irclog2html for #asterisk on 20050719

00:00.15harryvvopus, moh works here...btw, would you like to establish a link ?
00:00.44*** join/#asterisk ManxPower (~eric@stirprop-s4-0-0-21.ndcr2.datasync.net)
00:04.23mariogamboahow i can load first my E1 card and after load the tdm 400p with zaptel
00:04.24mariogamboa?
00:06.04*** join/#asterisk Ariel_ (~Ariel@dsl-20-177.cofs.net)
00:09.15mariogamboahow i can up my tdm card?
00:09.24mariogamboacause i don't know why is down
00:09.26mariogamboa?
00:13.27Ariel_hello everyone
00:13.35Kattyhihi Ariel_
00:13.39harryvvariel!
00:13.49mariogamboahi ariel
00:13.50Ariel_Katty, zap is due to the drivers were made by zapata
00:13.50*** join/#asterisk newmember (~newmember@S010600a0c93dce87.cg.shawcable.net)
00:14.03Ariel_mariogamboa, service zaptel restart
00:15.30harryvvariel, was able to get into my firewall
00:15.38mariogamboamm
00:15.43mariogamboai do it
00:16.11harryvvbut, I need to go and pickup some auto parts.
00:16.18mariogamboabut i don't recibe tone from the card and asterisk when put the command zap show channels is empty
00:16.27Ariel_harryvv, hello how are you doing
00:16.50Ariel_mariogamboa, how are they configured ztcfg -vvvv
00:16.52mariogamboai remember some here say me you need to load first you e1 and after the fxs fxo card but i don't know how
00:17.02mariogamboaok
00:17.05harryvvgood, well bad..wifes car is having problems my car is on the blocks.. you know how it is. I need to leave to get some parts.
00:17.11Ariel_oh you have not set them up yet?
00:17.21mariogamboachannel 1-31 e1  channel 32-33 fxo 34-35 fxs
00:17.28Ariel_harryvv, see you later then have fun.
00:17.38Kattytime to start workout
00:17.46Katty2 hours later, every muscle in my body shall ache
00:18.47mariogamboasorry
00:18.49DarthClueharryvv: i will have configured them in another couple of months,maybe.
00:18.59sig-;>
00:19.04mariogamboais inverse and in this moment works the fxs fxo
00:19.10mariogamboai need to check if e1 works too
00:19.42sig-Katty: zap zap it`s mean really nothing, but i like say zap zap ;>
00:19.43SarahEmmDarthClue: you should go to cluecon. that'll help.
00:19.50Kattysig-: k
00:20.00KattyDarthClue: yes, come to cluecon!
00:20.00*** join/#asterisk Sedorox (~Brandon@sedorox.staff.smartserv)
00:20.02DarthClueSarahEmm: nothing can help this problem.
00:20.03KattyDarthClue: ;P
00:20.32mariogamboaquestion in my case i want to learn more about asterisk is good to take the asterisk tranning course and dcap?
00:20.59DarthClueSarahEmm: we (meaning i) have been trying to get this particular company setup for nearly 8 months now.
00:21.17sig-No such extension 'katty' in context 'iaxout' =)
00:21.24*** join/#asterisk nathe (~lissa@silenceisdefeat.org)
00:21.40hermieDarthClue: are you related to Clued Skywalker?
00:21.47hermie(ba da da)
00:21.57DarthCluemariogamboa: no, use google and read the wiki and experiment.
00:22.05DarthCluemariogamboa: and come to cluecon.
00:22.20DarthCluehermie: come to cluecon.  we will be performing DNA tests at that time.
00:22.34KattyWHAT?!
00:22.40Kattyi am not getting a dna test done!
00:22.42Kattynonononono
00:22.50SarahEmmdarkskiez: meep! for what?
00:22.54SarahEmmDarthClue rather
00:22.56DarthClueKatty: not to worry.
00:23.01KattyDarthClue: k
00:23.34SarahEmmDNA, rather...
00:23.44mariogamboacluecon?
00:23.48mariogamboawhat that mean
00:24.27DarthClue~cluecon
00:24.27jbotwell, cluecon is http://www.cluecon.com - The Open Source Telephony Expo and Developers Conference featuring presentations by Mark Spencer, Greg Boehnlein, Ken Rice, Brian West, Vikrant Mathur, Craig Southeren, David Sugar, Bob Andreasen, Joshua Colp, Greg Merriweather, Brian Fertig, Peter Nixon, Marc Olivier Chouinard, and Anthony Minessale II.
00:24.41Kattyi....see.
00:25.04DarthClueKatty: we already know who you are related to, no need for testing.
00:25.22Katty...
00:28.06file[laptop]ooh dangerous fashion
00:28.08Kattyhi kram (=
00:28.08litagewhat FXO/FXS cards and other hardware should one start out with to play with and learn about Asterisk?
00:28.12Kattyfile[laptop]: exactly
00:28.21kramonly a wave??
00:28.24Kattyfile[laptop]: cower in fear!
00:28.30file[laptop]I'm watching two movies
00:28.34file[laptop]you're lucky to get a wave
00:28.40kram:(
00:28.45kramwell thanks for the wave :)
00:28.46file[laptop]watching them, simultaneously!
00:28.53krami'll take what i can get
00:28.55kramand share snuggles
00:28.59kramgreets to katty too!
00:29.27kramreps? :)
00:29.41Kattycurrently on teh dumb bell kickbacks (=
00:30.19opus_<PROTECTED>
00:30.22opus_hahahaah
00:30.53file[laptop]Cresl1n: ohhhhhhh sexy dude
00:31.08kramkatty: ooh, that's hot
00:31.52hmmhesaystime for a hair cut
00:32.03*** join/#asterisk SpaceBass (~sp@c-24-125-184-203.hsd1.va.comcast.net)
00:32.23Cresl1nhowdy
00:32.24Cresl1n:-)
00:32.30hmmhesaysgotta chop those spikies down
00:32.32Cresl1nDid kram get his camera working?
00:32.53file[laptop]Cresl1n: aha I have the power to summon thee!
00:33.16Kattykram: i have to keep the girly arms toned (=
00:33.38opus_got a picutre
00:33.39MiccIs there some type of keyboard that lets you type a lot faster than regular keyboards?
00:33.46opus_micc yeah
00:33.52Kattymmm, spikey
00:33.57Miccopus_, whats it called?
00:34.11opus_i'm looking
00:34.19Cresl1ndvorak?
00:34.24hmmhesaysoff to the haircutter
00:34.28opus_micc - its only slashdot like every year
00:34.32Cresl1nyou have to learn a new keyboard layout though
00:34.56opus_Dvorak, instead of QWERTY
00:34.56Miccyeah I've heard of that one. I need to learn the new layout.
00:35.12Cresl1nit takes  a few weeks to feel somewhat comfortable again
00:35.30opus_http://www.maltron.com/ that one is a fast qwerty keyboard
00:35.36opus_my friend uses it, I hate it because I can't use his computer
00:35.45*** join/#asterisk cpatry (~junky@69.156.123.178)
00:36.04SpaceBassand this will help http://www.artlebedev.com/portfolio/optimus/
00:36.12*** join/#asterisk dasenjo (~dasenjo@63.245.86.173)
00:36.26NukemizerKram, thanks for Doin * !  you have made the world a better palce :)  ( Nukemizer tips his hat )
00:36.36Nukemizerplace
00:36.41opus_old IBM keyboards are good
00:37.12Kattyhmmhesays: :<
00:37.42cpatrymoooo
00:38.46bkw_omg why learn a new keyboard layout?
00:38.54bkw_you can type and spell just as badly with qwerty
00:40.03opus_this is mykeyboard layout http://www.hypermaths.org/quadibloc/comp/images/kybcon.gif
00:40.04Kattyhaha
00:40.35opus_i rewrote asterisk to use APL
00:41.27Corydon76-homeThat's nothing... bkw_ rewrote Asterisk so it runs as part of emacs
00:41.44Corydon76-homeand consumes 300 times as much memory
00:42.20Kattyyeah well i didn't rewrite asterisk ata ll (=
00:42.22Kattyhow neat is that.
00:42.30Corydon76-homeAt a quarter the speed
00:42.31Nukemizersounds like a microsoft App
00:42.48Katty:<
00:42.52SedoroxHa! I re-wrote asterisk in .net and is a fully web-based application, as you need IIS7 to run it :p
00:43.06file[laptop]I rewrote asterisk in assembler, and it runs on my toaster
00:43.08Kattyis this he who has the biggest re-write wins?
00:43.11Kattyfile[laptop]: hottt
00:43.12Nukemizernow "theres your sign"
00:43.12Sedoroxlol
00:43.14denon.net isnt a lagnauge :)
00:43.36Sedoroxsorry... C#
00:43.37Sedorox:p
00:44.25Corydon76-homeUsing punchcards...
00:44.49Sedoroxdamn... I would hate to see what the moh punchcards look like
00:44.55SarahEmmLOL
00:45.10Sedoroxor do you just use those old recording piano ones?
00:45.23SarahEmmthey're round... and it's not punches, you carve a groove in a spiral in it...
00:45.38Sedoroxa.... record?
00:45.40*** join/#asterisk iq (~iq@70-59-161-101.omah.qwest.net)
00:46.26*** join/#asterisk doughecka (~Miranda@doughecka.user)
00:46.30iqhi
00:46.39voiperis there a way to turn the fake ring off ? I have a SIP UA making call through asterisk to a SIP softswitch (Nextone)
00:48.15*** join/#asterisk adiao (~adiao@219.82.155.128)
00:49.35*** join/#asterisk stkn (nobody@stkn-active-pdpc.developer.gentoo)
00:49.48SarahEmm?CLear
00:49.48adiaoi use a non-zaptel card ,why it's channel can't send voice to iax channel? thanks
00:49.49SarahEmmoops
00:52.16Sedoroxwhat do you mean you use a non-zaptel card??
00:52.37*** join/#asterisk dasenjo (~dasenjo@63.245.86.173)
00:53.17adiaoIt's a card of myself
00:53.40adiaoI add this card to asterisk by refer to chan_vpb.c
00:57.19SedoroxI dunno then
00:57.26Sedoroxthen again.. I'm not much help anyway
00:59.08*** join/#asterisk siffer (~foobar@69.156.123.178)
00:59.44cpatrysiffer!!!
01:00.40sifferwhat'up
01:00.47cpatrysky!
01:00.51Chujiwhat do you guys use for a win32 jabber client?
01:01.17SarahEmmdon't :)
01:01.20Twisterpandion
01:01.54ChujiI need to look at rolling it out to our IT department
01:02.04Chujiwe are getting spread out too much in the buildings
01:02.51SarahEmmooh, wait
01:02.54SarahEmmwe use that at work on win32 lol
01:02.56SarahEmmTipicIM
01:05.58Darwin35grrr
01:06.29ManxPowerthe people on #cisco are a bunch of jerks
01:06.50SarahEmmi dunno that i agree with that
01:06.50Twistersoo are the people when you call cisco
01:06.52SarahEmmi've had lots of help from them
01:06.58*** join/#asterisk pdracevich (~bob@210.54.249.228)
01:08.09pdracevichHello, all i have just updated my test asterisk box with the newest CVS-HEAD and I now have a major issues with IAX phone using the echo test, and other stuff sound like the jitter is a major problem
01:08.16pdracevichany ideas
01:12.00*** join/#asterisk Druken (~druken@CPE00121716da99-CM000e5cde4ca2.cpe.net.cable.rogers.com)
01:13.54_DAWChuji - Jabber, you tried JAJC?
01:15.45ManxPowerpdracevich, did you check today's mailing list messages?
01:15.56pdracevichyeap :(
01:16.32*** join/#asterisk SarahEmm (~sarahemm_@MTL-ppp-159032.qc.sympatico.ca)
01:16.41Twisterwb :)
01:17.22*** part/#asterisk |Fender| (~fender@fctnnbsch17-156034217059.nb.aliant.net)
01:19.36*** join/#asterisk PBXtech (~nik@001-723-259.area1.spcsdns.net)
01:20.26PBXtechanyone run astlinux?
01:23.01*** join/#asterisk TheEmperor (~TheEmpero@210.19.250.122)
01:24.42*** join/#asterisk Newbie___ (me@211.24.146.12)
01:25.12*** join/#asterisk juice (~juice@mo-69-69-247-217.dyn.sprint-hsd.net)
01:26.23*** join/#asterisk sudoer2 (~toy@c-24-60-183-102.hsd1.ma.comcast.net)
01:26.46*** join/#asterisk Convey (~chatzilla@208-216-127-234.cust.gti.net)
01:29.33Newbie___hi all, any idea on how do i remove the "fake" PSTN ringing tone generated by asterisk ?
01:30.08ManxPowerNifty.  I just got /kicked from #cisco
01:30.25Maxxedheh
01:30.49Maxxedhowd u managethat
01:30.55ManxPower*shrug*  They were not helping anyway.
01:31.00Maxxedoh :p
01:31.01Maxxedheh
01:31.02Maxxedwha cha need
01:31.03ManxPowerMaxxed, some auto /kick when they see "rofl"
01:31.21doughecka:)
01:31.21dudesManxPower - and that is a suprise?
01:31.26dougheckarofl
01:31.29ManxPowerMaxxed, Familiar with Cat 55xx running CatOS?
01:31.33Maxxed[20:31] <Maxxed> rofl ?
01:31.34Maxxedsee no kick
01:31.42ManxPowerMaxxed, Try EFNet #cisco.
01:31.44Maxxednuhuh
01:31.48Maxxedheh
01:31.54Maxxedwhata buncha asses
01:31.58ManxPowerNobody ever actually says anything on #cisco on this network
01:32.01Maxxedcome on now, rofl dosnt mean anything bad
01:32.04SarahEmmManxPower: oh! efnet!
01:32.07Maxxednah they just idle
01:32.09SarahEmmi hang out in FN #cisco often
01:32.44Maxxedi dont care for efnet, chan services have goten me spoiled
01:32.58ManxPowerMaxxed, I went there in desperation
01:33.18Juggiei've been ideling on efnet for 10 years now :P
01:33.42Maxxedlol
01:34.11Maxxeddewd its all about the tax deductable freenode :P
01:34.35Maxxed:p
01:35.32Twisteranyone ever had a problem with winblows xp home running out of buffer space after long voip conversations
01:36.03Twisterim too poor right now to buy a hardphone, even off ebay lol
01:36.03TheEmperorhello
01:36.22Maxxedbuffer overflows are good, inject some shell code :p
01:36.28Maxxed10 ?
01:36.32Maxxedu sellin em ?
01:36.36TheEmperori am wondering if anyone can tell me how to make the calls go through an asterisk server faster when using DISA?
01:36.38dougheckano
01:36.39dougheckafor a client
01:36.43Maxxednuts
01:36.46dougheckasetting up a phone system tomorrow
01:36.50Maxxedcool beans
01:36.52Maxxedhave fun :)
01:36.56Maxxedi no i allways do
01:36.57dougheckahopefully I can figure out my issue first
01:37.06Maxxedrollouts are easy
01:37.09dougheckaI get pops/clicks in the audio stream
01:37.11Maxxedissue?
01:37.13dougheckaduring a call
01:37.15Maxxedum
01:37.25dougheckaannoys the heck out of me, and its gonna bother the client
01:37.25Maxxedwhat image u runnin
01:37.29dougheckaimage?
01:37.33*** join/#asterisk wasim_ (~wasim@wasim.active.supporter.pdpc)
01:37.48TheEmperorcan anyone help?
01:38.12SarahEmmTheEmperor: with?
01:38.23TheEmperortrying to speed up dialling with DISA
01:38.42TheEmperorcall comes in via pstn into disa contect and it takes about 20 seconds for the call to go through
01:39.50dudesAnyone ever had IAX/SIP not have audio before?  Because the configuration is the same on this box as it is on a box that does the exact same tasks but it sound only works once in awhile and when it does it's choppy ... then it stops working.
01:39.59TheEmperorbut using softphone it only takes about 8 seconds...
01:41.36jskcrdudes yes and its usually a firewall issue
01:42.01jskcrerr maybee not did not see the choppy part :/
01:42.10dudesit's head from today
01:42.15jskcrwhat codec?
01:42.32dudesTried them all ... ulaw/gsm
01:42.56dudesMoreover it can't be bandwidth since the computers on a 5m/bit backbone
01:42.59jskcrany info when ya turn verbose debugging on
01:43.19Twisterhow do i start a conference?
01:43.31cpatryTwister: with meetme?
01:44.17Juggie5mbit? :)
01:44.23wunderkinjskcr, im getting a dedicated ld pri t1 from global crossing.. ill let you know how it is.. unless you were asking about voip
01:44.27Juggieit could be bandwidth if anything else happens on that network
01:44.32dudesTwister - make an extension that points to MeetMe(*room number*)
01:44.35jskcrwunderkin: its about the voip :)
01:44.39wunderkinok
01:45.21dudesJuggie - This is going to be a SIP gateway and no action on the network until this POS starts working
01:46.09dudesWe're going to try putting in an older head and see what happens
01:46.24*** join/#asterisk pr0m_ (~pr0metheu@24-75-196-70.chvlva.adelphia.net)
01:50.55*** join/#asterisk focks (~cbruender@12-220-210-26.client.insightBB.com)
01:51.21TheEmperorcould someone tell me how to speed up dialling with DISA?
01:51.58TheEmperor:)
01:52.25*** join/#asterisk kks (~kks@203.115.208.140)
01:52.31*** join/#asterisk tecnico (~tecnico@user-24-236-123-31.knology.net)
01:53.49kksAnyone use grandstream and what facility should i give to capture syslog in separate log file?
01:58.16*** join/#asterisk Inv_Arp (junya@adsl-156-139-229.mia.bellsouth.net)
02:02.27*** join/#asterisk Rowter (~SilverDra@201.133.210.80)
02:03.45*** join/#asterisk mcreedjr (~email@oh-65-41-206-34.sta.sprint-hsd.net)
02:04.22mcreedjrHi all, anyone wanna help an Asterisk n00b out? I have no dial-tone on my FXS port :(
02:04.51mcreedjrWhen I try and run the Asterisk executable, I get an error while trying to load the module chan_features
02:04.56mcreedjrI am sure that is why
02:06.01*** join/#asterisk heath__ (~heath__@12-215-32-56.client.mchsi.com)
02:06.05*** join/#asterisk tessier (~treed@222.253.74.83)
02:06.15heath__i can't find where to get older cvs head versions
02:06.27mcreedjrAnyone around that can help me out?
02:06.53drumkillaheath__: cvs co -D "15 days ago" asterisk
02:06.56drumkilla... seriously  :)
02:07.12heath__cool, thanks
02:07.45_DAWmcreedjr - you configured zaptel and zapata correctly?  What does zttool tell you?
02:07.48Kattymrow
02:08.15mcreedjr_DAW: zttool recognizes my card and the Alarms column says "Ok"
02:08.39_DAWyou have signalling set to fxo?
02:08.52mcreedjr_DAW: when I pickup my anaglog phone, the active column in zttool goes to 1
02:09.04mcreedjr_DAW: in zaptel.conf?
02:09.41mcreedjr_DAW: fxols=1 (1 is the ID of my FXS port)
02:09.44_DAWyes fxoks
02:10.11mcreedjrWill ls do? or does it have to be ks?
02:10.38_DAWkewl start is better
02:11.11_DAWmake sure signalling is fxoks in zapata as well
02:11.19mcreedjrokay... I will switch it.. then in my zapata.conf, it has these three lines: signalling=fxo_ls, context=main, channel=1
02:11.23TheEmperorso does anyone know how to make DISA send calls faster? or put the calls through faster?
02:11.45_DAWmcreedjr - is that all it has?  just those three?
02:11.47drumkillaTheEmperor: give it ESP!
02:12.08mcreedjr_DAW: yes, do I need more than that?
02:12.27_DAWloadzone and defaultzone I believe
02:12.32TheEmperordrumkilla: how do I do that?
02:12.47TheEmperoris that a special agi?
02:13.33jskcrlmfao
02:14.42drumkillayeah, I wrote it.
02:14.49jskcrYes the esp.c it knows what number your going to call before you do, it does not need to register or invite because of a patented quantum algorithim that predicts it
02:14.53drumkillait's the coolest AGI script ever!
02:14.57TheEmperorseriously though guys
02:15.16TheEmperori have having an issue where using disa it takes 30 seconds for the calls to go through
02:15.27TheEmperorusing a softphone (Such as firefly) it goes through in about 8 seconds
02:15.34jskcrethereal.
02:15.49TheEmperor?
02:15.51jskcrSee whats goin on at the packet level.
02:16.08TheEmperorbut how come with a softphone it goes through much faster?
02:16.45jskcrProbably its properly configured and the disa is not
02:16.57TheEmperorwhere can i go to get more info on disa?
02:17.23mcreedjrBah... When I run asterisk -vvvc, I get an error about undefined symbols while trying to load chan_features.so. Is that a config problem, or a compile issue?
02:17.27SarahEmmTheEmperor: the wiki
02:17.44TheEmperorok
02:17.51jskcrhttp://www.voip-info.org/wiki-Asterisk+cmd+DISA
02:17.58focksTheEmperor, also http://forums.digium.com/index.php
02:18.10TheEmperorthanks
02:18.39jskcrmcreedjr google site:lists.digium.com chan_features.so
02:18.55twistedTheEmperor, did you find the ESP AGI?
02:19.33TheEmperornot yet
02:19.33jskcrtwisted:  I thought it had been ported to C and merged into tommorow cvs
02:19.38mcreedjrjskcr: Google is your friend :) I will look, thanks!
02:19.39TheEmperorcan you give me the link where i can download it?
02:19.40twistedjsckr
02:19.40twistedno
02:19.56twistedTheEmperor, give me a few minutes to link it on my website
02:20.14TheEmperorgreat!
02:20.19TheEmperorlet me know when it's done
02:21.35mcreedjrjskcr: Didn't find much... got any other idears?
02:22.02_DAWmcreedjr = in zapata.conf do you have the lines listed with the context=main first?
02:22.15_DAWthen signalling
02:22.17_DAWthen channel?
02:22.36twistedhttp://www.indigent-networks.com/esp.html
02:22.55drumkillatwisted: LOL
02:23.02*** join/#asterisk mutilator (WebChat@i.think.napoleon.dynamiteblows.com)
02:23.26jskcrmcreedjr:  trey http://www.voip-info.org/wiki-Asterisk+Slimming
02:23.36mcreedjr_DAW:yes, those three lines, in that order
02:23.44twistedTheEmperor, ^^
02:24.03jskcrlmao
02:24.07_DAWmcreedjr - got me..
02:24.24*** join/#asterisk shmaltz (~chatzilla@69.28.255.210)
02:24.43TheEmperoryes twisted
02:25.01*** join/#asterisk znoG (~gs@200.115.216.109)
02:25.13mcreedjr_DAW: Thanks for the help... back to the drawing board
02:25.14twistedTheEmperor, http://www.indigent-networks.com/esp.html
02:25.52TheEmperorhey, that is a cute picture
02:25.55twistedhehe
02:25.57jskcrtwisted:  you should of spelled STFU
02:26.45TheEmperorso is this chat room here for newbies to be able to ask for some help?
02:26.56TheEmperoror it's for experts only with expert related problems?
02:27.02jskcrNo thats whats #ser is for
02:27.14twistedTheEmperor, yes, it is.
02:27.20*** join/#asterisk MikeJ[Laptop] (~ircatjerr@adsl-69-209-143-200.dsl.sfldmi.ameritech.net)
02:27.21twistedwe're just tired and bored
02:27.24jskcrHey Mike
02:27.30TheEmperori can imagine :)
02:27.40TheEmperorsorry for the lame questions, but i was stuck on that one.
02:27.41MikeJ[Laptop]we care alot
02:28.01twistedi would say check your digit timeouts and the length of your pattern matches vs. real extensions
02:28.30TheEmperorwell, what i am doing is authenticating via caller id to give access to disa to try and speed it up
02:28.32MikeJ[Laptop]you would if what?
02:28.57twistedMikeJ[Laptop], if i were actually speaking
02:29.21TheEmperorseems the call goes through and into disa context pretty quick when i look at the asterisk console, but the delay comes after that, terminating the call
02:29.50twistedyeha
02:30.04twisteder yeah, check your pattern matches vs. extensions  (length) and your digittimeout
02:30.18twisteddisa just drops you into a dialtone and waits for you to dial a full valid extension
02:31.12SarahEmmshido6, you here?
02:31.38*** join/#asterisk tessier (~treed@222.253.74.83)
02:33.23TheEmperortwisted: yeah, it gives a dialtone and then the number gets sent through
02:36.23jskcrTheEmperor: ya know caller id can be spoofed :)
02:37.20*** join/#asterisk HellAgony (HellAgony@200.121.242.95)
02:37.38TheEmperoryes, but caller id with password :)
02:40.24MikeJ[Laptop]spoof
02:41.23jskcror spoofalumpagus as its known on sesame street
02:41.53shmaltzjskcr, of course it can be spoofed, it's not news, way b4 voip you could spoof it
02:42.45*** join/#asterisk unixmonster (user@pcp986360pcs.northw01.in.comcast.net)
02:42.59dantand it can't be spoofed in most of the world
02:43.06MikeJ[Laptop]time to make the donuts...
02:43.11MikeJ[Laptop]time to fix the sourcecode
02:43.25twisteddude
02:43.29twistedour code isn't full of holes
02:43.31twistedwtf r u thinking
02:43.33jskcrlol
02:44.04MikeJ[Laptop]uhhhh
02:44.11MikeJ[Laptop]what code have you been reading ;)
02:44.54twistednsa_security.o
02:44.55twistedaer
02:44.56*** join/#asterisk bjohnson (~bjohnson@i216-58-64-17.igs.net)
02:44.56twisted.c
02:45.06twistedoh wait, this is asterisk
02:45.07twistedmy ba
02:45.07twistedd
02:45.13wunderkinmmm donutholes
02:45.15jskcrDont you know theres no such thing as the nsa.
02:45.17twistedi will calm my pinky spasms
02:45.37jskcrTheres no airforce base in groom lake nevada either
02:45.45twistedi know there's not
02:45.51twistedi've not seen it out of my airplane window
02:46.21Kattyscooby do
02:46.25Kattydoo
02:46.30Kattysomething like that
02:46.46twistedzoinks!
02:47.14jskcrno its a cvs -D head with lots of extra meat and fat
02:47.29jskcrI was a vegan for about two years
02:47.38jskcrThen the happy meal came
02:47.39*** join/#asterisk dca (~dca@c-24-9-31-63.hsd1.co.comcast.net)
02:47.47jskcrKnocked my ass right off the wagon
02:48.06*** join/#asterisk pbxxx (~pbxadmin@CPE0050bae8d02c-CM0011aea484a4.cpe.net.cable.rogers.com)
02:48.10twistedhahahah
02:48.11twistedhappy meal
02:48.12twistedroflmao
02:48.37jskcrAfter that I got a 20oz porterhouse, mmmmm
02:48.54twisted;)
02:49.09jskcrThe only problem with being vegan is you fart alot.
02:49.12twistedshining happy vegans eating plants
02:49.17file[laptop]I'm thinking of getting the in ear Apple headphones...
02:49.28twistedfile[laptop], i got badderasser ones than those
02:49.29Kattytwisted: like oreos
02:49.33Kattytwisted: and baked lays
02:49.40Kattytwisted: and eggrolls (without the egg)
02:49.49Kattyand mmm, spaghetti
02:49.49file[laptop]twisted: that made total sense to me ... NOT
02:49.51twistedKatty, hehe... i LOVE baked lays
02:49.58twistedesp sour cream & onion
02:50.03MikeJ[Laptop]ouch.. in the ear?
02:50.09Kattyi like sunchips
02:50.15twistedyeah, sunchips are good too
02:50.16Kattyoriginal sunchips, obviously
02:50.21MikeJ[Laptop]in the ear?
02:50.23Kattydunked in tofutti better than sour cream
02:50.25Kattyit's dreamy
02:50.27twistedKatty, there's other flavors? *snicker*
02:50.33Kattyyes
02:50.42jskcrhttp://www.indigent-networks.com/esp.html
02:50.48MikeJ[Laptop]I have a freezer full of deer meat to eat
02:50.56Kattythere's cheddar harvestor something
02:50.57jskcrIm gonna print that and put it up.
02:51.02Kattyand then the sour cream and onion ones
02:51.04MikeJ[Laptop]my father in law sent me home w/ a cooler full of meat
02:51.09Kattyand i htink they might have a bbq one too
02:51.10Kattynot sure
02:51.11MikeJ[Laptop]bear jerky anyone?
02:51.12twistedKatty, oh yea, those are good
02:51.26Kattythe only thing i actually miss is ravioli
02:51.28twistedbut I do love steak with my sunchips
02:51.31Kattywhich, i could probably make my own
02:51.42Kattywith some sort of soy sausage stuff
02:51.44twistedmedium rare
02:51.45file[laptop]dang nabbit I'm missing QaF
02:51.48jskcrWhen I lived in ohio they would have there cookouts where its was game meat, everything from ducks to woodchucks
02:52.05twistedi couldn't eat a woodchuck.
02:52.19Juggieeugh
02:52.20*** join/#asterisk doughecka (~Miranda@doughecka.user)
02:52.21Juggiegross
02:52.38twistedanywho
02:52.38dougheckaamen
02:52.39twistedi need a shower
02:52.44Kattythanks for sharing
02:52.47jskcr:)
02:52.52twistedKatty, you're welcome ;)
02:52.56_DAWPeople in racoon down here in Louisiana  :\
02:52.58shido6christ
02:53.01shido6i need sleep
02:53.07shido6datacenter whir ear
02:53.10dougheckashido6: :)
02:53.12shido6crap is back up
02:53.16shido6sorry about the delay folks
02:53.22jskcrPeople in racoons? thats nasty
02:53.31SarahEmmgo shido6! :)
02:54.20jskcrngrep sip
02:54.23_DAWI here it's greasy but good.  I not really trying to find out though.
02:54.25jskcroops wrong window
02:57.27*** join/#asterisk CyberSword (~Cyberswor@cablea0mle.cybercable.net.mx)
02:57.39CyberSwordhi this is the channel of Asterisk Home?
02:58.00jskcrthats #asterisk@home
02:58.08CyberSwordo tkz
02:58.20CyberSwordemtpy channel :S
02:58.24Supaplexhah
02:58.38Supaplexeveryone knows it's #asterisk2,000
02:58.49CyberSwordits a simple question that i have
02:59.12Sedoroxyes, you probably can
02:59.13Sedorox:p
02:59.19CyberSwordthe default password of the asterisk home dosent match whit the password of Asterisk home handbook
02:59.31Kattydoughecka: clicking is bad :<
02:59.35CyberSwordand i cant log in in the server
02:59.48dougheckaclicking in my voice calls
03:01.06jskcrhttp://davidguy.brinkster.net/computer/001.html
03:01.23jskcrlol nice micro computer
03:03.53mcreedjrI just compiled Zaptel from CVS and when I try to modprobe wctdm, I get an error that wcfxs isn't present. When I browse to my kernels module directory, it is certainly not there. Ideas?
03:05.55drumkillamcreedjr: you probably have an alias in your modules conf file that points wctdm to wcfxs
03:06.04drumkillawhich you should remove if you're using cvs head now
03:07.24mcreedjrdrumkilla: I found two lines like "alias wctdm wxfxs" in my modules file.. I commented them out.. should that do it?
03:07.37drumkillai hope so :)
03:07.41mcreedjryou 'da man
03:07.44drumkillagive it a try
03:07.46mcreedjrthat worked anyways :)
03:07.47Kattyhmmmm
03:07.50drumkillayay!
03:07.52MikeJ[Laptop]drumkilla,
03:07.52Kattythe air pressure is rising
03:07.56MikeJ[Laptop]yay
03:07.59Kattycause my arms are all tingly
03:08.06Kattywhich means it should rain within the hour
03:08.11drumkillaMikeJ[Laptop]:
03:08.41MikeJ[Laptop]how are you this evening
03:08.49Kattytwisted: mmm!
03:08.57MikeJ[Laptop]heh.. yeah sure you smell good
03:09.12MikeJ[Laptop]any solaris asterisk users around?
03:09.12Kattyhe eats meat
03:09.16Kattyso he can't smell /that/ good
03:09.37dougheckaanyone want to throw any new ideas out for crackling and popping on a TDM400P system?
03:09.51KattyDarthClue: mew
03:09.56DarthCluedoughecka: um, did you check for garden gnomes?
03:10.10dougheckayea
03:10.20Kattydoughecka: did you hug it?
03:10.27dougheckaI shot it
03:10.31jskcrchange its irq
03:10.31Katty...
03:10.33dantdoughecka, irq?
03:10.35Kattywell that's your problem!
03:10.41SarahEmmnini all
03:10.46Kattynini SarahEmm
03:10.50DarthCluedoughecka: fairies?  oompa loompas?
03:10.53dougheckaits on it own IRQ
03:10.58SarahEmmMe hugs katty nmini
03:10.59dougheckaactually I have 2 of them
03:11.02jskcrwhat irq?
03:11.09dougheckaand both are on seperate irqs
03:11.10*** join/#asterisk chris_d (~chris@210.21.226.3)
03:11.16dougheckaum, I think its 17 and 24
03:11.17Kattydoughecka: you might want to hug it too
03:11.28dantdoughecka, acpi
03:11.31dantdoughecka, eww
03:11.32chris_dQuick question
03:11.42dougheckaright now the server is in my car ready to travel 2 hours away tomorrow morning
03:11.47KattyDarthClue: it's going to rain soon :>
03:11.51chris_dIf I have two Asterisk servers in two locations, can I bridge a conference call between the two locations?
03:12.04dougheckadant: can I turn ACPI off?
03:12.05KattyDarthClue: :<
03:12.08DarthClueKatty: I'm still waiting...not a drop in sight.
03:12.13KattyDarthClue: nitenite
03:12.14dantdoughecka, sure
03:12.25chris_dIn other words, have some participants dial into Location A and some dial into Location B and all participants join the same conference room?
03:12.48dougheckachris_d: transfer conference a to conference b...
03:13.06drumkilladoughecka: that's too easy!  I would have dropped a call file, hehe
03:13.24chris_dI need this to be transparent to the callers.
03:13.41dougheckadant: would that solve my issue?
03:13.43chris_dSo, once they connect, they all dial the same conference number.
03:14.28chris_dI guess a dial plan on server B that routes all conference rooms to server A would work...
03:15.01dougheckachris_d: do you want to load balance or just make it accessable from both sides?
03:15.09chris_dSo, if I created a room with a number like 9000 on server A, and somebody on server B dialed 9000, the call would route over IAX to server A.
03:15.28chris_ddoughecka: Make it accessible from both sides with the minimum of fuss for the users.
03:15.51chris_ddoughecka: Load balancing is not an issue--I'll throw as much hardware at the problem as I have to. :-)
03:17.06dantdoughecka, I don't know, but you could try 'noapic' in your kernel options
03:17.20*** join/#asterisk doughecka (~Miranda@doughecka.user)
03:17.49dantdoughecka, I don't know, but you could try 'noapic' in your kernel options
03:18.32*** join/#asterisk mariogamboa (~caro@201.133.229.211)
03:18.54litagewhich FXO/FXS cards and other hardware should one start out with to play with and learn about Asterisk?
03:18.55MikeJ[Laptop]chris_d, with or without c coding?
03:19.16chris_dMikeJ[Laptop]: c coding?
03:19.28MikeJ[Laptop]writing asterisk code?
03:20.07MikeJ[Laptop]as in, you want to do it with the current functionality in asterisk, or extend asterisk to do it.
03:20.48MikeJ[Laptop]there is a nity hack way you could do it and have it all automatic with dialplan and call files and groupcount.
03:22.53*** join/#asterisk marc324 (~marc32344@206-248-158-254.dsl.teksavvy.com)
03:22.58mariogamboawell litage i use the digium tdm400p the only problem i have with it is how i can put in service some times i don´t know if for the position of the cards in zaptel or something
03:24.48*** join/#asterisk goldenolden (~goldenold@c-67-160-85-227.hsd1.wa.comcast.net)
03:25.03goldenoldenhi
03:25.12goldenoldenIs it possible to do a sip.conf specific caller ID for zap only channels on a per/user basis?
