04funxionbut I didnt like it
16:51.09Jedirlwhy ?
16:51.14funxionjedirl I've tried it
16:51.30vandienhas it CAPI support?
16:51.31funxionlost functionality not the easiest to navigate
16:51.42JedirlI don't think it has capi support
16:51.50funxionplus IM a control freak and u lose a bit of control with xorcom
16:51.56Jedirlhow can I specify multiple "nationalprefix = " in zapata.conf?
16:52.07vandienhm, i need (want) CAPI :)
16:52.19*** part/#asterisk K9DI_BSD_WrkStn (~k9bsd@207-246-185-168.EastVillage.ResNet.wiu.edu)
16:52.45newmedianEuro ISDN http://www.junghanns.net/asterisk/
16:53.15Jedirlvandien: bristufff
16:53.19vandienyes... but i dont get it to work with my normal debian & asterisk
16:53.39vandieni got CAPI to work, but seems like asterisk doesnt like it
16:53.40Jedirlvandien: you need a patched asterisk, patched libpri and patched zaptel
16:53.58vandienuhm... i used debian packages
16:54.09Jedirlthen you'll have to re-engineer packages
16:54.14vandienasterisk-chan-capi
16:54.23vandienwhat does that mean?
16:54.31vandien<-- not a linux / debian pro
16:54.33funxionvandien dont use debain packages for asterisk
16:55.00JedirlI've never used debian packages
16:55.07funxionthey're usually older and wont work with asterisk cvs
16:55.07Jedirlyou'd better use cmpiled packages
16:55.14Jedirlcompiled sources, sorry
16:55.27funxiondebian packages are ok for other things but not asterisk related
16:55.30funxionnot yet at least
16:55.35vandienargh, i have to go now.. are you still here in about 2 hours? i'd love to hear from you 2 :)
16:55.48vandien*more
16:55.48funxionit would be nice if debian packages were cvs head
16:56.08*** join/#asterisk map71 (~map@201.248.61.162)
16:56.08vandienare you? then cya later...
16:56.42vandienbye
16:57.49newmedianjob seen SarahEmm
16:57.56newmedianjbot seen SarahEmm
16:57.58jbotsarahemm <~sarahemm_@Toronto-HSE-ppp3704598.sympatico.ca> was last seen on IRC in channel #asterisk, 4d 13h 34m 32s ago, saying: 'twisted: i'll try to post a patch when i get to sjc (i have inet access there, and will be working on * inflight)'.
16:59.15*** join/#asterisk bofh42 (~bofh42@p54822E56.dip0.t-ipconnect.de)
17:10.58*** join/#asterisk doolph (doolph@201.226.146.178)
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17:15.25Jedirlhehehe
17:15.37*** part/#asterisk cpatry (~grepmoo@65.39.228.5)
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17:22.18JedirlI have two BRI channels, but when I do zap show channel 1 (one of these channels) I get:
17:22.18JedirlSignalling Type: PRI Signalling
17:22.42Jedirlbut it should be EuroISDN
17:22.57santiagoHimeko, I've a very strange problem with an *, some calls are dropped after May 23 18:35:43 DEBUG[219]: Auto destroying call
17:23.00santiagommm
17:24.16santiagoHimeko, I've a very strange problem with an *, some calls are dropped after some seconds and the only weird message that i found is: May 23 18:35:43 DEBUG[219]: Auto destroying call ... , anyone could help me?
17:25.38*** join/#asterisk fa__ (faceoff@devel.acdbddh.eu.org)
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17:38.45bewestwhat does this mean in *?: That's odd...  Got a response on a call we dont know about.
17:41.23NivexSomeone sent you a frame with a call ID that isn't in your table of active calls.
17:42.07santiago* = asterisk
17:42.59flotoxrotfl
17:43.05stilexanyone seen JerJer
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17:44.31*** join/#asterisk Chotaire (chotaire@chotaire.net)
17:46.16Qwellanybody willing to test a patch for voicemail in odbc?  bug 4403...could use a few more testers
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17:50.23newmedianjbot seen jerjer
17:50.26jbotjerjer <~JerJer@DSL-226.206-rt-bras.che.centurytel.net> was last seen on IRC in channel #asterisk, 15d 4h 38m 12s ago, saying: 'ManxPower:  sounds interesting, but i've still never used it'.
17:51.11Qwell~seen jerjer[mobile]
17:51.12jbotjerjer[mobile] <~jj@ip68-103-26-140.ks.ok.cox.net> was last seen on IRC in channel #asterisk, 2d 10h 22m 58s ago, saying: 'vlc fought pretty hard, but I won -  been streaming video to a friends place for most of the day'.
17:51.19newmedianahh yes.
17:51.54shido6.
17:52.07Qwellshido6: test my patch! :p
18:02.06tzangershido6: is jerjer MIA?
18:02.21*** join/#asterisk [TK]D-Fender (~joe@4.67.252.216.dsl1.colba.net)
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18:09.40mikewho2sup fellas
18:11.14[TK]D-FenderQuestion I could use som tips on :  I'm running Polycom IP 600's and the following wierd thing happens :  I go into voicemail, access a few option, place the call on hold, come back and digits don't work anymore.  Any guesses on why that is?
18:11.36hypa7iathat's really weirf [TK]D-Fender
18:11.51hypa7iaweirf... as in weird and freaky
18:14.51[TK]D-Fender:/
18:15.14[TK]D-FenderI am runnin 1.05 though, so I guess I'll see what happens when I upgrade to 1.07
18:15.27[TK]D-FenderNot sure if I should touch CVS-head.
18:15.36[TK]D-FenderFor production use that is.
18:22.48khmter
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18:24.45santiagoHi, I've a very strange problem with an *, some calls are dropped after some seconds and the only weird message that i found is: May 23 18:35:43 DEBUG[219]: Auto destroying call ... , anyone could help me?ls
18:25.15RoyKuse the source, luke
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18:29.40TheJudgehello all / evening / morning :)
18:30.52TheJudgecould someone help me with an asterisk problem namley I pass a call to it from my ser sip proxy, however asterisk does not seem to pass back if it has answered the call or not
18:31.08TheJudgeSo my ser server billis even if there was no answer on the asterisk box ?
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19:14.43shmaltzwhy would I get this:
19:14.44shmaltzUnable to create channel of type 'Zap' (cause 0)
19:15.02chaineyanyone live in los angeles want to sell me some/a voip/sip phones? will pay cash today ;)
19:16.29tzangershmaltz: in asterisk/include/asterisk/causes.h what's cause 0?
19:16.59*** join/#asterisk iq (~IQ@209-180-111-69.omah.qwest.net)
19:17.23Qwellchainey: I'll sell you my TDM400p :p
19:17.28tzangerI agree though it'd be nice to have all that cause shit spelled out... Unable to create channel of type 'Zap' (Cause 0: LP0 on fire)
19:18.25chaineyQwell: hehehe, sure, tempt me with your analog ;0
19:18.57*** join/#asterisk folsson (~filip@h82n1fls32o985.telia.com)
19:18.58shmaltztzanger, this is what I get in causes.h:
19:19.00shmaltz#define AST_CAUSE_NOTDEFINED 0
19:19.02chaineyalthough that would help with my tivo situation ;)
19:19.15shmaltznow, I know it's defined since it was working until now, and it's working now
19:19.28ManxPowershmaltz, what does "zap show channels" show"?
19:19.58tzangershmaltz: haha, ok well that's the problem
19:20.07shmaltznow its working and I get the right stuff
19:20.11tzangerI am guessing that whatever's causing it is bad logic or something whihc isn't SUPPOSED to fault out
19:20.21tzangerhello ManxPower
19:21.50shmaltztzanger, meaning?
19:22.01tzangermeaning someone zigged when they should have zagged
19:22.08tzangerI'm thinking the bad logic is something like
19:22.14shmaltzgoon
19:22.18shmaltzgo on
19:22.38shmaltzlike what? tzanger
19:22.41*** join/#asterisk DEEZED (deezed@adsl-065-006-189-182.sip.bct.bellsouth.net)
19:22.57tzangerif(result) { ast_log(ERROR,"Can't create channel of type \'%s\': %d\n", chantype, cause); }
19:22.59ManxPoweror perhaps he had Zap/1 and that port was in use.
19:23.00tzangerwhen they meant !result
19:23.10tzangerManxPower: that should not givre a cause code of 0
19:23.32ManxPowertzanger, yeah, a lot of things should not give a cause code of 0
19:23.53tzangeralternatively it coudl be something where the cause code is getting set in a deeply nested block of logic where it should be set earlier up
19:23.59tzangercause code of 0 is no cause
19:24.02tzangeror undefined cause
19:24.05tzangereither one is bad
19:24.23shmaltzso how can I avoid this problem in the future?
19:24.30shmaltzmaybe use stable?
19:24.37QwellManxPower: hows your tour going?
19:24.38tzangershmaltz: what fixed it?