03:25.54MikeJ[Laptop]sip.conf for zap only?
03:25.59goldenoldenyah
03:26.02litagemariogamboa: not sure what you meant by "how i can put in service some times i don´t know if for the position of the cards in zaptel or something"
03:26.15MikeJ[Laptop]sip.conf is for sip, not zap
03:26.52goldenoldenright I understand
03:26.58MikeJ[Laptop]I don't
03:26.59goldenoldenbut... even if I do something like set a variable in sip.conf
03:27.02goldenoldenthat passes to extensions
03:27.26goldenoldenI just want to retain user specific CID for interoffice calling but set their DID for outbound calling
03:27.38MikeJ[Laptop]oh.
03:27.43MikeJ[Laptop]setvar in sip.conf.
03:27.59MikeJ[Laptop]then in your outbound external, set callid based on that var
03:28.04SwKor set cid in the outbound to PSTN call via the dialplan
03:28.14MikeJ[Laptop]echo
03:28.22SwKtheres about 5 ways to set the variable for it too
03:28.30*** join/#asterisk Twister (Twister@216.30.232.108)
03:28.33harryvvis there a way to make a script that allows somone to chose which phone will connect them automaticly without hearing the ivr the next time around?
03:28.34*** part/#asterisk Cresl1n (~Cresl1n@207.111.174.1)
03:28.49MikeJ[Laptop]harryvv, sure
03:28.52goldenoldenMikeJ: will setvar be user specific though or only one for the entire group?
03:29.12goldenoldenSwK: that makes sense but what if I want the CID to DID specific per/user and not just one number for everyone?
03:29.12MikeJ[Laptop]read sip.conf sample
03:29.25harryvvmike?
03:29.32MikeJ[Laptop]yes... SwK and I are talking about hte same thing
03:29.35MikeJ[Laptop]harryvv?
03:29.44harryvvyou said sure but thats it?
03:29.46SwKgoldenolden: netsted variables ;)
03:30.12MikeJ[Laptop]yes
03:30.14SwK${CID{$EXTEN}}
03:30.19MikeJ[Laptop]you want me to right it for you?
03:30.26MikeJ[Laptop]write
03:30.44SwKthat would expand to ${CID2000} where EXTEN=2000 and CID2000=someCID String
03:30.48SwKso....
03:30.55MikeJ[Laptop]sooooo
03:31.07Twisterwhat would cause an extension to be muted in *
03:31.16MikeJ[Laptop]you say potato, I say potato, let's call the whole thing off
03:31.22*** join/#asterisk akrall (user@201.144.58.200)
03:31.23harryvvnot really im sure there is something aviable. say, the ivr gives a extention that caller presses, then ask the caller "what extention would you like me to dial every time you call in"
03:31.46akrallGuys, I need to debug a nat problem, which tools do you recommend? ngrep? ethereal?
03:31.52MikeJ[Laptop]harryvv, sure, then store it to a db, then check the callid and check db when they call next time
03:31.56harryvvThat way, I dont have to do it automaticly
03:32.19harryvvmanually type it in extentions.conf
03:32.20goldenoldenSwK... thanks, in extensions.conf any clue what the SETVAR string would look like
03:32.43MikeJ[Laptop]harryvv, are you asking if somone already did it?
03:32.46SwKlook for SetCID
03:32.49harryvvyes
03:33.02MikeJ[Laptop]oh, I dunno, shouldn't be hard to do tho
03:33.22SwKand setting a global vaiable for CID mapping like that is done in globals sections simple foo=bar
03:33.49goldenoldennow I am going in circles, sorry to be so thick
03:33.58twistedyou're thick?
03:34.04goldenoldenso I want to set a variable for each user, like 200=1234567890
03:34.09sylethick as a dick?
03:34.13twistedsyle!
03:34.21drumkillaMikeJ[Laptop]: !!!
03:34.26sylesup dude
03:34.32twistedsyle, watch it
03:34.34SwKgoldenolden: more like CID2000=Foo Bar <8885551212>
03:34.37MikeJ[Laptop]drumkilla, SwK, twisted!!!
03:34.45SwKMikeJ[Laptop]:
03:34.51twistedMikeJ[Laptop] drumkilla SwK!!!
03:34.59SwKtwisted: oMG!
03:35.01syletwisted: i can't really control my dick you know, appologies, he tends to get the better half of me
03:35.10MikeJ[Laptop]file[laptop]!!!
03:35.12MikeJ[Laptop]hmmmm
03:35.16twistedsyle, take it somewhere else then
03:35.42goldenoldenSwK: we're on the same page, thanks will give it a go :)
03:35.56chris_dMikeJ[Laptop]: Sorry for the delay in responding. I'd like to avoid coding completely. Doing it through dial plans is preferred.
03:36.00MikeJ[Laptop]then give it pickles
03:36.52file[laptop]hi
03:37.10MikeJ[Laptop]chris_d, best hints are, use the groupcount stuff and if you are creating the conf (going from 0 to 1 user) then run a script to drop a call file to the other meetme, then when it drops back to 0, softhangup
03:37.29drumkillayou kids and your laptops
03:37.37MikeJ[Laptop]drumkilla, SwK, twisted, file[laptop] !!!
03:37.41mariogamboai have a question is important the order in that the cards of fxs fxo e1 register in the configuration of zaptel.conf
03:37.47SwKOMG ITS FILE
03:37.52mariogamboa?
03:37.53file[laptop]OMG OMG
03:37.54MikeJ[Laptop]OMG becky
03:38.08drumkillaOOO MMM GGG
03:38.29akrallGuys, I need to debug a nat problem, which tools do you recommend? ngrep? ethereal?
03:38.38MikeJ[Laptop]sip debug
03:38.42MikeJ[Laptop]kram!!
03:38.44MikeJ[Laptop]he
03:38.59MikeJ[Laptop]drumkilla, SwK, twisted, file[laptop], kram!!!
03:39.00mariogamboaomg? object manager group_
03:39.04MikeJ[Laptop]yes
03:39.07SwKOMG IS KRAM!
03:39.12Juggiesomeone go commit 4403 i have another update to do :P
03:39.30MikeJ[Laptop]Juggie, no!
03:39.31twistedah yes, it is kram!
03:39.31SwKkram: how was the trip?
03:39.41twistedJuggie, as soon as people test it and it gets reviewed ;)
03:39.44mariogamboai return in a moment i restart the linux box
03:40.00MikeJ[Laptop]twisted, just say no to Juggie.
03:40.12kramit was cool mostly
03:40.22MikeJ[Laptop]as in chilly?
03:40.27SwKheard you were in London on the wrong day...
03:40.32MikeJ[Laptop]or as in nice
03:40.34Juggiemike :)
03:40.44MikeJ[Laptop]juggie is a drug
03:40.53twistedhe's a drug?
03:41.08MikeJ[Laptop]yes
03:41.11Juggienah, i just know of another odbc problem that needs to be fixed
03:41.18Juggiebut i dont want to combine any more stuff into 4403
03:41.26MikeJ[Laptop]please don't
03:41.36MikeJ[Laptop]it will make it take longer
03:41.40Juggieyah i know
03:42.02MikeJ[Laptop]cuz size does matter in asterisk bugs
03:42.30MikeJ[Laptop]hey file
03:42.53Juggiei combined in that other bug because it was related
03:43.10Juggieand most of what he did had already been implemented so it was just the mwi stuff
03:43.30chris_dMikeJ[Laptop]: Thanks. As long as I know it's possible...
03:43.33chris_d:)
03:43.41litagewhat's the difference between an FXO card and a T1/E1 card?
03:43.48Sedorox0_o
03:43.56litagewhen do you need a T1/E1 vs an FXO card?
03:43.59Juggielitage, analog vs digital?
03:44.05MikeJ[Laptop]litage, analog vs t1/e1
03:44.06Juggie4ports max vs 24-96
03:44.07MikeJ[Laptop]heh
03:44.41MikeJ[Laptop]you can't plug a pots line into a t1 card
03:44.51MikeJ[Laptop]you can't plug a t1 into a fxo card
03:44.54litagewhat do you plug into a T1 card?
03:44.59Juggiea t1
03:45.04MikeJ[Laptop]you can't plug a potato into either
03:45.39MikeJ[Laptop]I need more french fries
03:46.14*** join/#asterisk jeremywhiting (~jeremy@71-37-66-96.slkc.qwest.net)
03:46.22MikeJ[Laptop]hi
03:46.34litageif you can't plug a potato, POTS line, or FXO into a T1 card, what can you plug into a T1 card?
03:46.53chris_dlitage: A T1.
03:46.54MikeJ[Laptop]A T1
03:47.16litageand what connects to the other end of that T1 that's going into the T1 card?
03:47.25litageanother T1 card?
03:47.28MikeJ[Laptop]well, I suppose plug anything you want into it.. but it works best with a t1
03:47.35MikeJ[Laptop]litage, you don't know what a t1 is
03:47.39MikeJ[Laptop]go read
03:47.56MikeJ[Laptop]~docs
03:47.56jbotDocumentation 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
03:48.02litageMikeJ[Laptop]:  already read the wiki, but i still don't
03:48.20MikeJ[Laptop]a t1 is a digital line that supports 24 lines
03:48.24MikeJ[Laptop]in 1
03:48.34MikeJ[Laptop]e1 is basically the same, but 30
03:48.50MikeJ[Laptop]you order it from your phone company
03:48.58litageMikeJ[Laptop]: "supports 24 lines" as in "supports 24 POTS lines"?
03:49.06MikeJ[Laptop]it's not pots
03:49.15MikeJ[Laptop]but, 24 concurrent calls
03:49.15litagewhat sorts of lines are these 24 lines?
03:49.20litageah i see
03:49.25MikeJ[Laptop]phone lines
03:49.29gravemindyou might have a T1 coming in if you have many  lines to your company
03:49.37*** join/#asterisk HellAgony (~HellAgony@200.121.242.95)
03:49.43MikeJ[Laptop]perpendicular lines?
03:49.44litagewould you plug a channel bank into a T1 line?
03:49.49gravemindyou slap a t1 card in your system, point asterisk to it, and you've got the equivelant of up to 24 POTS lines
03:49.51gravemindbut digitalfied
03:50.04gravemindonly if the channel bank is meant for t1
03:50.06litageahhhh sooooo!
03:50.22*** join/#asterisk pdracevich (~bob@210.54.249.228)
03:50.24gravemindie your telco might give you one
03:50.36gravemindso instead of running 10 POTS lines you can have all 10 on one T1
03:50.43litagesounds good
03:50.58litagei know this varies, but how much does an average T1 line cost?
03:50.59pdracevichdoes the stable version of Asterisk run Realtime databases?
03:51.13gravemindhmmm
03:51.23gravemindi've never looked into the price of voice t1, just data
03:51.30gravemindand i've forgotten that even by now
03:51.33*** join/#asterisk menger (~menger@static-88.243.240.220.dsl.comindico.com.au)
03:51.37gravemindit'll vary by line count
03:51.38MikeJ[Laptop]litage, where?
03:51.50gravemindcall up your telco and get a quote
03:51.59litageMikeJ[Laptop]: i'm in australia, so...
03:52.02pdracevichdoes the stable version of Asterisk run Realtime databases??
03:52.20litagepdracevich: we heard you the first time. be patient and wait for a response
03:52.44MikeJ[Laptop]litage, local voice t1 in the us runs from about $500-$1500 depending upon market, distance to co, competion, ect.
03:53.00litageMikeJ[Laptop]: that's per month, right?
03:53.06MikeJ[Laptop]pdracevich, no
03:53.07jskcrIve seen them in the middle of that usually
03:53.14MikeJ[Laptop]litage. yes
03:54.38MikeJ[Laptop]for example, in detroit, I have a bunch in the $600\mo range, in atlanta, I just cancelled one because they wanted $1400 from the traditional bell if I didn't commit to 3 year...
03:54.39*** join/#asterisk mariogamboa (~mariogamb@201.133.229.211)
03:54.44pdracevichok then how do i roll back versions using cvs-head
03:54.49MikeJ[Laptop]now I feed that office did's from my other offices
03:54.56MikeJ[Laptop]pdracevich, roll back?
03:55.00mariogamboayahoo i return in my linux box ;)
03:55.05MikeJ[Laptop]as in check out a specific date?
03:55.20DaminAnyone ever tried "YATE"?
03:55.24pdracevichyeap.
03:55.27dudesT1's run 2500 here in Minnesota ... and they don't have Voice T1's
03:55.34dudesWheaton anyway
03:55.48pdracevichIs anyone having problems, with the latest CVS-HEAD?
03:56.03MikeJ[Laptop]yes
03:56.05gravemindno voice? run gsm to someone who does ;)
03:56.05dudespdracevich - we had issues with it
03:56.14MikeJ[Laptop]there are a bunch of bugs in mantis about it ;)
03:56.17litagehow would you install an asterisk system into an office that has 20 phones connected to an 'old' analog PBX?
03:56.21pdracevichhad???
03:56.33mariogamboawhat kind of troubles have it with the lastes cvs-head
03:56.35MikeJ[Laptop]litage, depends
03:56.42dudeslitage - get a T1 card and a channel bank
03:57.02gravemindlitage: you'd get a channel bank(or a metric butt-ton of FXS cards) and one(or two) t1 cards
03:57.18mariogamboayeah channel back to have 24 fxs or 24 fxo
03:57.22litagedudes: so the current 'old' PBX be incompatible with a T1 card?
03:57.33MikeJ[Laptop]I do legacy pbx connections to asterisk over t1 withno channel bank...
03:57.42dudeslitage - asterisk is a PBX ... why would you connect another PBX to it?
03:57.43MikeJ[Laptop]I don't like to go ananlog if I don't have too
03:57.50litagedudes: true  =)
03:57.58MikeJ[Laptop]dudes, invesment in previous phones?
03:58.01mariogamboafor integration maybe
03:58.04pdracevichIAX Softphone calling into meetme
03:58.07gravemindyou don't need FXO in the channel bank of course, the FXO is provided by the t1
03:58.14mariogamboain my case i use integration of asterisk with definity
03:58.16*** part/#asterisk akrall (user@201.144.58.200)
03:58.24mariogamboalike a media server gateway for ip
03:58.37Inv_Arpim trying * for callcenter use
03:58.42jskcrIm using ser + asterisk
03:58.44mariogamboayeah
03:58.47gravemindInv_Arp: great
03:58.50mariogamboacallcenter
03:58.53mariogamboaivr
03:58.54MikeJ[Laptop]Inv_Arp, congrats
03:58.59MikeJ[Laptop]?
03:59.03Inv_Arplol
03:59.42Inv_ArpThere is a comapny I seen online that does it already for around 200 agents... forgot the name tho...
03:59.49dudesInv_Arp - you're not the first ....
03:59.57*** join/#asterisk mog_home (~mogorman@user-24-236-84-48.knology.net)
04:00.08MikeJ[Laptop]I know of other 100's of agent callcenters...
04:00.11dudesInv_Arp - There are several that use the * PBX for call center applications
04:00.31mariogamboaany here from mexico?
04:00.35mariogamboahe he he
04:00.37pdracevichdudes: You had issues, what sort and what was the fix?
04:00.42Inv_Arpdudes: MikeJ[Laptop] : any revelant links  google not turning up much
04:00.49*** join/#asterisk Math` (~math@modemcable166.240-37-24.mc.videotron.ca)
04:00.55MikeJ[Laptop]no
04:00.56litageare the only ways to connect 20 analog phones to asterisk either?: (1) channel bank + T1 card, (2) many FXS cards
04:01.05mariogamboayep
04:01.14dudespdracevich - We could get voice on sip/iax ... so we went back to a older version and sure as it rains in california it worked.
04:01.17MikeJ[Laptop]litage, no
04:01.22mariogamboa2 fxs
04:01.22dudespdracevich - arg couldn't
04:01.30MikeJ[Laptop]ata's
04:01.37tzafrir_homeI'm trying to set up a sipphone account. Any way to make them generate a call to my phone?
04:01.38mariogamboathe maximum of modules of fxs is 4
04:01.44mariogamboaor ata's
04:01.47litageMikeJ[Laptop]: ah right, or (3) use ATAs
04:01.57mariogamboayep
04:02.04litageso those are the only 3 ways?
04:02.05MikeJ[Laptop]litage, media gateways?
04:02.17MikeJ[Laptop]sting and cans
04:02.24MikeJ[Laptop]potatoes
04:02.45*** join/#asterisk Corndawg_ (whoisit@c-66-176-249-51.hsd1.fl.comcast.net)
04:02.57MikeJ[Laptop]hi
04:03.08MikeJ[Laptop]my wife does not really care about russia.
04:03.14MikeJ[Laptop]so she says...
04:03.15gravemindMikeJ[Laptop]: sting and cans? hornet-based communications were banned 3 years ago.
04:03.19litagemedia gateway ?= voip gateway
04:03.24mariogamboayep
04:03.30mariogamboavoip gateway
04:03.36mariogamboa:)
04:03.41MikeJ[Laptop]echocancel=yes
04:03.43Qwell?=  ?  Thats a new one
04:03.44gravemindif you're hooking into POTS phones, 1 and 2 are your options
04:03.55gravemindQwell: its a query, not a statement
04:04.01Qwell!=, ==, <=, >= (and even =>), but never ?=
04:04.09MikeJ[Laptop]if you are looking into monkeys, spider is my suggestion
04:04.13mariogamboain e1 for cancel echo i need to put the echocancel=yes ?
04:04.14gravemindfoo ?= bar - is foo == bar?
04:04.21Qwelloh, and the ever popular |= and ~=
04:04.28gravemindpff
04:04.33gravemindyou're obviously forgetting //=
04:04.53QwellIs that a comparison operator?
04:04.56gravemindalthough //= is an assignment technically
04:05.02gravemindso let's stick to //
04:05.12MikeJ[Laptop]c++ comment?
04:05.18gravemindno, perl6 garbage :P
04:05.27gravemindlike || but only false for undefined i think
04:05.29MikeJ[Laptop]c99 comment really
04:05.40gravemind$foo // $bar would return $bar if $foo is undefined, but not if $foo is 0
04:05.55Qwellhmm
04:06.01gravemindwhere as $foo || $bar would return $bar if foo is anything evaluating to false
04:06.01QwellWhat would it return otherwise?
04:06.13gravemindeh?
04:06.15Qwellnevermind
04:06.23gravemind$foo // $bar returns $foo if $foo is anything defined, $bar otherwise
04:06.27Qwellyeah
04:06.35MikeJ[Laptop]cmon.. spidermonkey... work w/ me people
04:06.37*** join/#asterisk jeffik (~Jeff@toronto-HSE-ppp3985991.sympatico.ca)
04:06.46tzafrir_homeres_bf?
04:06.51MikeJ[Laptop]bf?
04:07.01gravemindbest friend
04:07.08MikeJ[Laptop]awww
04:07.16QwellI was thinking boyfriend...
04:07.17tzafrir_home~res_bf
04:07.18MikeJ[Laptop]res_js uses spider monkey....
04:07.19gravemindhandy module for hooking your best friend into the channel :P
04:07.29MikeJ[Laptop]~pbxfreeware
04:07.37MikeJ[Laptop]hmmm
04:07.43MikeJ[Laptop]www.pbxfreeware.com..
04:07.51MikeJ[Laptop]cool toy if you like js
04:07.55litagedoes a T1 card only hook up to a T1 line or a channel bank, or are there other devices that it can connect to?
04:08.07gravemindlitage: anything with a T1 card in it
04:08.20gravemindwhich is pretty much limited to computers and channel banks
04:08.26gravemindor routers.
04:08.27MikeJ[Laptop]battery dieing.. later all
04:08.31gravemind(for data t1)
04:08.35litagethanks for you rhelp MikeJ[Laptop]
04:08.46litagegravemind: gotcha. thanks for that
04:08.52mariogamboarhino channel bank
04:09.03litagegravemind: besides the obvious, what's the difference between a data and a voice T1?
04:09.12mariogamboapbxfreeware down ;(
04:09.27gravemindlitage: not really sure tbh
04:09.39harryvvthere needs to be more chatter between polycom integration and asterisk.
04:09.43tzafrir_homecan anybody call me to a sipphone number?
04:09.59tzafrir_home(be it an echo test)
04:10.17harryvvtzafrir_home you want a address to call like mine?
04:10.41tzafrir_homeharryvv, I need to know that incoming calls are handled
04:10.41gravemindtzafrir_home: i think so
04:10.50gravemindtzafrir_home: lemme check my dialplan to make sure i set up the prefix
04:11.18gravemindyeah, i can do that
04:11.18tzafrir_homemy number is 1747-6065631
04:12.04gravemindhrm
04:12.07gravemindmy phone didn't like that
04:12.16gravemindoh, low battery :P
04:13.30gravemindi get a busy but it oculd just be my pap2 being a wanker
04:13.56goldenoldenthanks SwK and MikeJ!
04:15.04CyberSwordyehaaa i set my first PBX !!!!!
04:15.05CyberSwordi am happy
04:15.28gravemindyou should be 8]
04:15.47litagehahah on channelbanks.com their link-buttons have no text in them  =P
04:16.31*** join/#asterisk MrBelvedr (~tt@ip68-227-218-250.dc.dc.cox.net)
04:16.41harryvvyea crapy huu
04:16.52twistedYOYO
04:16.54twistedhttp://www.weeklyworldnews.com/features/chamber/61606
04:17.09SwKhah
04:17.19SwKnew Tripping the Rift episodes coming!
04:17.44gravemindwell this is indeed odd
04:18.03gravemindtzafrir_home: i can't get the bastard to dial out
04:18.11gravemindit's probably the PAP2's fault
04:18.24mariogamboawhat sip phone recomend for linux
04:18.26mariogamboa?
04:18.38mariogamboasip phone soft :)
04:18.46gravemindphonegaim is supposed to be good, but i couldn't get it to work
04:18.50gravemindso i suffer with kphone
04:19.51*** join/#asterisk stkn (nobody@stkn-active-pdpc.developer.gentoo) [NETSPLIT VICTIM]
04:19.51*** join/#asterisk jtodd (~jtodd@garthim.fox-den.com) [NETSPLIT VICTIM]
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04:19.51*** join/#asterisk dwmw2_gone (~dwmw2@baythorne.infradead.org)
04:20.12harryvv12 more days before cisco takes over sipura
04:20.14sivanais something up with Nufone?
04:20.34litagegravemind: i haven't tried phonegaim, but kphone worked most of the time for me, though disconnecting would hang it
04:21.07litagestupid question, but i just want to make sure:  if you have a 2-port T1 card, 1 port can go into the T1 line and the other port into a channel bank, right?
04:21.19gravemindsure
04:21.46litageand asterisk automagically figures out which port does what?
04:21.55litages/does/is connected to/
04:22.27gravemindyou tell asterisk what to do with the lines on each port
04:22.35litageawesome possum
04:22.46litagethanks for your help, i really appreciate it guys and gals
04:22.51mariogamboayep
04:22.54mariogamboalitage
04:23.02litagewhat's up mariogamboa
04:23.18*** join/#asterisk Barmal (~info@c-24-30-75-206.hsd1.ga.comcast.net)
04:24.35BarmalWARNING: Chan_sip.c:8188 handle_response: Forbidden - wrong password on authentication for INVITE...  Is this a password issue??
04:26.51gravemindhrm...
04:29.08tzafrir_homeI haven't tried the new kphone. There's also linphone (yuck as well)
04:29.16tzafrir_homeI use iaxcomm nowadays
04:29.53gravemindtzafrir_home: what's the sipphone official echo number?
04:30.05gravemindor the one that reads your number back
04:30.55tzafrir_homegravemind, http://www.sipphone.com/numbers/ . But I see no options there to as for a call-back
04:31.12tzafrir_homeFWD has this option from the web interface and it's really handy
04:31.41gravemind<PROTECTED>
04:31.41gravemind<PROTECTED>
04:31.41gravemind<PROTECTED>
04:31.45gravemindcould be my fault
04:32.54dudesgravemind - take the 1 out maybe?
04:33.11dudesusally it means you need a 1 or don't
04:33.44*** join/#asterisk jake1932 (~jake1932@pool-70-16-132-177.phil.east.verizon.net)
04:35.53gravemindwait...
04:36.00tzafrir_homegravemind, could you please try again? there was much unrelated junk on my console
04:36.00gravemindit's dialing out the wrong proxy.
04:36.03gravemindduh.
04:37.04gravemind<PROTECTED>
04:37.04gravemind<PROTECTED>
04:37.04gravemind<PROTECTED>
04:37.04gravemind<PROTECTED>
04:40.06gravemindill try the lopback numbers
04:40.13gravemindloopback rather
04:40.48tzafrir_homeand sadly, I gravemind thanks. Does this mean they expect me to use stun or similar if I'm behind nat?
04:41.15graveminddunno
04:41.20gravemindi'd assume so
04:41.43tzafrir_homeAccording to the web interface I'm registered:  Public IP address   212.179.75.202:50109 (expires in 107 seconds)
04:42.12sivanaanyone else having issues with Nufone registration?
04:42.20gravemindi'm getting that 488 error doing ECHO too
04:42.31file[laptop]bad codec, please try again
04:42.35tzafrir_homeI managed to use echo, actually
04:42.48gravemindyou're probably right
04:42.58gravemindi've used my sipphone forwarding before, but it was thru kiax
04:43.03gravemindmy pap2 is probably locked on ulaw
04:43.12gravemindwhat does sipphone allow?
04:44.04Qwellmost softphones use at least 3-4 codecs
04:44.30gravemindbut apparently sipphone's PBX doesn't like whatever my pap2 is using
04:46.03tzafrir_homeI tried sending a call from a Local/ channel
04:46.13gravemindmaybe when i called you it dropped back to voicemail
04:46.19gravemindand then it rejected the codec
04:46.21tzafrir_homeand got the same results. So this is not a simple sip issue.
04:46.26gravemindi heard a bit of voice though
04:46.44tzafrir_homegot to go no, Thanks all
04:46.58gravemindlater
04:53.22*** join/#asterisk jake1932 (~jake1932@pool-70-16-132-177.phil.east.verizon.net)
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04:57.58jfonsecausaSome here?
04:58.29gravemindmoo
05:00.34*** join/#asterisk wt (~wt@be88a5fc72edbdaf.session.tor)
05:00.46wtanyone here?
05:00.50Barmalyep
05:01.25gravemindmoo
05:01.41wtanyone have recommendations for voip providers?
05:01.48Qwellwt: nufone, asterlink
05:01.49Barmalwhere us?
05:02.07wtnufone?
05:02.13wtI think I have heard of them
05:02.16Inv_Arpwt: voicepulse,iax.cc host of others
05:02.33wthere's the issue, is end-to-end QOS needed?
05:02.39Qwellneeded?  no
05:02.56Barmalsellvoip
05:03.03Barmalsellvoip.net
05:03.14Qwellsounds pretty shady
05:03.37wtand to get the end-to-end QOS, do I need to use a provider that also provides my internet?
05:03.48wtmy understanding is that I would
05:05.19BarmalQwell: how much asterlink charges per minute for SIP termination?
05:05.29Qwelllike 2c I think
05:06.32wtmaybe it would be better to go with a partial T1
05:07.04wtwhat about a place like bandwidth.com?
05:08.16*** join/#asterisk litage (~nick@ws01.5749.dsl.winshop.com.au)
05:08.58Barmalinternet resellers....
05:09.54*** join/#asterisk ZeeLax (~zeelax@rxgw.network.kz)
05:10.40wtanyone heard of nuvox?
05:10.58wtQwell, should I get end-to-end QOS though
05:11.09Corydon76-homeHah, heard of nuvox?
05:11.11wtQwell, I am worried that it would suck otherwise
05:11.18*** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk)
05:11.23Corydon76-homeThey're only one of the big CLECs in the US
05:11.52Qwellwt: depends on your ISP...if you've got a good connection, QoS isn't really needed.
05:12.10wtcorydon76-home: do they work well with asterisk?
05:12.37wtQwell: do you really believe that, like in your heart of hearts?
05:12.39Corydon76-homeUm, any provider with PRI works well with Asterisk
05:12.50wtQwell, It just seems kinda like gambling
05:13.01Corydon76-homeHaven't found a provider yet that screwed up PRI
05:13.07Qwellwt: I'll say it again - if you've got a good connection...
05:13.07wtCorydon76-home: would  you suggest PRI over SIP peering?
05:13.14wtor IAX peering, even
05:13.33Corydon76-homeUh, that's like IAX over SIP... makes no sense...
05:14.12wtQwell: would you recommend PRI or peering?
05:14.33wtCorydon76-home: I meant would you recommend PRI or peering?
05:14.48Corydon76-homeDepends on what you're trying to do
05:15.14Corydon76-homeI would never recommend SIP only for a business
05:15.23Corydon76-homebut for a home, it might be fine
05:15.46wtcorydon76-home: now we are getting somewhere :-)
05:16.10Corydon76-homeThe reliability isn't guaranteed... it's best effort only...
05:16.36wtcorydon76-home: that's why I am skeptical of peering without a ITSP
05:16.49Corydon76-homeand if there's a new Windows virus, trojan or worm out, you aren't going to get reliable traffic, no matter who the carrier is
05:16.56wtunless I can get QOS from me to the provider, I screw myself
05:17.19*** join/#asterisk brookshire (~matt@esbrooks3.traveller.com)
05:17.21QwellCorydon76-home: QoS or not, correct?
05:17.38Corydon76-homebut it's fine 99% of the time
05:17.39wtCorydon76-home: I know that asterisk can do PRI
05:17.53wtcan it do partial T1 on voice and partial T1 on data?
05:17.58Corydon76-homeYes
05:18.00brookshireyup
05:18.14wtand if so, does the T1 have to come right into the asterisk box?
05:18.35brookshireno
05:18.37Corydon76-homeAlthough providers like nuvox would prefer to break out the data on their own equipment, rather than letting you do it yourself
05:19.05Corydon76-homeWe actually do the breakout ourselves... but we have USLEC
05:19.19brookshireyou can use tdm over ethernet to pull the voice channels from the t1
05:19.22wtcorydon76-home: is USLEC good?
05:19.41Corydon76-homeYeah, they're fine, as well...
05:19.58Corydon76-homeOnce you're working with PRI, it doesn't matter who the provider is, really
05:20.07loudok so far, 7.5 brings no problems .. Useragent    : Cisco-CP7960G/7.5
05:20.24file[laptop]ooh user agent change?
05:20.39loudjust upgraded to 7.5
05:21.15wtcrap! echo cancellation doubles the price of the card?
05:21.23file[laptop]yes, yes it does
05:21.35Corydon76-homeAny of the clecs tend to be easier to work with than the RBOCs or companies like ATT...
05:21.36wtis it important?
05:21.38brookshirethere is software echocan :)
05:21.44*** join/#asterisk Inv_Arp (junya@adsl-156-139-229.mia.bellsouth.net)
05:21.55brookshirei think it sounds better with the hardware echo cancellation
05:22.15Corydon76-homebrookshire: don't forget full disclosure...
05:22.22brookshire??
05:22.22wtif I wanted to have say 10 extensions with about 4 lines, would a 2.8 GHz box be enough?
05:22.29Corydon76-homebrookshire: you work for Digium
05:22.34brookshireit's not a requirement
05:22.46brookshirecory: i'm just being truthfull though
05:22.53wtcorydon: I thought you did also
05:23.04Corydon76-homeNo, I work for a reseller
05:23.14wtcory: word
05:23.28brookshirewhich means you can probably get a better price from cory than me ;)
05:24.06Corydon76-homeand my ears aren't good enough to tell the difference between 64k and 192k MP3s, so let that be an advance warning about the quality of my hearing...
05:24.11wtanyone here like Sangoma?
05:24.23Corydon76-homebrookshire: nah, we charge the same as Digium
05:24.30wtcory: i met you a while back
05:24.41brookshirehaha.. ok
05:24.46*** join/#asterisk DA-MAN (~DA-MAN@66-215-92-29.pas-eres.charterpipeline.net)
05:25.14wtcory, brookshire: what makes the digium better than the sangoma?
05:25.36Corydon76-homeDigium puts a lot more money into Asterisk than Sangoma
05:25.42wtI have to admit the sangoma appears, on the surface, to be a better card
05:25.49QwellCorydon76-home: Mind if I ask what reseller?
05:25.54Corydon76-homeand Sangoma is actively trying to put Digium out of the hardware business
05:25.57brookshiresangoma is better with data.. digium is better with voice ;)
05:25.58wtcory: so it's more supporting the habit?
05:26.06wt;-)
05:26.12Corydon76-homeQwell: VCCH
05:26.29Corydon76-homewt: it's supporting more development
05:26.36wtright
05:26.37DA-MANwow, is sangoma asterisk compatible?
05:26.49brookshireif you like echo ;)
05:27.23DA-MANi do
05:27.59Corydon76-homeI have no experience with the Sangoma cards...
05:28.18Corydon76-homeSo I can't say anything about their performance
05:28.20tclarkCorydon76-home: is that an admisison you dont think digium can compete and make better gear :) ?
05:28.39Corydon76-hometclark: no, it isn't
05:28.51tclarkwhy the heck da mean ?
05:29.08Corydon76-homeI have no insider knowledge about the quality of Digium compared to Sangoma
05:29.19DA-MANi'm just kidding. Hell I got rid of my plantronics bluetooth headset because of bad echo
05:29.40tclarkwhy do you say 'Sangoma is actively trying to put Digium out of the hardware business' ?
05:30.02Corydon76-homeI've heard it said
05:30.16wtbrookshire: does digium have a good return policy?
05:30.43brookshirewt: our stuff is great, why on earth would you ever want to return it? ;)
05:30.44wtbrookshire: say I get one of each and try them and the Sangoma rules the day, would digium refund me?
05:30.57tclarkbizarre stmt the only ppl that will put Digium out of h/w biz is Digium ..
05:31.02Qwellwt: Thats just being a bad customer
05:31.25wtbrookshire: I have to admit, I am worried about the different 5.5 and 3 volt cards
05:31.32wtdoes that make it better
05:31.42Qwell5v and 3.3v
05:31.45gravemindthe cards fit into different slots
05:31.49brookshirewt: hardly any one uses 5v cards anymore
05:31.51gravemindso its not like you'll fry anything
05:31.58brookshiremainly only old motherboards
05:32.02gravemind(i think)
05:32.14Qwellgravemind: I think you're right.  it's slotted differently
05:32.17wtQwell: everything I read on static web pages indicates that the hardware handling of certain events makes the Sangoma far superior, but I like Digium
05:32.21gravemindlike one is backwards from the other
05:32.34Corydon76-homeNo, you won't fry anything.  The voltages come off completely different pins on the cards
05:32.54brookshirethe cards won't even fit in the wrong slots
05:33.07brookshirei mean.. they will.. but you would have to force it
05:33.10Barmalwhat about the prices on Sangoma? Are they better?
05:34.15wtQwell: i don't think its being a bad customer to want to get the best device
05:34.27brookshireabout the same
05:34.50Qwellwt: Thats being a horrible customer.  You're turning a new item into a used item, and expecting to pay nothing
05:35.07Corydon76-homewt: the Digium cards come with phone support for Asterisk...
05:35.15sylewhere can i get a damn channel bank in canada lol
05:35.22Corydon76-homeby the same people who wrote Asterisk...
05:35.22wtQwell: then point me to a place where I can get a good comparison
05:35.30Corydon76-homewt: Sangoma can't promise that
05:35.31wtthe voip-info.org wiki says sangoma is better
05:35.36brookshiredigium will not support any system with a sangoma card
05:35.37Barmaldoesnt look like you can buy sangoma direct from website...
05:35.44brookshiredigium will however support anything else
05:35.58Qwellbrookshire: even a clone x100p? ;]
05:36.00brookshireyup
05:36.02brookshire:D
05:36.07Qwellreally?
05:36.07sylebrookshire: well be damned if digium wants the world to pay higher prices for their own pocket book
05:36.18brookshireyeah.. they do.. not free of course
05:36.22Qwellsupported as in "try removing it from your system, and running this command."
05:36.24Qwellahh, I see
05:36.25wtbrookshire: honestly, I like the fact that digium has a better setup for the drivers, and I will prolly go with them
05:36.51brookshiresyle: are you serious, have you priced anything else?
05:36.51wtbrookshire: does digium ever plan to attempt to integrate the zaptel driver with the mainline kernel?
05:36.52Corydon76-homeActually, I think Digium set the pricepoint and Sangoma priced theirs to be competitive with Digium
05:36.54brookshiredigium is the cheapest
05:36.55Qwellwt: well, Digium wrote the drivers for its card for its pbx...
05:37.01brookshireas far as hardware goes
05:37.06sylebrookshire: try T1 cards
05:37.08brookshiresangoma is pretty much with us
05:37.16sylebrookshire: 15 bucks vs what 300
05:37.24brookshirebut jeeze.. try and buy a t1 avaya system
05:37.29brookshireor cisco
05:37.39brookshireyou're looking at paying $10,000 more
05:37.48QwellI groaned out loud on Friday...Avaya van was outside my work
05:38.10wtQwell: the sangoma drivers are open source also, and they use the zaptel drivers
05:38.13wasimand it still won't do half of what * does ...
05:38.22brookshirepeople who complain that our prices are too hight annoy me
05:38.26brookshirewe made it cheaper
05:38.29Qwellwt: Did Sangoma write Zaptel?
05:38.30brookshirewhat more do you want?
05:38.30wtthey just aren't in the zaptel driver source package
05:38.51Corydon76-homewt: and they won't be, either
05:38.57wtQwell: what does that matter if they implement an indentical interface?
05:38.57syleis it that hard to make a cloan of a digium card and release the same driver
05:39.08brookshirenope... digium wrote zaptel.. sangoma has their own zaptel drive that is based on digium's zaptel work
05:39.35brookshiresyle: that would be illegal
05:39.35sylewell as i hear it google is paying students to develop asterisk over the summer
05:39.45brookshire;)
05:39.46wtbrookshire: right, it presents the same userspace interface
05:39.59Corydon76-homesyle: yes, it's a lot of work... most of the challenge is in the card firmware, which is not readable from the card
05:40.08wasimand the firmware is not GPL
05:40.09wtsyle: what projects were accepted for the summer of code?
05:40.26brookshirewt: i know of a few off the top of my head
05:40.33brookshireweb-based gui, and srtp
05:40.44Qwellsrtp?  sexy
05:40.47wtcory: not to mention that they do some stuff in hardware the digium does in the firmware
05:41.07brookshirewt: ??
05:41.16brookshirefirmware is hardware
05:41.16QwellI like firmware features. :D
05:41.17brookshiresilly
05:41.19Corydon76-homewt: how do you know that, if just a few minutes ago, you weren't sure which was better?
05:41.26sylewt: http://www.iaxprovider.net/modules.php?op=modload&name=News&file=article&sid=93
05:41.42wtI had researched them before...I still don't know which is better
05:41.59wtlike I said, I will prolly end up with the digium because of driver support though
05:41.59syleactually the projects were approved, i lost the link to what the 4 were
05:42.21sylei think 1 was the codec support
05:42.38BarmalI think if digium will have about the same prices as Sangoma people wll always choose digium
05:42.42syleanother was encryption based
05:42.46wtdid the SIP over TCP thing get accepted?
05:42.56syleyeah that was one
05:43.29brookshirethe only thing sangoma does in the hardware is hdlc
05:43.29brookshirewhich really gives you no advantage from doing in software
05:44.00wasimbrookshire: what about echocan?
05:44.19brookshirewhat about it?
05:44.33wasimdoes sangoma do it in hw?
05:44.36brookshireno
05:44.38brookshirenot yet
05:44.41Corydon76-homeIn fact, there's a distinct disadvantage about doing things in hardware... if you find a bug, you get to send it back to the OEM to have it replaced instead of simply getting a software upgrade...
05:44.52brookshirethey do not have a product out that does hardware echo cancellation at this time
05:45.04Barmalhow much is TDM400p theese days?
05:45.21brookshireBarmal: depends on what you get ;)
05:45.29Barmal?
05:45.36brookshireit's modular
05:45.47brookshireso.. it starts at around $130
05:45.53Corydon76-homeTDM400P is the underlying card... there are 4 slots and 2 different modules...
05:45.58wasimBarmal: the tdm400p is the base card
05:46.02brookshireup to about $300 i would guess
05:46.11wtyou are right about the price though, sangoma is only nominally cheaper
05:46.23Corydon76-homebrookshire: somewhere closer to $350 if you get all FXO modules
05:46.37brookshirecory: probably
05:46.38brookshiremy bad
05:46.39brookshire:D
05:46.48wtCorydon: the sangoma is field upgradable, apparently it is FPGA based
05:46.50Barmalok so its the same price as it was from beggining
05:47.00Corydon76-homeHowever, there's a major problem with the TDM drivers currently in regards to fax or modem connection...
05:47.01Barmaland I heard digium lowered prices
05:47.11wtbrookshire: is the digium upgradable like that
05:47.20brookshirewt: digium card w/ generation 2 firmware is field upgradable
05:47.40wtbrookshire: cool that alleviates that fear
05:48.53brookshirewt: http://www.digium.com/index.php?menu=press/pr_2gen_firm
05:49.21Barmalyou can't buy directly from digium anymore?
05:49.59wtg.729 compression is 8kbps?
05:50.04brookshireBarmal: you can, but prices are generally cheaper from a distributor or reseller
05:50.11wasimwt: yes
05:50.19sylewt: yeah but it sucks , you can barely hear the caller, use gsm instead
05:50.49Barmalbrookshire: I remember in old website you could see the prices there, not anymore
05:50.50Corydon76-homeGSM isn't as tight as G.729
05:50.53wtsyle: you have to pay for g.729 anyway
05:51.00syleno you don;t
05:51.08brookshirelegally.. you do
05:51.08Corydon76-homesyle: yes you do
05:51.13syleonly if you plan on distributing asterisk under GPL
05:51.18wtsyle: legally speaking
05:51.18Qwellbrookshire: I don't quite get that press release...
05:51.19syleso no
05:51.23Corydon76-homesyle: you do anyway
05:51.26Qwellsomething doesn't make sense to me
05:51.29Corydon76-homesyle: it's patent law
05:51.55wtyou have to license the patents to use G.729
05:51.57Corydon76-homeDon't phuck with the patent lawyers... they don't like it when people try to evade their licensing fees
05:52.03Qwellnevermind
05:52.03*** join/#asterisk bzbw (~wlwzhang@68-190-232-229.mpk-mres.charterpipeline.net)
05:52.08wtdoesn't matter what copyright covers the cod
05:52.11sylewell they should have come up witha  good codec to patent lol
05:52.13wte
05:52.29bzbwanyone know what is the following error from the console:
05:52.30bzbwJul 18 23:51:51 ERROR[3217]: utils.c:509 tvfix: warning negative timestamp -267672.-941250
05:52.51sylepeople get wrong impression: i should pay for g729 because it costs money so it should be good!...try it for a week and kick themselves in ass for not using another codec
05:52.55wtbrookshire: that press release sounds like the new digum cards are far superior to the prior ones
05:53.12wtbrookshire: is the current 4 port T1 card second generation?
05:53.22Corydon76-homeWe use G.729 for most off-network traffic...
05:53.25wtbrookshire: or is that only the echo canceling ones?
05:53.26brookshirewt: it's been out for about a month
05:53.31brookshireall cards
05:53.37sylecorydon: then i;m glad i;m not on your network :)
05:53.43Corydon76-homeIt's the only compressed codec which is fairly universal
05:53.48Qwellkram: y0
05:54.04kramsup qwell!
05:54.06Corydon76-homeThe only other two universal codecs tend to be ulaw and alaw, which aren't compressed
05:54.21sylewhats wrong with gsm?
05:54.24Corydon76-homeWell, other than G.723.1 which isn't an option anyway
05:54.25Qwellkram: Idiots left another package at the front office. ;]
05:54.27Barmalwhat codeces is enough for fax?
05:54.34Corydon76-homesyle: most commercial providers don't support GSM
05:54.41wasimBarmal: any of the inlaws
05:54.46sylethats bullshit corydon and you know it
05:54.49QwellWhen will these shipping companies learn?
05:54.52bzbwemm, looks like none knows what the error "Jul 18 23:51:51 ERROR[3217]: utils.c:509 tvfix: warning negative timestamp -267672.-941250
05:55.02Corydon76-homesyle: I've tried.  Trust me, I've tried.
05:55.12sylecompany i;m using now does gsm
05:55.13wtbrookshire: I was making sure that the cards I was looking at were under that umbrella
05:55.16Corydon76-homesyle: they support ulaw, alaw, G.729, G.723.1
05:55.17syleso does one i am switching to
05:55.23syleso i have no idea what your talking about
05:55.31Corydon76-homesyle: on a Quintum?
05:55.52brookshirewt: if you buy from a reseller, just inquire if they are gen2
05:56.02bzbwkram: give me a hint on this error, thx a lot!
05:56.03bzbwJul 18 23:51:51 ERROR[3217]: utils.c:509 tvfix: warning negative timestamp -267672.-941250
05:56.06Corydon76-homeor god-knows-whatever-else-we-have-to-connect-to
05:56.12brookshireour resellers should all have them by now
05:56.17brookshirethey have been out for a while
05:56.22Corydon76-homeTry getting GSM codec from ATT
05:56.27syleisn;t quintum a asterisk competor?
05:56.47Corydon76-homeYes, but you have to interface with the competition...
05:57.07Corydon76-homeYou don't have the luxury of only negotiating with other Asterisk providers
05:57.30wasimnot yet, not till our well laid plans are more implemented :)
05:57.54Corydon76-homeYou'll get thrown out on your ass if a customer learns you can't interface with someone they're already doing business with
05:58.18Corydon76-homeInstead, they'll sack you and get a Quintum
05:58.20Inv_ArpBroadoice still only support ulaw?
05:58.33Inv_Arpbroadvoice rather
05:58.42brookshireulaw is awesome though :)
05:59.01SwKlpc10 is better then ulaw
05:59.02sylevoipjet is not bad
05:59.06SwK(or something)
05:59.10sylethey do about 4 diff codecs
05:59.25sylebut i had a problem with them today, after 5 rings they cut my call off
05:59.30syleso wasn;t impressed
05:59.45wtWell, I guess if I go PRI, I won't have to worry about codecs outside my network for now
05:59.54Corydon76-homeI do, enjoy, however, getting tasked with implementing a new provider and learning they use Asterisk too, so we quickly switch the protocol from SIP to IAX2...
05:59.55brookshirelpc10 sucks
05:59.55brookshirelol
06:00.03SwKhah
06:00.04brookshireulaw has the best quality
06:00.17Corydon76-homebrookshire: and the worst bandwidth
06:00.17sylewt: if your not doing SIP or IAX and loosing more money on your PRI sure hehe
06:00.27SwKbrookshire: lpc10 sounds great (if you like sounding like MrRoboto)
06:00.36brookshirehehehe
06:00.51wtsyle: first step of implementation
06:00.59wtsyle: one bleeding incoming line
06:01.13SwKpersonally I wish iLBC was more widely implemented
06:01.15wtI can't do both on one line
06:01.16syledepends what your doing wt
06:01.16tzafrirbrookshire, are there any phones that support lpc10? I wonder if it could be useful if you have a dial-up line or something similar
06:01.23SwKok on the bandwidth and sound quality
06:01.44brookshiretza: i have no idea about that
06:01.49wttzafrir: is speex any good as a codec?
06:01.53syleif your doing a telepersonals line a PRI is great, if your adding customers from every diff area code in the world , hope you have millions for all those PRI's
06:02.04SwKjust dont bother trying ulaw over dialup it doesnt work very well
06:02.09wtthat was meant to be a question to everyone
06:02.20wtspeex sounds good with prerecorded stuff
06:02.24tzafrirwt, it eats CPU. I have no idea about quality. It should have some nice features. But I'm no codec expert
06:02.33brookshireswk: you did give josh his stuff right?
06:02.40SwKbrookshire: yeah he got it
06:02.47brookshirecool..
06:03.13syleoww brother is here, time to use my xbox for super mario bros 3 hehe
06:03.14syle18r
06:03.29Barmaldammm 2am time to sleep :(
06:08.17*** join/#asterisk fenlander (~neils@82.152.81.57)
06:09.54bzbwdamn, this error just puzzles me, kram, help!!!
06:10.16bzbwJul 18 23:51:51 ERROR[3217]: utils.c:509 tvfix: warning negative timestamp -267673.-31250
06:10.38bzbwwhenever there is a call, this error keep filling up the log.
06:12.38bzbwi'm using CentOS, thought someone had the same error.
06:16.34*** join/#asterisk JooZoo (~chatzilla@kultapossu.yok.utu.fi)
06:18.44*** join/#asterisk clive- (~pirch@rndf-146-34-67.telkomadsl.co.za)
06:21.45bzbwanyone out can help??
06:22.34bzbwJul 19 00:22:30 ERROR[3217]: utils.c:509 tvfix: warning negative timestamp -204621.-90875
06:22.41tzafrirbzbw, don't spam
06:22.47tzafrir~pastebin
06:22.48jbot[pastebin] a place to paste your stuff without flooding the channel - try http://pastebin.ca
06:23.11tzafriranyway, do you notice anything wrong besides that message?
06:23.13bzbwI'm despeerate
06:24.53*** part/#asterisk CyberSword (~Cyberswor@cablea0mle.cybercable.net.mx)
06:25.05tzafrirbzbw, have you looked at utils.c? any idea what's the problem there?
06:25.25bzbwtzafrir: yeap, but no clue.
06:25.36tzafrirwhat is tvfix?
06:25.55bzbwI don't know, this seems new to me.
06:26.47bzbwbut I believe it's related to RTP, only when RTP audio gets generated when it shown
06:27.28tzafrirlong delays? clock not synced?
06:28.12bzbwbeats me:(.  this is pretty normal call, close to LAN environment
06:28.20tzafriranyway, my copy doesn't have it, so it's probably something new in HEAD. Which reminds me you have not stated your version of *
06:28.42bzbwthe most latest one I guess, I cvs out today
06:29.33bzbwAsterisk CVS-HEAD built by root@localhost.localdomain on a i686 running Linux on 2005-07-19 00:09:07 UTC
06:29.52bzbwusing CentOS 3.3
06:30.18*** join/#asterisk sig- (~sig@gnook.org)
06:30.31sig-;>
06:31.23kaldemarhi all, i'm trying to compile libpri, but: http://pastebin.ca/18024. any hints?
06:32.17bzbwsomehow I think it might be related to clock reference, i have a clone x100p, maybe the clock is not right.
06:36.19sig-kaldemar: http://lists.digium.com/pipermail/asterisk-users/2003-August/017452.html
06:36.40sig-cvs update when?
06:38.16kaldemarfive days ago, i think.
06:38.56bzbwok, I give up. maybe someone will figure it tomorrow.
06:41.56sig-kaldemar: try cvs now and compile..
06:43.51kaldemarsig-: took the current CVS version, still getting the same error.
06:44.32*** part/#asterisk bzbw (~wlwzhang@68-190-232-229.mpk-mres.charterpipeline.net)
06:46.44*** join/#asterisk limbique (~limbique@nl-ifw-oss.orcagroup.com)
06:46.50limbiquegood morning
06:48.58tzafriralways blaming the poor x100p, and not even trying ztspeed (or is it zttest?)
06:50.54*** join/#asterisk The_Ball (~alex@static-227.35.240.220.dsl.comindico.com.au)
06:51.26The_Ballis the country code for the us(hawai) +01 ?
06:52.23*** join/#asterisk srt (~nobody@gw0-cgn.reucon.net)
06:56.17Qwell+1
06:57.34*** join/#asterisk JerJer[mobile] (~jj@pcp0010833999pcs.detrtc01.mi.comcast.net)
06:57.40JerJer[mobile]mooo
06:58.30sig-kaldemar: dunno in this way,try or old or wait for next cvs..
07:00.15kaldemarsig-: i'll try to figure something out. thanks for your help.
07:01.34*** join/#asterisk kore (kore@mindwipe.org)
07:03.55*** join/#asterisk pif (ldm@zenon.apartia.fr)
07:05.22JerJer[mobile]people, don't type at the same time, I cannot keep
07:05.23JerJer[mobile]up
07:07.39*** join/#asterisk ZeeLax (~zeelax@rxgw.network.kz)
07:12.20heath__i can't get dtmf to work thru this path sipura->asterisk->iax trunk->asterisk
07:12.26heath__tried inband and that didn't help
07:12.35*** join/#asterisk CyberSword (~Cyberswor@cablea0mle.cybercable.net.mx)
07:13.30DA-MANi use rfc style dtmf on my sipura
07:13.31DA-MANworks fine
07:14.10CyberSwordhi i have a little problem, first, i add extensions, and 2 clients are connected ( in my home network ) same network that asterisk server, and u can log in and communicate each other, but i connect using adsl conection and i try to log in in to my server and its okey , but i cant talk whit the other guy its only say connected
07:14.15CyberSwordi am using a xlite client
07:14.40DA-MANdtmfmode=rfc2833
07:15.23DA-MANCyberSword, are you behind nat?
07:15.48DA-MANis the box running asterisk providing nat
07:17.27CyberSwordyes
07:17.54CyberSwordwell i dont know if is nat but i have a router
07:17.55*** join/#asterisk tuxinator_linuxM (~spabin@ip68-109-146-168.ph.ph.cox.net)
07:18.31CyberSwordbut i set DMZ rlz on the router for my server so the server have all the ports free
07:19.22*** join/#asterisk tafazzi (~Dario@81-208-60-200.fastres.net)
07:19.23JerJer[mobile]dmz is not necessary or desired
07:19.36JerJer[mobile]heath__:  does the first asterisk in the chain see DTMF?
07:19.38tzafrirthat is: that -v implies not daemonizing
07:19.56JerJer[mobile]run safe_asterisk
07:20.11JerJer[mobile]with a console - even if you don't ever log on to the 'console'
07:20.44tzafrirJerJer, this is a crude workaround. And if I don't like that wrapper?
07:20.51heath__jerjer: yes
07:21.10tzafrirIt adds an extra layer of complexity that sometimes gets in my way
07:21.45JerJer[mobile]heath__:  now check the second asterisk's iax2 debug for a DTMF frame
07:21.56CyberSwordi gonna put it directly to the internet
07:21.57CyberSwordbrb
07:23.50*** join/#asterisk bjohnson (~bjohnson@i216-58-15-199.igs.net)
07:26.19*** join/#asterisk W|NGNUT (~wingnut-n@128.80-203-103.nextgentel.com)
07:26.57pifhi, anyone using bristuff?
07:28.37kaldemarpif: yep.
07:29.21pifkaldemar : how do you return a busy to the telco when all B channels are in use?
07:31.34kaldemarpif: if i remember right, with bristuff you jump to n+201 if DIALSTATUS if CHANUNAVAIL, so try to set Busy() to n+201.
07:32.26*** join/#asterisk litage (~nick@ws01.5749.dsl.winshop.com.au)
07:32.33pifok, I'll try that, thanks
07:35.06*** join/#asterisk FryGuy- (fryguy@c-24-10-47-136.hsd1.ca.comcast.net)
07:36.09wtwhat's up everyone?
07:38.04*** join/#asterisk hellop (~hellop@cpe-70-95-18-61.hawaii.res.rr.com)
07:40.45*** join/#asterisk rabelais (~blank@ca-stmnca-cuda4-gen2c1-171.vnnyca.adelphia.net)
07:42.56pifkaldemar : but when B channels are busy the call does not enter any context
07:43.42pifsince it's coming from outside
07:45.16tzafrirpif, me, nort-of
07:45.32tzafriroh, ok
07:45.47pifyes?
07:45.52tzafrirnm
07:46.29pifwhen both channels are busy, asterisk says "Ignoring callwaiting SETUP on channel 0/0 span 4 0"
07:46.35*** part/#asterisk JooZoo (~chatzilla@kultapossu.yok.utu.fi)
07:46.59pifand the caller hears only silence for some time, then a telco error
07:47.14pifwhich sounds not very professionnal
07:53.37tzafriranybody here using sipphone? I can't seem to accept calls from them: only initiate. . 'sip debug' shows no trace of incoming call. But I do seem to be registered
07:57.36limbiquei do have a SIP phone
07:57.43limbiquework perfectly
07:57.57limbiquesnom 190 and snom 220
07:59.39*** join/#asterisk ZeeLax (~zeelax@rxgw.network.kz)
08:01.39*** join/#asterisk Jas_Williams (~Jason@host86-130-10-146.range86-130.btcentralplus.com)
08:04.43tzafrirlimbique, I meant: sipphone.com
08:05.05limbiqueok, sorry :P
08:05.15limbiquea softphone?
08:06.07limbiquewe're trying to make this device work: TDM400P   n/a TDM400P (Quad Analog FXO/FXS)
08:06.35limbiqueHow does these channels called?   zap?
08:11.01*** join/#asterisk pdracevich (~bob@210.54.249.228)
08:11.18pdracevichI have a JB resyncing issue, can anyone help?
08:11.28Math`I got "Rejected connect attempt from 69.90.176.122, who was trying to reach '100@incoming'" and I DO have context=incoming in my iax.conf
08:11.50Math`the calling server says No authority found
08:12.46*** join/#asterisk oob (~oob@203-173-146-89.bliink.ihug.co.nz)
08:12.58pdracevichThis is what it came up with:    chan_iax2.c:659 jb_warning_output: Resyncing the jb. last_delay -2, this delay -1717983010, threshold 1000, new offset 1717947644
08:15.24*** join/#asterisk Blake0PS (~blake@blakeops.com)
08:15.37*** join/#asterisk litage (~nick@ws01.5749.dsl.winshop.com.au)
08:15.43Blake0PSWith NuFone does anyone get "No one is available to answer at this time"
08:22.16tzafrirlimbique, yes, those are zap channels
08:23.46pdracevichThis is what it came up with:    chan_iax2.c:659 jb_warning_output: Resyncing the jb. last_delay -2, this delay -1717983010, threshold 1000, new offset 1717947644
08:23.57Jas_WilliamsMath`: Sounds like a password problem on incomming
08:38.13Math`Jas_Williams: nah the name of the iax client wasnt the same on both sides lol
08:40.26*** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de)
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08:54.54*** join/#asterisk RealFairPlayer (~3e8ae707@lx02.formativ.net)
08:54.58RealFairPlayerhi to all
08:57.12RealFairPlayercan somebody help me with a issue between asterisk and sipura2000 (dialtones)
09:05.47*** join/#asterisk tobiasWolf (~konversat@195.162.255.10)
09:11.54*** join/#asterisk indra (~indra_wat@microinfo.rain.fr)
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09:13.38*** join/#asterisk bikokola (~bikola@202.67.85.158)
09:14.42bikokolaanyone know if VOIPBuster supports IAX
09:16.28sig-use google.
09:18.10bikokolaim looking at ther site and they provide absoloutly free calls to around 20popular countries, it however looks to be too much as a service provider to customers which simply call and recieve
09:19.22*** part/#asterisk JooZoo (~chatzilla@kultapossu.yok.utu.fi)
09:19.36sig-free calls?
09:20.21bikokolaye
09:20.45sig-and u try it?
09:20.48bikokolaas in all you do is top up with $1, and you get free calls to landlines to countries like, US canada, australia etc
09:21.07bikokolavoipbuster.com
09:21.28bikokolai want to try it witha dollar and see what they tell me when i ask if they provide iax
09:21.50bikokolathere isnt any other way to contact them, they seem to caught up in the whole beta testing thing
09:21.53*** join/#asterisk newl (~newlook@203-59-217-50.dyn.iinet.net.au)
09:22.01sig-i know, but they support some other conection then using they windows client ?
09:23.39*** join/#asterisk kFuQ (~somedude@c-67-185-114-199.hsd1.wa.comcast.net)
09:23.43bikokolaye, thats what sucks
09:25.27*** join/#asterisk teapot (~tandrews@mail.grok.org.za)
09:25.28bikokolabbl
09:25.32tootbikokola yeah i'm using it
09:25.38bikokolaoh ok
09:25.47bikokolaso is it jus a win client
09:25.48tooti'm quite happy with it
09:25.58sig-toot: but with this windows client ?
09:25.59teapothi
09:26.02tootna i just setup asterisk to use it - there was a howto somewhere
09:26.04sig-or u connect using asterisk?
09:26.11sig-where? :P
09:26.15bikokolawow, i think im so signing up
09:26.34tootexten => _8X.,1,Dial,IAX2/SomeName@voipbuster/00${EXTEN:1}
09:26.39teapotIs anyone here in the UK ?
09:26.51teapotI want to know if anyone else is using a-law ?
09:26.53sig-ok i have ;>
09:27.09sig-213.61.187.150 here ;>
09:27.14sig-iax
09:29.14sig-teapot: and u know any free voip prov where i can try alaw? fwd support only ulaw
09:29.31*** join/#asterisk darkskiez (~darkskiez@194.247.78.146)
09:30.12teapotsig-: sorry I don't :p
09:33.08clive-teapot howzit
09:33.14teapothi clive-
09:33.29sig-toot: u register in iax that voipbuster?
09:33.58*** join/#asterisk lters (~lters@mrtcdsl-034.mis.net)
09:35.37*** join/#asterisk loick (~loick@per92-7-82-236-197-96.fbx.proxad.net)
09:35.41bikokolasig_: im looking thru a few forums, a bunch of people have allready tried it and claim it works for only one euro
09:37.39bikokolaanyway, iam also trying to learn to use asterisk as the same time
09:38.27sig-me too ;P
09:38.42sig-brb
09:39.29bikokolaim a little confused in exten => _8X.,1,Dial,IAX2/SomeName@voipbuster/00${EXTEN:1} something seems weird to me, what is up with _8X., does that only connect to voipbuster if the number dailed is 3numberd and has 8 as the middle term?.
09:39.30*** join/#asterisk mulvane (~mulvane@c-66-177-45-2.hsd1.fl.comcast.net)
09:40.39ltersyes
09:41.06tootsig- eh?
09:41.06mulvaneHere is a question that may seem weird but is a rising problem I am facing as well as others I know.. I'm cell phone based only. Me and my wife both have cell phones and no land line. Using something like asterisk and with phones that support it, could my cell phones be plugged into a asterisk system and then provide home phone service with say distincive and multi line support?
09:41.16lterswell, the period is a wild card search so 8+number+anything
09:41.24tootinstalled softphone - registered, changed reg to asterisk in iax.conf and extensions.conf - jobs a goodun
09:41.36bikokolawhy _
09:41.40bikokolawhats that do
09:42.06tooteh?
09:43.15ltersbikokola, means it is a wild card special matching diallan
09:43.28bikokolaah ok, thx
09:45.52mulvaneFrom what I am reading, if I could get a cell phone capable of answer on ring that would be able to signal the pbx, this design would work and I could create a cellphone cradle and provide home phone service via my cell phones and have multiple phones throughout the house with no need to worry about cell phone charging or losing the phones cause of the kids.. Any advice?
09:53.13*** join/#asterisk sig- (~sig@gnook.org)
10:00.43ltersanyone here run * with -p ?
10:00.55ltersor nice -n -19 ?
10:02.29*** join/#asterisk nextime (~nextime@213-140-6-96.fastres.net)
10:05.13tzafrirlters, me
10:05.24tzafrirwith -p
10:05.42ltershow many calls avg ?
10:05.51tzafrirnot much
10:05.51*** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl)
10:06.00lters50 ~
10:06.09puzzledmorning
10:06.26tzafrirlters, it shouldn't harm your performance, naturally
10:06.39tzafrirhi
10:06.51lterswhy is it not in safe_asterisk by default ?
10:09.43tzafririf asterisk hangs the cpu (with -p) the system will hang badly
10:09.51puzzledyup
10:10.01ltersthat is what I fear too...
10:10.40ltersstill puzzling over "max retries"
10:11.28puzzledI'm trying to figure out why the first second or so of an incoming/outgoing call is inaudible. using yesterday's cvs HEAD, chan_capi and a sip phone with ulaw. any ideas?
10:11.53puzzledhehe
10:11.59puzzledwell, I was thew first one
10:12.03lters:)
10:12.53ltersthe max retries, seems to be related to sp
10:12.55lterssip
10:14.14puzzledyes, search google. there have been some discussions about it
10:14.53*** join/#asterisk lehel (~asd@82.79.20.17)
10:14.55lehelhello
10:16.00puzzledhi
10:16.56Wonkapuzzled: how did you get chan_capi to compile?
10:17.25clive-puzzled, are you using the sourceforge chan_capi ?
10:17.30puzzledwith make? what didn't work for you?
10:17.35puzzledyes 0.5.4
10:17.43Wonkahups.
10:17.46clive-good
10:17.49Wonkanow it compiles...
10:17.54puzzledhehe
10:18.03Wonkabut it's against 1.0.9
10:18.11*** join/#asterisk gordonjcp (~gordonjcp@cpc3-broo2-5-0-cust232.renf.cable.ntl.com)
10:18.13Wonkadoesnt matter, it compiled
10:18.15clive-wonka I think it should work with both
10:18.16puzzleddunno about that. I use yesterday's HEAD
10:18.28puzzledclive-: yes it is supposed too
10:18.33Wonkalast time i tried, it didnt :)
10:18.45puzzledWonka: using 0.5.4?
10:18.55Wonka0.3.5
10:19.09Wonkalemme get 0.5.4
10:19.10puzzledWonka: get 0.5.4 for sure. many issues fixed
10:19.14tzafrirwhat kernel-level support is required for chan_capi?
10:19.25puzzledcapi support
10:19.42tzafrirany idea if any of the default debian sarge kernels have it?
10:20.01puzzledno idea. I use CentOS and FC
10:20.04Wonkais there a newer version of chan_misdn too?
10:20.19puzzledcheck beronet
10:20.54Wonkai take that as no.
10:21.08puzzlednope, I just don't know
10:21.30leheltzafrir yes it has
10:21.39leheli'm using debian sarge
10:21.42lehelwith capi compiled
10:21.52Wonka0.0.3-rc6 is on beronet, and that didn't compile against cvs some weeks ago
10:21.57*** join/#asterisk Theuni (~ctheune@alphastar.gocept.com)
10:21.59leheland it is with capi support
10:22.00TheuniHowdi
10:22.08puzzledtzafrir: for best capi support you need a recent 2.6.x kernel
10:22.12tzafrirgood. any idea how does chan_capi-tm interacts with bristuffed asterisk?
10:22.14*** join/#asterisk apardo (~apardo@85.Red-81-39-78.pooles.rima-tde.net)
10:22.16TheuniI just got a grandstream bt 100 for some testing
10:22.20TheuniI'm getting  WARNING[562]: chan_sip.c:611 __sip_xmit: sip_xmit of 0x80ff684 (len 455) to 10.1.1.216 returned -1: Invalid argument
10:22.22puzzledtzafrir: it doesn't
10:22.25sig-somone can say why i get this error?
10:22.26Theuniand Jul 19 12:21:14 NOTICE[562]: chan_sip.c:7691 handle_request: Registration from '<sip:ctheune@10.1.1.40>' failed for '10.1.1.216'
10:22.28sig-http://pastebin.ca/18045
10:22.29Theunifrom the asterisk error
10:22.31Theuniany hints
10:22.32Theuni?
10:22.32Wonkaand where do i find chan_capi 0.5.4? not on http://www.junghanns.net/asterisk/downloads/
10:22.46leheltzafrir what do you mean bristuffed asterisk?
10:22.49puzzledWonka: sourceforge
10:22.50tzafrirWonka, http://sourceforge.net/projects/chan-capi/
10:23.08Wonkaah, chan-capi. i tried chan_capi
10:23.11puzzledTheuni: user or pass are wrong?
10:23.14Theunihmm
10:23.16tzafrirasterisk with the bristuff patch, from junghanns as well
10:23.20WonkaO.o
10:23.23lehelWonka what card do you have?
10:23.49tzafrirpuzzled, you mean it won't build? won't work? or doesn't care?
10:23.50Wonkalehel: one hfc-s and one avm b1 isa
10:24.08puzzledtzafrir: chan_capi doesn't care about the existence of bristuff
10:24.24tzafrirWell, I'll give it a shot...
10:25.00puzzledsig-: no idea what condition 14 is. search the source, google, ask when the US guys are awake or send an email to the list
10:25.06af_there is a stun server (for nat traversal of grandstream's, for example) that run on linux?
10:25.28lehelpuzzled you have any idea why asterisk wont run as 'asterisk', when i loading capi?
10:25.45sig-puzzled: tnx,
10:26.11puzzledlehel: that does not sound logical. without chan_capi installed it does run as asterisk?
10:26.57lehelpuzzled i have chan_capi installed.. fritz.. la la.. when i start asterisk as 'root'.. all OK
10:27.09puzzledok
10:27.18lehelpuzzled Contr1: 2 B channels total, 2 B channels free.
10:27.23Ahrimanesaf_: yes
10:27.32*** join/#asterisk kswail (~kyndar@modemcable244.73-81-70.mc.videotron.ca)
10:27.34Wonkalehel: look for some files it tries to open, which belong to root
10:27.39lehelbut i want to run asterisk as asterisk
10:27.43Wonkalehel: strace -eopen asterisk -U asterisk
10:28.06Wonkalehel: look for open failures there and check file ownership and permissions
10:28.23af_Ahrimanes: do you know the name of this project, or where I could find binaries/sources of it?
10:28.49Ahrimanesaf_: 2 sec and i'll send you url of the one i'm using
10:28.59af_thanks Ahrimanes
10:29.23Ahrimanesaf_: http://www.vovida.org/ <- in the left menu, choose stun server
10:30.08af_just to speak about it: suppose I have an * on a public ip address, if I have sip clients in a lan under nat (adsl router, for example) I need this stun thingy?
10:30.26lehelWonka: do you see somthg strange here?: http://pastebin.ca/18046
10:30.26af_oh vovida
10:31.08puzzledlehel: check http://www.voip-info.org/tiki-index.php?page=Asterisk%20non-root. it says that you need to do: chown asterisk /dev/capi20
10:31.59newlaf_: if it's A (asterisk) -> (internet) <- B (router) <- LAN, no, you shouldn't need a STUN server.
10:32.38af_you mean A have public ip addreess?
10:32.55af_and a client on(LAN) has a private one?
10:33.42lehelthanks puzzled!.. i missed that this time!;)
10:33.42Wonkalehel: no, looks quite ok
10:34.00af_so A(asterisk on public ip) -> (internet) <- B (router does NAT) <- LAN (client with private ip, sip ua)
10:34.08lehelthanks Wonka it was the /dev/capi20 owner problem
10:34.12af_this case I don't need stun?
10:35.26lehelWonka did you compiled chan_capi?.. i have a fritz card, and i don't know that 'hfc'.. 'avm' in the kernel support
10:35.28Ahrimanesaf_: if the client is behind nat, stun can help
10:35.41af_good, what I suppose
10:35.59Wonkalehel: i did not compile it yet, and i don't have a passive card
10:36.14af_Ahrimanes: I guess stun is importanti if I have more then one sip client under the same nat/router, that's right?
10:36.16Wonkalehel: my avm b1 is an active card...
10:36.30*** join/#asterisk Carion (~wayne@rndf-146-50-58.telkomadsl.co.za)
10:36.49Ahrimanesaf_: hm havent tried this yet actually.. but i've found that using video with * stun becomes ever more useful
10:37.11af_viedeo?
10:37.24Ahrimanesvideocalls
10:37.34lehelWonka where is your problem begins?
10:37.46af_asterisk supports video calls?
10:37.54Ahrimanesaf_: yep
10:37.57af_oh
10:38.07af_Ahrimanes: why stun server needs two ips?
10:38.31Ahrimanesaf_: guess it's for better accuracy.. not sure though, havent read the rfc
10:38.39Wonkalehel: my problem was that chan_capi 0.3.5 didn't compile against the asterisk cvs from weeks ago.