19:24.39ManxPowershmaltz, Maybe. 8-)
19:24.58shmaltztzanger, the same thing that created it
19:25.00shmaltznothing
19:25.04tzangershmaltz: ahh
19:25.07tzangeryeah that's no fun
19:25.11ManxPowerQwell, The weather today sucked, but I hope with the addition of some alcohol to my system shortly I will feel better about it.
19:25.13shmaltzexactly
19:25.22shmaltzManxPower, where you located?
19:25.32shmaltzcause the weather here is beautifull
19:25.38cypromishere it is 32C
19:25.45tzanger32C?  holy fuck
19:25.45ManxPowershmaltz, right now?  Antwerp, next week Amsterdam, the week after that, Madrid
19:25.46tzangerthat's hot
19:25.58shmaltzAntwerp weather always sucks
19:25.59tzangerit's like 18C here and it's beautiful
19:25.59ManxPowerlast week Stockholm
19:26.09cypromisstockholm had nice weather
19:26.30shmaltzthey say that in Antwerp it rains 9 months a year, and the other 3 months it pours
19:26.36tzangershmaltz: :-)
19:27.26shmaltz~antwerp
19:27.27jbothmm... antwerp is a city where it rains 9 months a year, and the other 3 it pours :)
19:29.56[TK]D-Fenderhey shmaltz, how goes?  I've got the Poly IP 600 working ok, am still a little peaved at how * queues work but have another interesting problem you probably have experience with.  When I call voicemail DTMF works fine, I put the call on hold and come back, it doesn't.
19:30.27shmaltzhmmmmmmm, let me test this
19:30.43[TK]D-FenderTried DTMF inbad and RFC for that....
19:30.51[TK]D-Fenderinband*
19:31.51shmaltzD-Fender, you using DTMF inband?
19:31.58[TK]D-Fendertried both.
19:33.54[TK]D-FenderNVM, RFC seems to work now.... wonder what I did wrong....
19:34.40*** join/#asterisk Andrezo (~www@217.129.208.124)
19:34.46[TK]D-Fender*sigh*
19:35.56tzangerhttp://www.pointlesswasteoftime.com/games/wargames.html
19:35.57tzangertoo funny
19:38.54shmaltzD-Fender, I guess when you first tried DTMF it wasn't really applied, either on the phone side or * or both
19:39.34[TK]D-FenderAs log as it workss.  I also have to upgrade to 1.0.7.  That's got me a little more nervous than it should.
19:39.39[TK]D-Fenderlong*
19:41.33Blissextzanger: good pointer :-)
19:41.42tzanger:-)
19:42.03heath__we got 2 te410p cards in a server, and the second one doesn't get any interrupts, anyone know how to fix this?
19:42.33tzangerheath__: did you adjust teh ID switch?
19:42.54heath__i'm not familiar with that, is it on the board?
19:43.12tzangeryes
19:43.18tzangerit's fairly obvious :-)
19:43.27heath__i got a dude a the co-lo right now
19:43.35heath__what should he look for?
19:43.49heath__a=at
19:43.59tzangerhttp://www.mixdown.ca/~asterisk/TE405P-3.3v-2.jpg
19:44.14tzangeryou see, near the bottom right of hte pic
19:44.27tzangerthe red thing with 0-9A-F and "Ident" written near it?  :-)
19:45.47heath__i see it
19:45.53heath__i bet they are both set to 0
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19:47.12heath__it should probably be 0 for one, and 1 for the other?
19:49.19heath__yeah i guess he already has the ids set right
19:49.20tzangeryup
19:49.48heath__cat /proc/interrupts shows an interrupt for one card, but then 0 for the other
19:50.13dudesYou'd think a Dual P4 Dell Server wouldn't be a such a ......
19:51.40*** join/#asterisk _santiago_ (~santiago@200.24.109.199)
19:52.10heath__i should only have to do modprobe once right?
19:52.26cluecon[file]my mother is freaking out again, and it's getting on my nerves
19:52.43Qwellcluecon[file]: What'd you do this time?  heh
19:52.44dudeswell, tell her to shut up then
19:52.56cluecon[file]I decided something for myself, OMG
19:53.13Qwellcluecon[file]: what was that "something"?  It matters, heh
19:53.23dudeswhat'd you do?  Get a piercing?
19:53.30cluecon[file]ha no
19:55.59cluecon[file]needless to say she won't speak to me now without swearing at me
19:56.12Qwellcluecon[file]: sounds like my mom
19:56.23Qwellexcept less psycho
19:56.28dudesDid you have unprotected sex or something?
19:56.54cluecon[file]lol no
19:57.02Qwellsomething simple, like...
19:57.17cluecon[file]buttttt I guess I won't be getting my passport application done tonight
19:57.18dudestrack alittle dirt to your bedroom
19:57.28*** join/#asterisk dca (~dca@c-67-166-37-218.hsd1.co.comcast.net)
19:57.43Qwellno, no, it has to be something he did "right"
19:57.50cluecon[file]I was in a great mood too until I got home
19:57.55Qwelllike...taking out the trash without being told, and then forgetting to put a bag back in.
19:58.13dcawhat is the command in iax.conf to tell it not to cache dns?
19:59.05dudeshaha
20:00.07Juggiewhy dont you move the hell out
20:00.09Juggiejesus
20:00.20cluecon[file]I plan on
20:00.35Juggiei know rent aint that expensive in newbrunswick :)
20:00.38cluecon[file]I was expecting her to flip out soon though
20:00.45cluecon[file]she's been too... nice
20:00.56Juggiebest thing i ever did was move out of the maratimes
20:01.01Juggieso damn boaring
20:01.03Qwellcluecon[file]: If she ever "kicks you out", leave...its worth it
20:01.08Qwellshe'll never forgive herself :D
20:01.19Juggieseperated from the rest of canada by QUEEEEbec
20:01.30cluecon[file]well she's already highly insulted me but saying I'm exactly like my brother
20:01.38Qwellcluecon[file]: ha, I've heard that one a few times
20:01.44Qwellmassive insult
20:01.57tzangerthat'd be a compliment for me
20:02.00tzangermy bro's the smart one
20:02.40Juggiefood is easy sheesh
20:02.42Juggiehow old are u file?
20:02.49tzangeryeah you're in the maritmes
20:02.55cluecon[file]18
20:02.57tzangerreach into the ocean and grab yourself a lobster for chrissakes
20:03.10Juggiei was living in ottawa when i was 18
20:03.14Juggie3000km from home
20:03.17Juggiesuck it up :)
20:03.20*** part/#asterisk alerios (~Santiago@200.24.109.199)
20:03.27cluecon[file]I'm not of legal age yet
20:03.33cluecon[file]makes things difficult
20:03.44*** join/#asterisk santiago (~santiago@200.24.109.199)
20:03.49Qwell18 != 18?
20:03.54dudesCluecon - enjoy it while you can
20:03.55Juggiesays who
20:03.57Qwellor, 18 < 18?
20:04.02cluecon[file]Juggie: government
20:04.07*** join/#asterisk shido6 (~shido@d57-87-253.home.cgocable.net)
20:04.12Juggieisnt it like 16
20:04.14cluecon[file]nope
20:04.15cluecon[file]19 here
20:04.18shido6my windows box just took a shit
20:04.24Qwellshido6: not uncommon
20:04.25shido6anyone have a spare winlogin.exe ?
20:04.27Qwellcluecon[file]: uhh
20:04.34dudesthat's suprising that a windows box took a crap ....
20:04.34inspiredcluecon[file]: where do you live?
20:04.35Juggieyour still a kid anyways :P
20:04.38shido6for xp sp2
20:04.40cluecon[file]New Brunswick
20:04.40Qwell19?  You sniffing glue?
20:04.55Ariel_what do you mean a spare winlogin.exe
20:04.59cluecon[file]age of majority my dear people
20:05.04shido6I need to replace mine
20:05.13shido6for xp sp2
20:05.13*** join/#asterisk Sedorox (brandon@sedorox.staff.smartserv)
20:05.16cluecon[file]http://canada.justice.gc.ca/en/ps/sup/steps/s2c.html
20:05.19Ariel_xp w2k home/pro?
20:05.21Qwellcluecon[file]: What kind of sick ass backwards country is that?
20:05.22shido6pro
20:05.28Ariel_xp pro
20:05.36cluecon[file]take a look and you shall see, what countries are 18 or 19
20:05.39cluecon[file]er provinces
20:05.50shido6or...
20:06.25Qwellcluecon[file]: stupid USD > CAD conversions...makes everything higher
20:06.33shido6why cant I write to a samba drive if I'm logged into it as the user who has write permissions :)
20:06.44shido6mountes as rather
20:06.46tzangershido6: you using user-level auth?
20:06.47shido6mounted as
20:07.25*** join/#asterisk alerios (~Santiago@200.24.109.199)
20:07.33Juggiefile, thats the age to drink
20:07.41*** join/#asterisk RoyK (~roy@host-n39-137.homerun.telia.com)
20:07.41Juggieit has nothing to do with being considered an adult
20:08.03dudesWe have two dell servers both with 2 t410p digium cards firmwire:09.  We have one card set to 0 and the other 1.  We can't seem to get a interrupt on the second card.  Would a firmwire upgrade fix this issue?  Or might there be another option?
20:08.06*** join/#asterisk imagmo (~imagmo@c-24-20-249-117.hsd1.or.comcast.net)
20:08.21cluecon[file]Juggie: uh, I can't even get a credit card without my mother
20:08.23Sedoroxswitch the cards?
20:08.26Sedoroxsometimes that works...
20:08.31cluecon[file]cause she has to cosign
20:08.32dudeshmm.
20:08.33*** join/#asterisk moonaddict (~moonaddic@213.129.253.62)
20:08.37Qwellcluecon[file]: simple solution, move to Ontario :p
20:08.42cluecon[file]haha
20:08.54cluecon[file]I'm half tempted to run upstairs, but I don't feel like getting yelled at for a third time
20:08.55Sedoroxor in the bios.. switch the OS PNP option to whatever the opp. is.. like is its yes.. change no.. and if no.. change to yes
20:09.03cluecon[file]makes me depressed
20:09.12Qwellcluecon[file]: shrug it off, heh
20:09.21QwellI learned to just ignore the psycho-bitch
20:09.32cluecon[file]Qwell: I have learned the same
20:10.21Juggieyou'll eventually marry a psycho-bitch
20:10.23Juggieyou cannot avoid them
20:10.25Juggielearn to deal
20:10.25Qwellwalking upstairs would do nothing, except be background noise
20:11.04dudesI find that women like money ... give them some and they f'n shut up
20:11.28shmaltzshido6 you around?
20:11.32hypa7iammm, misogyny
20:11.35shmaltz~seen shido6
20:11.37jbotshido6 is currently on #asterisk (7m 30s).  Has said a total of 10 messages.  Is idling for 4m 50s
20:11.42tzangerhypa7ia: :-)
20:11.47dudesSedorox - that's for the advice
20:11.51shmaltzanybody from nufone around?
20:11.57tzangerI have a daughter I'm not capable of mysoginy
20:11.59dudessedorox - thanks*
20:11.59Sedoroxdid it work?
20:12.02hypa7iadudes, how about you do the other half of humanity a favor and remove yourself from the gene pool :-)
20:12.25[TK]D-FenderSee : Darwin Awards
20:12.43dudeshypa7ia - I'm not a asshole, I'm just saying the truth.  So ...
20:12.49tzangerheh
20:13.30dudesSedorox - switching the cards didn't make a difference.  they're checking for the PNP in the bios
20:13.45hypa7iadudes, forget what i said about removing yourself from the gene pool.  with an attitude like that you already have :-)
20:15.00tzangerhahaha
20:15.03tzangerhow's the arm RoyK
20:15.20RoyKgetting better
20:15.29dudeshypa7ia - I'll tell you.  I'm 20, and a few of my friends have one, two, three kids ... I don't, perhaps for a good reason.  But, I don't have 300-400/mth child support, I don't have a *women* telling me how much of a loser I am (like I need a reminder ... =p)
20:16.39hypa7iadudes, as far as i know i'm one woman not *women*
20:16.41aleriosHi to all. I have a problem with an asterisk, many calls just hungup after some seconds of having started. I enabled debuging and got many messages like this: "Auto destroying call ..." and "Scheduling destruction of call ...",  anyone can help me?
20:16.52tzangerahh I see you're all better
20:17.02hypa7iaand my point is: keep your misogyny out of the channel, please.
20:17.25tzafrir_laptopalerios, what happens just before that?
20:17.37dudeshypa7ia - didn't mean to offend you.  Though, I will say that I've always found computer chicks to be a lot less like the lot.
20:17.37tzangerdudes: I'm 29 and have two kids.  Divorced but managed to negotiate an amicable divorce/custody/support system that is fair.  Personally (psychos aside) people just need better social skills
20:17.54dudestzanger - I agree
20:17.56tzanger~lart royk
20:18.00tzangerwhat the hell
20:18.08tzanger~lart royk
20:18.17tzangerit has to be random seeded on the nick
20:18.18tzangerI mean hell
20:18.29tzangerhe's always downright ABUSIVE to me, and he does these little pansy-ass things to you
20:18.42tzangerdudes: so, if you agree, how the hell is what you said anywhere near reality
20:18.52tzafrir_laptoptzanger, RoyK , flods should go to the pastebin
20:18.54hypa7iadudes, well, you succeeded in doing so.  and i think that it has more to do with most people sucking in general, not women :-)
20:19.19aleriostzafrir_laptop, hi. nothing strange ¿you wanna see a part of the debug file?
20:19.36tzangertzafrir_laptop: I wasn't flooding
20:19.46RoyKops
20:20.00tzafrir_laptophow obvious and expected
20:20.09tzangerand how do you get him to do that so quickly, I get the "someone just said that x seconds ago" if I repeat too fast
20:20.26dudestzanger - people should be able to get along and such ... but reality is, most people don't rate *good social skills* as a priority in their lifes.
20:20.35cluecon[file]Qwell: she didn't even speak to me lol
20:20.40vandienre
20:21.08tzangerdudes: again, what's that got to do with it being all the fault of "psycho-bitches" ?
20:21.17tzafrir_laptopalerios, not sure I can help, but it won't hurt
20:21.21tzangercluecon[file]: personally I preferred the silent treatment sometimes
20:21.23tzangerbut my kids hate it
20:21.37tzangerthey're pretty good though, I rarely have to raise my voice
20:21.45dudestzanger - I said nothing about "psych-bitches."
20:22.03cluecon[file]it's nicer for me, I don't have to hear her picking on people at her work, saying gossip, etc
20:22.08*** join/#asterisk herag (~herag@adsl-69-234-154-117.dsl.irvnca.pacbell.net)
20:22.11tzangerdudes: apologies, that was qwell
20:22.16aleriostzafrir_laptop, where do you want it pasted?
20:22.17tzangeryou were talking about giving them money to shut them up :-)
20:22.29tzafrir_laptopalerios, pastebin, or PM
20:22.31Qwelltzanger: I was talking about mothers, not wives. :p
20:22.39tzangerQwell: heh
20:22.49tzangermy mom and are are very strong willed... led to some interesting times
20:23.03*** join/#asterisk clive- (~pirch@rrba-146-66-199.telkomadsl.co.za)
20:23.05tzangerwe both love each other to never live together :-)
20:23.14tzangerI'd love to build her a granny flat next to my place though
20:23.16dudestzanger - yes, I said money can shut a women up (some not all.)  but a better part than not.
20:23.18tzangerjust not under the same roof
20:23.30tzangerdudes: bribes never shut anyone up for long
20:23.30aleriostzafrir_laptop, pastebin.ca?
20:23.42tzafrir_laptop~pastebin
20:23.43jbot[pastebin] a place to paste your stuff without flooding the channel - try http://pastebin.ca
20:24.07dudestzanger - that is true.  When they finish shopping/getting their nails done, bagering you to their friends ... your back at square one.
20:24.09*** join/#asterisk file[mac] (~jcolp@mctn1-3462.nb.aliant.net)
20:25.03heragif the invite line for an incoming call reads as INVITE sip:310*******@164.***.***.*** SIP/2.0  is 310******* the user it's trying to connect to?
20:25.24tzangerherag: I believe so
20:26.00heragso then where do I have to have an entry for this 310*******? does it become an extension? a context? or an entry in my sip.conf?
20:27.21tzanger[310******]
20:27.24tzangertype=user
20:27.26tzangercontext=somecontext
20:27.29tzangeretc., etc.
20:27.51file[mac]good try, wrong though
20:28.00tzangerme?
20:28.03file[mac]yes
20:28.06tzangerhmm
20:28.35file[mac]]the number on the INVITE line is the extension being dialed
20:28.41file[mac]the username and stuff are in the authorization header
20:28.58tzangerahh mine's set up different
20:28.59file[mac]so therefore you have to figure out what context the call is going to, and add the extension there
20:29.01tzangermy apologies, herag
20:29.23heragok, so I think my problem lies in the auth area
20:29.23*** part/#asterisk stilex (~occ@pcp01850015pcs.danbry01.ct.comcast.net)
20:29.34heragcause I keep getting an "auth failed for user..."
20:30.07heragin the invite header, the from entry reads...From: "7145280904" <sip:telasip@4.79.19.58>
20:30.24*** join/#asterisk MikeJ[Laptop] (~ircatjerr@pcp02795302pcs.roylok01.mi.comcast.net)
20:30.27heragI'm guessing the thing in quotes is cid
20:31.01heragand the thing in brackets is the spi uri of where the call is coming from?
20:31.59*** join/#asterisk delphiuk (~richard@host81-155-71-192.range81-155.btcentralplus.com)
20:32.59delphiukhi, if I have a x100 clone card and i do modprobe zaptel, should I get another module loaded other than zaptel?
20:33.34cluecon[file]delphiuk: nope...
20:34.01aleriostzafrir_laptop, http://pastebin.ca/12974
20:34.01cluecon[file]nooooooo
20:35.13delphiukmmm, i am getting a "unable to specify channel1: no such device or address when trying to run?