10:38.57puzzledwhich doesn't matter as you need to use 0.5.4 anyway :)
10:39.25Wonkajep
10:39.26*** join/#asterisk Kraven (kraven@inetandmore.com)
10:39.31Wonkacompiling that now
10:39.32Kravenhello
10:39.37puzzledhi
10:40.03WonkaVIA C3 processors are a pest...
10:40.18Wonka_everywhere_ there's "PROC=`uname -m`"
10:40.18leheli have another problem.. mine main problem now.. who knows about IAX2
10:40.24af_not so much doc, I am seeing
10:40.33Wonka_everywhere_ i have to put PROC=c3 behind that
10:40.37KravenCan someone explain me to get * working with innovaphone ip400? I've compiled *, oh323 and this stuff... but I can't hear the voice of the samples
10:40.39puzzledWonka: yeah, make sure you compile everything for i586 or it won't work on C3
10:40.47Ahrimanesaf_: google for stun rfc
10:40.47*** join/#asterisk vuvie (~vuvie@bb219-74-46-99.singnet.com.sg)
10:40.55Wonkapuzzled: for c3, then it does too
10:41.03puzzledoh, didn't know that
10:41.13lehelpls take a look: http://pastebin.ca/18049 < IAX call stopped
10:41.18af_Ahrimanes: I am reading it now, is linked from the vovida.org site
10:41.38Ahrimanesaf_: ok
10:41.46leheli can call that number.. but i can't call from that number
10:41.57puzzledWonka: to confuse things more, iirc the latest nehemiah core does support proper i686 instruction set so doesn't need PROC=i586/c3
10:42.39Kravennoone?
10:42.51Wonkai got an Ezra here
10:43.22af_still, I don't understand why the server needs two ips
10:43.31lehelanyone .. IAX2 ?? fixlocalprefix
10:43.59puzzledKraven: most people avoid h323 like the plague and you are not providing any information. cut 'n paste your configs, error messages to pastebin.com and ask again
10:44.44puzzledlehel: I don't see that your IAX call stops
10:45.19*** join/#asterisk cmk (~cmk_@p54A3CFBF.dip.t-dialin.net)
10:45.23lehelpuzzled on the other "end" nothing happens
10:45.46af_mmhhh
10:45.50KravenThe only conf that I've edited is: oh323.conf: http://pastebin.com/316140
10:46.13puzzledlehel: dunno, is that boxc-peer registered?
10:46.30*** join/#asterisk jpcarvalho (Jeff@201.30.193.135)
10:47.40sig-anybody know any iax provider that give some free amount to calls? (voipbuster) something more?
10:47.47jpcarvalhoHello all : I need to process my CDR (postgresql) based on a tarriff rate ... someone have any idea? I looked for all based OS appz and don't reach what i'm looking for
10:48.04sylesig-: voipjet
10:48.15puzzledKraven: I'm not sure but don't think that asterisk supports silence suppression so you may want to remove that. Also afaik it sends 20ms frames and not 10ms with ulaw/alaw
10:49.28lehelisn't ringing.. and i have enabled iax debug.. but nothing; how would u [what do you mean] register?
10:50.07Kravenpuzzled: Still no sound...
10:50.39tzafrirwell, packaged. Will it work? I really hope it will
10:51.41puzzledKraven: no sounds is usually rtp/udp related. check firewall etc.
10:52.13puzzledlehel: no idea. I just use sip phones and chan_capi. no boxA iax BoxB stuff
10:52.26Kravenpuzzled: the ip400 in in the same subnet as the *-box and there is no firewall between them
10:53.06puzzledKraven: paste some output of the asterisk console when you dial from the phone
10:53.10puzzledin pastebin.com
10:53.24sig-syle, tnx.
10:53.37lehelpuzzled: how do you configure the capi channels?
10:54.05Kravensec
10:54.13puzzledlehel: the only thing I do is add an MSN and change the context. that's it
10:54.45Kravenhttp://pastebin.com/316142
10:55.43lehelpuzzled: sounds simple;).. what do you mean "MSN"?.. you add in the extensions.conf?
10:55.45mulvaneWith asterisk, I can intercept knwon numbers, and handle them differently..Correct? Lets say I know when my command tries to call me, its gonna always be from (904)-270-xxxx. I can handle all the 270 calls to auto forward to say my cellphone?
10:56.03Ahrimanesyeah sure
10:57.07Wonka*grmpf*
10:57.08Wonka<PROTECTED>
10:57.11WonkaLoading module app_capiFax.so failed!
10:57.20puzzledlehel: no, in capi.conf. MSN is a multisubscriber number. my ISDN has 4 numbers (MSNs)
10:57.28Wonka$ grep capidebug chan_capi.so
10:57.28WonkaBinary file chan_capi.so matches
10:57.30Kravenpuzzled: any idea?
10:58.15puzzledKraven: I dont know if the "+" can be used so replace that one with a "_" or "-" and solve the first error: == Starting OH323/Sascha+Growe@192.168.10.2-c5f5 at default,41,1 failed so falling back to exten 's'
10:59.33lehelWonka: noload app_capiFax (?)
10:59.50Wonkalehel: that's a workaround, not a solution... :)
11:01.07Wonkabut that workaround works. so let's look for a real solution
11:02.21*** join/#asterisk zeedo (~zeedo@www.reboot-robot.net)
11:04.05lehelWonka ...hmmm.. i have 0.3.5, and it is not included the capiFax, if you find out the problem i'll install the 0.5.4;)
11:04.22Wonkanaah, i don't need it yet
11:04.27Wonkamaybe i'll play with it later
11:04.37Wonkadon't get into trouble for that
11:04.49Jas_WilliamsWonka: which version of * are you running
11:05.28WonkaJas_Williams: 1.0.9-BRIstuffed-0.2.0-RC8j
11:06.07Jas_Williamswonka using capi ?
11:06.23Jas_Williamsor chan_zap ?
11:06.31*** join/#asterisk limbique (~limbique@nl-ifw-oss.orcagroup.com)
11:06.49limbiquehi
11:06.50WonkaJas_Williams: both. capi for the avm a1, zap for the hfc-s
11:07.06Jas_WilliamsFun :)
11:07.38Wonkahm. now i shot my * down trying to dial out with CAPI
11:09.21*** join/#asterisk chiggins (~chris@213-94-240-94.b-ras1.srl.dublin.eircom.net)
11:09.23sig-syle: how voipjet use caller id? i just saw number from personal mobile range.. and in evry dial i see other number..
11:09.47sig-and caller id in extension it`s diff then number that i see in mobile..
11:10.48limbiquei can't get this device installed, TDM400P, can anyone help me?
11:11.01limbiqueTDM400P (Quad Analog FXO/FXS)
11:11.09tzafrirand the problem is?
11:11.18limbiquei have no dialtone on it
11:11.33sylesig-
11:11.45syleexten => _1NXXNXXXXXX,1,SetCallerID(2223334444)
11:11.49sylethen dial
11:12.14tzafrirlimbique, do the channels appear in 'zap show channels'?
11:12.45limbiquenope :(
11:12.52limbiqueonly see the bri channels
11:12.59syleyou should see number you passed, if you don;t , then they don;t support caller-id passthrough, dump them and get a new provider
11:13.06tzafrirso they are not defined in zapata.conf
11:13.41tzafrirlimbique, do they appear in cat /proc/zaptel/*
11:13.46tzafrir~genzaptelconf
11:13.46jbotextra, extra, read all about it, genzaptelconf is a script to general zaptel.conf and zapata.conf snippet for TDM cards, availble from http://tzafrir.org.il/genzaptelconf or from the Rapid zaptel packaged. Ask tzafrir about it. ignore warning about missing ast-cmd.
11:14.01Kravenpuzzled: http://pastebin.com/316147 <- still not working... :(
11:14.09tzafrirgenzaptelconf now supports zaphfc as well
11:14.11limbiquethey are not defined in the zapata.conf
11:14.14lehelWonka i compiled chan_capi0.5.4 ;p
11:14.20sig-syle: then i know that they dont support ;P u know any that support?
11:14.28W|NGNUTHi all. Does anyone know how to execute logic _during_ a call? I need to do this for a prepaid-app where I want to check remaining credit during the call...
11:14.33limbiqueonly see switchtype=euroisdn
11:14.35lehelWonka and it's working with that capiFax
11:14.40Wonkahm
11:14.46limbique:q
11:15.04Ahrimanesanyone done fax over iax?
11:15.22sig-when i change from 911 to 2223334444 i just get unknown caller id
11:15.23limbiqueonly quadBRI PCI ISDN Card 1 Span 4 [TE]
11:15.32lehelnot yet Ahrimanes;P
11:15.38Ahrimaneslehel: hehe
11:15.40Wonkasomething's broken here. i've commented out all noload => app_capi*, and now it breaks with that:
11:15.44Wonka[app_capiHOLD.so]/usr/local/usr/lib/asterisk/modules/app_capiHOLD.so: undefined symbol: get_ast_capi_MessageNumber
11:15.44tzafrirlimbique, so it seems that the module is not loaded. What version of *?
11:15.51Ahrimanesi have all my pstn at another location but want fax to reach my * here
11:15.57limbiquehow to see?
11:16.13tzafrirasterisk -rx 'show version'
11:16.15Theunipuzzled:  yeah. somehow authentication was weird. everything fine now!
11:17.03limbique# Asterisk 1.0.2-BRIstuffed-0.2.0-RC2 built by nnrelman@compile.office.netland.nl on a i686 running Linux
11:17.12Ahrimaneseeek
11:17.17Ahrimanesplz to update
11:17.20Wonkascrolling back, it seems * didn't load chan_capi before that? wtf?
11:17.27Kravendamn this oh323 is freaky... it works but still no sound I can do what I wnat...
11:17.30*** join/#asterisk gambolputty (~gambolput@cblmdm72-240-241-108.buckeyecom.net)
11:18.07puzzledTheuni: cool, enjoy!
11:18.44puzzledWonka: did you add the 2 lines in modules.conf?
11:19.08Wonkaif i give it an explicit load => chan_capi.so, it works?
11:19.11Wonka*gr*
11:19.29syleguys what are you using for billing?
11:19.34limbiquetza: is that version up to date?
11:19.34sylewhat software
11:20.32Ahrimanessyle: cdr_pgsql.so
11:20.35sylesig-: tons support
11:20.45Wonkapuzzled: which lines?
11:20.48tzafrirlimbique, that's ancient :-)
11:20.53limbiquelol
11:20.58sylecdr_pgsql hehee
11:21.06Wonkaah there, those from INSTALL
11:21.07syleif i actually liked postgres maybe
11:21.09tzafririf you have a new card, it's pci id is only identified by recent zaptel
11:21.10limbiquethere is an other company that build this system
11:21.13Wonkai do have those now
11:21.24puzzledWonka: http://pastebin.com/316152
11:21.25limbiquewe're not that good with linux etc.
11:21.33tzafriryou could patch it to add the relevant PCI IDs. Shouldn't be difficult, but I nevel tried
11:21.37*** join/#asterisk jonathh (~asd@host217-46-145-65.in-addr.btopenworld.com)
11:21.43lehelWonka in modules.conf: context: [global] chan_capi.so=yes
11:21.47limbiqueit is a pilot for us, to do asterisk
11:21.55Wonkai got those now
11:22.10leheland upper: load => res_features.so        load => chan_capi.so
11:22.12Wonkaonce i get to read the doku, it mostly works :)
11:22.12Wonka<- hates reading doku, even more hates writing doku
11:22.53limbiqueand i'm not that good with linux.. but i'm willing to learn
11:23.11limbiquewe always did windows.. with tapi etc.
11:23.23lehelWonka.. so now is loading?
11:23.26sig-syle: tons? tons.com some shit, they have some website?
11:23.49limbiquetza, patch what?
11:24.10limbiqueok, so the older version doesn't recognize it
11:24.32sylesig-: www.voipjet.com www.nufone.com etc etc
11:24.37W|NGNUTOkay, trying again; How do I stream a file to a call (preferably only to the caller) without shutting the call down?
11:25.23Kravenpuzzled: no further idea?
11:25.33sylebasically all caller-id passthrough means is the fuckers didn;t set callerid themselves for you in their iax or sip.conf like it should be
11:25.34puzzlednope
11:25.48lehelW|NGNUT if i would know the answer..
11:25.56puzzledW|NGNUT: show application Dial
11:26.06Kravenbut thx for your help
11:26.17sig-syle: voipjet dont support changing caller-id in my country, i just get few number on my mobile but not this same that i use in callerid ;>
11:26.38puzzledKraven: check back in a couple of hours when the US is awake or mail the asterisk-users list
11:27.10W|NGNUTpuzzled: Thanks, only I need to do it asyncronous during the call, possibly several times...
11:28.02puzzledW|NGNUT: no idea, perhaps try the asterisk-users list?
11:28.40W|NGNUTpuzzled: Yeah. Maybe the M(x) option is a solution... thanks anyway!
11:29.45Wonkalehel: it is loading
11:29.57Wonkahah. my new plaything just arrived
11:30.16lehelnice Wonka;P
11:30.46Wonkaa Netgear WGT634U
11:34.08Wonkahrm. asterisk still segfaults dialling out
11:34.14Wonkaon capi
11:37.41sig-debug core
11:39.33Wonka*CLI> dial 008001507090
11:39.33Wonka*CLI> SIMPLE DIAL (NO URL)
11:39.33WonkaSegmentation fault (core dumped)
11:40.18Wonkawith more -v, it also says:
11:40.19Wonka<PROTECTED>
11:40.25Wonka<PROTECTED>
11:41.08sig-and when u change to OSS ?
11:41.46WonkaALSA should be ok, it works with sip and h323
11:43.23sig-then dunno. wait for usa guys ;P or just try debug core,
11:48.21*** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net)
11:50.46cpatry~realtime
11:50.46jbotrealtime is, like, http://www.voip-info.org/wiki-Asterisk+RealTime
11:52.36lehelbye all
11:55.10*** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com)
11:55.37*** join/#asterisk stkn_ (nobody@stkn-active-pdpc.developer.gentoo)
11:59.43sig-bellester down?
11:59.54*** join/#asterisk Thus0 (~Thus0@158.111.102-84.rev.gaoland.net)
12:00.59*** join/#asterisk Pcharky (~khagen@cust.12.215.adsl.cistron.nl)
12:01.29Thus0Hello
12:01.34*** join/#asterisk memic (~memic@chicago089.server4free.de)
12:02.02PcharkyHi
12:02.05sig-hi Thus0
12:04.17PcharkyHey there, got a little question maybe one of you can help me?
12:04.34*** join/#asterisk lathos42 (~lathos42@65-42-27-66.dowdingindustries.com)
12:04.41sig-maybe if we can..
12:06.14PcharkyGreat, I've just set up an asterisk, and I want to use it 'beside' my lucent index. Do you know what settings my zaphfc card should have?
12:06.27PcharkyAs in BRI/PRI NT/TE?
12:06.40sig-sorry i dont use zap*
12:07.32PcharkyApart from the zap* any idea? (general)
12:07.59*** join/#asterisk bikokola (~bikola@211.27.1.87)
12:08.17sig-u have one hfc card?
12:08.34Pcharkycurrently yes.
12:08.54bikokolahey guys quick question. How do i go about installing a USB adsl modem on Linux, i have Linux fedora core 4. I'm new to this.
12:09.08bikokolai'm currently just burning the ISO
12:09.09sig-then use bri with signalling = bri_net_ptmp
12:09.23Pcharkysig-, thanks alot.
12:09.38sig-Pcharky: n/p
12:10.46sig-bikokola: gogle with u modem infor.. and i think that u find any answer..
12:11.07bikokolathx sig
12:11.37sig-but it`s not answer only clue ;)
12:11.50bikokolayep, ill give it a tryt
12:12.43*** join/#asterisk pif (ldm@zenon.apartia.fr)
12:15.04tzafrirThe zaptel I've built (1.0.9) does not appear to show colors when I connect with 'asterisk -r'?
12:15.12tzafrirbikokola, #fedora ?
12:15.54tzafrirbikokola, it also depends on the type of ADSL connection (or: how braindead is your ISP and what tunnels do you have to go through just to get a connection)
12:18.20tzafrirbikokola, however you really should get a better adsl modem ;-)
12:18.24sig-i dont put -n and i dont have too colors.. :>
12:18.43Darwin35Good morning who is serving the Latte this am
12:18.54sig-:D
12:19.09sig-o somone wakeup
12:19.12sig-(=
12:19.33Darwin35Dbl mocha almond with a hint of maple
12:19.39bikokolatazfrir, why do u say i shld get a better modem, should it auto detect
12:19.43tzafrirsig-, I don't believe I put -n
12:19.51sig-Darwin35: i just eat dinner :>
12:20.15tzafrirbikokola, generally the adsl usb modms I know are bad. You should use an ethernet oen....
12:20.31sig-bikokola: buy modem/router in this way u dont need configure with u pc..
12:20.42bikokolai can use ethernet, but i choose usb, i only have one ethernet slot on my comp
12:20.48sig-tzafrir: like sagem@? :>
12:21.31bikokolai have a modem router
12:22.01sig-with only usb? without ethernet?
12:22.03bikokolanetcomm modem router with ethernet, but i dont know exactly what to expect after i install linux
12:22.19bikokolano with ethernet, but atm, im using usb
12:22.24tzafrirbikokola, however this is a Q for a general linux channel. try aome australian linux channel. There must be one. the little details there differ greatly from country to country
12:23.15bikokolawow, how did u know aussie
12:25.25*** join/#asterisk blop (~blop@213-193-176-12.adsl.easynet.be)
12:25.36blophi all :)
12:25.38Darwin35anyon eknow how to unlock the linksys vonage  voip router
12:27.31tzafrirbikokola, whois(1) on your IP address. Nothing special
12:28.13bikokolaim not so into mirc,
12:28.21*** join/#asterisk oej (~oej@apollo.webway.se)
12:29.09*** join/#asterisk klapzin (~klap@200-161-186-110.dsl.telesp.net.br)
12:30.04tzafrirbikokola, it's not the irc command whois. it is an information lookup in the domains registry.
12:32.03*** join/#asterisk meppl (~mephisto@84-245-182-211.ipool.celox.de)
12:32.23*** join/#asterisk cjk (~cjk@80.92.64.103)
12:32.31bjohnsonDarwin35: yes, but they aren't telling
12:32.34cjkhi, anyone here using voicemail odbc storage
12:33.29bjohnsonDarwin35: well, that's not exactly true.  They're not sharing the Linksys/Sipura developers software required to do it
12:39.12Darwin35grrrr
12:39.12*** join/#asterisk awk_ (~ht@BS.iNES.RO)
12:39.22Darwin35I have one that a friend gave me lastnight
12:39.30Darwin35he just dropped vonage
12:39.39Darwin35I want to use it but not with them
12:45.31*** join/#asterisk coldfeet (~cold@dsl-80-46-109-145.access.as9105.com)
12:45.50coldfeetmornin all
12:47.30Darwin35anyone played with chan_vox yet
12:47.38tzafrircolors again: I have colors when I run asterisk non-daemonized. But asterisk -r does not show colors. Should it?
12:48.15tzafrirwhat is chan_vox?
12:48.47Darwin35I know I saw it
12:48.59Darwin35grr suppost to give voice control
12:49.31*** join/#asterisk Invisible_Magi (~abc@165.154.121.241)
12:49.35*** part/#asterisk awk_ (~ht@BS.iNES.RO)
12:50.42*** join/#asterisk florz (~florz@2001:1a50:503c:0:0:0:0:1)
12:50.45Darwin35the is a app_formatvox
12:50.50Darwin35grr
12:51.08Darwin35I want voice control over asterisk for handicapped users
12:51.09*** join/#asterisk Gunnar (~gunnar@bkkb-gw.bitcon.no)
12:51.17Darwin35so they can say dial and the nmbr
12:52.13Darwin35I have 3 friends that are quads and this would be great for them
12:52.14florzAre there any countries left that can't reach international 15-digit numbers?
12:53.11*** join/#asterisk |Vulture| (~V@c-66-177-92-4.hsd1.fl.comcast.net)
12:53.55PcharkyChiao
12:53.56*** part/#asterisk Pcharky (~khagen@cust.12.215.adsl.cistron.nl)
12:56.10iCEBrkrDarwin35: I'm working on integrating Sphinx.
12:56.30iCEBrkror is chan_vox totally different?
13:01.37rikstaiCEBrkr: i have a good link for you then
13:02.32rikstahttp://turnkey-solution.com/asterisk-sphinx.html
13:05.23hmmhesaysheh turnkey
13:05.39tzangerI always read "turkey"
13:05.50hmmhesayspretty much
13:06.25hmmhesaysthe 'I know nothing gimme something to sell' solutions
13:10.06Kravenhrm dam h323 won't work... :(
13:10.51*** join/#asterisk Katty (~angela@68.112.15.110)
13:12.16hmmhesayssure it will
13:12.22Kattymew
13:12.24hmmhesaysyou must know the right voodoo
13:12.34Kattyvoodoo? who do?
13:12.43hmmhesaysthat you do, so well?
13:12.47*** join/#asterisk benthos (~n_o_b_o_d@69.90.85.25)
13:12.49Kattyway to goof it up :<
13:12.54hmmhesaysha
13:12.56Kattyvoodoo?
13:12.58Kattywho do?
13:12.59Kattyyou do!
13:13.00Kattydo what?
13:13.03Kattyremind of the babe!
13:13.49Kattydid they shave off all teh purrty spikes?
13:14.06Hmmhesayssomewhat, it is pretty short now
13:14.20Hmmhesaysshould be back to spikey length in a week or so though
13:14.27Katty:>
13:14.43Hmmhesaysi'm contemplating dying the tips blue
13:14.55Kattynummy
13:15.38Hmmhesays:D
13:15.50*** join/#asterisk B0ngFrOg (~wsmith@c-24-9-253-203.hsd1.co.comcast.net)
13:15.51Kattyi get to redo my dial plan today!
13:15.56Kattysorta.
13:15.59B0ngFrOggood morning....
13:16.04sylethere goes half your day katty
13:16.07Hmmhesaysever get the transfer stuff figured out?
13:16.12Kattysyle: heh, not quite (=
13:16.22KattyHmmhesays: attended transfer to call parking?
13:16.52Hmmhesayshrm, blind transfer to a phone showing up like an outside call
13:16.59Kattyoh, that
13:17.04Kattyno, not yet
13:17.08B0ngFrOgAnyone have any recomendations for a 20 phone system ?  Motherboards, server etc.
13:17.25*** join/#asterisk Ariel_ (~Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net)
13:17.28nDuffmmm, cute
13:17.33dtwilsonyes a server with a motherboard would be good :b
13:17.33Hmmhesaysi'll give you some pointers some time
13:17.34KattyAriel_: mew (=
13:17.35DelvarB0ngFrOg: call center?
13:17.37KattyHmmhesays: k
13:17.48RaYmAn-BxB0ngFrOg: a start would be to be more specific...voip phones or analog phones for example...
13:18.12HmmhesaysB0ngFrOg: dell poweredge something, with 7960's or ip501's
13:18.23Hmmhesays3ware raid controller, done deal.
13:18.55Ariel_morning everyone
13:19.01Ariel_morning Katty hope all is well.
13:19.05Hmmhesaysmorning Ariel_
13:19.26Invisible_MagiI'm looking for recommendations for a 12 phone extenstion system utilizing VoIP
13:19.37darkskiezthe new cisco sip firmware plays the sip audio channel and overlays its own ringing sounds, so if I call an external number, i hear the ringing twice, I can disable the outside ringing noises with the r Dial parameter, but i'd like to disable the phones instead..Its not possible though is it ?
13:20.04Hmmhesays<repeat>
13:20.19Hmmhesayswhat firmware is that?
13:20.24darkskiez7.5
13:20.44iCEBrkrriksta: that's all good and dandy, but my EAGI isn't spewing audio.
13:20.44Hmmhesaysseems unlikely that you can't disable something like that
13:20.47darkskiezI cant wait till asterisk adds proper remote-party support, as the phone supports updating the display with who you are connected to
13:20.55iCEBrkrriksta: So I wonder if the EAGI audio descriptor is broken.
13:21.05darkskiezso *8 could be rewritten to who you picked up and if your call is transferred etc
13:21.11iCEBrkrriksta: on top of that, who wnats to use perl?!?!
13:21.40Hmmhesaysperl is a fine language
13:21.43*** join/#asterisk salvini_fs (~felipesal@201.32.231.174)
13:21.49rikstaiCEBrkr: you don't have to use perl, you can use the logic from his code
13:21.55rikstaand use whatever language you want
13:22.01*** part/#asterisk salvini_fs (~felipesal@201.32.231.174)
13:22.02rikstaperl aint bad for that kinda stuff anyway
13:22.09iCEBrkrHmmhesays: Thers's the problem with most opensource projects.  IT REQUIRES TOO MANY DAMN PIECES FOR THE DAMN PUZZLE TO WORK
13:22.18Theuniouch
13:22.30iCEBrkrInstall these 12 RPMS just to get something simple to work.
13:22.32iCEBrkrIt's F'n retarded.
13:22.38Hmmhesaysheh, use debian
13:22.41iCEBrkrJust like AMP
13:22.49Hmmhesaysdon't use amp
13:22.52iCEBrkrHave you seen thre requirements for AMP?
13:22.55darkskiezmodular != bad
13:23.08rikstaiCEBrkr: no pain no gain, use windows if you aren't happy with that
13:23.12iCEBrkrAMP is cool, but at the sametime it's JUNK cuz it relies on too much shit.
13:23.12Hmmhesaysdownload aah and quit yer b1tchin
13:23.15B0ngFrOgI am mainly fishing for Recomendations on server (other than DELL) that have worked for people .... I have just used what was around in the past but need to move to a consistant platform for this cust......
13:23.30Hmmhesayswhat is wrong with poweredge hardware?
13:23.37syleyeah no shit
13:23.40KattyAriel_: yes, thanks (=
13:23.42Kattysyle: :<
13:23.48iCEBrkrriksta: See that's the idea.  See, that's why Windows is on the desktop and linux hasn't even started to become a desktop os.  it's to F'n clumsy.
13:23.57florzB0ngFrOg: I'd suggest computers as a consistens platform.
13:24.00rikstaiCEBrkr: so? i don't care about that
13:24.02Kattysyle: why not say something else more colorful next time
13:24.08Kattysyle: like purple people eater
13:24.08rikstai don't care what "other people" are using
13:24.23B0ngFrOgEach server will have 1 pri, iax trunks, polycom phones, 1-4 fxo line
13:24.28Hmmhesaysknoppix has gotten to be a pretty good 'desktop' os
13:24.29sylelol katty you must have kids appologies
13:24.32iCEBrkrriksta: Hey man, I used to be that Linux Elitest too man.. Just open your eyes.. and you'll see
13:24.37Kattyme!
13:24.37rikstaknoppix is pretty sweet
13:24.38Kattykids!
13:24.38Kattyomg
13:24.40Ariel_argh why are distro so different from each other.  I feel this is the reason there it's so hard to deploy more linux systems.
13:24.43HmmhesaysB0ngFrOg: what is wrong with dell hardware?
13:24.50Kattyk, all better.
13:24.53sig-yeah, we are kids mom ;P
13:24.53rikstaiCEBrkr: my eyes are fully open, i'm just not concered about everyone else right now
13:24.57B0ngFrOgflorz good morning to you too
13:24.57darkskieziCEBrkr: elitests of any cause tend to be fuckwits
13:25.09iCEBrkrriksta: Besides, I'm 1st in line to bash and hate on Microsoft.. The line starts back there, buddy..
13:25.18florzB0ngFrOg: Yeah, gmorning =:-)
13:25.19darkskiezexactly my point
13:25.21B0ngFrOgHmmhesays IRQ, IRQ, IRQ
13:25.24iCEBrkrdarkskiez: I'm noticing. :D
13:25.36Hmmhesaysheh, get the right poweredge and you'll be fine
13:25.44rikstaiCEBrkr: I understand things need to work easily for the average joe. Actually i'm working on that right now for my latest java project for asterisk
13:25.53rikstayou'll see it realeased next week sometime
13:25.57rikstareleased
13:25.58iCEBrkrriksta: So who in their right minds wants to dick with 15 RPMs to get a product working?
13:26.07darkskiezriksta: what is it? an amp type thing?
13:26.14Hmmhesaysdebian debian debian
13:26.15iCEBrkrJava.. LOL
13:26.19rikstaiCEBrkr: no-one if possible, i'm just saying that i dont mind it
13:26.20iCEBrkrHmmhesays: Debian isn't the answer.
13:26.22crash3mlathos42: get 2, they are worth it :)
13:26.29tzafrirAriel_, this is a reminder for you to donate some funds to the LSB...
13:26.34Hmmhesaysapt, it has super cow powers
13:26.38tzafrir42 is the answer
13:26.39Ariel_tzafrir, lsb?
13:26.43darkskiezwhat distro uses apt ?
13:26.44iCEBrkrriksta: I put up with it too.. But when I design a 'product', I attempt to make it self contained.
13:26.47rikstadarkskiez: kinda of, its called Asterisk Desktop Manager, its got quick dial from clipboard, bluetooth presence redirection when you walk out of the office, and tons more crap
13:26.48tzafrirLinux Standards Base
13:27.03B0ngFrOgHmmhesays I have recommended Dell for most apps but not being in control of IRQ is a little bit of a pain ... some installs use both t-1(pri) and fxo lines
13:27.04florzB0ngFrOg: I mean, really, what could be a sensible cause for using only one kind of hardware? "If anything fails, everything fails"?!
13:27.05rikstaiCEBrkr: it's java and swt, to allow it to work on linux mac windows and others.....because people have been begging me to do so
13:27.08iCEBrkrriksta: Lemme see screenshots! I want screenshots!!
13:27.18Ariel_tzafrir, oh I see. I have a customer that has one system with Gentoo, another one with Mandrake and one done with Slackware
13:27.29Ariel_none is the same there all messed up by this customer.. argh
13:27.35rikstaiCEBrkr: the old getk2-perl version at adm.hamnett.org has a flash video example
13:27.48tzafrirThis is the problem of that customer, then
13:27.50rikstayou'll all get to try it next week anyway
13:27.54B0ngFrOgflorz no there are 5 sites that I want to be the same for ease of maint
13:27.55iCEBrkrriksta: People who deman Java are retarded LOL!!  Cuz I personally, HATE java. unless it's the project I'm working on for my Sidekick
13:28.36Ariel_tzafrir, not if I am going to be doing there network work.
13:28.36florzB0ngFrOg: Ah, so it's only for that particular application? IC, that's far more sensible indeed :-)
13:28.45rikstaiCEBrkr: java is perfect for this instance where it needs to work on all platforms, and anyway you wouldnt even be able to tell this app is java, because it uses the swt look and feel...i used to hate java too..i was quite ignorant
13:28.56tzafrirAriel_, why is that diversity there?
13:29.00iCEBrkrriksta: Rock'n
13:29.04B0ngFrOgcows fart tooo much
13:29.28Ariel_tzafrir, they have had 6 different people work on there system in the pass 6 months.
13:29.39rikstaiCEBrkr: you can hate java.....but i could easily see you ending up using my application :)
13:29.45iCEBrkrriksta: LOL!! Please tell me you don't have that RegExp box in the Java version?
13:29.53Hmmhesaysi bet not one of them documented a damn thing
13:30.01rikstaiCEBrkr: no that version was for my own personal use
13:30.05tzafrirriksta, ther eare quite a few cross-platform toolkits. good ones, that is
13:30.17Ariel_OK Frist firewall/router going to be changed to m0n0wall, 2nd all others are going with CentOS.....even the Asterisk box.
13:30.18rikstatzafrir: i had to use SWT because ADM uses the system tray
13:30.25rikstait's very difficult to handle
13:30.34*** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.res.rr.com)
13:30.34*** mode/#asterisk [+o anthm] by ChanServ
13:31.03iCEBrkrriksta: Java's runtime just sucks cuz it's slow as shit to start up.. But like I said, it's fine on my sidekick.  I wrote an app for my phone in Java, kinda painful, but it works well
13:31.33rikstaiCEBrkr: yeah i'll agree with you totally on that. but like i said, there is no more perfect language for this particular application
13:31.57rikstait doesn't have to be ligtening fast to fire up, and once it's running its fine
13:32.04*** join/#asterisk MikeJ[Laptop] (~ircatjerr@mi.origenfinancial.com)
13:32.08iCEBrkrriksta: Not to bash your manager, but I think a lot of these call managers suck ass.. At least yours has a nice interface.
13:32.34rikstaiCEBrkr: i know, im fully aware of that, the thing you are looking at right now is totally different this java version is 10x cleaner
13:32.37rikstapromise :)
13:32.42iCEBrkrI don't think most of these programmers who write call managers know what a end-user call manager is supposed to do.
13:32.45rikstaanyway wait till i release it then you can bash me :)
13:32.51iCEBrkrNice!!
13:32.55iCEBrkrhaha
13:32.56Hmmhesaysfop does a pretty good job
13:33.02iCEBrkrHmmhesays: who?
13:33.12Hmmhesayswww.asternic.org
13:33.19Hmmhesaystis what I use
13:33.28rikstaiCEBrkr: i got some really nice simple interfaces, for managing the automatic lower volume while on call and stuff
13:33.31iCEBrkrOh god. not this piece of shit
13:33.34rikstaits got some really really nice features
13:33.38rikstaFOP is bad news
13:33.48iCEBrkrFOP is junk
13:33.56rikstai'm convinced that ADM will be a really solid, simple piece of s/w
13:34.02rikstathats the aim anyway
13:34.08syleJul 19 06:30:33 NOTICE[9624]: rtp.c:281 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389).  Please turn off on client if possible
13:34.15syle=how do i go about doing this
13:34.20Hmmhesays<shrug> fop works fine for what I use it for
13:34.46iCEBrkrriksta: Check out my prototype..
13:34.48iCEBrkrriksta: http://tinyurl.com/8dvfp
13:35.02iCEBrkrHmmhesays: It takes up the entire screen!!!
13:35.04rikstalookin
13:35.10iCEBrkrHmmhesays: What if you have 500 extensions?
13:35.15rikstafor starters FOP is FLASH, and web based ;0
13:35.22iCEBrkrwhat riksta said!
13:35.29Hmmhesaysdoesn't have to take up the entire screen
13:35.40Hmmhesaysand you can change button size
13:35.42iCEBrkrHmmhesays: Prove it.
13:35.47rikstaahh iCEBrkr i used a program exactly like that for windows in a company a while ago, it was by intertel
13:35.50riksta:)
13:35.53cypromiswhat if you don't want every user to be able to do all the stuff ?
13:35.59rikstabut basically my app is providing more or less all this functionality too
13:36.05iCEBrkrriksta: Shoretel maybe?
13:36.06rikstacall histories n such
13:36.13rikstano intertel definitely
13:36.16Hmmhesaysheh, you can change the entire layout, nothing is set in stone
13:36.17iCEBrkrok
13:36.39rikstayour app looks good, i have all those features in my todo, nearly done with most
13:36.43DannyFiCEBrkr, looks nice
13:36.50rikstaiCEBrkr: i take it you arent x-platform
13:37.24DannyFiCEBrkr,  does it work reliably?
13:37.30iCEBrkrHmmhesays: I shouldn't have to change the layout..
13:37.32*** join/#asterisk likwid-- (~likwid@nc-205-240-44-39.dyn.sprint-hsd.net)
13:37.39iCEBrkrriksta: Yea, unfortunately, it's for Wintendo.
13:37.44rikstaoh well :)
13:37.50DannyFdarn...
13:38.02HmmhesaysiCEBrkr the point is, you can.
13:38.02iCEBrkrDannyF: That version in the screenshots worked pretty good.  But I lost the sourcecode.
13:38.20DannyFanyone know what happened to the leo project?
13:38.22iCEBrkrHmmhesays: again, who wants to dick around with layout?  Sysadmins and shit like that can hardly install Windows.