20:35.39Qwelldrumkilla: y0
20:35.46[TK]D-FenderBroken record time : Is there a way so that *'s call queue doesn't call agents who are on the phone?  I see that it doesn't if they are already on a queue call, but not a regular one.
20:35.54drumkillahey Qwell
20:36.33Qwelldrumkilla: iax2/guest@24.50.66.194/s :)
20:36.57tzangerdelphiuk: yup
20:37.27delphiuktzanger, am I being stupid?
20:37.38Qwellassuming it still works with head
20:37.42tzangeryou bought a clone card and are asking for support, yes.
20:38.01heragwhat is the user is it referring to when it says  "Failed to authenticate user "714*******" <sip:telasip@4.79.19.58>;tag=as03c8ac8e"
20:38.11drumkillaQwell: this is awesome!!!!!!!!
20:38.20delphiuktzanger, I will be purchasing digium cards as soon as they are certified for UK use which will hopefully be very soon now :)
20:38.23Qwelldrumkilla: I have two more.  one sec
20:38.33drumkillaQwell: hold up
20:38.37tzangerthe clone was certified for uk?
20:38.48aleriostzafrir_laptop, well, actually I don't know exactly the right moment where a call gets dropped. It's totally random
20:38.50delphiuktzanger, as far as I am aware yes.
20:38.51bkw_its just a 6 dollar modem
20:38.57tzangerdelphiuk: wow
20:38.57bkw_even the one digium sold was really just a modem
20:39.09tzangerbkw_: I'm well, well aware of that
20:39.15bkw_cluecon[file], yo yo yo
20:39.23tzangerdelphiuk: the zaptel driver is just like a "generic services" driver
20:39.23bkw_tzanger, the modem should be fine for the UK
20:39.33tzangeryou need the specific hardware driver now
20:39.59tzangerbkw_: should be... line impedance should be alright if he bought a UK modem from the UK
20:40.11delphiukyes, was from the uk
20:40.53delphiuktrust me, as soon as the tdm400 is certified for the UK, that will be the only card I use for pstn
20:41.17tzangerdelphiuk: I told you waht you have to do now
20:41.33delphiuktzanger, sorry, did I miss something?
20:41.40*** join/#asterisk Marlow (~marlow@159-134-145-42.as1.mvw.galway.eircom.net)
20:41.57tzangerdelphiuk: the zaptel driver is just like a "generic services" driver
20:41.59tzangeryou need the specific hardware driver now
20:42.14tzangerthink of the zaptel driver as the shared library for all the zaptel hardware
20:42.28delphiukah, so I need to modprobe an additional module?
20:42.56Marlowdelphiuk : jup .. depending on the hardware you want to use ..
20:43.46drumkillaQwell: ok, next!  :)
20:43.50drumkillaI wanted to show mark
20:43.50Qwelldone
20:44.13delphiukas, I see. ok, so if I have plugged into the pstn, I need wcfxo?
20:44.47Qwellthat was the first one I did.  not so great
20:44.51Qwellnext one is a bit better
20:45.07*** join/#asterisk zotz (~zotz@208.196.247.140)
20:45.41*** part/#asterisk roamer323 (~sing@toronto-HSE-ppp4075317.sympatico.ca)
20:45.51drumkillaQwell: ok, next, hehe
20:45.54Qwellagain
20:46.26Qwellthe first one is by far the best though
20:46.53Qwell66.11, its s :p
20:47.00*** join/#asterisk chaoscon (~ph33r@chaoscon.ceo.smartserv)
20:47.05shmaltzanybody from nufone?
20:47.11shmaltz~seen shido6
20:47.14jbotshido6 is currently on #asterisk (43m 7s).  Has said a total of 10 messages.  Is idling for 40m 27s
20:47.58drumkillaQwell: yes, tetris was absoluely awesome.
20:48.06Qwelldrumkilla: yeah..
20:48.18drumkillagimme gimme gimme, hehe
20:48.23Qwellheh
20:48.37aleriostzafrir_laptop, looks like this happens only whene there's heavy load on the server. Do you have an * v1.0.7 server working with many users?
20:48.37drumkillaI could hack it into zonedata.c, haha
20:48.55Qwellheh
20:49.41clive-scmalts look for jerejr
20:49.44clive-jerjer
20:50.08*** join/#asterisk r0d3nt (anonymous@soveliss.luniac.com)
20:51.49heragthe console displays this error for an incoming call, what is the user is it referring to when it says  "Failed to authenticate user "714*******" <sip:telasip@4.79.19.58>;tag=as03c8ac8e"...is it telasip, or the 714******* number?
20:52.20alerioshello everybody. I have a problem with an asterisk, many calls just hungup after some seconds of having started. I enabled debuging and got many messages like this: "Auto destroying call ..." and "Scheduling destruction of call ...",  anyone can help me?
20:52.28Qwelldrumkilla: I am so surprised it turned out so well, really
20:52.48*** join/#asterisk [hC] (~hardcore@c-66-176-181-76.hsd1.fl.comcast.net)
20:52.50bewestalerios, there's probably a message before that that says why
20:52.58bewestalerios, what level of verbosity are you?
20:53.05bewestalerios, at least 3 is useful
20:53.07[hC]What is unique about the cisco PoE implementation, that this PoE switch i have works on all my phones EXCEPT my cisco's?
20:53.20[hC]yet, the cisco will take a standard power injector that the other phones like, too?!?
20:55.20aleriosbewest, I was on level 5
20:55.30[hC]ah. I see. polarity.
20:55.31[hC]doh.
20:56.17*** join/#asterisk roamer323 (~sing@toronto-HSE-ppp4075317.sympatico.ca)
20:56.43aleriosbewest, ¿is level 5 good or bad?
20:57.46shmaltzanybody from nufone?
20:57.52bewestalerios, it's fine
20:58.04bewestthe trick is to watch for the reason it's being hung up
20:58.16bewestit might say something about unable to get a frame
20:58.25bewestor can't translate between codecs
20:58.39bewestor can't bridge or something
21:00.20aleriosbewest, q
21:01.11aleriosbewest, i mean,.. I have a little snap from the debug file in here: http://pastebin.ca/12974
21:01.34aleriosbewest,
21:01.39*** join/#asterisk wrtchd (~dentont@c-24-0-114-212.hsd1.tx.comcast.net)
21:02.00*** join/#asterisk jas_williams (~Jason@host217-43-100-176.range217-43.btcentralplus.com)
21:02.01*** part/#asterisk Samoied (~Samoied@200.157.225.223)
21:02.41bewest#
21:02.41bewestMay 24 17:38:59 NOTICE[358]: Peer 'CORTIZ' is now UNREACHABLE!
21:03.24wrtchdGood morning all.  I have been at this for a few hours.  I have zaptel and ztdummy confgured.  Now I am trying to download Asterisk and i am stumped. cvs checkout -r v1_0_rc_2 asterisk  Gives me no such tag.  I don't understand CVS at all so can someone give me a clue.  Netware is so much eaiser...lol
21:03.47drumkillawhy would you try to checkout that tag
21:04.08wrtchdTo be honest that is what the doc tells me to do.
21:04.18drumkillaum ... ha, ok
21:04.25drumkillajust use -r v1-0
21:04.44wrtchdok.  is there a way to list what the CVS has for me to use?
21:04.53drumkillaprobably, heh
21:04.54wrtchdor do you all just know by the force
21:05.01drumkillaI use the force.
21:05.13drumkillaI AM the force...
21:05.21wrtchdgreat..I am all out of flordians.
21:05.22drumkilla(I make the tags, hehe)
21:05.34hardwireyou deserve it
21:06.16jas_williamswrtchd: just do cvs checkout -r v1-0
21:06.29drumkillawrtchd: or if you want to be paranoid, you can use -r v1-0-7
21:06.31jas_williamswith asterisk on the end of cource
21:06.33wrtchdit is sucking it down now. thanks folks..
21:07.26drumkillai'm kind of worried about this 'doc' that you are using, haha
21:07.38wrtchdhttp://www.asteriskdocs.org/modules/tinycontent/content/docbook/current_v1/docs-html/x460.html
21:07.41*** part/#asterisk moua (david@men75-2-82-66-50-159.fbx.proxad.net)
21:08.30*** join/#asterisk TheSin (~TheSin@iphost-64-56-130-194.edm.wiband.net)
21:08.56TheSinlo all
21:09.25TheSinjust wondering if anyone has mib files for a digium te11xp for linux net-snmp 5 ?
21:09.33wrtchdit has been dead on so far..This is not for production just trying to learn for now so I can install with authority in a couple of weeks when I am in Bangalore..I am a Netware/Windows convert that is going to make an all linux call center in india.  They have no linux there, thus I get to keep my job..
21:10.33wrtchd* no offense to any Indian participants in the room.
21:10.55[hC]okay, so who can tell me the pinout for a cisco poe patch cable, since mine doesnt seem to like this.
21:11.06[hC]I tried the voip-info reccomendation of the polarity switch, but that didnt seem to help
21:11.25blitzrageanyone know zoa's bandwidth calculator link?
21:11.39dudeswrtchd - an all linux dialer huh ...
21:11.55drumkillablitzrage: you got an asteriskdocs.org compliment!
21:12.36TheSindigium is closed today :\
21:12.57cluecon[file]it's a holiday.
21:13.06blitzragedrumkilla: oh yah?
21:13.14TheSinOr right
21:13.18TheSinwe had it last weekend
21:13.22drumkillablitzrage: yeah, wrtchd said he was using a guide from there and that it has been dead on so far
21:13.24wrtchdyeap/ GNUDIALER/Asterisk---FC3 for file, print, DHCP, DNS, and Proxy.  Thinstation on the desktop booting pexe..
21:13.36blitzragedrumkilla: nice! the book will be way better
21:13.48wrtchdI think blitzrage gave me the link last night.
21:13.58drumkillawrtchd: ha, nice.
21:14.00wrtchdmikej did, I forget..
21:14.04drumkillablitzrage: i will be looking forward to it
21:14.06TheSinI bet mib files are on that asterisk business card they send me that I can't use since I have a slot loading drive :D
21:14.11blitzragedrumkilla: me too - need to get the damn thing done
21:14.12dudeswrtchd - gnudialer?  Does it suck?
21:14.21drumkillaTheSin: I don't think we have anything like what you are looking for
21:14.25wrtchdI don't know.
21:14.43wrtchdI am getting to get * running then on to GNUDIALER..
21:14.47blitzragedrumkilla: you don't know zoa's BW calc site do you?  No link on astertest.com - but its there somewhere
21:14.55drumkillablitzrage: nope ... google maybe?
21:14.58TheSindrumkilla, really?  darn, I had every port maped on my Cisco Router, oh well, no monitoring :D
21:15.10drumkillasorry TheSin
21:15.11blitzragedrumkilla: pfffffft... what the hell does google know?
21:15.11wrtchdI asked if anyone could recomend a predictive dialer last night but no one could ante up a suggestion..or clue.
21:15.24drumkillaTheSin: Digium does custom development, though!  hehe
21:15.26TheSindrumkilla, np, I like the TE11xp better anyhow
21:15.32blitzrage~google site:astertest.com bandwidth calculator
21:15.44drumkilladenied!
21:15.47TheSinthanks anyhow drumkilla
21:15.50blitzrageouch
21:15.57*** part/#asterisk TheSin (~TheSin@iphost-64-56-130-194.edm.wiband.net)
21:15.58blitzragehrmm
21:16.21dudeswrtchd - predictive dialing in linux using asterisk ...
21:16.23cluecon[file]blitzrage: http://www.asteriskguru.com/bandwidth_calculator.php
21:16.25*** join/#asterisk jsolares (~jsolares@200.30.141.86)
21:16.25cluecon[file];)
21:16.27blitzragedoh!
21:16.29wrtchdyes dudes
21:16.33blitzragecluecon[file]: same guy, different site :)
21:16.41dudeswrtchd - I know about dialing ...
21:16.55cluecon[file]drumkilla: nooo run around in circles
21:17.09wrtchdreally.. I am desktop guru..but have been tasked with a call center in a box..
21:17.13cluecon[file]drumkilla: good enough
21:17.20dudeswrtchd - you'll need to learn mysql/asterisk to use gnudialer
21:17.34wrtchdmysql I have.
21:17.48dudeswrtchd - what kind of CRM are you using?  Or are you using the included CRM package?
21:17.51wrtchdI am on the asterisk curve at the moment.
21:18.08dudeswrtchd - what version are you going to use of GnuDialer?
21:18.14wrtchdI have a pre-built CRM.  it is the AEGIS APP..I work for Aegis Com group..
21:18.28wrtchdI am not sure what version of GNU yet..
21:18.55wrtchdI am not married to GnuDialer..I would use another if it is suggested.
21:19.35dudeswrtchd - I won't recommend another one ... personally anyway.  But I will say I've used Gnudialer to dial before and it worked great.
21:19.47wrtchdah great..
21:19.56wrtchdthen I am on the right track..that helps greatly..
21:21.33wrtchdI have to make stuff work in India with no money, so I am building it to secure my job.  I am really excited about ThinStation..it is Gods gift to Desktop guys like me.  I am hoping Asterisk is the same thing.  I am tired of Avaya, and Nortel that I have to work arround now..
21:21.38dudesI can't think of a better alternative.  Though, when we will be releasing version 2 shortly
21:22.15dudesAvaya ... ecks... Same with dialogic and SER ....
21:22.16delphiukdudes, is version 2 that close then?
21:22.23wrtchdUnfortunatly I have a 4 week timeline.  as is everything in the call center it has to be done yesterday.
21:22.29dudesdelphiuk - we're about done with it
21:22.36delphiukcool
21:22.55wrtchddo you work for/with GnuDialer?
21:23.01dudesdelphiuk - we're just debugging currently, so anyone willing to do testing let me know and I"ll get you the source
21:23.18dudeswrtchd - A friend and I made gnudialer
21:23.18wrtchdhow ever the opensource terminology works..
21:23.23wrtchdwell.
21:23.29wrtchdCool.  i will tell you how it works.
21:23.44wrtchdit will be dialing 100,000 leads in its first week for me.
21:23.51wrtchd20 agents..
21:24.00dudeswe'll have gnudailer.org up and running tomorrow at the lastest
21:24.05wrtchdif all goes will by July I will have a 1000 seater running it.
21:24.42dudeswrtchd - if you do that make sure and send me a e-mail so I can post it on our site.
21:25.12dudesWe'll be doing some dialing later with week on some mortgage leads.
21:25.19wrtchdI am part of a team that will be supporting the Olympics in china..we will be using what ever solution I come up with.
21:25.48dudesheath used to work for aegis
21:25.51delphiukdudes, your all doing a stunning job on Asterisk
21:25.57wrtchdreally.
21:26.06dudesdelphiuk - thanks =)
21:26.07wrtchdwell the olympics is not aegis..it is a side venture..
21:26.17wrtchdheath? which aegis branch?
21:26.32wrtchdhow long ago?
21:26.34dudeswrtchd - He did providen (spelling) years ago
21:26.43dudesback in 2000 or so
21:26.45wrtchdwow that was a while ago..
21:27.14wrtchdit has changed since then..we are owned by Essar and indian company that is choking us to death..
21:27.15dudesI'm pretty much still hungover so excuse my typeing.  Plus I worked a lot weekend.
21:27.35wrtchdthat is cool..where are you located?
21:28.01dudesheath and I live in Wheaton MN.  He worked in Browns Valley MN
21:28.10wrtchddrop me a line with the address you want me to send my results to.  tommydenton@gmail.com
21:28.31dudesjames@gnudailer.org / heath@gnudialer.org and richard@gnudialer.org
21:28.35wrtchdbrowns valley..wow..I remember hacking that novell server out of the tree when I started in 2002..
21:28.41delphiukstop now
21:28.50delphiukwhoops, wrong window ;)
21:28.53dudeshaha
21:29.11*** join/#asterisk Ariel_ (~Ariel@dsl-20-177.cofs.net)
21:29.24dudeswrtchd - they used a Unix SER Dialer with terminals if I recall correctly
21:29.44Ariel_there seems to be some major internet problem down in my area.
21:29.53wrtchdthere they might have. we have DataGeneral with a home grown dialer now..
21:30.20wrtchdRemember Data General and informix went out of style about 10 years ago..
21:30.48dudeswrtchd - they've upgraded to a newer SER dialer with windows 98 stations
21:31.07wrtchdwhere?  we don't have a browns valley anymore..
21:31.19dudeswrtchd - they told me it was "impossible" to get on the internet at work ... took me 10 minutes to get it working.
21:31.42aleriosasterisk bloody sucks
21:31.46wrtchdoh..yeah..well we have since locked down so that I can't get on.
21:32.01Ariel_alerios, no it does not.
21:32.02*** join/#asterisk jalsot (~tamas@abacus.eworldcom.hu)
21:32.13Ariel_asterisk is great does allot. You just need to set it up correctly.
21:32.32dudeswrtchd - it's owned by a lady here.  Wendy Henning.
21:32.35aleriosAriel_ I have a problem with an asterisk, many calls just hungup after some seconds of having started. I enabled debuging and got many messages like this: "Auto destroying call ..." and "Scheduling destruction of call ...", maybe you can help me
21:32.53bewestlol
21:33.03bewestalerios, I already pointed out the exact spot it was doing it
21:33.05wrtchdah..Aegis Communications must have sold it off sometime ago..lol..
21:33.10bewestpeer is unreachable
21:33.20bewestI noticed you are both using private IP's
21:33.25bewestare you on the same lan?
21:33.34dudeswrtchd - heath and I have been keeping a VP at synergy informed on GnuDialer.  Hopefully once we release 2.0 we'll be able to convert them to open source.
21:33.59wrtchdcool..I am hoping to do the same at Aegis.