13:38.28blopwhich port is used for provisionning an iaxy ? can i do that through internet ?
13:38.36rikstaoh well, im off to do some more work....keep a lookout on the lists for ADM next week sometime....i'd appreciate some feedback
13:38.36HmmhesaysiCEBrkr: again, not the point
13:38.48DannyFblop aint that in the iax_pro*.conf?
13:38.53iCEBrkrHmmhesays: It's the point of making a usable, scalable and deployable product.
13:39.06iCEBrkrOtherwise, it's a waste of time.
13:39.09Hmmhesaysheh, gotten completely off the original statement
13:39.15nDuffiCEBrkr, hmm, cute. How does it integrate w/ *? (Having software that needs admin access to the * server makees me wary).
13:39.17DannyFdid the heat issues with the iaxy get fixed btw?
13:39.33rikstanDuff: via asterisk manager api i presume
13:39.37blopmaybe on the new model
13:39.47iCEBrkrnDuff: It uses the Asterisk Manager port.  Well, a Proxy to the port.
13:39.51DannyFold iaxy was scary
13:40.08HmmhesaysiCEBrkr, you said 'it takes up the whole screen', I said 'it doesn't have to' ... moving on.
13:40.15iCEBrkrHmmhesays: Ok, so it works for you. It won't work for 90% of the others.
13:40.21blopi got an old one , running fine
13:40.33rikstait's just a bit messy and a bit AOLish (FOP)
13:40.36iCEBrkrHmmhesays: screenshot your control panel and lemme see how 'cool' it is.
13:40.51Hmmhesayslooks just like FOP, but the buttons are resized
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13:41.00nDuffthat way the gateway itself has admin privileges, but the software on users' desktops doesn't
13:41.07Hmmhesaysyou seem to think i'm claiming something I'm not
13:41.18iCEBrkrriksta: You can't knock FOP's clean UI, but it's just too damn big and I shouldn't have to customize it.  Desktop realestate is expensive.
13:41.36iCEBrkrHmmhesays: I'm just waiting to see proof of your statements.
13:41.38rikstayeh it is, i wouldnt say the UI is nice tbh
13:41.44Hmmhesaysproof that you can resize the buttons?
13:41.58iCEBrkrHmmhesays: how do you resize the buttons?
13:42.15iCEBrkrSo I have to go in and modify Actionscript?
13:42.30Hmmhesayssizes for all of the objects are in op_style.cfg
13:42.35*** join/#asterisk ellvis (~ellvis@adsl-data-148.84-47-83.telecom.sk)
13:42.39ellvishi people
13:42.51iCEBrkrThat's the other thing.  Textbased config files.. UGH..
13:42.53rikstaiCEBrkr: laters.
13:42.56iCEBrkrWhen will they go away.  There n
13:42.59iCEBrkrriksta: later
13:43.08iCEBrkrriksta: lemme know when you get that call manager working :D
13:43.18rikstait'll be on the lists
13:43.24iCEBrkrcool, I'll check it out.
13:43.31iCEBrkrEven if it IS java :D
13:43.31rikstaor sign yourself up to watch it o feshmeat its called ADM in projects
13:43.44rikstaon my god what happened to my typing then?
13:43.44ellvisi am trying to dial a number over isdn line, but i don't know how to set-up the Dial command correctly. i am abble to use only extension 's' and don't know how to change it.
13:43.48rikstas/on/oh/
13:43.50ellvisanyone can help me, please?
13:43.57DannyFriksta, know the feeling ;))
13:44.28ellvisit mean, that it's always dialing 's' and no number
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13:44.40iCEBrkrMy Call manager lowers/Raises WinAmps volume on inbound/outbound calls.
13:44.42Hmmhesayshttp://www.voip-info.org/wiki-Asterisk+cmd+Dial
13:44.59rikstaiCEBrkr: mine raises/lowers the mixer you select in the prefs dialog :)
13:45.06ellvisHmmhesays: i need info about DOD
13:45.19ellvisHmmhesays: and there's nothing about it on wiki :(
13:45.22rikstagradually :)
13:45.23iCEBrkrriksta: I wanted to do that, but I want to keep the system volume up so I can play ring.wav into programmers headphones :D
13:45.42nDuffiCEBrkr, lots of us silly UNIX-background types think text-based config files are a Good Thing, and are waiting for the fad of using anything *else* to go out of style.
13:45.44rikstacant you get asterisk to send the beep
13:45.51rikstait does if you use queues
13:46.23iCEBrkrriksta: The phone is hung up.. the phones volume is turned off, and it rings through the speakers.
13:46.35iCEBrkrriksta: I'm a programmer, I jam with my headphones on, I miss calls if I don't see the thing flashing.
13:46.51iCEBrkrSo I wanna make it ring through the speakers/headphones, whatever.
13:46.54rikstai get ya
13:46.57iCEBrkrSo I just lower the sound of Winamp
13:47.01rikstayou could have a popup balloon
13:47.11iCEBrkrWell, the app will pop/slide up on inbound call too
13:47.15Hmmhesaysyou could have it turn your screen upside down
13:47.21iCEBrkrHmmhesays: LOL
13:47.39Hmmhesaysthat'd be pretty hard to miss
13:47.42rikstayeah mine is doing that, and cuz i have a cisco 79xx i can telnet into it and make it press the speakerphone key, that way the popup comes up and  i can click ANSWER
13:47.45rikstaand i can use handsfree :D
13:47.46iCEBrkrriksta: I'm modeling this thing off the Shoretel call manager.
13:47.48rikstasweeet
13:47.59iCEBrkrhaha
13:48.58rikstabetter go
13:48.59rikstapz
13:49.22*** join/#asterisk The_X (chris@true.fiberpimp.net)
13:49.30The_Xwhat is asterisk call manager for?
13:50.14HolosThe_X: Managing calls?
13:50.15*** join/#asterisk D1ng0 (~D1ng0@210.213.168.137)
13:50.22iCEBrkrThe_X: Basically, I hate touching my phone unless I have to.
13:50.31*** join/#asterisk RooTchO (~BrainBug@fw2gw1.eprogress.bg)
13:50.35iCEBrkrThe_X: It's nice to have a pop-up with CallerID on my desktop so I don't have to look at my phone.
13:50.37HolosThe_X: You can see what is happening with calls, as well as place calls with it.
13:50.49*** part/#asterisk RooTchO (~BrainBug@fw2gw1.eprogress.bg)
13:51.02rikstaiCEBrkr: yeah i got the same thing rockin here
13:51.05The_Xshould I waste time with it?
13:51.06iCEBrkrThe_X: Also, having a "Transfer to VM' button for when you can't take the call is nice.. not having to take your hands off your keyboard/mouse
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13:51.26iCEBrkrriksta: Where the hell is the projects page?!?! I can't find it
13:51.29rikstai got a sliding popup, with Answer / Busy
13:51.39rikstahttp://freshmeat.net/projects/adm ?
13:51.44iCEBrkrriksta: I like the AIM 'away message' thing
13:51.47iCEBrkroh.
13:51.55rikstaiCEBrkr: yeah that is a kool feature
13:51.58iCEBrkrriksta: I'm looking on voip-info.org
13:52.03rikstai'm most proud of the bluetooth presence
13:52.13iCEBrkrYeah, I know a lot of people who'd use that.
13:52.17rikstaas soon as you walk out of the office, your box knows to redirect your calls to your cell (if you told it to)
13:52.25iCEBrkrYup yup!
13:52.34ellvisok, i've got it
13:52.35ellvisciao
13:52.49iCEBrkrellvis has the clap?
13:52.54rikstaalso i'm planning on being able to retreive phonebook data from the call to use for the caller id's if it cant be matched locally
13:53.01rikstas/call/cell/
13:53.09rikstableh
13:53.18iCEBrkrriksta: I haven't engineered my phonebook yet. Not sure how I'm gonna do that.
13:53.44rikstawell, with my p800 cellphone, i can do a remote sync, which dumps the vcards into a mysql db
13:53.44iCEBrkrFirst thing that comes to mind is LDAP.  But LDAP administration sucks ass
13:53.44coldfeetguys has anyone installed the bristuff patch
13:53.59rikstaiCEBrkr: i'm going to be supporting mysql/flat file/vcard/ldap
13:54.12iCEBrkrriksta: Yea, MySQL was my next option.
13:54.22rikstavcards are nice, you can have binary photos attached
13:54.24iCEBrkrJust have to figure out what's the least painful to admin.
13:54.28rikstapop up your g/fs face when she calls
13:54.31iCEBrkrhehe
13:54.42rikstamind you, i wouldnt want to pop up YOUR g/f's face :)
13:54.45rikstalol j/k
13:54.49iCEBrkrriksta: you gotta think about company-wide type stuff. Not just your own personal phonebook
13:54.57rikstaiCEBrkr: i know...hence ldap
13:55.00iCEBrkrI don't have a GF :(
13:55.03rikstait's all in my todo
13:55.06iCEBrkrhehe
13:55.18rikstasrt: is helping majorly on this project
13:55.21rikstahe is being my mentor
13:55.25riksta:)
13:55.29iCEBrkrCool
13:56.07iCEBrkrDamnit, I can't get to my machines at my apartment. :(
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14:21.47tzafrirI'm trying to configure an E100P ccard we have here. Is the dchan channel no. 16? not channel 31?
14:21.47MattB2astoria... you about?
14:21.53astoriaMattB2: yo yo
14:21.58astoriaMattB2: got my echo problems fixed
14:22.16MattB2cool
14:22.18MattB2how did you do it
14:23.17astoriaI was editing the wrong file before (zaptel.h) rather than zconfig.h.. so i changed it back to it's original state and made the requisite changes to zconfig, and it seems to have solved my problem so far.
14:23.31MattB2ah
14:23.31MattB2cool
14:23.41MattB2think i got my ech oproblems solved as well
14:23.46astoriaI've take probably a dozen calls that typically would have had echo, and it trained the cancellor within the first second.
14:23.50MattB2really weird - io started from scratch, ie turned off all echo cancellation and training
14:23.56MattB2resorted back to mark2 echo
14:23.59MattB2then turned on echocancel
14:24.01MattB2and it's great
14:24.07astoriaYeah, thats pretty much what I did.
14:24.12MattB2i think the big difference is that i do NOT have echocancelwhenbridged turned on
14:24.16astoriaThe AGRESSIVE seemed to solve my problem.
14:24.23astoriaI do not either.
14:24.28MattB2agressive was causing the sound to cut out on mine
14:24.35MattB2but the bridged thing solved it
14:24.37MattB2so i'm happy :D
14:24.46MattB2just got to figure out how to stop our damn grandstreams from crashing now!!!!
14:24.49astoriaI don't really care, but my co-workers are happy.
14:24.53MattB2hehe
14:24.57astoriawe all have polycom ip500s here.
14:25.23MattB2we went the GS route for cost!
14:25.38MattB2they're great 99% of the time, but occasioally one will just crash and it'll need a reboot
14:25.41astoriaHmm. Probably cost you more in time in the long run.
14:25.52astoriaPolycoms occasionally need a reboot as well.
14:26.11astoriaI wish there was a way I could just cron reboot them all from the boot server.
14:26.19MattB2i can do that with the GSs
14:26.27MattB2ni fact i did that every night when we had registration problems
14:26.36lathos42astoria: You could always cut main power to the building
14:26.36astoriaWhoa, thats cool.. There's probably a way to do with the polycoms, but I haven't figured it out yet.
14:26.37MattB2linux script that logs into the html page and runs the reboot
14:26.39lathos42:)
14:26.39tzafrirastoria, why can't you?
14:26.48iCEBrkrMattB2: I probably would have gonne the Sipura route :D
14:26.54astoriaMattB2: oh yeah, didn't think about that.. thats an easy perl deal
14:26.57MattB2we had two sipuras in our UK office
14:26.58MattB2they're terrible
14:27.00MattB2very quiet
14:27.03MattB2awful sound quality
14:27.07iCEBrkrMattB2: What?!?!?!
14:27.13MattB2been back to the suppliers once
14:27.19*** join/#asterisk coppice (~chatzilla@161.195.17.210.dyn.pacific.net.hk)
14:27.24iCEBrkrMattB2: There are gain control options in the sipura.. and don't just JUNK analog phones :D
14:27.33astoriaanalog phones are sooo 1986
14:27.34iCEBrkrI love my Sipura2ks
14:27.38iCEBrkrastoria: hehe
14:27.47MattB2where are the gain controls?
14:27.52astoriaI have had great experience with sipura 1xxx series.
14:28.06MattB2sipura ip phones that is
14:28.08MattB2we don't use ATAs
14:28.23astoriaha ha. I could make an extension that would reset all the phones.. ha, that would be sweet.
14:28.24iCEBrkrMattB2: I wish I could tell ya, but I don't have access to mine at the moment, my cable seems to be on the fritz in my apartment and I can't get into my machines there
14:28.25MattB2we had no analog phones in the beginning
14:28.28lathos42I have a SPA-841 here that we bought for testing.. and my biggest problem with it is that there isnt a dedicated button for transfers
14:28.30iCEBrkrMattB2: Ohhhhhh!
14:28.48MattB2yeah 841s are the ones we bought
14:28.52iCEBrkrMattB2: The Sipura2k ATA just rocks.
14:28.56*** join/#asterisk Katty (~angela@68.112.15.110)
14:28.57astoriaextension 666 kills all the phones :)
14:29.04MattB2;)
14:29.13MattB2also i hate the screen on the 841 - no backlight?
14:29.18iCEBrkryuck
14:29.19Kattymew
14:29.22MattB2unless that's another hidden option!
14:29.24syleyeah but the linksys pap2-na is cheaper :)
14:29.33Hmmhesaysyeah its called a mag lite in your pocket
14:29.47KattyMEW
14:29.47lathos42MattB2: If you find that option, let me know :)
14:29.47MattB2hehe very useful :S
14:30.01MattB2i love the GS GXP2000s - big bold screen, nice backlight
14:30.09MattB24 SIP channels
14:30.19Kattyk, all better
14:30.25MattB2and GS are pretty good at fixing bugs and glitches, and even adding new features on our request
14:30.38MattB2and they're not as ugly as many other phones!
14:31.03iCEBrkrMattB2: I just hope that phone is better than the K-Mart special Budgetone 101
14:31.08MattB2;)
14:31.11MattB2we have a bunch of those as well
14:31.20MattB2el cheapo
14:31.27MattB2for the people who rarely make/receive calls - they do the job fine
14:31.36iCEBrkrMattB2: I mean, it's a phone to tinker with.. ya know for the VoIP hobbiest.. But I wouldn't put those on users desks.
14:32.02MattB2once GS sorted the registration problem, the BT101s have been solid & reliable
14:32.05syledefine hobbiest
14:32.19MattB2and they have a backlit screen!!!
14:32.19syleis there anyone actually not making money off it
14:32.39lathos42I was going to try out the Uniden UIP-200 until I found out it only has one call appearance
14:32.48astoria<-- polycom fanboy
14:32.55MattB2oh btw
14:32.56iCEBrkrsyle: Hobbist.. Meaning, someone like me who has Asterisk setup as a glorified answering machine at his house.
14:33.00MattB2can anyone recomend a good conference phone?
14:33.10MattB2not so bothered about ip conference phone coz of the price so analog with ATA will do fine
14:33.15jake1932cisco 7960
14:33.23jake1932ok
14:33.23iCEBrkrsyle: Who'd spend $200 on a Polycom phone just to test and play with VoIP???
14:33.28astoriai DID!
14:33.36jake1932me 2
14:33.37MattB2like the look of the polycom 2w ex or whatwever it is - the wireless one, as we have a shared conf room in our building the wireless would be v useful
14:33.41sylehaha me to, complete with ivr answer, and music on hold for dialing with some kick ass mp3's lol
14:33.45iCEBrkrastoria: Dummy.. Now throw your money at me, instead of in the wind!!!
14:33.59jake1932actually not a hobby
14:34.03astoriaiCEBrkr: i wanted to test with something that I would be installing for my clients, so I knew what it was doing..
14:34.17iCEBrkrastoria: See, that's different.. That's a business investment.
14:34.18jake1932and the phone is the only one that actually works well with asterisk
14:34.37astoriajake1932: thats a strech, lots of phones work great with *
14:34.41iCEBrkrastoria: I'm playing with Asterisk cuz it's fun.  I'm a geek, and I want to explore VoIP stuff. :D
14:34.41jake1932tried the 841 and it sucked
14:34.54jake1932spa 300 wasn't good either
14:34.57jake19323000
14:34.59iCEBrkrastoria: Still, you can thrown any extra money my way. I'm trying to get a downpayment on a house :D
14:35.16iCEBrkrastoria: http://www.cyberdyne.org  Please donate!
14:35.18iCEBrkrLOL
14:35.48astoriaHa ha, I never said I'm rich! I just have credit card debt!
14:35.54lathos42Is the IP301 worth looking at, or should I go right to the 501?
14:35.55iCEBrkrOh.
14:35.56jake1932anyone else sharing my luck with Sipura?
14:35.57sylei';m working with ibell.us right now, call seems to go through every 4th time if i;m lucky, no callerid passthrough, stuck with g729 only, and now i find out they don;t even run asterisk yet are on the asterisk site listed as a provider
14:35.57iCEBrkrhaha
14:36.06iCEBrkrjake1932: ???
14:36.30Holoslathos42: Do you need speakerphone duplex? or listen only?
14:36.32iCEBrkrsyle: Why not NuFone or VoicePulse?
14:36.47sylebecause they do 1 cent a minute
14:36.54sylecheapest i;ve found so far
14:36.54astoriaYou get what you pay for...
14:37.01lathos42Holos: I believe our current phones are Duplex, so probably that's what they'd want to stay with
14:37.09sylewell i tend to search the net alot and find the cheapest and try them all out
14:37.26iCEBrkrsyle: You get what you pay for :D
14:37.29greg_workjake1932: i have 841's and a 2000 .. the 2000 is great, the 841s are .. okay..
14:37.32iCEBrkrwhat astoria said heheh
14:37.43Holoslathos42: 501 then.. Get the 1.5.x firmware and you're set.. i'm on a 501 right now and I love it.
14:37.46sylevoipjet i tried yesterday, and they are good cept they seem to hangup on you after 5th ring to a person
14:37.53Holoslathos42: You in Canada?
14:37.57iCEBrkrsyle: VoicePulse is a whopping 0.24/min OH NOZ!
14:38.12lathos42Holos: Nope, the US, Michigan to be exact
14:38.20syleicebrkr : and where am i gonna profit selling SIP accounts at that rate?
14:38.49iCEBrkrsyle: Where are you going to get customers if you can't provide a reliable service?
14:38.53astoriawell presumably if your selling sip accounts, you should probably be offering pstn termination, not just forwarding sip/iax..
14:39.14Holoslathos42: Lots of US Polycom resellers.. Get with a good one and they'll keep you updated with firmware and updates. I also got two 501s to play with and demo before placing any order.
14:39.15astoriayou don't want to be the guy that takes the heat when your provider screws up..
14:39.21sylei do pstn termination for inbound and local but for long distance calls i;m talking about
14:39.25iCEBrkrsyle: what astoria said.
14:39.40astoriasyle: oh, that makes sense.. especially for outbound, where you can re-route if something goes down..
14:39.51astoriasyle: sorry. you're right :)
14:39.53jake1932my 841 - the volume goes up and down constantly... maybe it's a feature (AGC) but it's RFA (real f-n annoying)
14:40.08jake1932the 300 detect's my voice as touch tones
14:40.11jake19323000
14:40.21MattB2must have a good singing voice
14:40.24jake1932i called tech support and e-mailed and they don't respond
14:40.41jake1932guess they're too big now
14:40.51lathos42Holos: I'm hoping i'll find someone that will treat me good, and give me a good discount for a purchase of 70 or so phones :)
14:41.22jake1932actually had better luck with the DTA310 adapter ($30)
14:41.43*** join/#asterisk mut (~animenodv@65.111.201.79)
14:41.53Holoslathos42: Go with a big reseller.. We went with Bell Canada, who resells for CCPin.com and they gave us a great price $10 more then CCPin's cost, demo units, and quick service.
14:42.57sylelathos
14:43.00jake1932guess you get what you pay for - the 7960 has given me 0 problems once it was setup
14:43.14sylei know a guy that can get you half prices on cisco stuff with that many phones hehe
14:43.26Juggiejake1932, 7960 is a fustrating phone
14:43.36Juggiei prefer mitel 5220's
14:43.39jake1932frustrating in what way?
14:43.43lathos42syle:  Does the sales meeting start with "psst, over here in my trunk"? :)
14:44.01Juggiejake1932, nuisance to upgrade... they have problems with tftp...
14:44.05greg_workHolos: i have horrible stories about bell canada
14:44.10sylehaha no it starts with a conferance call to cisco;s sales dept
14:44.34jake1932juggie - ok - that's understandable - but in regular use - it's been solid
14:44.50Juggiejake1932, thats fine, i have one on my desk too, but i would not want to manage 100's of them
14:44.55jake1932i'd rather have a solid phone than a bunch of features
14:44.57Juggiethe mitel however is a dream :)
14:45.10jake1932is the mitel as good quality?
14:45.13Juggieyes.
14:45.23jake1932have you used the SIPURA 841?
14:45.29af_what should I put as "Outbound proxy" non GXP2000?
14:45.30Juggiethey have been making business phones and switches for a long time
14:45.31JerJer[mobile]simply write a perl script and web front end to manage 79XX devices
14:45.36Juggiejake1932, no
14:45.37JerJer[mobile]it is not hard if you know WTF you are doing
14:45.48greg_worklike how they suddenly tell you "you can't cancel, you have a contract" .. "oh? can you fax it over, i don't seem to have a copy" .. "uh, no, we can't do that"
14:46.01lathos42syle: Well, my boss really would like to get Cisco phones, but the initial cost estimates put it a little high..  we're competing with the fact that we can buy our current phone system from Avaya for a little over $10k
14:46.10greg_work"oh, wait, it was a verbal contract" ... "with who?"  .. "we don't have that information"
14:46.13*** join/#asterisk gaggaman (~leo@host-82-135-28-39.customer.m-online.net)
14:47.07Holosgreg_work: For buying phones?
14:47.11gaggamanHi everybody!
14:47.15jake1932i'm going to check out the mitel 5220
14:47.31MattB2af_: af_: the ip of asterisk
14:47.35Juggiejake there is the 5220 and 5215 not sure what the dif is
14:47.36*** join/#asterisk PBXtech (~nik@67.107.241.3.ptr.us.xo.net)
14:47.45Juggiemake sure you get the dual mode phone...
14:47.58greg_worki've heard that happen to other people though .. it will never go external, (ie, they won't take you to collections or sue you) but they will come after you via subsidiaries
14:48.03*** join/#asterisk Newbie___ (me@211.24.146.12)
14:48.06greg_workHolos: no that was for DSL
14:48.10jake1932yep - will do
14:48.17Juggiethe original model only supported mitel protocol, but the dual mode supports minet & sip
14:48.33PBXtechanyone running a soekris 4801 board?
14:48.38greg_worki know of a company that had a T1 with them, and same thing happened. except to them, bell cut off all their phone lines and cell phones
14:48.41Juggieit does ulaw/alaw/g729
14:48.42jake1932right - the 5220 looks like it's the one
14:48.42darkskiezso, two phones on my TDM card have gone down, they now ring, but keep ringing, with clicks thru the speaker when they are picked up
14:48.56Newbie___hi, how do i disable the default ringing tone generated by asterisk ?
14:48.59greg_workwe didn't have any other srevices with bell, so nothing happened
14:49.05darkskiezand I dont want to powercycle the server as its got a PRI line too being handled
14:49.11astoriadarkskiez: sounds like some kind of voltage issue.
14:49.20greg_workeventually a manager called (months later) and said "we want your business, how about this, we'll waive this outstanding balance..."
14:49.23Juggiethe only feature the cisco has that the mitel doesnt, is the ability to register lines against more then one server
14:49.26darkskiezast: they worked fine for a few weeks
14:49.36Juggiebut, i dont consider that much of a feature anyways.
14:49.44greg_workso how legit is it if 1) they never came after us for it, and 2) they just waived it without a second thought  ?
14:49.48darkskiezast: they receive the callerID, but dont seem to register being picked up
14:49.49jake1932newbie - ring out tone?
14:50.02astoriadarkskiez: that doesn't mean something than go wrong. You're probably going to want to powercycle and see if that works.
14:50.45Newbie___jake1932: when i dial out ie to any country, i am getting a standard ringing tone. not the country's PSTN tone
14:51.39jake1932ok - i would think that is passed by the VOIP provider
14:52.24jake1932provider sends trying, then connect once it is connected
14:52.29Newbie___thats what i though, but they said i am the only one having the problem.
14:52.41Newbie___they said something like i am not requesting the tone from asterisk
14:52.56Newbie___since there is no request, a standard tone is provided
14:52.58gaggamannewbie r u using a sip phone on * ?`
14:53.26Newbie___not really, is SIP to h323 signalling
14:53.34Hmmhesaysfun times
14:53.48Newbie___asterisk -> SIP -> h323 -> telco
14:54.00gaggamantelco via POTS?
14:54.09Newbie___no, telco via h323
14:54.11Hmmhesaysh323 termination
14:54.18Newbie___i am getting a 180 ringing from SIP
14:54.26*** join/#asterisk mariogamboa (~sudaikdd@201.138.151.155)
14:54.31mariogamboahi all
14:54.40Hmmhesayshola mariogamboa
14:54.49mariogamboahola hmmhesays
14:54.51mariogamboahe he he
14:54.53Newbie___even when the phone on the other side is busy, no busy signal, and it keeps ringing
14:55.54Newbie___there was a post like this in asterisk mailing list, but no answer
14:56.08mariogamboaany here have a phone it have the ignorepat function build-in because i have grandstream and the support say me is not supported for this moment
14:56.18Hmmhesaysthe majority of the people who use asterisk don't use h323
14:56.32MattB2or did use h323 and ran away crying
14:56.32mariogamboaonly for gatekeeper
14:56.36mariogamboa:)
14:56.37PBXtechis putting 2 TDM4 card in a system to much load?  (ie 8 FXO ports)
14:56.48Holosgreg_work: Dealing with a small time call center agent, and a dedicated sales rep for your area is night and day. If you are placing a large order with them (20k - 100K) your service level goes way up!
14:56.52mariogamboano
14:56.56HmmhesaysI use it in a few cases, but don't have any real problems
14:56.58*** join/#asterisk mkrufky (~mk@68.160.103.77)
14:57.01Newbie___i know, but telco ONLY insist H323 beacause they do not know SIP
14:57.15Hmmhesaysso... get a new one
14:57.17*** join/#asterisk Pcharky (~khagen@cust.12.215.adsl.cistron.nl)
14:57.22Hmmhesaysthere is no shortage of termination services
14:57.23mariogamboayep
14:57.34Newbie___i use yate for the SIP-H323 signalling, pretty neat
14:57.35mariogamboaupgrade
14:57.42Pcharkyhi
14:58.01Hmmhesaysare you married to the owner of this telco or something?
14:58.11mariogamboamaybe i think hmm
14:58.16Newbie___Hmmhesays: lol, is easy is states, but not here
14:58.18mariogamboahe he he
14:58.29HmmhesaysNewbie___ why not there? no paypal accounts?
14:59.00Newbie___beacuse we have a close link, PING time 83ms
14:59.14Newbie___companies in states is like > 200ms
14:59.19HolosPBXtech: 8x fxo's shouldbe ok, you may have glitches under full load, but from what I hear that is the limit before going to a channel bank. Would a fractional t1/e1 circut not be cheaper in your area?
14:59.20Hmmhesayswhere are you?
14:59.23Newbie___malaysia
14:59.44Hmmhesays200ms isn't to bad
14:59.56*** join/#asterisk dasenjo (~dasenjo@63.245.86.173)
15:00.10PBXtechpossibly thx
15:00.14*** join/#asterisk sig- (~sig@gnook.org)
15:00.49MattB2talking of machine specs, is there a recmomended spec for certain tasks
15:00.52Newbie___been seaching for the stupid ring tone thing for the last week, i think i have search everything in google
15:00.54Newbie___haha
15:01.04MattB2ie spec for tdm400p, 4 fxo, 3x IAx connections, 20 SIP phones etc
15:01.18mariogamboacool matt
15:01.22MattB2there are vague specs on the wiki but nothing very quantitative
15:01.29darkskiezmy * isnt detecting two phones going offhook for no reason all of a sudden :/
15:01.52darkskiezI've restarted *, but not the whole server, I cant afford that downtime at the moment
15:02.02Newbie___darkskiez: it happens to me all the time. SIP phone i mean.
15:02.09Hmmhesayscould probably get by with a p4 2.8ghz machine with 1-2gigs of ram
15:02.31MattB2darkskiez: have you rebooted the phones?
15:02.33darkskiezNewbie___:  its not SIP tho, its a TDM analogue board, with 3 ports
15:02.42gaggamanthe ring tone is generated by the phone, so your problem is that you get no or wrong busy signalling on the asterisk<->h323 side.
15:02.46MattB2Hmm: so my PIII-450 with 256MB isn't up to par then?!
15:02.58limbiquecya
15:02.59iCEBrkrMattB2: hehe
15:02.59Newbie___darkskiez: i am doing E1, never try TDM
15:03.20MattB2icebkr i'm being serious!!! it runs fine except we had a conference with 3 external calls and 3 internal and it buzzed like mad
15:03.21darkskiezthe fax is on one, and it works fine, but the two other ports dont notice the phones being picked up
15:03.22HmmhesaysMattB2 you could get by with that if you don't overload it with calls, lol
15:03.24MattB2apart rfvom that it's solid!
15:03.29darkskiezNewbie___: i'm got an E1 too
15:03.35darkskiezI've got...
15:03.40HmmhesaysI can get 2-3 sip calls on my p233
15:04.14Newbie___i have to beg the telco to sell me E1
15:04.26darkskiezcan I restart the zap card ?
15:04.42*** join/#asterisk jansen (user@p54879634.dip0.t-ipconnect.de)
15:05.09darkskiezif I use the ztcfg to shutdown, it will take down both the TE110P and the TDM400 and wont bring them back till after  a reboot
15:05.22Newbie___guys, know any used Voip with 2 E1s that i can buy ? dont tell me ebay, there is none
15:05.45darkskiezNewbie___:  what are you looking to buy?
15:06.01darkskiez"Used voip with 2 E1's"  means nothing
15:06.06Newbie___2 E1 with SIP and h323 capabilities
15:06.29darkskiezthats still not a noun
15:06.38Newbie___errr
15:06.48lathos42So am I asking for trouble thinking that I can run an office of 70+ phones on a Dual Xeon 3.0 with 1-2GB of RAM? :)
15:06.53Newbie___looking at sysmaster
15:06.56HolosNewbie___: You mean a server, that can run linux (or BSD), with two 1x T1/E1 cards or 1x 2-T1/E1 cards?\
15:07.32Newbie___i have that
15:07.45HolosNewbie___: So what do you need?
15:07.46Newbie___something like clarent 100
15:07.50Newbie___sysmaster
15:08.09darkskiezyour in the wrong place
15:08.26Newbie___darkskiez: just trying my luck, since most of you in telco business
15:08.41tclarklathos42: your ok, depending number of concurrent called & you codecs, 70 all ulaw no issue there at all, 70 concurrent all g729 might be an issue dpending on what else the box is doing
15:09.40lathos42tclark: Well, it'll definitely be a dedicated Asterisk server.. and We'll have one ISDN PRI.. in a company that gets by with about 12 analog lines at the moment
15:09.45Darwin35ok whats going on here
15:09.46syledoes asterisk make good use of SMP?
15:10.17gaggamanlathos as you are on ISDN, you will use ulaw anyway.
15:10.29PcharkyHello, does anybody use a HFC card?
15:10.31gaggamanso there shouldn't be a problem.
15:10.39coppicesyle: yes
15:10.41tclarkso myu thta you been no more than 12 concurrent calls ?, well that box is over kill then eeven with g729
15:11.24lathos42Well, i'm thinking for future growth, and we have 2 plants, so our employees do call each other quite a bit
15:12.05sylewhats better the PAP2-NA or SPA-2002?
15:12.07Newbie___lathos42: i am using a Dell single xeon, 1GB RAM  1 E1 X 100 phone. never a problem since day 1
15:12.16*** join/#asterisk pingywon (~mike@pcp0010034410pcs.reding01.pa.comcast.net)
15:12.17HolosAre there any good SIP based utils. for testing how many calls can be sent across a single link without degrading quality?
15:12.23AhrimanesNewbie___: how many concurrent calls?
15:12.34Newbie___max 20-30
15:12.36gaggamanlathos * is just forwarding packets as long as you don't change the codecs.
15:12.39astoriaHolos: no but you could probably do the math yourself..
15:12.50AhrimanesHolos: yes, setup another asterisk server and script it to call up the other asterisk
15:12.59astoriaHolos: oh yeah.
15:13.23Newbie___and there are some ATAs sitting in remote offices
15:13.23tclarkyah we have predictive dial all ulaw sip voip termination 50 agents up all day on just p4 3ghz, tuned box
15:13.38AhrimanesNewbie___: ok, and what's the typical load on that machine?
15:13.45HolosAhrimanes: starting trunking calls back and forth eh... I'll try benching the line by ftp'ing some large files first.
15:14.15Newbie___Ahrimanes: i never check on that, sorry
15:14.26AhrimanesHolos: no not trunking.. dont use iax if you want to test sip, then sip register and dial sip/XXX - then no trunking is used
15:14.29AhrimanesNewbie___: ok
15:14.48lathos42That definitely sets my mind at ease that if anything the server should be overkill
15:15.46tclarkand on a dual p4 3ghz 120 agents up & 200+ channels peak prdictive dial time
15:16.25mariogamboaany here have a phone it have the ignorepat function build-in because i have grandstream and the support say me is not supported for this moment
15:17.05InfraRedi take it asterisk still doesnt support generating Remote-Party-ID
15:17.12InfraRedam i correct ?
15:17.20Ahrimanesremote-party-id ??
15:17.28InfraRedyes
15:17.42InfraRedit's like callerid
15:17.46Ahrimanesyou want to generate and id for the person calling in?
15:17.47tclarklathos42: and if your going digium & not sanoma for your t1 gear make sure you have a mother boards where you can control irq assigments to the pic slots
15:18.11InfraRedcalling out
15:18.12*** join/#asterisk redder86 (~lee@gateway.howardsilvan.com)
15:18.20InfraRedits the 'external' callerid number
15:18.23InfraRedfor outgoing calls
15:18.24lathos42tclark: I was thinking Sangoma for the T1 card, just for that very fact
15:18.42Ahrimanesthat's just callerid, it's up to your upstream provider if they pass it on or not
15:18.50InfraRedno
15:18.55InfraReddifferent thing
15:19.25Ahrimanesit sure is the same thing here.. my callerid is passed on to pstn no problem
15:20.15InfraReddepends on the provider
15:20.35InfraRedmine is asking for Remote-Party-ID
15:20.57*** join/#asterisk mrun (~Escape@85.14.32.34)
15:21.16Ahrimaneshm all providers i have worked with use callerid... but would be the same thing just with a different name, should be easy
15:22.03mariogamboastupid question asterisk can works with amd64?
15:22.53Kattyuhmm, yeah
15:22.55coppicemariogamboa: basically yes, except I am having trouble with an old Tormenta 2 card
15:23.03Kattythat's a silly question
15:23.05Ahrimanesmariogamboa: yes, there's been some work to make use of 64-bit even
15:23.18Kattypractically like all my questions
15:23.27mariogamboahe hehe
15:23.33Hmmhesaysha!