21:34.29dudeswrtchd - they pay 10,000 a month for the dialer they have, plus they have to pay 2,000 per year per agent (two licenses are required for inbound)
21:34.37dudeswrtchd - it's insane what they charge.
21:34.55aleriosbewest, that's not the problem. There are many extensions and channels being used, even with PSTN
21:35.30aleriosand nat, and ata's, etc
21:35.36wrtchdthat is why we paid a fellow to write us one..it does anything you want and gives you change. but we are married to him no matter as the source code is in his head..
21:35.39Ariel_alerios, there are many reasons that happens.  more information is needed.
21:36.03wrtchdwe don't have to pay licenses, but we have to pay avaya and nortel out the wasue..
21:36.17aleriosAriel_ I have a little snap from the debug file in here: http://pastebin.ca/12974
21:36.25wrtchdRegardless.  I have got to get.  My wife is calling. Thanks for the info and I will see you all around!
21:36.41dudeslater man
21:37.10dudesI msged you the infor for our test server if you want to checkout the new admin stuff.  Note that it's a actual working setup so be nice =)
21:38.03aleriosIt seems (not sure) that with v1.0.3 the problem never occured
21:38.33Ariel_alerios, first let me ask you the following. Your problems look like nat reinvite problems. do you have in the sip.conf for these users canreinvite=no?
21:40.08aleriosAriel_, yes, I do
21:43.33*** part/#asterisk wrtchd (~dentont@c-24-0-114-212.hsd1.tx.comcast.net)
21:44.39aleriosAriel_, Could there be any other reason for this problem? I'm gonna get wacko with this
21:45.43aleriosth dark side is calling,.. maybe I  should buy a cisco pbx or run asterisk on windows  :S
21:45.57*** join/#asterisk emakris2 (~emakris2@c-24-131-136-49.hsd1.ma.comcast.net)
21:46.50seanI've done it.
21:46.55seansort of (-:
21:47.17seanI run coLinux on my windows workstation.. so it's asterisk on linux on windows
21:47.41seanand it's not good for anything but building/testing -- can't handle calls.
21:47.56*** join/#asterisk speakman (daniel@81-234-74-26-o945.telia.com)
21:47.59speakmanhi people!
21:48.12speakmaneveryone idleing today? :D
21:49.04*** join/#asterisk CoaxD (coax@shell1.cornernet.com)
21:50.02Ariel_alerios, how many codec's do you have setup? I would recommend you getting the system back to basic.  From your small sample of debug it seems like your having nat and different reinvite problems.
21:50.25[hC]Anyone here know how to pin out a PoE cable for a cisco phone from a switch that does 802.3af?
21:50.52*** join/#asterisk bjohnson (~bjohnson@66.11.188.215)
21:50.54Drukenspeakman: today?? we idle all day everyday
21:52.15aleriosAriel_, gsm, ilbc, alaw, ulaw with disallow=all
21:52.47Ariel_ok alerios lets set them for a test to just have ulaw.
21:54.17Drukenalerios: do you have a allow in there?
21:54.28aleriosAriel_, can I test with gsm or ilbc?
21:54.59aleriosDruken, of course
21:55.01Ariel_alerios, both of those will require your system to do some transcoding. Which is what I think is the problem.
21:56.03shmaltzanyone from nufone around?
21:56.28*** part/#asterisk makhtar (~ageller@mail.bulletinnews.com)
21:57.19aleriosAriel_, ok,.. I'll test it over the week, with normal load,.. right now the server is not being used, since is a holiday on my country
21:57.37jas_williamsalerios: also turn of the qualify= and see if tha makes any difference
21:59.24santiago:O
21:59.27Drukenyou think you need to pay 2 million for my wife? ok :)
21:59.30speakmanDruken: ;)
21:59.43speakmanDo anyone here use ZapRAS or PPPD commands?
21:59.51jas_williamsnah you need to pay me 2 million to take your wife for 24hrs :)
21:59.51santiagoDruken, do you have any picture of she?
21:59.52speakmanOr even uses data calls through Asterisk?
22:00.13speakmanOr really short: How do I make Asterisk initiate DATA calls, instead of VOICE?
22:00.14Marlowspeakman : data calls through asterisk are no problem ..
22:00.21Marlowspeakman : as long as they are zap to zap
22:00.30speakmanzap to zap? I'm dialing my ISP :)
22:00.32Drukenjas_williams: nah... for 2 mill, i can handle her :)
22:00.40Marlowspeakman : jup .. that should just work ..
22:00.46speakmanTheir RAS server changes behaivor pending on my call..
22:00.59Marlowspeakman : i have a isdn router on the inside interface, that just dials through my asterisk
22:01.02santiagoDruken, do you have any picture of her?
22:01.03*** join/#asterisk wasim (~wasim@203.81.201.188)
22:01.14speakmanIf it's voice call, it will try V.90, and if it's DATA call it will accept PPP communication
22:01.24Drukenof course i do.. but i'm not given no one pictures :)
22:01.32Drukennot without seeing the cash first
22:01.35Drukenhehe
22:01.43speakmanMarlow: What ISDN protocol are you using?
22:01.50Marlowspeakman : euroisdn ..
22:01.52santiagommmm :S
22:01.59Marlowspeakman : dss1
22:02.06speakmanMarlow: great, same here ;) hdb3 though.. ;)
22:02.22Marlowspeakman : shouldn't make a difference ..
22:02.33speakmantrue, true..
22:02.41*** join/#asterisk Rez (lorez@lorez.staff.freenode)
22:02.51Marlowspeakman : anyhow .. there is also a "D" tag for the dial command to tag a call data .. i'm not sure if it's "d" or "D" though ..
22:02.51speakmanMarlow: Do you think it's possible to make a .call-file initiate a DATA call?
22:02.57Marlowspeakman : it's specified somewhere ..
22:03.06aleriosjas_williams, I already have the qualify option set to yes, but without a time value, just qualify=yes
22:03.07speakmanHm! YOu're right!
22:03.19speakmanBut how do I make pppd take over when using the DIal command?
22:03.35speakmanSince a .call-file will use Dial command itself...
22:03.37Marlowspeakman : never tried .. i just hooked the isdn router up to a hfc card in NT mode and let it dial through the box .. without configuring anything ..
22:03.44speakmanhehe ok
22:03.50Marlowspeakman : zap to capi works not, though
22:03.52speakmanBut U do have a point in Dial command..
22:04.07jas_williamsalerios: try setting qualify=no as a test see if it is the qualify packets causing a problem
22:04.12Marlowspeakman : there is a flag there, but i don't use it
22:04.12speakmanI'm having a E1 PRI which I like to connect to my ISP through..
22:04.15aleriosjas_williams, oh, you said turn it off,...  jejeje,  I misunderstood,....  ok I'll test it
22:05.31*** join/#asterisk JunK-Y (~junky@modemcable174.107-81-70.mc.videotron.ca)
22:09.49Drukencan someone verify what the data flag is
22:09.50Druken?
22:09.53Drukend or D ?
22:10.01Qwellshow application dial
22:12.24MarlowDruken : it's "d"
22:12.48speakmanBut how to make pppd handle the data?!
22:13.05Qwell<PROTECTED>
22:14.01MarlowQwell : in the dial command ?
22:14.12QwellThats what show application dial says
22:14.22Marlowhmm
22:14.30Drukeni belive qwell is right
22:14.43Drukeni belive d is the dial extension after answered thing
22:15.23Marlowahh .. found it ..
22:15.29Marlowexten => _X.,1,Dial(Zap/g1d/${EXTEN})
22:15.41Marlowthe "d" goes on the back of the group ..
22:15.48Marlowfor isdn calls
22:16.37*** join/#asterisk kankan (kankan@frenchcancan.net)
22:16.48speakmanlol! ok.. then it's easy to make it in .call-files!
22:16.49kankanhi there
22:17.04speakmanMarlow: big thanks, m8! :D
22:17.17kankanwhat is the purpose of this channel?
22:18.07Marlowit forces the channel to be bridged digitally
22:18.13Marlowend-to-end digital call
22:19.01speakmananother thing; why can't I rmmod wcxxxx (the driver for TE110P) ?
22:19.07speakmanIt hangs my server BIG TIME! :)
22:19.34altkankan: Asterisk PBX - www.asterisk.org
22:19.43kankanalt: thank you
22:19.52heragthe console displays this error for an incoming call, what is the user is it referring to when it says  "Failed to authenticate user "714*******" <sip:telasip@4.79.19.58>;tag=as03c8ac8e"...is it telasip, or the 714******* number?
22:23.05Marlowherag : i would say that the authuser provided was the 714 one ..
22:23.53Marlowherag : you can have a user and a authuser in sip
22:24.13heragMarlow: I see, so teleasip is sending me the 714 number as the authuser?
22:24.36heragMarlow: that 714 number is really just callerid, if the cid is blocked, it comes in as Anonymous
22:24.45heragregardless, that's the authuser?
22:25.15Marlowherag : i would say so ..
22:25.31*** join/#asterisk matobago (~matobago@einstein.transtelco.com.mx)
22:25.49heragMarlow: ok, that sounds reasonable, so how do I tell asterisk to accept any authuser that telasip might send me (after all, I want to receive calls from everyone)?
22:25.52Marlowherag : try to turn debug for sip on ..
22:26.07heragMarlow, I have   http://pastebin.ca/12832
22:26.13matobagoany CISCO users out there knows if it is possible to
22:26.13matobagoChange the locations of the BUTTONS along the bottom of the screen.
22:26.25matobagoin a cisco 7940
22:26.29[hC]Anyone here know how to pin out a PoE cable for a cisco phone from a switch that does 802.3af?
22:26.55jas_williamsmatobago: No it is not possible using SIP
22:27.18matobagothanks jas_williams
22:27.47matobagobut disable one?
22:28.06Marlowherag : usually any not authenficated user should go the default context, if that exists
22:28.18Marlowherag : specified in your sip.conf ..
22:29.25jas_williamsherag: or you could add insecure=yes to the telasip peer
22:29.27*** join/#asterisk jhava (~icechat5@200.58.26.21)
22:29.50heragMarlow: jas_williams I have tried insecure = yes and insecure = very, no success
22:30.04heragMarlow: and I do have a default context defined
22:30.08*** join/#asterisk shmaltz (~chatzilla@69.28.255.210)
22:30.12heraglet me get the link for my sip.conf
22:30.25jas_williamsherag: ok post away
22:30.26jhavaHello all, anybody has experienced Segmentation Fault when recording an incoming message on voicemail ?
22:30.46heraghttp://pastebin.ca/12877
22:30.48Qwelljhava: yes, but only when I was screwing around debugging
22:30.52mishehuI've experienced a segmentation fault while eating taco bell, which later became a very fast core dump.
22:30.52matobagojas: si ti possible disable one button or many?
22:32.04Marlowherag : have you tried taking that "telasip" section out there ..
22:32.18Marlowherag : it should not matter at all, since you are sending it to the same context
22:32.20jas_williamsherag: you need a type=user entry for telasip
22:32.55jas_williamsas well or change to type=friend
22:33.14Marlowjas_williams : friend would just mean that it acts as peer and user at the same time
22:33.14jas_williamstype=peer is for outbound only
22:33.22Marlowjas_williams : would not change anything ..
22:33.31heragjas_williams: I have tried creating a new entry of type = user
22:33.34Marlowjas_williams : and he wants to send users in regardless of username
22:33.41heragand I have also tried type = friend
22:33.50*** join/#asterisk MikeJ_ (~ircatjerr@pcp02795302pcs.roylok01.mi.comcast.net)
22:34.00heragMarlow: let me try commenting out the telasip entry altogether, leaving only the register
22:34.34Marlowherag : exactly .. when having trouble try to minimize your config to the bare limit and see, what happens ..
22:34.36jas_williamsherag: also it might be worth specifying an extension on  the end of the register command
22:34.39Marlowherag : then add on again
22:34.43herag!!
22:34.46heragMarlow: that did it!
22:34.58heragit accepted the call!
22:34.59heragwheeeeeeee
22:35.16Marlowherag : it tries to authentificate against the specified telasip user ..
22:35.22Marlowherag : that's what went wrong ..
22:35.25heragMarlow: I see
22:35.36*** join/#asterisk rene- (~root@201.144.11.57)
22:35.39heragMarlow: so how do I get it to not check against the peer entry?
22:35.47heragMarlow: cause without the peer definition, I can't dialout
22:36.18jas_williamsherag: set the type=friend and insecure=yes
22:36.34Marlowherag : peer is ok ...
22:36.48Marlowherag : but take the "fromuser" out .. that is for inbound ..
22:36.55Marlowherag : specify the "authuser" instead
22:37.51heragMarlow: what do I set as the authuser, the same as my username value?
22:38.06Marlowherag : sure .. if that is your user to authentificate with .
22:38.18heragah ok....yes, it is the same
22:38.34*** join/#asterisk rue_mohr (~dw1@h24-207-80-55.cst.dccnet.com)
22:38.37heragMarlow: you are a god...
22:38.39Marlowherag : and you don't need to set the username
22:38.40heragMarlow: that worked
22:38.50heragMarlow: no username for the peer entry?
22:38.51Marlowherag : the section name = username
22:38.56Marlowherag : allways
22:39.10Marlowherag : you only need to set it, if you want it to be different
22:39.15*** part/#asterisk rue_mohr (~dw1@h24-207-80-55.cst.dccnet.com)
22:39.25heragMarlow: ah, I see, otherwise it will just inherit the authuser entry?
22:39.44Marlowherag : no ... it will take it from the [telasip]
22:39.52dcawhat is the command in iax.conf to tell it not to cache dns?
22:39.55Marlowherag : the authuser would also inherit it from there normally
22:40.08heragMarlow: "there" is what? the register line?
22:40.14Marlowdca : it does not cache dns as such ..
22:40.23dcasure it does
22:40.41Marlowdca : eh .. it only does the lookup once when the module is loaded :)
22:40.54Marlowdca : it doesn't do it on every call
22:41.02dcaright, but i believe there is a something= so that it won't cache at all
22:41.51Marlowherag : the register line has nothing to do with your [telasip] section
22:42.02Marlowherag : it's something entirely different ..
22:42.27Marlowherag : register is for inbound, your [telasip] peer section is for outbound
22:42.49Marlowherag : or better .. register is to tell others that you want inbound traffic :)
22:43.08heragMarlow: right, that's sorta how I interpreted it, so when you said that it'll be pulling the username from somewhere, where was that "somewhere"?
22:43.18Marlowherag : like gw3.telasip.com in this case
22:43.41Marlowherag : you start your peer section with [telasip]
22:43.59Marlowherag : that is interpreted as you username, if you don't override it with authuser/username
22:44.35Marlowherag : so if "telasip" not is your username, you should specify them both
22:44.35heragoh, I see
22:45.17tzangerI always hated that register was its own entity
22:45.31tzangerand that you couldn't say register => [someuser]
22:45.46Marlowtzanger : that would make more sense though ..
22:45.52Marlowtzanger : i agree on that ..
22:45.55heragMarlow: where did you read about all these little details? where do I find documentation at this level?
22:46.14Marlowherag : ehh .. in the sample sip.conf ? :)
22:46.22Marlowherag : matter of interpretation
22:46.52Drukenhas anyone ever actually gotten the privacy manager to work?
22:46.58Marlowherag : or when you after having been poking around with asterisk finally see the light ..
22:46.58*** join/#asterisk visik7 (~ciao@visik7.user)
22:47.24Marlowherag : ;o)
22:47.43heragMarlow: ok...I think I sorta have a place to work with...I've been banging my head against the wall for almost a week trying to solve this incoming call problem
22:48.00Marlowherag : some people needed years :o)
22:48.09heragMarlow: though I screwed it up again, I think I can figure it out...thanks a ton for your help
22:48.27Marlowherag : just go back to the simplest approach again, if it fails ..
22:48.33Marlowherag : and build up from there ..
22:48.37Marlowherag : never fails :)
22:48.52heragMarlow: ya, I'm gonna go back to just the register line
22:49.14heragthanks again
22:51.04aleriosgood bye and thanks for the help
22:52.10Marlowthey are a hassle to open
22:56.16blitzrageanyone want to host a website for me? :)
22:56.31Marlowblitzrage : for what purpose ?
22:56.54*** join/#asterisk zonezero (~zonezero@59.b167.bendtel.net)
22:56.58blitzrageMarlow: I need to run www.dotproject.net for myself - would only be me and maybe a couple of friends accessing it
22:58.02*** part/#asterisk roche (~roche@216.194.173.2)
22:58.57zonezeroanyone use asterisk and voicepulse connect recently?
22:59.31Ariel_zonezero, yes I am using it.
23:00.19zonezeroI just setup an account but didn't get the asterisk config in email...noticed they send that on the wiki entry.
23:00.47*** join/#asterisk davep_infoseeker (~davep@S010600a0cce01511.cg.shawcable.net)
23:01.04zonezerothe faq link they sent with my number also doesn't work...is it something simple?
23:01.51davep_infoseekerhi all just installed the zaptel drivers and asterisk from cvs . . cant get the drivers to load right
23:02.14davep_infoseekerusing the tdm400 card and the wctdm driver
23:03.00davep_infoseekerwhen I modprobe wctdm it just sits there doing nothing so I bg it and take a look at the lsmod
23:03.22Mavviecheck /var/log/messages
23:03.23davep_infoseekerlsmod shows that zaptel, wctdm and crc_ccitt are loaded
23:04.40davep_infoseekermessages shows
23:04.41davep_infoseekerMay 30 16:53:31 localhost kernel:  <6>Zapata Telephony Interface Registered on major 196
23:05.04davep_infoseekerrunning asterisk dies though
23:05.23Mavviecheck your asterisk logs
23:05.29davep_infoseekerMay 30 17:04:47 ERROR[6705]: chan_zap.c:6541 mkintf: Unable to open channel 1: No such device or address
23:05.30jas_williamsdavep_infoseeker: type ztcfg -vvv
23:06.04davep_infoseekershows the channel map with two channels configured
23:06.20davep_infoseekerand an error
23:06.20davep_infoseekerZT_CHANCONFIG failed on channel 1: No such device or address (6)
23:06.31Drukenblitzrage: how much bandwidth and what do you need it for?