15:23.34AhrimanesKatty: dont mock him ;))
15:23.54Ahrimanesmm coffe
15:24.38Hmmhesaysthe guys who wrote lartc seem to have done a good job
15:24.43redder86I'm having a CallerID problem, too, and I've had it for a very long time.  I've got one PSTN T1 PRI connection to Asterisk which has a TE405P card in it.  I have SIP and IAX clients, and  I've got one of the other ports on the TE405P connected to another T1 device (could be a T1 modem or a T1 channel bank) that does not set CallerID.  So in the dialplan I set it with SetCallerID.  That seems to work fine for calls that go from that internal T1 to
15:25.28coppicetclark: I should ask sangoma for some cards. they keep offering them
15:26.08coppicespandsp seems to go like a rocket on my X2 machine. I can't figure out why
15:26.46redder86hi coppice
15:26.56tclarkyah i'd think dougv would drop few on you in heartbeat
15:27.49Ahrimanesredder86: you only described what works.. not a problem?
15:27.51tclarkwhat up with upencall site its up/down like a yoyo laste few weeks
15:27.56tclarkerr opencall
15:28.23PcharkyHello people, can i use an asterisk for an ivr/call-logger connected to an index-trunk, and having the index forward specific calls to the asterisk?
15:28.46redder86Ahrimanes: the CallerID that is set with SetCallerID when calls originate from the internal T1 is not being passed to the PSTN as is CallerID for calls that originate from SIP or IAX clients.
15:29.07Ahrimanesredder86: hm ok
15:29.46redder86Ahrimanes: I don't use SetCallerID with SIP or IAX clients because they "just work" with their settings in sip.conf/iax.conf
15:30.19redder86Ahrimanes: in fact, the CallerID works when the call goes from internal T1 to SIP/IAX.  It's just not working from internal T1 to PSTN.
15:30.20Ahrimanesredder86: ah yes, might it be possible to set callerid in zapata.cof then?
15:30.43redder86Ahrimanes: I tried setting it in zapata.conf and it did the same thing.
15:30.43Ahrimanesi dont have much experience with the zaptel bits of * yet
15:30.46Ahrimanesok
15:31.07coppicetclark: duh! opencall went down months ago
15:32.02redder86Ahrimanes: hrmmm... perhaps I need to speak with the telco about this.  Although, it would seem that Asterisk is treating T1-to-T1 CallerIDs different than SIP-to-T1.
15:33.00redder86coppice: did you recently check on the unicall t31 modem to see if it's still working or not?
15:33.03Ahrimanesredder86: yeah does seem to indicate something like that..
15:33.23coppiceredder86: nope
15:33.29tclarkhehe so what is the current site where your stuff is buried :)
15:33.31redder86coppice: okay :-(
15:33.32*** join/#asterisk mhnoyes (~mhnoyes@user-2ivfmv1.dialup.mindspring.com)
15:33.59coppicetoo many things on the go for anything to get finished right now :-(
15:33.59*** join/#asterisk outtolunc (outtolunc@adsl-66-218-53-170.dslextreme.com)
15:34.10redder86coppice: I understand that
15:34.20PcharkyHello people, can i use an asterisk for an ivr/call-logger connected to an index-trunk, and having the index forward specific calls to the asterisk?
15:34.41*** join/#asterisk badboyz (~badboyz@adsl-70-128-78-22.dsl.stlsmo.swbell.net)
15:35.04badboyzis there a way to get more details as to what the manager is doing?
15:35.07badboyzi see '  == Parsing '/etc/asterisk/manager_custom.conf': Found '
15:35.12badboyzi want to know what commands are being sent ..
15:35.47Pcharkybadboyz: add another v? ;-)
15:35.52Ahrimanesadd 10 :D
15:35.54tzafrirbadboyz, this is just the module reading its configuration at startup. nothing more
15:35.57*** join/#asterisk Dave-- (~a@choloconsultancy.plus.com)
15:36.33badboyzwell it occurs when i dialout from outlook using astapi
15:36.38tclarkhttp://www.soft-switch.org/ duh
15:36.44PcharkySo has anybody got a clue on the case I just wrote?
15:36.46badboyzand that pops up, id like to know what commands are being passed in
15:36.59badboyzpcharky asterisk -vvvv
15:37.02badboyzright?
15:37.13Pcharkybadboyz: I ques.. ;-)
15:37.16badboyzlol
15:37.30tzafrirbadboyz, 'set verbose 20' #or whatever
15:37.34tzafririn the CLI
15:37.39badboyzi cranked it up
15:37.43badboyzdidnt give me any more info
15:38.06tzafrirbadboyz, second place to look at: /etc/asterisk/logger.conf
15:38.21tclarkcoppice: still loads like its a web server running on a old 8086 web server
15:38.31mariogamboaguys i have a problem with the tdm400p and te110p in this moment upgrade my asterisk to 1.0.9 i see the 2 card is up because the green leds is on but in the tdm400 doesn't have tone for the analog phone what is the problem?
15:38.49darkskiezmariogamboa:  Thats just happened to me
15:38.53badboyztza: that whole file is commented out
15:39.10tclarkcoppise; and the mysql server dont work either :(
15:39.18badboyz.. except this line: full => notice,warning,error,debug,verbose
15:39.36tzafrirgrep -v '^;' /etc/asterisk/logger.conf
15:39.39darkskiezmariogamba: im running CVS Head from 1 June 05, its been fine for a month or so, then it just went bam
15:39.41*** part/#asterisk redder86 (~lee@gateway.howardsilvan.com)
15:40.04*** join/#asterisk angler (~angler@angler.digium.sponsor.pdpc)
15:40.22tzafrirbadboyz, also, read the comments. See if you can make anything useful there
15:41.35badboyzkk lemme look around
15:41.43*** join/#asterisk wunderkin (kev@12-215-218-160.client.mchsi.com)
15:41.49Juggiehey if anyone is intreasted in storeing voicemail in database, or correntally runs that configuration, can you test 4403.
15:41.50Dave--anyone here from the uk and had "fun" with the zapata confing getting inbound callerid working?
15:42.49mariogamboahow i can up my tdm400p first and my e1 after
15:42.51mariogamboa?
15:43.18funxionanyone using queuing with announcment overide?
15:43.46darkskiezanyone else noticed the cisco 7.5 ringing weirdness ?
15:44.28Hmmhesayspoke through the menus darkskiez
15:44.44coppicetclark: Its on a new machine, and I forgot to copy the mantis database over. now the old machine in a land-fill somewhere :-\
15:47.05darkskiezHmmhesays: traversed the lot, didnt see anything about it.
15:48.22Hmmhesayssend me the firmware
15:48.25HmmhesaysI can poke around a bit
15:53.07darkskieztheres nowt in the menus
15:54.30HolosAnyone used Sipp or asterisktest?
16:01.20gordonjcpwooo
16:01.46Theuni;)
16:01.55Theunistop me when i'm getting l4m3
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16:18.18Theunidarn
16:18.20Theunimisdn crashes
16:19.12*** join/#asterisk [illuminatus] (~illuminat@cpe-65-189-245-76.woh.res.rr.com)
16:19.24[illuminatus]is there a # method to put people on hold with asterisk?
16:19.27*** join/#asterisk teknoprep (teknoprep@pcp01717570pcs.nash01.tn.comcast.net)
16:19.29teknoprephi all
16:19.37teknoprepwow asterisk owns all PBX's
16:20.05teknoprepits just as good as any PBX i have worked with
16:20.08teknoprepbut its free
16:20.28loudnods, they owned my two meridians 11c fast.
16:21.09teknoprepall i can say is wow
16:21.20teknoprepnow to get this setup with an H.323 provider
16:21.27mariogamboaanyhere know how make the zaptel load first the module for te110p and after load the module for the tdm400p because the problem for me is the zaptel load fist the module of fxs fxo and consider the channel 1 of e1 like fxo how i can to obligate to zaptel to make that?
16:21.43tzangermariogamboa: you just load them in the order you want
16:22.05teknoprepi had a question yesterday that i didn't understand much of the answer.... when i have an H.323 provider.. how do i have multiple inc and outgoing call's .. does the provider just take care of that.. do i need to get a special provider that handles mutltple calls... or will most any do
16:22.13*** join/#asterisk SpaceBass (~sp@c-24-125-184-203.hsd1.va.comcast.net)
16:22.32tzangerthe automatic module loading goes by some predetermined PCI order and IIRC the only way to really get it to automatically reverse the order is by either setting up a "false" dependency for the incorrect module loading, or to swap PCI slots
16:22.52teknoprepstop loading modules automatically lol
16:23.01teknoprepthey are kernel mod's you are talking about right?
16:23.21loudyou should ask your provider on how many clear channels you get from them.
16:23.26*** part/#asterisk Pcharky (~khagen@cust.12.215.adsl.cistron.nl)
16:23.30teknoprepahh
16:23.42teknopreplets say i need 120
16:23.52teknoprepdo i just ask them for a 120 channels
16:24.00teknoprepand then they give me a price for it
16:24.03mariogamboain my case i need the channels 1 to 31 is for e1 and 32 to 35 is for fxs fxo in this case i need to modprobe first wcte11xp and second wcfxs is correct
16:24.04mariogamboa?
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16:24.25*** join/#asterisk brookshire (~matt@207.111.174.1)
16:24.28tzangermariogamboa: correct
16:24.53teknoprepmariogamboa are you not using this with a fully VoIP network
16:25.04tzangerso either set up a false dependency on wcfxs which tells it to load wcte1xxp first, or swap PCI slots on the two cards which *should* swap the automatic load order...  if it's by PCI vendor ID that won't help though:-)
16:25.12brookshireyay new 2 port T1/E1 cards from digium: http://www.digium.com/index.php?menu=product_detail&category=hardware&product=TE210P&tab=details
16:25.17teknoprepi think that would be the only way to doit in my opinion.. i hate working with older phone systems
16:25.40teknoprepVoIP is just easy to setup in my opinion
16:26.18Dave--lol voip is easy, integrating it with other things is a bitch ;)
16:26.36teknoprepwhat would you want to integrate it with
16:26.56Dave--lol, BT!
16:27.03teknopreplol BlueTooth?
16:27.17Dave--British Telecom
16:27.19teknoprepoh
16:27.31teknoprepthank god i live in the US... even tho i hate it
16:27.35Dave--my sipgate inbound and outbound + all features works a treat
16:27.41teknoprephehe my new log off line
16:28.00Theunihmm
16:28.05Dave--my inbound and outbound with BT is ok, its just the callerid thats a pain in the rear
16:28.18teknoprepwhat is caller id such a pain?
16:28.25Dave--i have no idea
16:28.32teknoprepit should just work.. no?
16:28.39Dave--ive found dozens of different ways of fixing it
16:28.42Dave--and not one of them works
16:28.45Dave--lol, apparently so
16:29.01teknopreppretty much .. this PBX... you dial a number.. get a welcome msg.. dial party ext.. get that person?
16:29.24darkskiezDave: you have definately got callerid on your line?
16:29.32teknoprepcan you do <specific number> -> <specific ext>
16:29.37Theunisomeone who knows mISDN around?
16:29.42darkskiezDave: if you sign up to BT privacy at home you can get callerid for free these days
16:29.45Dave--darkskiez - thats what im actually trying to find out at the moment.......
16:29.50Dave--darkskiez - exactly what i did
16:29.52Dave--a week ago
16:29.56Dave--that said 3 days to put in place
16:30.15Dave--the only "analogue" phone i've got that does callerid is currently sat in pieces on my knee
16:30.17darkskiezchanged the CIDRINGS to 2 ?
16:30.39Dave--no, i couldnt find where
16:30.47Dave--you know where?
16:30.48darkskiezzaptel.h i think
16:30.48Dave--lol
16:31.10[TK]D-FenderDoes IAX2 have a different feature set as compared to SIP?  AKA : Features like blind/consultative transfer, server based MOH, DND, etc?
16:31.57Theunihmm. maybe the isdn card shouldn't share the interrupt with 2 other cards ...
16:31.58Theunigrr
16:32.07darkskiezDave: do u have call waiting?
16:32.13Dave--darkskiez - in the kernel source?
16:32.15Dave--no
16:32.23darkskiezdave: in the module source
16:32.39darkskiezif u had call waiting you could tell if you had callerid by calling in the middle of a call
16:32.44darkskiezyou'd hear the ZZZZt
16:33.07teknoprepthe "ZZZZt"
16:33.16Dave--lol
16:33.21*** join/#asterisk RaYmAn-Bx (rayman@x1-6-00-40-63-da-39-3f.k191.webspeed.dk)
16:33.21darkskiezv23bis burst
16:33.26darkskiezor whatever it is
16:33.42*** join/#asterisk sliechti (~mone@adsl-213-200-243-230.cybernet.ch)
16:34.11teknoprepis most of the actual administration done through GUI or cmd line
16:35.01teknoprep?
16:35.16Dave--./usr/src/modules/zaptel/zaptel.h doesn't contain CIDRINGS anywhere in it
16:35.36darkskiezdave: cvs head ?
16:35.51Dave--i dont cvs stuff
16:35.59*** join/#asterisk shido6 (~greg@d57-87-253.home.cgocable.net)
16:36.01eKo1real men use the cli
16:36.05Dave--lol
16:36.48brookshirethere are guis out there ;)
16:36.55Darwin35real men plug and go and dont worry about cli
16:37.05*** join/#asterisk Hmmhesays (negative3k@66.173.103.108)
16:37.34darkskiezdunno
16:37.44darkskiezdont have stable handy
16:37.49sliechtiHi Guys, does someone of you have time to help me with 3 questions about AVM B1 and Asterisk? My head is hurting..
16:37.51darkskiezI only remember reading about that CID rings thing
16:38.06teknoprepanyone use the SIP Load Balancer with success from Vovida
16:38.11Dave--yeah, ive seen a couple of mentions about it whilst digging trying to find a solution
16:38.41*** join/#asterisk trickyrick77 (~rsegrest@207.111.174.1)
16:38.43*** join/#asterisk Maxxed (~max@cpe-70-114-238-9.houston.res.rr.com)
16:39.45teknoprepok better question.. anyone suggest a gui admin interface for Asterisk
16:41.50Hmmhesayshrm tbf doesn't seem to be working
16:42.05*** join/#asterisk r0d3nt (~RatMan@wsip-24-234-241-84.lv.lv.cox.net)
16:43.00sliechtiquestion? can anyone tell me if the AVM ISDN-Controller B1 works with asterisk, yes, no?
16:43.01[TK]D-Fenderteknoprep : What are you intending on using it for?
16:43.22*** part/#asterisk Thus0 (~Thus0@158.111.102-84.rev.gaoland.net)
16:43.24teknoprepVoIP PBX... for 25 phone
16:43.55[TK]D-Fenderteknoprep :How technical is the person responsible for administering the system?
16:44.06Dave--i think (after my op) im gonna have to get back in to linux development and start fucking with asterisk ;)
16:44.14teknoprepwith linux .. more then needed
16:44.33teknoprepjust want to make it easy to admin if i am not around
16:45.05*** join/#asterisk jfonsecausa (~jfonseca@12.42.141.34)
16:45.17[TK]D-Fenderteknoprep : in that case, crew GUI's.  You're probabaly better off using the CLI and maintaining the conf files then.  Most GUI's take away more than they give and are a hassle to set up.
16:45.39teknoprepconf files are not an option for most ppl
16:45.53sliechtiI'll just buy the card an test then :)
16:45.55sliechticheers
16:45.59NuggetLinux is poo.
16:46.03*** part/#asterisk sliechti (~mone@adsl-213-200-243-230.cybernet.ch)
16:46.06teknoprepi do everything through conf files.. while all our other admins want to use webmin
16:46.07[TK]D-Fenderteknoprep : My company wanted a "turn-key" solution so I found one manufacturer that makes a * GUI that is sufficiently powerful but I wouldn't CHOOSE to use it if I had a choice
16:46.21Dave--webmin does nasty things to the config giles
16:46.22[TK]D-Fenderteknoprep :Thats why I asked who'd be administering it :)
16:46.30teknoprepDave-- no it doesn't
16:46.35teknoprepmore then just me
16:46.42Dave--heh
16:46.54teknoprepi want a central .. easy to use .. web admin
16:47.38teknoprepi have yet to get my linux box from where i used to live yet.. or i wouldn't be asking questions
16:47.40eKo1use webmin
16:47.46teknopreplol
16:47.48[TK]D-Fenderteknoprep : what kind of phones/functionality do you expect the GUI to give you?
16:47.51teknoprepwebmin for asterisk
16:47.53teknoprepi hate webmin
16:48.02teknoprepi just want something for Asterisk
16:48.24teknoprepi am sure i saw a system that allows for call transfers from a web gui
16:48.30[TK]D-Fenderteknoprep : Got a budget?
16:48.30teknoprepthat would be more end user
16:48.51teknoprepnot my budget ... client budget.. and this solution will be much cheaper then what they are using now
16:48.53eKo1web GUIs are overrated. use vi and be done with it
16:48.55teknoprepor going to go with
16:49.01teknoprepPBX's are expensive as shit
16:49.44[TK]D-Fenderteknoprep : Only gui I would suggest for anyone seriously needing one : www.scopserv.com
16:49.45teknoprepeKo1 like i said.. not for me man.. and i perfer nano for conf files..
16:49.51teknoprepty
16:50.02teknoprepfinnally someone answered the question without another one.. or a smart ass comment
16:50.28tzangerI just told the only ohter guy I know who uses nano that he's not alone in the world
16:50.36tzangerand that I'd have to kick his ass for evangelicizing :-)
16:51.31eKo1nano is too simple for my tastes
16:51.36tzangergimme vim
16:51.43tzangerI even install gvim on my win32 boxes
16:51.47tzangerrightclick/edit with vim
16:51.52tzangerit's the new hotness
16:51.53astoriatzanger: me too, I LOVE that!
16:52.02shido6i miss bbedit
16:52.09tzangerI am gonna have to figure out how to slipstream that in
16:52.11astoriatzanger: makes editing ascii files and csv files a breeze
16:52.16tzangerbbedit, heh there's a blast from the past
16:52.17eKo1right click?!
16:52.19tzangerastoria: yup
16:52.22eKo1real men don't use a mouse
16:52.22[TK]D-Fenderteknoprep : Asking why you want it and what you expect of it is a very real point.  And as for cost, the free ones don't really let you do that much and are "iffy".  ScopServ's is AMAZING for what it does, but comes at a cost.
16:52.32tzangereKo1: nonsense.  real men value the utility of a good tool
16:52.41tzangerwhether that be a rifle, mouse or cock ring
16:52.43eKo1yeah, it's called a keyboard
16:52.50[TK]D-Fenderteknoprep : But I'd still just do it all myself if I could (AKA if they'd LET ME)
16:52.55astoriasomeone should invent a cock-ring/computer interface..
16:53.00tzangerhahaha
16:53.03tzangerfufme.com
16:53.15teknoprep[TK]D-Fender its not you.. it was other ppl
16:53.17tzangerI run linux on my laptop here
16:53.23astoriaI just took a class on human-computer interaction at msu.edu, and i had to invent something. Damnit! Why didn't i think of that!
16:53.25teknoprep[TK]D-Fender they will let me do what i want
16:53.36teknoprep[TK]D-Fender but i still in the end have to have a GUI interface for them to use
16:53.46teknoprep[TK]D-Fender and i want an opinoin on a good one to use..
16:53.50*** join/#asterisk shmaltz (~chatzilla@38.117.213.17)
16:54.16shmaltzhelllo every1
16:54.35lathos42I do have to run Windows though for Lotus Notes.. so I installed VMWare
16:54.42teknoprep[TK]D-Fender how do you purchace this software
16:54.49teknoprep[TK]D-Fender there is no buy options.. it says comming soon
16:54.54tzangerlathos42: that seems like overkill
16:54.56[TK]D-Fenderteknoprep : Then you might be better off with AMP.  ScopServ FULLY rebuilds all * conf files.
16:54.58tzangerdid wine not work?
16:55.06shmaltzteknoprep, what software r u talkig about?
16:55.19[TK]D-Fenderteknoprep : I'm buying a full turnkey server from them soon, and they are preparing for mass production.
16:55.27teknoprephttp://www.scopserv.com/products.php
16:55.36lathos42tzanger: Yeah, it worked, but having Windows on here does come in handy for troubleshooting user issues
16:55.46teknoprep[TK]D-Fender i just want the software
16:56.34[TK]D-Fenderteknoprep : E-mail them and ask.  Not sure what they'll sell it for solo.
16:56.48teknoprep[TK]D-Fender you can download it tho .. lol
16:56.52teknoprep[TK]D-Fender heh
16:56.55[TK]D-Fenderteknoprep : I met with their CTO & Sales mgr
16:57.09[TK]D-Fenderteknoprep : its a limited download in some whay, not sure on the details though.
16:57.22tzangertrue
16:57.24teknoprep[TK]D-Fender 15days
16:57.35shmaltzteknoprep, [TK]D-Fender, how much is that?
16:57.36[TK]D-Fenderteknoprep : that'd do it :)
16:58.09teknoprep[TK]D-Fender how much you paying for that box
16:58.44*** join/#asterisk razu (~razu@80-235-89-85-dsl.prn.estpak.ee)
16:59.10[TK]D-Fenderwell for our CFG, around 13,000 for a seriously redundant server w/ support.
16:59.34shmaltzwho is that scopserv anyhow?
16:59.40shmaltzanybody that we know from here?
17:00.10[TK]D-Fendershmaltz : Nope, just a Quebec based company gearing for amss production for the distribution chain.
17:00.34[TK]D-Fendershmaltz : They're not on IRC, but they employ so serious firepower programming-wise
17:01.03teknoprep[TK]D-Fender this asterisk will work as a fully VoIP PBX system, yes?
17:01.07shmaltzyou know them? d-fender?
17:01.26eKo1serious programming firepower? vim + gcc?
17:01.33shmaltzwhat is the price range here?
17:01.34[TK]D-Fenderteknoprep : it is technically a full * implementation, but all the conf files are build by their gui.  Absolutely EVERYTHING.
17:01.45[TK]D-FendereKo1 : I meant workforce.
17:01.58eKo1ah
17:01.59teknoprep[TK]D-Fender i just mean asterisk in general.. you can use this as a VoIP PBX
17:02.02[TK]D-Fendershmaltz : for the software alone, no idea.
17:02.15[TK]D-Fenderteknoprep : * IS a full fledged IP-PBX
17:02.17teknoprep[TK]D-Fender no special cards... just network cards.. and VoIP Phones
17:02.18*** join/#asterisk Corydon76-home (black@Corydon76-home.sustaining.supporter.pdpc)
17:02.44*** join/#asterisk ChkDigit (~mike@static65-87-228-18.regina.accesscomm.ca)
17:02.46[TK]D-Fenderteknoprep : Yeah you acan use it with VoIP only gear (termination & phones (soft/hard))
17:03.03[TK]D-FenderIt fully supports every kind of trunking * does.
17:05.55[TK]D-FenderThey use their custom GUI database to regen the files though so custom stuff is much more difficult to implement (currently).  By customer I do mean "serious" stuff.  The base functionailty of the GUI for call processing is the best of ANY for any product (Avaya, Nortel BCM, etc) that I've ever seen.
17:07.03*** join/#asterisk file (~jcolp@mctn1-3854.nb.aliant.net)
17:08.01teknoprep[TK]D-Fender i wouldn't be asking this if my linux test box for home was here .. but i moved recently... is everything conf file setup
17:08.08Theunidarn
17:08.16Theunithat misdn/misdnuser stuff is severely unstable
17:08.17teknoprep[TK]D-Fender this seems really easy to setup by hand
17:08.52teknoprep[TK]D-Fender i have worked with meridian systems before.. and some dip switch PBX which is no comparison.. but conf files are easy
17:09.27teknoprepskrew it
17:09.32teknoprepi am installing VMWare
17:09.39teknoprepi can't wait anymore
17:09.49Hmmhesayswait for what?
17:09.59[TK]D-Fenderteknoprep : Seriously.... if you don't need a GUI for a technologically challenged boss, then by God don't get a GUI.
17:10.46teknoprep[TK]D-Fender i know man
17:10.50teknoprep[TK]D-Fender saves my job
17:11.08[TK]D-FenderSo you need it for "boss" reasons?
17:11.11teknoprep[TK]D-Fender i am my own boss.. its the ppl i work with
17:11.18teknoprep[TK]D-Fender we have no real "boss"
17:11.35teknoprep[TK]D-Fender we are a firm that works together for small - med business to save them money and make us more
17:11.38[TK]D-FenderOk, tell them "too bad" and just config it yourself and screw the GUI.
17:12.00[TK]D-FenderNot worth the trouble/cost it'll incur
17:12.03teknoprep[TK]D-Fender we don't do that.. we build custom webmin modules all the time
17:12.21teknoprep[TK]D-Fender for our clients.. we built the firewall clustering module for iptables in webmin
17:13.09[TK]D-FenderWell them, AMP may be the way to go, but all GUI's suck to a point.  ScopServ's is by FAR the best, but at a price.
17:13.21[TK]D-Fenderif you specialize that much, then build your own :)
17:13.27HmmhesaysAMP  so/so if you use it right
17:13.44teknoprep[TK]D-Fender i wrote a custom c program for NAT / IP Clustering for virtually any server to be clustered at an ip level at the firewall lvl instead of relying on just fail-over... this provides the ability that LVS does with a much easier configuration and no software to be run on the server .. just on a redundant ip clustering box
17:13.57[TK]D-FenderHmmhesays : Glowing review :)
17:13.57teknoprep[TK]D-Fender i find it much more reliable then LVS
17:14.18Hmmhesaysthe problem with amp is most people don't read the documentation and they don't know how the configuration files work in the first place
17:14.32[TK]D-FenderI know remarkably little about clustering, networking, and Linux as a whole :)
17:14.53teknoprep[TK]D-Fender i am looking at this for an approach that i can proceed to use this in every environment we work on and make it easy to use on a factor from looking at how stupid the avg user is with computers
17:16.12*** join/#asterisk j4m3s (~debbie@216.207.245.18)
17:16.20astoriaIs there a specific url for polycom phones that one should pull to reset the phone from a bootserver?
17:16.40teknoprep[TK]D-Fender so that is why i am asking these questions.. i want a cheaper solution that can still do what is needed to be done.. and have a selling point of easy to use
17:16.50[TK]D-FenderAverage users?  Uh oh... thats a dangerous grouping...
17:16.51astoriaOop. NEver mind, found the wiki script.
17:16.57teknoprep[TK]D-Fender lol
17:17.13teknoprep[TK]D-Fender i see they have already made a proxy for load balancing SIP.. which owns..
17:17.14Holosastoria: There is a perl script :)
17:17.32teknoprep[TK]D-Fender going to be setting that up within a virtual environment this week
17:17.48astoriayay for community sharing!
17:17.57JunK-U~agi api
17:17.57jbotfrom memory, agi api is at http://home.cogeco.ca/~camstuff/agi.html
17:18.01[TK]D-Fenderteknoprep : * doesn't lend itself to "quality" GUI's, its TOO FLEXIBLE.  So thats why they are mediocre at best, or extremely rare and pricey
17:18.27[TK]D-FenderScopServ's is as turnkey as it gets, but I don't know what they'll sell the S/W for solo...
17:18.35tzafrir_homehi folks
17:18.40astoriaWait until the real-time stuff gets all debugged and everything..
17:18.44*** join/#asterisk loick (~loick@APuteaux-151-1-4-40.w82-120.abo.wanadoo.fr)
17:19.36teknoprepgod i love owning a copy of vmware.. its way better then virtual pc
17:19.47Theunianybody knows what this means when starting asterisk:
17:19.52Theunidebug_init: debug_mask = 0
17:19.52TheuniNo Upper ID port:1
17:19.52Theuniinit_stack: Function not implemented
17:20.43tzafrir_homeMy toy * server has an FXO card connected alongside a real analog phone. when a call comes in someone maay try to answer it using that phone. Is there a way for Asterisk to detect that the line has been picked up elsewhere?
17:21.13Dave--lol, is it common when trying to configure asterisk for this type of reaction?
17:21.25Theunilooks like it ... *G
17:21.46Dave--the wiki argument says to change the #DEFAULT_CIDRINGS 1 line in chan_zap.c
17:21.49tzafrir_homeDave--, no. I don't believe * has any desk motion detector
17:22.04Dave--and NO!!!, i dont have a ruddy chan_zap.c
17:22.17teknoprepVoIP is the solution to all your problems
17:22.22teknoprepwith a VoIP provider
17:22.23Dave--so i re-extracted the source i compiled from
17:22.26teknoprepyou can't go wrong
17:22.32Dave--and there isn't one in there
17:22.34Dave--lol
17:22.40Theunihmm. pitty. gotta run ... if someone should remember something ... qry me please.
17:22.44Theunithanks for * anyway :0
17:22.51teknoprepwhy would ppl still use analouge over VoIP
17:23.05Dave--hehe, it works fine if apart from callerid
17:23.07tzafrir_homeDave--, what's exactly the problem?
17:23.32Dave--tzafrir - callerid never gets to asterisk
17:23.57tzafrir_homeDave--, how do you know you get caller ID from the provider?
17:24.12Dave--tzafrir - no there is the problem ;)
17:24.14Dave--now
17:24.25Dave--i am assuming i am, because i requested it turned on a week ago
17:24.30Dave--and they said it would take 3 days
17:24.41Dave--(this is with BT in the UK btw)
17:25.57tzafrir_homeDave--, do you have a phone that can display it or something?
17:26.00Dave--i've just worked out one of my problems
17:26.05Dave--tzafrir - it died :(
17:26.07Dave--so no
17:27.15Dave--i think i need to download the source and build it myself
17:27.51tzafrir_homeback to my question. I wonder if it actually possible. If I pick up a phone and start talking, asterisk cannot tell that this is not coming from the other side of the line, right?
17:28.02InfraReddave
17:28.07InfraRedtheres a page bout uk callerid
17:28.12tzafrir_homeDave--, what card?
17:28.12InfraRedwww.voip-info.org
17:28.14Dave--http://www.voip-info.org/tiki-index.php?page=UK+Asterisk+Details
17:28.15Dave--there
17:28.29Dave--urm, urm, urm, urm, urm, urm
17:28.37Dave--2 secs, i cant remember
17:28.42tzafrir_homeand there are actually binary packages that have that patch
17:28.50Dave--ooooh
17:28.51Dave--where?
17:29.13Dave--<PROTECTED>
17:29.14tzafrir_homecurrently http://tzafrir.org.il/rapid108/unstable
17:29.30tzafrir_homecurrently http://tzafrir.org.il/rapid108/unstable/ , that is
17:29.38Dave--lol
17:38.52*** join/#asterisk [TK]D-Fender (~joe@4.67.252.216.dsl1.colba.net)
17:41.07[TK]D-FenderJust to reiterate for those who may have missed my previous question : How does IAX2's call feature list differ from SIP's?  (CID, MOH, Tranfers, etc)
17:41.41*** join/#asterisk Beave (~beave@vistech.org)
17:41.41JunK-Usee drafts?
17:44.35DaminMethinks there is memory leaks in Stable..
17:44.35Damin<PROTECTED>
17:44.36Damin<PROTECTED>
17:44.58*** join/#asterisk zeedo (~zeedo@www.reboot-robot.net)
17:46.44teknoprepor your memory is bad
17:46.52teknoprepor your linux distro is installed poorly
17:48.42Beavewhats the uptime of asterisk?
17:50.08bkw_did you guys see this http://bugs.digium.com/view.php?id=4735
17:50.19denonyeah
17:50.24denonpretty cool
17:50.33bkw_Damin, na I got asterisk processes with 800 meg memory footprints
17:50.36NuxiDamin, yes, but they are stable memory leaks.
17:50.42bkw_hahahahaha
17:50.45bkw_Nuxi, good point
17:50.52bkw_it leaks and does it very reliably
17:51.34Beaveohhh.. nifty
17:52.37Juggiemy * is a week old and consuming 66megs
17:52.48Nuxibesides, shouldn't you be able to restart asterisk every half hour and get the memory back?
17:53.37Nuximake a crontab and call it mop_up_leaks.sh
17:53.54Nukemizerusing a Digium T1 card, can I make an E&M Wink  T1 have Dial Tone for the user if the user accesses the trunk to make a call ?
17:54.14mariogamboammm
17:54.31mariogamboai still with the problem i can recive from my tdm400 tone
17:54.34mariogamboaany sugestion
17:54.59Nuximy asterisk has been up for a couple of days and has 55 mpg123 processes.
17:55.35jalsothi
17:55.54astoriawow, thats a lot of spice girls playing at once!
17:56.02Juggieheh
17:56.08jalsothow can I run a dialplan application from .call file? any example?
17:56.19Hmmhesayscall a local extension
17:56.26Hmmhesaysdo a little dance
17:56.39jalsotoh, great idea!
17:56.39Hmmhesaysvideo it, and post it for us to see
17:56.41jalsotthanks!
17:56.47*** join/#asterisk maggit (~maggit@dsl-200-67-147-190.prod-empresarial.com.mx)
17:56.47Hmmhesaysnp
17:57.39*** join/#asterisk dwmw2_gone (~dwmw2@baythorne.infradead.org)
17:57.54Kattyi could put a big debian swirl on my credit card.
17:58.00Kattywouldn't that be hottt
17:58.01NuxiHmmm, how do I make mpg123 not do that????
17:58.28twisted[asteria]i want to put a nice big cow on mine
17:58.32twisted[asteria]for gentoo ;)
17:58.47twisted[asteria]and an X from OS X branded in his backside
17:58.53*** join/#asterisk harryvv (~none@S010600a0c93f6f7e.vs.shawcable.net)
17:58.54blitzragemaybe its just too early... but is there a way I check if my bindport in sip.conf got parsed from the CLI?
17:59.03Kattyor maybe one of those LINUX INSIDE swirly things
17:59.07twisted[asteria]hehe
17:59.08Kattythat'd be cute on a credit card
17:59.15twisted[asteria]Katty, did you find a place to get customized credit cards?
17:59.15Hmmhesayshaha
17:59.23Hmmhesaysthere is one that is close
17:59.28Kattytwisted[asteria]: my bank does a photo expressions thing
17:59.33twisted[asteria]nice.
17:59.40twisted[asteria]i'll have to check with mine
18:00.21twisted[asteria]lol
18:00.31Hmmhesaysvan gogh's starry night reminds me of the debian swirl
18:00.31astoriaAnyone ever get a polycom freezing at the Welcome! screen?
18:00.41Kattyhttp://www.bankofamerica.com/creditcards/photoexpressions/images/card_pe.jpg
18:00.47Ariel_astoria, yes just unplug it and start it again.
18:00.55astoriaAriel_: tried that about five times..
18:00.59Kattythat'd be so cute with a little tux on it
18:00.59twisted[asteria]Katty, ohh, you're with bank of america?
18:00.59Hmmhesaysahh bank of america, the original visa people
18:01.08astoriaAriel_: i just switched to ftp from tftp..
18:01.16*** join/#asterisk ginvent (~joseph@adsl-68-125-225-164.dsl.sndg02.pacbell.net)
18:01.19twisted[asteria]bank of america tried to fuck me
18:01.27funxionanyone using the announcment override option for queueing?
18:01.28Hmmhesayslube or no?
18:01.31ginventWhat are you zapata.conf gains set to. I can't seem to get the fax to work.
18:01.50mariogamboamm what happen with this card?
18:01.54Ariel_astoria, when it comes up press setup and make sure you have the right settings under server there.
18:01.59twisted[asteria]Hmmhesays, no
18:02.01anthmdid it buy you a drink first?
18:02.03Hmmhesaysouch
18:02.05twisted[asteria]Hmmhesays, they tried to tell me i commited atm fraud
18:02.08twisted[asteria]for depositing a check
18:02.19ginventAnyone know their gain settings offhand?
18:02.23astoriaAriel_: my boot settings appear to be correct, and it is updating the .log file and everything looks okay - no errors..
18:02.29Hmmhesayswas it a check written in crayon on a piece of construction paper?
18:02.37twisted[asteria]THEN, they held my account for 3 weeks before I could even close it, which made me default on my rent
18:02.50twisted[asteria]THEN, they wanted to keep the amount they said i defrauded them by
18:02.58Hmmhesaysgood thing I don't have a checking account through them
18:02.58twisted[asteria]only for me to get the cleared check back in the mail a week later
18:03.27ginventDoes the RX gain have to be high?