23:06.57jas_williamsdavep_infoseeker: post you zapata.conf to pastebin it has an error
23:07.11speakmandavep_infoseeker: got the same message with my TE110P
23:07.34speakmandavep_infoseeker: after a "powerdown" on the server, and turning it back on, the message is gone...
23:07.44speakmanvery strange behaviour
23:08.20davep_infoseekerzapata.conf or zaptel.conf?
23:08.35*** join/#asterisk Xander77 (~Alex@exten-halls-243.soton.ac.uk)
23:08.41speakmanHave to go to bed now, cya people! :D
23:10.08davep_infoseekerzpata.conf is posted
23:10.28davep_infoseekerspeakman: I just rebooted and got the same thing
23:10.37blitzrageDruken: haha... thanks, Marlow has already kindly donated me a vserver (whom I thank greatly!)
23:10.40Drukenspeakman: no... the digium cards don't like soft reboots, they need a "powerdown"
23:11.06Drukenblitzrage: aight no prob :)
23:12.08blitzrageDruken: thanks for the offer, I appreciate ti
23:12.09heragMarlow: or anyone else who has an idea, I now have a very curious problem...I can get it to dialin just fine, but I have to comment out the "secret = <pass>" line in my [peer] definition, and if I do that, then I can't dial out...what is going on?
23:12.22jas_williamsdavep_infoseeker: post your zapata.conf
23:12.39Marlowherag : isn't there something like authsecret in sip.conf ?
23:12.46Marlowherag : i would have to check though ..
23:12.55davep_infoseekerjas: already did under infoseeker on postbin
23:14.57*** join/#asterisk welby (~welby@70.84.53.138)
23:15.08*** join/#asterisk Tabitha (~moi@pcp0010097148pcs.levtwn01.pa.comcast.net)
23:17.16heragMarlow: no, unfortunately there is no authsecret in sip.conf....but the lightbulb finally went off, and I figured out how to make the "type = user" entry properly, and that solves everything, cause now asterisk matches incoming calls against the [user] entry, and outgoing calls are free to go out through the [peer] section
23:17.51davep_infoseekerwould it give me problems loading asterisk if the phone line is not plugged into the card?
23:17.51Marlowgood good ..
23:18.23Marlowherag : it's usually just a matter of poking around .. i have to figure it out every time from new, because i don't poke around in the config often ...
23:18.25jas_williamsdavep_infoseeker: are you sure your modules are in slot 4 and 1 post the relevant section from var/log/messages
23:19.22davep_infoseekerMay 30 16:53:31 localhost kernel:  <6>Zapata Telephony Interface Registered on major 196
23:19.31davep_infoseekerthat is the only message in /var/log/messages
23:20.12davep_infoseekerjas_williams: not sure if the modules are loaded right because the modprobe just hung
23:20.24davep_infoseekerjas_williams: i ran modprobe wctdm
23:21.02davep_infoseekerjas_williams: I think thats the right module for my card
23:22.00davep_infoseekerjas_williams: I was using this same card and zapata.conf on another box and it was working fine
23:23.14davep_infoseeker?
23:23.27jas_williamsDruken: Got your privacy manager working yet ?
23:23.56jas_williamsdavep_infoseeker: did you modprobe zaptel first ?
23:24.44davep_infoseekerjas_williams: no I just modporbe wctdm it loaded zaptel by itself
23:25.30jas_williamsdavep_infoseeker: try loading the modules your self zaptel first then wctdm then do a ztcfg -vvv
23:25.40davep_infoseekerok
23:26.17davep_infoseekerhumm cant seem to unload the modules now
23:26.22davep_infoseekerguess I will reboot?
23:26.32jas_williamsguess so
23:28.21*** join/#asterisk davep_infoseeker (~davep@S010600a0cce01511.cg.shawcable.net)
23:29.23davep_infoseekerjas: looks like that worked ztcfg gave me no error that time
23:29.52*** part/#asterisk rene- (~root@201.144.11.57)
23:30.48davep_infoseekerthanks gotta run
23:31.35*** join/#asterisk crash3m (crash3m@crash3m.user)
23:33.07*** join/#asterisk faqu (pn@OL187-166.fibertel.com.ar)
23:34.15faqui've problem linking 2 asterisks through IAX when i want to call any extension i got Call rejected by REMOTEIP: No authority found, any idea?
23:37.12*** join/#asterisk RoyK (~roy@179.80-203-29.nextgentel.com)
23:42.38*** join/#asterisk Paul[NOC] (~paul@66.195.243.254)
23:43.01Paul[NOC]Alright... I have eth0 as a public ip and eth1 as a internal IP. Phones are registering fine and ring ok
23:43.08Paul[NOC]But no voice RTP between
23:43.10Paul[NOC]ANy ideas?
23:44.44*** part/#asterisk kankan (kankan@frenchcancan.net)
23:45.13MarlowPaul[NOC] : no voice between what ?
23:45.35MarlowPaul[NOC] : internal phones together, internal phone to outside, be specific
23:45.55Paul[NOC]Outside > Internal
23:46.03*** join/#asterisk dr123 (~temp@12-202-51-38.client.insightBB.com)
23:46.40Paul[NOC]5004 is the RTP Port and I assume it's UDP
23:46.42MarlowPaul[NOC] : canreinvite=no ?
23:46.59dr123Question does anyone know why on a Grandstream BudgetTone 100 or SJPhone under SIP protocal when transfering to Parked Call I dont hear the extension the call was parked on... something about a blind transfer.... NEED HELP!!!
23:47.40Paul[NOC]Marlow,  yes
23:47.55faqui've problem linking 2 asterisks through IAX when i want to call any extension i got Call rejected by REMOTEIP: No authority found, any idea?
23:48.00RoyKkram: ping
23:48.55MarlowPaul[NOC] : and the devices are set to use the asterisk box as outgoing proxy ?
23:49.55*** join/#asterisk rene- (~root@201.144.11.57)
23:49.58rene-hey
23:50.29*** join/#asterisk gordonjcp (~gordonjcp@cpc3-broo2-5-0-cust232.renf.cable.ntl.com)
23:50.32mrtwisterquestion. oh323 vs h323. latest or cvs head.. what is better. h323 using latest pwlib ans openh323,.. what about fast/slow start, g245 tunneling, what about f723 codec?
23:50.34gordonjcphello
23:51.20*** join/#asterisk apardo (~apardo@80.26.160.180)
23:52.23Marlowmrtwister : in general => h323 is the wrong choice .. but if it absolute has to be h323, whichever you can get working ..
23:53.13Paul[NOC]<PROTECTED>
23:54.26mrtwisterlol :) thanks, but have to use h323. wholesale market is h323 based. i do not like h323, but must use.
23:54.49gordonjcp?
23:54.49Marlowmrtwister : then try both .. whatever works  best for you ..
23:54.54mrtwisteruntil now i play with oh323, but maybe someone have good expierence with h323?
23:55.00gordonjcpwho the hell uses h323 these days?
23:55.02Marlowmrtwister : usually i had the best experience with chan_h323
23:55.08Qwellgordonjcp: people stuck in the 80s
23:55.16gordonjcp'lo Qwell
23:55.31Marlowgordonjcp : old fashioned telco's with no clue ..
23:55.34gordonjcph323 is dead
23:55.49gordonjcpno-one supports it any more
23:55.55zonezerot38modem uses h323 to talk wiht the cisco
23:55.59mrtwisterlet discuss ? :)
23:56.21mrtwisterplease try to get sip routes there www.forumvoip.com
23:56.25Marlowzonezero : t38modem talks h323/t38 to anything, that will terminate/receive fax
23:56.29zonezerobut then maybe this is why t39modem dont work so good
23:56.30Marlowzonezero : via t38
23:56.33mrtwisterwholesale market is h323
23:56.35[hC]so.. Who here has some experience with cisco PoE? :)
23:56.38Marlowzonezero : but it works with cisco, yes ..
23:56.55mrtwisterand - big problem - 3rd world is h323 g723.1
23:57.11mrtwisterlike bangladesh, african routes, india etc
23:57.34gordonjcpmrtwister: h323, forget it
23:57.43gordonjcpget them using proper protocols
23:57.56gordonjcpif VoIP was an airline
23:58.08gordonjcpH323 would be the Lockheed Constellation
23:58.09mrtwisterhehe
23:58.26mrtwisteryou know cost of 64kbps 1:1 in bangladesh or nigeria?
23:58.37gordonjcpyes
23:58.54gordonjcpwhich is why I wonder why you waste it with H323
23:59.23mrtwisterwhy me
23:59.35mrtwisterlet say we buying routes
23:59.51mrtwistersaudi arabia, kuwait, egypt, bangladesh, nigeria etc

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