18:03.28twisted[asteria]me too
18:03.33twisted[asteria]my local bank rocks
18:03.41twisted[asteria]bank of america told me never to bank with them again
18:03.42Hmmhesaysyeah, mine is pretty good to me too
18:03.44twisted[asteria]so i said "fine"
18:03.46ginventMy local bank is awesome also.
18:03.55ginventI use it for me and my co.
18:04.00Kattytwisted[asteria]: yes
18:04.05Hmmhesaystheir online banking leaves something to be desired
18:04.10twisted[asteria]Katty, get out while there's still time ;)
18:04.17twisted[asteria]my local banks' online banking OWNS
18:04.21twisted[asteria]i can do everything from it
18:04.24Hmmhesaysbank of america has been good for my credit card though
18:04.27twisted[asteria](well, except wash my butt)
18:04.43Hmmhesaysyou need to have a web enabled bidet installed
18:04.47twisted[asteria]lol
18:04.54twisted[asteria]so THAT's what I'm missing
18:04.54Kattytwisted[asteria]: uhh, no (=
18:05.08Hmmhesaysx11 anyone?
18:05.08ginventcan anyone help me with my zapata gain?
18:05.10twisted[asteria]Katty, hehe..  I've just had very bad experience(s) with BofA
18:05.13mariogamboahow i can upload my cards?
18:05.21twisted[asteria]Katty, and so have a lot of people I know
18:05.28twisted[asteria]Katty, but if you're happy, cool :)
18:05.30Kattyk
18:05.44HmmhesaysI only have a credit card through them, and they've been fine
18:05.52bkw_ITS TWISTED!!!!
18:06.00Hmmhesaysexcept for raising my limit all the time, that kind of irritates me
18:06.28Hmmhesaysdog pile!
18:06.40ginventPawn take queen... EVERYONE ON THE QUEEN!!!
18:06.53Corydon-wqueens
18:07.13ginventQuote from history of the world...
18:07.14teknoprep<PROTECTED>
18:07.17teknoprep<PROTECTED>
18:07.20ginventfor those non mel brooks fans.
18:07.30twisted[asteria]bkw_, yeah, it's me
18:07.35Hmmhesays'its good to be da king'
18:07.46twisted[asteria]Hmmhesays, no lie there ;)
18:08.01Corydon-wtwisted[asteria]: are you sure?  Everybody claims to be 'me'...
18:08.14Hmmhesayshaha
18:08.17twisted[asteria]well, 'you' are not the king
18:08.45anthmoh piss boy!
18:08.52twisted[asteria]piss boy? LOL
18:08.59Hmmhesaysclassic
18:09.15Hmmhesaysso 3 people have seen that movie, lol
18:10.01*** join/#asterisk bzbw (~wlwzhang@ip67-153-142-109.z142-153-67.customer.algx.net)
18:10.16Kattyhttp://www.thinkgeek.com/images/products/additional/large/noplace-babydoll.jpg <- i should put that on the card
18:10.36bzbwAnyone knows what causes following error:
18:10.37funxionis there anyway to record to mp3 using the record command?
18:10.37twisted[asteria]Katty, THAT's HOT
18:10.38bzbwJul 19 11:59:55 ERROR[3217]: utils.c:509 tvfix: warning negative timestamp -251249.-366000
18:10.51Kattytwisted[asteria]: (=
18:10.54twisted[asteria]bzbw, update cvs.
18:10.59Hmmhesaysyou should put 1 Hmmhesays on your card
18:11.15bzbwtwisted: just cvs it out yesterday.
18:11.23twisted[asteria]bzbw, update cvs.
18:11.29KattyHmmhesays: or not (=
18:11.32Hmmhesayslol
18:11.35bzbwtwisted: ok
18:11.38Kattyi could get a picture of everyone at cluecon
18:11.41Kattyand then put that on the card
18:11.45twisted[asteria]damn!
18:11.48twisted[asteria]now I have to go!
18:11.55Kattybut my like bkw_ would be playing with file in the picture >.<
18:11.55mariogamboai need to know if i need to put firs port of my card for the fxs and fxo and the rest for the t1
18:11.56HmmhesaysI'll be the one with the giant novelty chicken suit on
18:11.59Kattys/like/luck/
18:12.09twisted[asteria]Katty, nah, he only does that 'under cover'
18:12.12Kattyk
18:12.17twisted[asteria]heh
18:12.18twisted[asteria]smoke break.
18:12.22Kattyboo, smoke
18:12.28Kattyi'll do that
18:12.30Kattyit'll be great.
18:12.36KattyDarthClue: k
18:12.49Kattyi'll have to borrow someone's camera
18:12.49Hmmhesaysnoooooooooooo!
18:12.58*** join/#asterisk iCEBrkr (icebrkr@rrcs-24-129-130-158.se.biz.rr.com)
18:12.58Kattycause i'm sure 50 billion people will want a copy
18:15.17Kattytwisted[asteria]: you have nextel?
18:16.03ginventCan someone help me with my ztcfg
18:16.29funxionany suggestions as how to loop a recorded message (recorded using record) to musiconhold?
18:17.20funxionginvent what kind of card?
18:17.36ginventfunxion a x100p clone
18:18.10funxiondamn
18:18.23funxionI dont have any experience with fx0 cards
18:18.26funxionsry
18:18.46*** join/#asterisk mDuff (~cduffy@fwext1-ext.isgenesis.com)
18:18.52funxionI can try but I might not be too accurate
18:19.08Ariel_funxion, what do you mean loop like play part then play another recording then back to the orginal?
18:20.54funxionbasically trying to make something to call into and record a message. then the recorded message wwould be played like musiconhold for a queue
18:21.57Ariel_OH ok your going to need the power of the extensions.  like you dial exten 77 you record then play it back you acept it then it's your main greeting for the ivr.
18:22.07funxionI thought I might be able to acheive somehting like you described using announce but I found that announcments are played to the agent ass the call is answerd
18:22.44harryvvAriel_ how long are you going to be here today?
18:22.55funxionAriel_ I've already got the recording part its adding it to the music on hold or the ability to barge into the musiconhold and play the recordied message thats got me bent
18:23.16Ariel_at least 2 or 3 hours I am working on two servers. one Zoneminder the other one a new AAH for a customer.
18:23.46harryvvohh fantastic, I need to get my zoneminder up and running then present it to this complex that has a theft problem.
18:24.22harryvvbut anyway, need to take the wifes car into a dealer so will be away for a while.
18:24.40Ariel_harryvv, I have the new LiveCD that is giving me a problem. I might have to bakc to the older version. I don't like mandrake it's set too.
18:24.55Ariel_harryvv, you and your car that was what you said yesterday
18:25.11HmmhesaysI dislike radius some days, I really do
18:25.15harryvvyea, problem wih wifes car not fixed :)
18:25.27*** join/#asterisk darkskiez (~mhb@host-84-9-71-189.bulldogdsl.com)
18:25.39*** join/#asterisk astoria (~haydenth@66.235.201.217)
18:25.42lathos42Hmm.. I was not aware of Zoneminder.. looks like something I might want to check into
18:25.42astoriaI hate nickserv.
18:25.44*** join/#asterisk wizhippo (~wizhippo@Quebec-HSE-ppp233869.qc.sympatico.ca)
18:25.53harryvvAriel, the idea is to get past this nat problem so if you did want to contact me, you can do so though my asterisk box then to my cell phone from the fxo card.
18:25.54blitzragenickserv hates you
18:26.02astoriayeah, i know!
18:26.36funxionAriel_ any suggstions?
18:26.38wizhippoI saw an example like exten => s,n,Dial....  Can I use "n" instead of a number
18:26.40Kattytwisted[asteria]: do you BEEP BEEP
18:27.00Inv_ArpAriel_: still moving?
18:27.45Ariel_Inv_Arp, yes in a few months I hope
18:27.58Ariel_funxion, I will have to think about that one more.
18:28.05funxionk
18:28.27*** join/#asterisk xeet2 (~xeet3@bwi1-br1-gig3-1.jsci.net)
18:28.27DannyFwizhippo, yes if you use CVS HEAD version
18:28.59xeet2yay, verizon's single, non-redundant mainframe that handles all provisioning and LNP requests was fried by lightening yesterday
18:29.07Inv_ArpAriel_: bummer... we need skilled asterisk peeps down here
18:29.19DannyFxeet2, urk
18:29.22xeet2yeah
18:29.24Ariel_Inv_Arp, well I need more work too.
18:29.29xeet2and I have 15 pending LNP orders
18:29.32xeet2that are just plain down
18:29.35DannyFyay
18:29.43MeatyAnyone can help me ?
18:29.44MeatyRecently we made our own .wav files to play in asterisk. In any .wav player on windows, it sounds perfect at 8000hz mono pcm wav format...  As soon as asterisk plays it with Background(), we hear a big "click" and the end of each file played...that's pretty anoying... any idea? We recorded the original wav files with pro-tool on a mac.
18:29.49wizhippoDannyF: thank you, that saves me numbering problems, I saw s+2, that means?  where can I read how it works?  I don't see it int the readme
18:30.17DannyFwizhippo, took a peak on the wiki?
18:30.22*** join/#asterisk Romik_ (~romik_@212.143.5.146)
18:30.23xeet2inv_arp: where are you?
18:30.26Inv_ArpMeaty: smooth out the ending or fade it out
18:30.39Inv_Arpxeet2: south florida
18:30.42DannyFMeaty, how about in gsm format?
18:30.44Romik_hello! somebody can advice source of Dead packets in command "iax2 show stats"?
18:31.06xeet2inv_arp: what do you have going on?  I know some asterisk-capable people down there
18:31.17DannyFRomik_, what firewall you running?
18:31.26*** part/#asterisk bzbw (~wlwzhang@ip67-153-142-109.z142-153-67.customer.algx.net)
18:31.42Romik_DannyF: pf
18:31.50DannyF...
18:31.53DannyFk
18:31.58DannyFon same box eh?
18:32.04MeatyDannyF -> http://www.voip-info.org/wiki-Convert+WAV+audio+files+for+use+in+Asterisk   With this help.  But wav do the same noise when played with asterisk.
18:32.18Inv_Arpxeet2: we have clients that may need pbx integration... I mainly do linux/networking...  but I may be able to push *
18:32.28Romik_dannyf: no, on different box
18:32.32Romik_dannyf: Outstanding frames: 10 (8 ingress, 2 outgress)
18:32.32Romik_Packets in transmit queue: 1 dead, 0 final, 2 total
18:32.48DannyFRomik_, you running Head?
18:32.50MeatyInv_Arp : If i record a silence, we also hear the noise
18:33.09Romik_dannyf: 1.0.7-BRIstuffed-0.2.0-RC7k
18:33.11r0d3ntDell 2850, T101P, PRI, Cisco 3560 switches, 15 - cisco 7940G phones.....  Why would there be echo ???
18:33.16DannyFahhh
18:33.37Romik_dannyf: so bad?
18:33.44Ariel_Inv_Arp, well I am still here and will be here allot.  My parents and rest of family is here.
18:34.03astoriar0d3nt: are you using echo cancellation?
18:34.04Inv_ArpMeaty: select last part in software..... and silence that part out in software
18:34.05Ariel_Inv_Arp, besides we are looking now also to the Boca area as well.
18:34.14DannyFif you have a spare box try a later verion an run the sounds through it and see it's the same, if it is, then you should take a closer look at the recordings themselfes
18:34.21Ariel_xeet2, where are you at?
18:34.27r0d3ntastoria, all enabled, when bridged, echo training yes, echo cancel yes..
18:34.29DannyFjust a thought anyways
18:34.33MeatyInv_Arp : Ill try
18:34.48astoriar0d3nt: do you have AGRESSIVE cancellation on?
18:34.53r0d3nti do not.
18:34.55astoriar0d3nt: that solved my recent echo problems
18:35.00r0d3ntisn't that a compiled option ??
18:35.03Inv_ArpAriel_: ahh and it may be alot of ssh'ing so even remotely shouldnt be a prob
18:35.09wizhippoDannyF: I'm looking tere right now under extensions.conf. Dosn'r show it there.  Did you see it on the wiki? If so would you mind saying where?
18:35.16astoriar0d3nt: yeah, you're going to have to recompile, and edit the zconfig.h file.
18:35.21r0d3ntok
18:35.23r0d3ntthanks...
18:35.26ginventSo after I change gains in zapata.conf why doesn't ztmonitor rx go up?
18:35.36r0d3nthas anyone used Digium's Express installation service ???
18:36.02tzangerginvent: re-run ztcfg -v
18:36.03DannyFwizhippo, search for sox on wiki and convert to gsm
18:36.14tzangerr0d3nt: not me, no
18:36.33ginventtzanger, that doesn't change because of zapata.conf?
18:36.40DannyFwizhippo, sorry answer to wrong guy ;)
18:36.54DannyFjust a sec I'll see if i cant find it
18:37.22wizhippowhew, thougt i was going crazy.  I was wondering ;)
18:38.36wizhippoDannyF: just found it in the example extensions.conf..thanks for hte help
18:41.10*** join/#asterisk Gunnar (~gunnar@34.80-203-209.nextgentel.com)
18:41.50Kattytwisted[asteria]: come back!
18:41.55*** join/#asterisk w4kpm (~w4kpm@67-23-53-144.chvlva.adelphia.net)
18:41.59twisted[asteria]Katty, i'm back!
18:42.08shido6ok dokey
18:42.32twisted[asteria]i never really leave, I just have to do some real work occasionally :P
18:43.01MikeJ[Laptop]yeah, whatever :P
18:43.05twisted[asteria]haha
18:43.08twisted[asteria]BEEP BEEP
18:43.45DannyFwizhippo, here are some more http://weblog.barnet.com.au/edwin/000097.html
18:43.50shido6twisted
18:43.58shido6make some calls on our network and give me feedback - BLOW IT UP
18:44.03shido6when u get a chance
18:47.21tzafrir_homedoes 'asterisk -r' supposed to show colors?
18:47.36InfraReddid you ask the man pae?
18:47.38InfraRedpage
18:49.20tzafrir_homeInfraRed, yes, I did. It does not refer to colors other than the fact that a certain switch disables them
18:49.38*** part/#asterisk trig (~jb@xob.neospire.net)
18:52.12*** join/#asterisk corpo (~aeiou@h-67-100-132-81.phlapafg.dynamic.covad.net)
18:53.13corpodoes anyone in here use asterisk with broadvoice?
18:53.19MikeJ[Laptop]tzafrir -r has nothing to do with colors...your term settings do
18:53.38MikeJ[Laptop]there is discussion at length on mantis, and the lists on this
18:53.44MikeJ[Laptop]search closed bugs in mantis
18:54.02tzafrir_homeMikeJ[Laptop], so here I get colors when asterisk is started from the console, but not with asterisk -r
18:54.02corpoMikeJ: to whom are you speaking?
18:54.08corponvm
18:54.14tzafrir_homeok
18:54.22MikeJ[Laptop]yeah, I know...
18:54.24*** join/#asterisk scratchrf (~ryan@63-226-200-214.tukw.qwest.net)
18:54.43InfraRedtzafrir_home: means its switched on by default
18:54.50InfraRedlook in your terminal settings
18:55.37scratchrfanyone know where to find phpvoipmail?  all the link i come across are dead
18:57.25corpoanother related question: has anyone configured asterisk to support "external" extensions - e.g. x64 dials someone's cell phone
18:57.38InfraRedscratchrf: try google cache?
18:57.57scratchrfyeah, tried that...
18:58.32astoriacorpo: sure, i do that all the time.
18:58.36InfraRedbugger
18:58.40tzafrir_homeInfraRed, this too. When I switched from rxvt-unicode to xterm, colors appeared in a "local" colsole (but still not in -r)
18:58.55astoriacorpo: we have extensions for our clients to transfer callers, etc..
18:58.55tzafrir_homerxvt is so much faster than uxterm...
18:59.35corpoastoria: okay...so typically, how is that done? do you know of an easy way using amp?
18:59.45corpoastoria: although i have no particular aversion to editing conf's
18:59.52astoriacorpo: no. I don't use amp, but you can do it really easily in extensions.conf
19:00.27*** join/#asterisk Tili (~Tili@202-133-65-205-dialup.sat.net.pk)
19:00.46InfraRedscratchrf: try emailing asterisk-users list
19:01.10corpoastoria: so what do you do? something like exten => 64,2,Dial.....?
19:01.10*** part/#asterisk w4kpm (~w4kpm@67-23-53-144.chvlva.adelphia.net)
19:01.25astoriacorpo: yeah, pretty much..
19:01.39astoriacorpo: it's just like any other extension, but uses your outgoing trunks.
19:02.05corpoastoria: gotcha...have you had any experience with broadvoice?
19:02.25astoriacorpo: nope, i use nufone
19:02.26scratchrfthanks infrared
19:02.28InfraRedhow do you switch off sip debug command
19:02.47corpoastoria: since, what i'm really trying to accomplish is to use broadvoice's call transfer, so we can transfer a call out to a cell, in the process freeing up the trunk the incoming call was using
19:02.57corpoastoria: and the outgoing, for that matter
19:03.02astoriacorpo: make sure you use the t or T options if you want your external extension to be able to transfer
19:03.09*** join/#asterisk |dennis| (~dennis@200.32.215.82)
19:04.16corpoastoria: alright...where does that option go?
19:06.39astorialathos42: is that a 517 area code, they do
19:06.44astorialathos42: i have a 517 with nufone
19:06.55astoriacorpo: at the end.. look on the wiki
19:07.10lathos42astoria: yep, 517 with a 663 exchange
19:07.11corpoastoria: k, thanks
19:08.05astoriacorpo: corpo: http://www.voip-info.org/wiki-Asterisk+cmd+Dial
19:08.20astorialathos42: oh, i don't know about that.. mine is a 679
19:08.41*** join/#asterisk zagaya972 (~d2s-compa@APointe-a-Pitre-102-1-2-49.w81-248.abo.wanadoo.fr)
19:08.42corpoastoria: that's even better, thanks a lot man
19:09.20lathos42astoria: If that's a Lansing exchange, then it would at least be a local call from here.. not that we have alot of local customers :)
19:09.53astorialathos42: i suspect whomever his carrier is, will probalby have eaton rapids if they have lansing.
19:11.05lathos42astoria: you'd be surprised.. alot of the VoIP providers that offer Michigan DIDs seem to have everywhere but here in ER
19:11.26astorialathos42: really? i wonder why, you guys still running electromechanical switches or something? :)
19:11.46*** join/#asterisk mrun (Escape@mrun.telecom-bg.com)
19:11.49tzafrir_homeMikeJ[Laptop], are you familiar with that color problem? any idea why not apply http://bugs.digium.com/view.php?id=4695 ?
19:12.11lathos42astoria: Well, when I called SBC for a PRI quote, they said we'd be charged extra for Distance from a real CO
19:12.47MikeJ[Laptop]first off.. -rc makes no sense at all
19:13.04tzafrir_home-r . -c is ignored
19:13.09MikeJ[Laptop]second off, the problem is with your term settings
19:13.18MikeJ[Laptop]if you set them right, it will work
19:13.30tzafrir_homeswitched to a different tewrm for the time being.
19:14.05tzafrir_hometerminal shows colors.
19:14.08astorialathos42: if youre looking for cheap PRIs, check out acd.net
19:14.20astorialathos42: i could get a full PRI, with colo at the EL facility for ~$450
19:14.30MikeJ[Laptop]yeah..
19:14.32MikeJ[Laptop]like I said..
19:14.55*** join/#asterisk sig- (~sig@gnook.org)
19:15.07lathos42astoria: Well, SBC quoted me that it would be no higher than $492 a month..  And we're in a contract with them right now so we dont have alot of options
19:15.41lathos42astoria: I may end up giving them a call next year when our Contract is up though
19:16.21*** join/#asterisk n0rf- (~n0rf-@ip27.66.1311D-CUD12K-03.ish.de)
19:16.32n0rf-hey guys
19:16.46astorialathos42: they do alot of * stuff down there in the el facility
19:17.10*** join/#asterisk |dennis| (~dennis@200.32.215.82)
19:17.11*** join/#asterisk dacleric (~dacleric@p54828F22.dip0.t-ipconnect.de)
19:17.14astorialathos42: they can be a pain to get things done though.
19:17.28n0rf-any AMP users around?
19:17.39lathos42astoria: well, that wouldnt be any different than getting things done through our current SBC Account Manager
19:18.01astorialathos42: thats probably pretty true..
19:19.32daclerichi
19:19.36lathos42astoria:  My boss was telling me that before I started working here, it took them over a year to get all of the stuff provisioned that falls under our current contract
19:20.45*** join/#asterisk brettnem (~brettnem@207.90.232.34)
19:21.29daclerici would like to set up a fax by voip server (ipp for sending and mail for receiving) is Asterisk the right tool for this task?
19:22.38InfraRedwith the right addons
19:22.44InfraRedand the sacrifical goat
19:22.48InfraRedand 12 virgins
19:23.02daclericuhm
19:23.13daclericwhere should i get the 12 virgins?
19:23.18InfraRedexpect loads of hair pulling
19:23.25InfraRedtesco
19:23.40daclericfrom the virgins or the goat ?
19:23.44daclericor my hair?
19:24.22*** join/#asterisk Tili_ (~Tili@202-133-65-124-dialup.sat.net.pk)
19:24.27InfraRed[x] all of the above
19:24.50*** join/#asterisk L|NUX (~linux@202.5.145.58)
19:24.50*** join/#asterisk jarrod (anon@juniperyour.net)
19:25.16*** join/#asterisk peglax (~peglax@p54B22707.dip0.t-ipconnect.de)
19:25.28jarrodhey why does my cisco 7960 show REGISTERED to the correct sip proxy:port, but the SIP proxy does not report it as registered using the credentials found in the phones SIP config file
19:25.45jarrod1     111  REGISTERED     3595        3490         x.x.x.x:5060
19:27.21*** join/#asterisk shmaltz (~chatzilla@38.117.213.17)
19:27.28shmaltzanybody here using voxee?
19:27.49loudwhat credentials
19:27.58jarrodSIP user / pass
19:28.09jarrodwhen registering with the proxy
19:28.12Holosshmaltz: ya I am..
19:28.19loudthey wont show up in sip show peers
19:28.22mrunhi, i got a Voice Blue gsm gateway connected to a SmartNode, a gatekeeper and a couple of phones in the office. I want when I dial the number of the gsm gw to be able to pass some extra digits, e.g. 13 for person 1, 14 for person 2 and so on, in order to reach the office phones. anyone know how to achieve that? thanks in advance
19:28.23*** join/#asterisk lathos42 (~lathos42@65-42-27-66.dowdingindustries.com)
19:28.24loudjust user
19:28.24shmaltzHolos, you like them?
19:28.39jarrodno im using SER.. and unlike my other phones it does not show as registered
19:28.46loudah
19:28.58jarrodbut the PHONE reports it as registered
19:28.58Holosshmaltz: Yup.. good quality, cheap rates, always works for me.. They just added another NY Gateway too..
19:29.01jarrodwhich is very very weird
19:29.13loudcan you dial ?
19:29.19shmaltzHolos, SIP or IAX?
19:29.22*** join/#asterisk ginvent (~joseph@adsl-68-125-225-164.dsl.sndg02.pacbell.net)
19:29.42jarrodyes it will send digits to the proxy, but the phone is unidentified
19:29.44ginventCan anyone help me figure out why my rx and txgain settings don't seem to change anything?
19:29.53jarrodso it is handled as such per my dialplan config
19:30.34Holosginvent: Did you reload the module, kill asterisk and then restart it?
19:30.41ginventholos, I did.
19:30.53lathos42astoria:  So, does acd.net do most of their VoIP service through Asterisk?
19:31.00Holosginvent: did you check the ordering and location of the gains?
19:31.07ginventWhat I am doing is running ztmonitor and I see an rx signal, but it never goes up and down...
19:31.09ginventordering?
19:31.11ginventlocation?
19:31.21ginventin the zapata.conf file... order matters?
19:32.20Holosginvent: are you calling a 0dB milliwatt test line?
19:32.56ginventNo, I am just looking at my dialtone...
19:33.07ginventIs that the wrong way to test this?
19:33.27ginventI just want to see the rx signal go up and down.
19:33.35ginventwith a change in rx/tx gain.
19:34.21ginventOK, got it... ordering does matter!
19:35.21Holosginvent: yup.. To get the echo set correctly you should find a milliwatt test # and set it based on the values.. It will solve a lot of problems.
19:38.01*** join/#asterisk Murtuza (Murtuza@adsl-69-236-55-182.dsl.irvnca.pacbell.net)
19:38.05ginventHow do I find a milliwatt test #
19:39.12Holosginvent: What country? Find a Phone tech that does lots of installing, or call your CO and ask them
19:39.21ginventK, I am in the USA
19:39.44*** join/#asterisk bzbw (bwz@ip67-153-142-109.z142-153-67.customer.algx.net)
19:40.04Holosginvent: Try XXX-958-XXXX Range.. mine ends in 0901...
19:40.51bzbwtoday's cvs update produce one way audio, looks like * does not understand the G711u packet, anyone has such experience?
19:41.03*** join/#asterisk corpo (~aeiou@h-67-100-132-81.phlapafg.dynamic.covad.net)
19:42.06Hmmhesaysreinstalling windows on my laptop. oh noes
19:43.09*** join/#asterisk shmaltz (~chatzilla@38.117.213.17)
19:43.11Kattytwisted[asteria]: what should i store you as?
19:43.14Kattytwisted[asteria]: twisted?
19:43.28Hmmhesaysheh
19:43.45twisted[asteria]Katty, sure, or Josh if you prefer
19:43.47Kattyk
19:43.59twisted[asteria]yeah
19:44.05MurtuzaI am trying to route analog DID calls through * to external analog PSTN numbers (in US with SBC) and have been having problems, has anyone tried that before?
19:44.06Kattyjosh what?
19:44.09twisted[asteria]R
19:44.09Kattytwisted josh!
19:44.10Kattyk
19:44.11twisted[asteria]that works
19:44.22twisted[asteria]twisted josh is cool ;)
19:44.44twisted[asteria]lol
19:44.51twisted[asteria]it says something like "LOOK OUT!!"
19:45.06file[laptop]file josh is cool too!
19:45.09Kattyi just made you some insane ring
19:45.12Kattyfile[laptop]: that last ring
19:45.29file[laptop]'tsk 'tsk
19:45.55Kattytwisted[asteria]: just testing ring
19:45.59twisted[asteria]what last ring?  huh?  am I missing something?
19:46.10Kattytwisted[asteria]: talking to file at same time (=
19:46.14twisted[asteria]ahh
19:46.51twisted[asteria]file[laptop], you on mike?
19:46.57Kattytwisted[asteria]: k
19:47.00file[laptop]twisted[asteria]: Mike doesn't exist in Atlantic Canada
19:47.01Kattytwisted[asteria]: beep beep me plskthx
19:47.07twisted[asteria]file, ahh
19:47.23twisted[asteria]Katty, done ;)
19:47.27file[laptop]it's sad :(
19:47.32Kattytwisted[asteria]: you did not beep beep!
19:47.39Hmmhesaysheh
19:47.41twisted[asteria]i alerted
19:47.43twisted[asteria]oops
19:47.47Kattyyeah
19:47.48Kattythat's what i thought
19:47.49KattyOOPS
19:47.53file[laptop]Katty is having fun with her phone... 'tsk 'tsk
19:47.58twisted[asteria]heh
19:48.00twisted[asteria]me too
19:48.01Kattyfile[laptop]: shh
19:48.06Kattyfile[laptop]: i get my real phone later
19:48.14file[laptop]shhhhh!
19:48.16Kattyfile[laptop]: yay for playing with loaners ;)
19:48.27twisted[asteria]oh, so that's not your real DCID?
19:48.36twisted[asteria]or are you on a simmed phone?
19:48.41Kattytwisted[asteria]: no, it will be
19:48.46Kattytwisted[asteria]: i'm getting a new phone, with two lines on it
19:48.52Kattytwisted[asteria]: my dc number will be the same (=
19:48.52twisted[asteria]ahh ok
19:48.56file[laptop]my desk just beeped
19:48.56twisted[asteria]yay
19:48.59twisted[asteria]i have an i730
19:49.01Kattytwisted[asteria]: which one do you have?
19:49.03Kattyah k
19:49.07twisted[asteria]whoa, i'm psychic today
19:49.11Kattyi'll be getting either the 836 or the 275
19:49.13*** join/#asterisk vandien (~stephan@p50906154.dip.t-dialin.net)
19:49.23twisted[asteria]i'm thinking of getting the new 800 series
19:49.36Kattyjust got the little 205 right now
19:49.42twisted[asteria]but i'm not so sure I want to blow my wad on it just yet, as my 730 is working great, albeit a little beat up
19:49.59*** join/#asterisk Nethab (~chatzilla@adsl-67-113-141-170.dsl.sntc01.pacbell.net)
19:50.04Kattysomeone needs to make an iax java applet for teh phone :<
19:50.07n0rf-anyone with a bit of experience with AMP round here?
19:50.19twisted[asteria]Katty, i'm in the motorola developer network thingy :)
19:50.30twisted[asteria]i have an 800 page manual on the java stack in these things
19:50.35Kattytwisted[asteria]: :>
19:50.48twisted[asteria](well, not quite 800 pages, but it's friggin HUGE)
19:50.50Kattytwisted[asteria]: i shall pester you for nifty applets later.
19:51.09twisted[asteria]i haven't yet written an applet for it... haven't had the time/resources
19:51.13Kattyk
19:51.17greg_workn0rf-: #amportal
19:51.27twisted[asteria]but an iax2 applet would be nice, using untethered data connections
19:51.32n0rf-greg_work: ah cheers :)
19:52.10Hmmhesaysfunny this channel is
19:52.17twisted[asteria]Hmmhesays humor we like
19:52.30twisted[asteria]nooo!
19:52.35twisted[asteria]not my 10 dot!
19:52.36Kattyyes, pls
19:52.37Hmmhesaysheh
19:53.00Kattythere should be a way to let me use your intarweb service
19:53.38twisted[asteria]heh
19:53.46twisted[asteria]10.153.10.51
19:53.57Katty;)
19:54.05Nethabso what's wrong with this channel
19:54.14twisted[asteria]but good luck, dhcp is rockin on these things ;)
19:54.23Kattytwisted[asteria]: i'll pester yo ulater when i get my 275
19:54.29twisted[asteria]hehe, okay
19:54.38*** join/#asterisk StealthMethod (~nelsonx@adsl-070-148-141-013.sip.mia.bellsouth.net)
19:54.38Kattyyou can Fix It(tm) for me
19:54.46Kattytwisted[asteria]: we
19:54.51Katty'll do it at cluecon <- twisted[asteria]
19:55.05twisted[asteria]?
19:55.11Kattyyou can fix my phone at cluecon
19:55.21twisted[asteria]oh, heh.  I don't think i'm going to be there...
19:55.22Kattyi should print business cards too
19:55.25Katty:<
19:55.29Kattyk
19:55.37file[laptop]I should too
19:55.43twisted[asteria]thing is, i can't get time off
19:55.50file[laptop]attack of the business cards!
19:55.54Kattyoh noes!
19:56.08twisted[asteria]but I will HOPEFULLY be at astricon 2k5 in anaheim
19:56.26Kattyshan't be there
19:56.48file[laptop]I had our graphic designer make me business cards, but like he left and now I don't know where the templates are...
19:56.55file[laptop]makes me sad :(
19:57.03Kattyfile[laptop]: make new ones
19:57.10file[laptop]the ones he made were great
19:57.11Kattyfile[laptop]: mine mostly suck
19:57.50Kattytwisted[asteria]: where do you go to put in settings for interwab
19:57.54Kattyi mean web
19:58.03file[laptop]the interweb is crashing
19:58.15Kattyi bet you tell all the girls that
19:58.23*** join/#asterisk Corydon76-home (two@Corydon76-home.sustaining.supporter.pdpc)
19:59.19Hmmhesaysheh you'll have plenty of people to fix whatever you want at cluecon i'm sure
19:59.33Hmmhesayspeople stand in line to help you ;)
19:59.37Wonka(.)(.)
20:00.10*** join/#asterisk Xen^ (linux@202.5.146.154)
20:00.12Kattyfile[laptop]: peanut butter!
20:00.33twisted[asteria]Katty, they do that OAA
20:00.47twisted[asteria]kbytes reset is how many times you're net usage has been reset I think
20:00.53Kattyk
20:01.29twisted[asteria]file[laptop] has seen them ;)
20:01.33Kattyyeah i should go make some
20:01.35file[laptop]yup
20:01.36Kattytwisted[asteria]: post gifs
20:01.40twisted[asteria]Katty, hehe
20:01.44file[laptop]http://asterlink.com/samples
20:02.16Kattyi think i'll make mine like this | rather than like this --
20:03.11twisted[asteria]hangon, i don't think i have any gifs/pdfs
20:03.13Kattyfile[laptop]: purrty
20:03.38Kattyyeah i should definately go make some
20:03.45file[laptop]yup
20:04.10Kattyi'm sure everyone will want one :<
20:04.20Kattyso they can go BEEP BEEP
20:04.27Kattycause i like to beep beep
20:04.33file[laptop]lathos42: who knows
20:04.35*** join/#asterisk Romik_ (~romik_@1.fix.netvision.net.il)
20:04.49Katty:>
20:04.52twisted[asteria]heh
20:05.08file[laptop]I need to get a copy of those... hrm
20:05.47*** join/#asterisk asteriskDOTbz (~logger@pbxtech.com)
20:05.47asteriskDOTbz<PROTECTED>
20:06.50jarrodstupid 7960 shows as registered, but it is not registering
20:08.22harryvvneeds to be more chat on ipphone support.
20:09.20*** join/#asterisk fugitivo (~ajf@201.255.104.140)
20:09.23fugitivohello
20:09.32wizhippocan i use #include in and included file?
20:09.50wizhippoand=an
20:10.05RaYmAn-Bxtry it :)
20:13.01*** join/#asterisk wulfy814 (~lorentz@dsl093-061-214.pit1.dsl.speakeasy.net)
20:13.11*** join/#asterisk zotz (~zotz@24.231.36.100)
20:13.39wulfy814afternoon folks (or at least for those of us in the US on the East Coast)!
20:14.15wulfy814I installed the latest CVS version of * on my ClarkConnect box last night and was highly successful, for the most part
20:14.30Inv_Arpwulfy814: great :)
20:14.31wulfy814I'm getting: ERROR[14676]: utils.c:509 tvfix: warning negative timestamp -63127.-230875
20:14.48MikeJ[Laptop]wulfy814, update to current cvs
20:14.48wulfy814when I place a call on hold from extension to extension
20:15.01wulfy814from sip to sip both GS 2000
20:15.05MikeJ[Laptop]fix went in already, but today
20:15.31wulfy814MikeJ[Laptop]: sorry I'm not that familiar with CVS
20:15.36wulfy814would I do another checkout
20:15.42MikeJ[Laptop]yes
20:15.46wulfy814recompile: make clean, make, make install
20:15.49MikeJ[Laptop]wait
20:15.55MikeJ[Laptop]you said you just updated...
20:15.59MikeJ[Laptop]just do that again
20:16.10wulfy814no, just did initial install
20:16.19wulfy814via: checkout asterisk libpri zaptel
20:16.22MikeJ[Laptop]oh, yeah, just do it again
20:16.30MikeJ[Laptop]don't do a make samples
20:16.37wulfy814ok, just for asterisk though right?
20:17.27MikeJ[Laptop]yeah, should be fine from last night
20:17.34MikeJ[Laptop]for the others
20:19.34wulfy814compiling now :-)
20:19.41wulfy814and who says IRC isn't helpful!
20:19.47anthmka-loo kahn
20:19.57gaggamancould maybe somebody help me with an issue with a Cisco 7970 ?
20:20.24Hmmhesayswhat issue
20:21.19gaggamanvery easy: I had to do a factory reset, because it refused to register, because it was locked to its old Cisco CallManager
20:21.20*** join/#asterisk doughecka_ (~Tad@doughecka.user)
20:21.34gaggamanI followed Ciscos advisory...
20:21.51gaggamanand ended up with the firmware deleted :-)
20:22.20Hmmhesaysheh, that is quite the issue
20:22.35gaggamanso now it is trying to tftp the firmware, which I don't have.
20:23.29gaggamanyes. true :-(
20:23.41louds/clear
20:24.36gaggamanany Ideas where I could get it?
20:24.56fugitivofrom cisco?
20:24.58loudcco of course.
20:25.21doughecka_hah, cisco advice?
20:25.36gaggamanI already own the rights, as I bought the phone, but they want me to make a service contract to get into their firmware download area.
20:25.44doughecka_gaggaman: correct
20:25.44loudDownload : P0S3-07-5-00.zip  (Right click on this link and use 'Save Link As' or 'Save Link Target As')
20:25.46loud:>
20:25.55fugitivogaggaman: welcome to cisco
20:26.13Nethabno one is talking
20:26.42gaggamanwhich link?
20:27.05doughecka_to be fully licenced you need a sip license, and a smartnet contract to be able to download firmware
20:27.06gaggamanis it the 7970 sw? sounds like 7960 or something
20:27.09loudim playing with you
20:27.28doughecka_actually, you can get a universal licensed phone for all firmware
20:27.40gaggamanthere is no sip for 7970
20:27.56gaggamanjust want the normal, standard sccp fw back :-(
20:29.46gaggamannobody here with a 7970 and firmware?
20:30.19gaggamanor is there any way to download the fw from a working 7970?
20:31.09*** part/#asterisk Holos (~asdf@72.1.197.10)
20:31.45syleanyone got a PAP2-na config
20:34.40KattyHmmhesays: mew?
20:34.42shmaltz~seen dca
20:34.42jbotdca is currently on #asterisk (17h 47m 3s)
20:34.58Hmmhesayswhats up?
20:35.14KattyHmmhesays: iax2/ip/ext help
20:35.21KattyHmmhesays: my syntax is all borken
20:35.25KattyHmmhesays: call me!
20:35.32Hmmhesayscall you where?
20:36.51sig-;>
20:37.20wulfy814ok onto the next issue, trying to dial out using a POTS line connected to the PSTN of a SPA3000
20:37.41wulfy814I'm getting : "No channel type registered for '192.168.30.70:5061'"
20:38.15wulfy814I have this working with another install and don't understand the problem
20:38.23file[laptop]your dial line is wrong
20:38.30file[laptop]you're not specifying a technology
20:38.59*** part/#asterisk Ariel_ (~Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net)
20:40.24*** join/#asterisk colinm_ (~colol@VDSL-130-13-8-95.PHNX.QWEST.NET)
20:40.29*** join/#asterisk meppl (~mephisto@87.193.6.132)
20:40.33twisted[asteria]Katty, AI is crashing all over itself
20:40.43wulfy814<PROTECTED>
20:40.52twisted[asteria]Katty, which means I can't open the illustrator document so that I can create the image
20:40.53wulfy814exten => s,1,Dial(SIP/${EXTEN}@${PSTN_GW})
20:40.53Kattytwisted[asteria]: ai?
20:40.56Kattytwisted[asteria]: oh
20:40.57Kattytwisted[asteria]: k'then
20:40.58twisted[asteria]adobe illustrator
20:41.03Kattyno big (=
20:41.07Kattyi'll beep beep you anyway
20:41.14Kattyi have unlimited nationwide
20:41.15Kattybuwahaha
20:41.17Kattyk
20:41.18Kattyover it
20:41.20twisted[asteria]hehe ok, as long as you're prepared to sing me a song or tell me a story ;P
20:41.25Kattyk
20:42.47file[laptop]wulfy814: uh that's not the extension being executed
20:44.59lathos42If I wanted to help out with Asterisk, should I learn C or C++?
20:45.04file[laptop]C
20:46.13*** join/#asterisk clive- (~pirch@rrba-146-106-190.telkomadsl.co.za)
20:46.29wulfy814file[laptop]: got it, damn syntax :-)
20:49.54anthmslearn C and then come to cluecon to learn how to make an app and how to debug both topics are covered
20:52.17*** join/#asterisk Ariel_ (~Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net)
20:53.18*** join/#asterisk Inv_arp (~junya@adsl-156-139-229.mia.bellsouth.net)
20:53.22*** join/#asterisk jdg (~jdg@CA03F93D.adsl.mana.pf)
20:53.28*** join/#asterisk pcharky (~khagen@ip503c6061.speed.planet.nl)
20:53.51pcharkyHello people
20:54.53InfraRedhi
20:55.18pcharkyhi, can you help me wit a little newbie something?
20:55.30InfraReddepends
20:55.33shmaltzanybody know how to set the alert_info for a cisco phone that has only the default chirp1 and chirp2?
20:55.44InfraReddid you do reasearch on www.voip-info.org?
20:56.01astoriapcharky: sure, go.
20:56.03*** join/#asterisk clive-- (~pirch@rrba-146-117-223.telkomadsl.co.za)
20:56.19shmaltzYeah I check the wiki but it doesn't work
20:56.33pcharkyWell, i'm trying to use a hfc isdn card to make an ivr/callrecording system.
20:56.38Nethabhello
20:56.38lathos42anthm: I'm trying to figure out my best option for learning C at the moment
20:56.42*** part/#asterisk Nethab (~chatzilla@adsl-67-113-141-170.dsl.sntc01.pacbell.net)
20:56.49pcharkyBut for some reason the *nm card won't go into BRI mode.
20:57.08anthmdont use asterisk code =>
20:57.25pcharkylathos42: "summit C Programming FAQ's" is a real good bootk.
20:57.42astoriapcharky: thats probably a good mailing list question :)
20:57.51*** join/#asterisk ArkyLady (ArkyLady@adsl-66-142-125-19.dsl.ltrkar.swbell.net)
20:58.03*** join/#asterisk LeeColleton (~lc@dsl254-021-168.sea1.dsl.speakeasy.net)
20:58.05lathos42pcharky: is that a good beginner's book?  I have no programming experience whatsoever
20:58.10pcharkyastoria: Okay, I'll post it in the morning.
20:58.25astoriapcharky: you're more likely to find someone using that kind of card there..
20:58.42pcharkylathos42: It's okay if you're somewhat into programming, find some tutorials on the web..
20:58.52LeeColletondoes anyone have experience with the iaxtalk company or used their phones? (AT-320EE)
20:58.52anthmyou may want to install some version of unix with the full packages of TFM
20:58.53pcharkylathos42: And whatever you do, don't buy dummies books.
20:59.17pcharkyastoria: Thanks.
20:59.56ginventI keep getting this error: /usr/src/zaptel/ztdummy.c:41: error: syntax error before '<<' token when trying to compile zaptel
21:00.05eKo1c is so unforgiving, not like scripting languages.
21:00.39pcharkyginvent: did you get a stable release?
21:00.44ginventI think so.
21:01.07pcharkyginvent: how did you acquire the source.
21:01.22ginventI will have to recheck.
21:01.40*** part/#asterisk jdg (~jdg@CA03F93D.adsl.mana.pf)
21:01.48lathos42Thanks for the advice everyone.. Its time for me to get out of here..  I hope everyone has a good night/day depending on where you're at
21:01.57anthmlooks like a failed merge
21:02.08twisted[asteria]anthm, i second that notion
21:02.21pcharkylathos42: Thanks, good luck learning C..
21:02.34gaggamanshmaltz you have to tftp the tones. which phone?
21:03.49shmaltzgaggaman, I'm trying to use the defaults (bellcore-dr1) and so on
21:04.56gaggamanoh, you are talking about the ringtone you hear when you call somebody?
21:05.10filewhy do people always call when I'm making food
21:05.15gaggamanor about the "bell" ?
21:06.16pcharkyfile: did you search for a secret webcam in you kitchen? ;-)
21:06.16gaggamanfile: connect your asterisk to the oven and forward the caller to vm if the oven is on.
21:07.00filebah
21:07.43gaggamanthe exten's for that should be a no-brainer :-)
21:08.12fileDial(SIP/fridge&SIP/oven&SIP/microwave)
21:08.14*** join/#asterisk Blissex (~Blissex@82-69-39-138.dsl.in-addr.zen.co.uk)
21:08.20pcharkygaggaman: quite cook.. ehh cool though ;-)
21:08.26*** join/#asterisk SarahEmm (~sarahemm_@MTL-ppp-159032.qc.sympatico.ca)
21:08.47shmaltzgaggaman, I'm talking about changing the what can be changed with alert_info (bellcore-dr1 thru 5)
21:09.47*** part/#asterisk Nebukadneza (~daddel9@i3ED6E58B.versanet.de)
21:09.53peglaxI have a single HFC card in NT mode and get lots of buffer overflows/underruns from zaphfc and hear a crackling noise. Tried the Florz patch, which doesn't help.
21:10.13peglaxAny ideas?
21:10.45heath__how do i make it so that a zap line get transferred to an extension as soon as i pick up the phone?
21:10.49pcharkypeglax: Just a question of my own, is you HFC card in BRI? If so how did you manage to do that?
21:11.17pcharkypeglax: Sorry, no ideas on your problem (extreme newbie, just started this afternoon.)
21:12.08*** join/#asterisk mithro (~tim@lester.mithis.com)
21:12.17SarahEmmhihi
21:12.50pcharkyhey
21:13.06cjoh, please help
21:13.08cj:)
21:13.23cjSarahEmm: You like helping n00bs, right? :)
21:13.53clive--anthm would ast_bridge stopping and starting mess up timeouts ?
21:14.23SarahEmmcj: err, maybe. why?
21:14.45cjSarahEmm: 'cuz I am lost..
21:14.48SarahEmmwith?
21:14.48anthmstarting and stopping from what ?
21:15.04cjHow does my extensions.conf know to fire off the [demo] context?
21:15.16cjhow do I tell it to use something else?
21:15.16SarahEmmcj: because something points it there.
21:15.25SarahEmmcj: use something else when you do what?
21:15.46clive--anthm something wierd, when dtmf digits get pressed I get ast_bridge saying bridging stopped, and then I guess it troes to re-bridge the channels after the dtmf tone
21:15.51pcharkycj: Unless I'm mistaken, somewhere in the driver config.
21:15.57anthmoh native bridge
21:16.01cjSarahEmm: I want it to present a different message than "You have successfully installed Asterisk"
21:16.10anthmthat is part of the sucky way a call works
21:16.20pcharkycj: in HFC (zapata.conf) i have context=demo
21:16.32anthmeach channel has it's own brifge func
21:16.35SarahEmmcj: when what happens? zap calls come in? SIP? IAX?
21:16.36peglaxpcharky, I don't really get what you mean, I guess by using zaptel it is a BRI
21:16.46anthmif you do a bridge on 2 channels the same codec and the same type
21:16.55anthmit runs that code instead
21:16.57nathewhat's HFC?
21:17.07*** join/#asterisk malandro (~awefawe@201.240.131.254)
21:17.08af_it's a chipset
21:17.09anthmbut since all the funky features are not implemented in native bridge
21:17.11natheah
21:17.12natheok
21:17.24anthmwhenever you dial a digit it exits back for 1 loop
21:17.32natheI know it as Hybrid Fiber Coax
21:17.45malandrodoes anyone tell me where can i download asterisk for freebsd?
21:17.47Kattytwisted[asteria]: it would help if i put the paper into the bypass the correct direction
21:17.59cjSarahEmm: zap calls come in
21:18.09pcharkypeglax: Oh, if i check out: zap show channel 1, it says Signalling Type: PRI Signalling
21:18.11cjpcharky: thanks, I'll look at that
21:18.21nathe:P
21:19.03pcharkypeglax: shouldn't that be BRI?
21:19.51*** join/#asterisk Carp1 (carp_xigon@ip-204-97-151-81.modem.logical.net)
21:19.51clive--anthm, if it exits for one loop and doesnt re-bridge, then the timeouts wont work...does that sound probable?
21:20.00pcharkypeglax: Another thing I don't understand is when i load the module in NT mode, the 'zap' command isn't there, and no channels are available through: 'show channels'
21:20.07Carp1is there any full documentation that is at least 90% up-to-date?
21:20.19anthmis it the L
21:20.23anthmcall limit
21:20.23shmaltzguys thanks for the help, I had to set _ALERT_INFO, and not what the wiki said ALERT_INFO.
21:20.39SarahEmmcj: in zapata.conf change the context
21:20.52ginventI got the latest zaptel from cvs, but it still will not compile.
21:20.59anthmif the flag is set for call limit via L() in app_dial
21:21.10anthmthen native bridge should be impossible
21:21.16pcharkyginvent: what os are you running?
21:21.22ginventubuntu
21:21.30ginventkernel 2.6
21:21.36anthmit was designed to skip native bridge when you use L (by me)
21:21.47pcharkyginvent: aren't there binary packages available for ubuntu? there are for debian.
21:21.57anthmif that doesnt work, then someone broke it
21:22.13peglaxpcharky, I have PRI Signalling too. zap show channels works here in NT mode also.
21:22.25ginventThere are binary packages... but I wanted to compile the latest.
21:22.42Carp1is there any full documentation that is at least 90% up-to-date?
21:22.47SarahEmmlol
21:22.50SarahEmmwell...
21:22.52SarahEmmthe wiki, sort of.
21:22.55*** join/#asterisk bdunn (~bwdunn@c-24-0-49-250.hsd1.tx.comcast.net)
21:22.57SarahEmmother than that, no.
21:23.44bdunnHi all.  What kind of computer would I need to consider for 30 SIP phones and 16 voice lines?  Is that a Dual Xeon or just a fast Pentium 4 with lots of RAM?
21:23.57clive--anthm yes it is the dia(L) option
21:24.20anthmit should *never do a native*
21:24.27anthmnever
21:24.31anthmif L is on
21:24.41ginventI have zaptel working, but incoming faxes don't work... it picks up, sees the fax, switches over, but doesn't work after that...
21:24.53pcharkypeglax: I guess I needed a rest this afternoon, the 'zap show channels' work now (just changed to NT mode).
21:24.54anthmbecause native bridge never returns till you hit dtmf
21:25.03anthmso you never land on the code to test the timeout
21:25.15anthmi know it worked when I added the feature
21:25.25pcharkyDoes anybody know anything about asterisk cooperating with lucent index PBX's?
21:25.52*** join/#asterisk colinm_ (~colol@VDSL-130-13-8-95.PHNX.QWEST.NET)
21:25.58cjthanks, SarahEmm
21:26.22cjOkay.  Now that I have the context down, how do I record and select audio snippits?
21:26.28cj.ogg?
21:26.29NukemizerIs it possible to make an E&M channel provide Dial Tone to an end user ?
21:26.31cj.mp3?
21:26.51*** part/#asterisk bdunn (~bwdunn@c-24-0-49-250.hsd1.tx.comcast.net)
21:26.57SarahEmmcj: err... using the Record function, you mean?
21:27.09pcharkycj: http://www.voip-info.org/tiki-index.php?page=PBX%20Call%20Recording
21:27.09SarahEmmNukemizer: err.. you have a phone with an E&M interface?
21:27.13cjThanks, pcharky
21:27.29Nukemizeryes :)  connected to a Legacy PB
21:27.34NukemizerPBX
21:27.38clive--anthm I am not sure why its not working when dtmf is pressed..., but at least the bug has been narrowed down to a specific area
21:27.38SarahEmmerrr...
21:27.39SarahEmmokay..
21:27.44SarahEmmwhat kind of phone is that?
21:28.26NukemizerSarahEmmwhen a trunk is accessed on the PBX, the users need to hear dialtone form *
21:28.28anthmi pretty much told you =), the code is not making a native bridge impossible
21:28.51anthmit should be
21:28.55anthm0 native bridge
21:29.00SarahEmmahh.
21:29.38SarahEmmlol
21:29.43SarahEmmwho also sucks
21:29.48SarahEmmbut has blackberries that work internationally
21:30.06Beirdowell...  the $366 cell bill for June... blech
21:30.20SarahEmmheh
21:30.23SarahEmmyuck.
21:30.26BeirdoI'll be changing LD plans...  and if they can't fix that... I'll call through asterisk at home
21:30.34Beirdofuck them and their ridiculous rates
21:30.51twisted[asteria]whoa
21:31.05twisted[asteria]bell mobility still exists?
21:31.05BeirdoI'll give JerJer 4c/min or so instead of Bell Mobility 99c/min or whatever...
21:31.20Beirdoleave it to me to call Puerto Rico so much from Canada :)
21:31.21Beirdoyeah
21:31.32twisted[asteria]i thought they got divvied up by verizon/cingular
21:31.41Beirdono, this is Bell Canada
21:31.44twisted[asteria]oooh
21:31.47Beirdo:)
21:31.56Beirdosorry for the confusion :)
21:32.02*** join/#asterisk cjk (~cjk@80.92.75.43)
21:32.07twisted[asteria]heh, it's okay.  I keep forgetting that kanada has a bell
21:32.11SarahEmmBeirdo: DISAing through your home box?
21:32.11Beirdotheir roaming rates are ass too
21:32.14cjkhi, anyone here using odbc voicemail storage?
21:32.29BeirdoI will soon be calling out through home, I think
21:32.34*** join/#asterisk apardo (~apardo@85.Red-81-39-78.pooles.rima-tde.net)
21:32.53*** join/#asterisk Jinxy (~UNIX@toronto-HSE-ppp4006295.sympatico.ca)
21:32.57BeirdoI already have it all setup
21:33.09cjkcj: why  are your pokeing me?
21:33.12*** join/#asterisk RRaven (~reignrave@69.15.83.145)
21:33.17clive--anthm thanks for your pointers....
21:33.23Beirdonow what would be absolutely ideal...  a IAX2-terminated DID with a Puerto Rico number
21:33.27Beirdobut Noooooo
21:33.44Kattytwisted[asteria]: BEEP BEEP
21:34.08Beirdoif any of you guys are providers and have 939 or 787 DIDs, please drop me a /msg :)
21:34.12Beirdohehe
21:34.43Beirdomeanwhile, she's getting an IAXy in August (assuming it arrives from the eBay dude before that)
21:34.55file[laptop]yikes momentary lapse of memory
21:34.59file[laptop]I thought it was Monday
21:35.19Beirdoit is for me :)
21:35.50BeirdoI had to empty my storage locker in the sauna-like heat yesterday
21:36.12twisted[asteria]Katty, heh.. i didn't quite understand what you were talking about :P
21:36.35pcharkytwisted[asteria]: something with keys to a jeep?
21:36.44twisted[asteria]pcharky, huh?
21:36.56twisted[asteria]omgwtfbbq
21:37.02file[laptop]O.o
21:37.06pcharkytwisted[asteria]: never mind.
21:37.10Beirdosigh.  so time to go talk to a Bell Mobility dweeb to change LD plans.  blech
21:37.22file[laptop]Beirdo: DISA is your friend
21:37.36Beirdodefine DISA?  my memory sucks
21:37.45Kattytwisted[asteria]: i was all, once upon a time there was a copier who thought it wanted to print on 5.5 x 8.5 paper
21:37.51file[laptop]direct inward dial system?
21:37.55Beirdotoo many acronyms in my brain
21:37.57Beirdoahh
21:38.01twisted[asteria]Katty, ahhhhh
21:38.11twisted[asteria]Katty, t'was a story indeed
21:38.14file[laptop]think callback with internal dialtone, hrm yes
21:38.15Katty;)
21:38.18RRavenHi, First of all let me say that I am just in the process of exploring Asterisk.  It seems like a great PBX platform.  Can anyone tell me if Asterisk can route calls to Agents based upon the location of the caller...?  I know that SwitchVox says that their software can, but it is just built on top of Asterisk, right?
21:38.21Beirdowell, I have it setup so I can authenticate by PIN and dial out
21:38.30Beirdobut that's kinda a nuisance :)
21:38.40Beirdoheh
21:38.49BeirdoI think I'll change plans anyways
21:38.52Kattyfile[laptop]: how many business cards should i print out?
21:39.05*** join/#asterisk Romik_ (~romik_@1.fix.netvision.net.il)
21:39.07file[laptop]Katty: you should print me some tooooo
21:39.18sig-and me too..
21:39.19BeirdoI don't need a repeat of this lousy bill. :)
21:39.19sig-=)
21:39.37Beirdoseeya dudes later.
21:41.11*** join/#asterisk Zaw (zaw@zaw.subneural.net)
21:41.24pcharkyDoes anybody know anything about asterisk cooperating with lucent index PBX's?
21:41.36Kattyfile[laptop]: gotta give this 15 grand color copier a workout
21:41.37cjSarahEmm: So if I set up an extension to Record(foo), I can record by dialing that extension?
21:41.50clive--anthm mind if I send you a priavte message?
21:41.54*** join/#asterisk outtolunc (outtolunc@adsl-66-218-53-170.dslextreme.com)
21:41.55SarahEmmcj: err, record like a voicemail, yeah
21:42.08cjHow 'bout recording a prompt?
21:42.59Romik_somebody can explain me what is dead package in "iax2 show stats" ? it packages out of order and not hannled by jitbuffer?
21:43.19*** join/#asterisk smani (smani@ool-45736978.dyn.optonline.net)
21:43.54*** join/#asterisk ZX81 (matt@222-153-16-23.jetstream.xtra.co.nz)
21:44.04ZX81hi all
21:44.06pcharkycj: As in IVR? Taking input from the keypad?
21:44.16ZX81anyone know why this is happening repeatedy on compile:
21:44.21ZX81rm -f include/asterisk/version.h.tmp
21:44.41pcharkyZX81: probably because it's .tmp
21:45.28ZX81ok
21:45.31*** join/#asterisk Twister (Twister@24-179-84-004.dhcp.chtn.wv.charter.com)
21:45.36smaniI have Asterisk server running on a DMZ. Successfully configured local IAX clients. But having problems with outside IAX clients registered this server. When dialing extension says "Connect attempt rejecteed".
21:45.37ZX81but it's looping endlessley
21:45.38cjkhi, i configure unixodbc to work with mysql. is there any sort of command line client i could use to test if i did everything right (launch some qurries)
21:45.39drumkillaHEY!  everyone is stealing my circles thing
21:45.46ZX81maybe the thing it is compiling is the problem
21:46.01Twisterif a line has call waiting on it, how does * handle that
21:46.57Twisterfor example, I have an initial menu, and the line has call waiting
21:47.18Carp1Do you prefer analog phones for an office phone or IP phones?
21:47.27Twisterdoes the line just ring for the second caller till the first line is finished
21:47.33*** part/#asterisk pcharky (~khagen@ip503c6061.speed.planet.nl)
21:47.37DarthClueCarp1: IP
21:48.23smanican somebody please help with IAX client configuration ?
21:49.22Carp1I want to read docs on asterisk but I think they are out of date, am I right?
21:49.28SarahEmmread the wiki
21:49.31Carp1Want to print and read away from computer
21:49.55SarahEmmprint the wiki
21:49.56SarahEmm;)
21:49.58SarahEmmit's the docs, right now.
21:50.04DarthClueCarp1: the wiki, the wiki.
21:50.05*** join/#asterisk harryvv (~none@S010600a0c93f6f7e.vs.shawcable.net)
21:50.09*** part/#asterisk mkrufky (~mk@68.160.103.77)
21:52.13*** join/#asterisk Sedorox (~Brandon@sedorox.staff.smartserv)
21:53.07*** join/#asterisk heison (~heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com)
21:56.17*** join/#asterisk peglags (~peglax@p54B21B8B.dip0.t-ipconnect.de)
21:56.27*** join/#asterisk Uther_P (~uther_p@66.180.120.82)
21:56.54*** part/#asterisk Uther_P (~uther_p@66.180.120.82)
21:59.35*** part/#asterisk colinm_ (~colol@VDSL-130-13-8-95.PHNX.QWEST.NET)
22:03.29*** join/#asterisk SarahEmm (~sarahemm_@MTL-ppp-159032.qc.sympatico.ca)
22:04.03harryvvthis is new..whats with the long dialtone pules rather short ones when picking phone up
22:06.39*** join/#asterisk shido (~shido@d57-87-253.home.cgocable.net)
22:14.28*** join/#asterisk Pfhorge (~pfhorge@rrcs-24-172-161-65.central.biz.rr.com)
22:15.02PfhorgeCan anyone help me with a Digium TDM400 installation problem?
22:17.47Corydon-wWhy don't you ask your question, and see if anybody in here knows the answer?
22:18.46Pfhorgewell, I'm trying to get 2 TDM400 cards working under linux. I can modprobe the drivers, and I get a log message like "Zapata Telephony Interface Registered on major 196"
22:18.47shido:)
22:18.54NuggetIs it OK if I ask a question?  If I ask a question, will anyone answer?  Does anyone here know the answer to the question I'm going to ask?
22:19.03NuggetLet me know when I can ask my question.
22:19.06wulfy814anyone know of a site with a walkthrough configuring Telix for SIP with * ?
22:19.17shidoand?
22:19.19wulfy814Teliax that is
22:19.28RaYmAn-BxNugget: the answer is moo. :P
22:19.32Nuggetyay
22:19.33Syncrosthat's 4 question already
22:19.41Pfhorgewhen I set up my zaptel.conf file and do ztcfg, I get a message like "ZT_CHANCONFIG failed on channel 1: No such device or address (6)"
22:19.43shidoPfhorge, and?
22:19.48shidook
22:19.50shidowell
22:19.50Pfhorgetyping :)
22:20.02shidouse pastebin.ca to post your zaptel.conf
22:20.05shidoand zapata.conf
22:20.12Corydon-wWhich drivers are you loading?
22:20.27MikeJ[Laptop]Corydon-w!!
22:20.41Pfhorgedoing "modprobe wctdm", which loads zaptel and wctdm
22:21.15Corydon-wTry typing 'dmesg' and see if there are any error messages in the kernel log
22:21.21Pfhorgethere aren't
22:21.27Syncrostry channel 4
22:21.38shidoshido use pastebin.ca to post your zaptel.conf
22:21.38shidoshido and zapata.conf
22:21.49*** join/#asterisk mariogamboa (~sudaikdd@201.138.151.155)
22:21.55Pfhorgepastebin.ca?
22:22.02shidohttp://pastebin.ca
22:22.10mariogamboahi i found the problem with the load of my e1 and tdm400p
22:22.13shmaltzis there a way to change the DTMF mode for sip at runtime?
22:22.13mariogamboacool
22:22.22mariogamboais the zaptel init script
22:22.55mariogamboai don't see how to make the script load first the wcte11xp and after the wctdm is how i need it
22:23.00Carp1Who wrote AMP?
22:23.21mariogamboammm
22:23.27mariogamboai don't remember
22:23.28Pfhorgezaptel.conf at http://pastebin.ca/18088
22:23.28Romik_somebody could explain what is mean dead package in "iax2 show stats" ?
22:24.57*** join/#asterisk XamoDoug2 (~doug@216.36.186.53)
22:25.03*** join/#asterisk Cresl1n (~Cresl1n@207.111.174.1)
22:25.09shmaltzok, sipdtmfmode does just that, any examples?
22:25.51mariogamboaany know how to make the zaptel script can load firest the module wcte11xp and after the module wctdm?
22:25.54*** join/#asterisk puowvip (ircuser@the-legacy-never-dies.diamond.org)
22:26.06puowvipwhumpus.
22:28.15*** join/#asterisk litage (~nick@ws01.5749.dsl.winshop.com.au)
22:28.28wulfy814ok, I'm working on the SIP connection to Teliax, inbound first
22:29.12wulfy814I'm getting Failed to authenticate user "8005551212" <sip:8005551212@x.x.x.x>;tag=as6d2e30db
22:29.27wulfy814when I call my Teliax DID
22:30.07smanican somebody help me with IAX conf to do a FWD kind of setup? Single hosted asterisk server with multiple remote clients ?
22:32.14*** join/#asterisk jskcr (~jskcr@jskcr.user)
22:33.05twisted[asteria]ooooh can't you see
22:33.07twisted[asteria]you belong to me
22:33.12twisted[asteria]blah balh blah
22:33.27twisted[asteria]</singing>
22:33.43jskcrhy all
22:33.44*** join/#asterisk adval40 (~adval40@83-98-240-43.teleadsl.nl)
22:33.49Twisterhey twisted[asteria]...share whatever your smokin with the rest of us :)
22:34.11Twisterbah!
22:34.20twisted[asteria]don't ask if you don't want it.
22:34.22Twisterdont ya know those thingz will kill ya
22:34.31Twisterand everyone else around
22:34.31twisted[asteria]yeah, but so will living.
22:34.36TwisterLOL!
22:34.41Twisteri guess you ot a point
22:35.53twisted[asteria]of course I do
22:36.11adval40i have a question about AAH
22:36.26*** join/#asterisk lters (~lters@mrtcdsl-034.mis.net)
22:37.31adval40when i call outside i must dial 0031 and the number of  the person
22:38.14adval40in asterisk i have modify extensions.conf so that  i call only the number of the person
22:38.26PfhorgeTrying to get 2xTDM400 working. Modprobe works, zaptel.conf is at http://pastebin.ca/18088 , getting error "ZT_CHANCONFIG failed on channel 1: No such device or address (6)" from ztcfg. Any ideas?
22:38.56adval40where can i paste the 0031 in a config ?
22:39.29adval40i'm sorry but i can't say it betterin english
22:40.23tootPfhorge - i did suffer that problem recently - my resolution was crap - i upgraded gcc, etc, compiled a 2.6.12 kernel and it worked (using HEAD)
22:41.02Pfhorgehmm. I'm on Gentoo, so I'm just using their package, which looked pretty recent. I'll try the CVS stuff
22:42.54*** join/#asterisk _Sam-- (~sam@207.245.79.253)
22:44.12*** join/#asterisk festr_ (~festr@ns.regnet.cz)
22:44.23festr_hello, anyone using latest cvs head?
22:45.06festr_i have problem with MOH -> SIP
22:45.28festr_sound is choppy
22:45.30_Sam--can someone point me in the right direction....i want asterisk to call my cell phone ( i get free incoming minutes) and then have it connect to the part i want to call....like dial my cell phone, wait 10, dial calling number, then connect both....
22:45.46_Sam--but i cant figure out the way to do it
22:46.06festr_playback work correctly
22:46.41eKo1_Sam--: i remember reading about an agi that can call too parties and connect them.
22:46.48eKo1s/too/two
22:46.56festr_dont know what to debug. stable cvs works nicely on the same box (i have tdm card loaded)
22:47.06festr_maybe trouble with timer?
22:47.10*** join/#asterisk bjohnson (~bjohnson@i216-58-63-93.igs.net)
22:49.24_Sam--this better sums up what i want:  "Can I use asterisk to call party A then call party B and finally connect
22:49.24_Sam--party A to party B, so they can talk to each other?
22:49.25_Sam--"
22:50.39eKo1search google for an agi that does that
22:50.48_Sam--ok i'll give it a shot, thanks
22:52.37Druken_Sam--: you want to make 2 calls and have them connected? or you want to do like a 3way call ?
22:53.02*** join/#asterisk tripleFFFasdjgif (~TripleFFF@modemcable131.156-131-66.mc.videotron.ca)
22:53.24_Sam--what im looking for is to have my A* call my cell phone, wait for me to answer, then dial my calling party, and finally connect me and the calling party -- i wouldnt use any cell phone minutes since i get free incoming
22:53.37*** part/#asterisk tripleFFFasdjgif (~TripleFFF@modemcable131.156-131-66.mc.videotron.ca)
22:53.45_Sam--so more like mak2 calls and have them connected
22:54.15*** join/#asterisk tripleFFFasdjgif (~TripleFFF@modemcable131.156-131-66.mc.videotron.ca)
22:54.28*** part/#asterisk tripleFFFasdjgif (~TripleFFF@modemcable131.156-131-66.mc.videotron.ca)
22:54.39Drukenwhy not just do a 3way call on a voip phone, first to your cell, then to the person ya want to answer...
22:54.56Drukenwhen you hang up the voip phone, your cell and the other person should stay connected
22:55.17_Sam--because ideally i could do it remotely...call from my cell to A*, and tell it the number i want to call...then hangup.  it would then call me back and call the calling party
22:55.36_Sam--if i could figure out how to connect the 2 parties i think i could do the first part easy
22:55.44festr_hmmm. :(
22:55.45festr_Jul 20 00:55:11 DEBUG[3007]: chan_sip.c:9679 sipsock_read: Failed to grab lock, trying again...
22:56.00festr_cvs head asterisk pretty unstable
22:57.59BeirdoGAH
22:58.15Beirdomy asterisk box just logged me off the console, wonder what I hit?
22:58.34Beirdoand do I remember the root passwd?  noooo
22:59.55jskcrctrl alt delete
23:00.01jskcrthat will do it
23:00.02Beirdoheh
23:00.06Beirdono it won't
23:00.21Beirdothat won't make me remember the password
23:00.33jskcrprobably ctrl D then
23:00.33*** join/#asterisk mcreedjr (~mikejr@oh-65-41-206-34.sta.sprint-hsd.net)
23:00.33Beirdoand it's on serial console, so there's no Ctrl-Alt-Delete
23:00.42_Sam--thats what you get for using root too much...should login as yourself and SU as needed!
23:00.47Beirdoit killed screen and everything.
23:00.53BeirdoI do, silly
23:01.00_Sam--im just being a jerk, sorry
23:01.06Beirdoand then I ran screen
23:01.15Beirdoand several months later it logged me off
23:01.23festr_anyone here using latest cvs?
23:01.24mcreedjrI have a Sipura phone which dials it's voicemail extension when you press the voicemail button. What kind of extension logic do I need to have it execute the VoiceMailMain command instead of calling myself?
23:02.14festr_mcreedjr: read some basic docs
23:02.30festr_mcreedjr: or type show applications in asterisk CLI
23:02.37loudtry 15
23:03.18festr_it seems that i have something broken in my upgraded asterisk to head. iax doesnt work (iaxcomm) SIP moh is choppy :(
23:03.44mcreedjrfestr_: I've looked around, thats why I am asking... I will keep digging. The stuff I tried didn't work.
23:03.53mcreedjrfestr_:Thanks...
23:03.59festr_mcreedjr: what did you tryed
23:05.25mcreedjrfestr_: I tried to filter based on caller ID when my SIP extension was rung. For instance exten => 200/200,1,VoiceMailMain(200)
23:05.37BeirdoARGH
23:05.42BeirdoI remembered the password
23:05.47Beirdobut su is busted
23:05.47Beirdoheh
23:06.13festr_mcreedjr: sort your exten into specified context
23:06.36mcreedjrfestr_: What do you mean?
23:07.21*** join/#asterisk buddah (~hnic@208.179.86.5)
23:07.21*** join/#asterisk Corydon-w (red@vcchgate.vcch01.springfield.tn.us.vcch.net)
23:07.36festr_mcreedjr: or make some default number to go in right voicemail. for example voicemail => 15,1,voicemailmain(${CALLERIDNUM})
23:08.06buddahis there a way for clients to check their voicemail for their sip phones from say a cell phone?
23:08.22_Sam--buddah:  of course, just define an extension
23:08.24festr_buddah: yes
23:08.42buddahlike exten = > xxx,1,VoiceMailMain
23:08.43buddahl,ike that?
23:08.46_Sam--exten => 8500,1,VoicemailMain
23:08.48buddahjust instead of the default one?
23:08.59mcreedjrfestr_: exten => 300,1,VoiceMailMain(200) -- I just set that up, but I wanted that cool little voicemail button to work :)
23:09.01buddahso i just use a DID to set up a call in
23:09.25festr_mcreedjr: dont understand
23:09.27_Sam--or if you are using IVR you could just dial the extension
23:09.31festr_mcreedjr: where is problem?
23:09.33buddahnot an ivr
23:09.50buddahhmm, i'll play around with that, thanks
23:10.12mcreedjrfestr_: Nevermind... Its working. Thanks for your help.
23:10.17festr_ok
23:10.31festr_:)
23:10.48_Sam--where is that AGI type files need to get written to get automagically run?
23:11.12_Sam--has to connect through a local socket, or could leave a file in a spool type directory?
23:11.19*** join/#asterisk adval40 (~adval40@83-98-240-43.teleadsl.nl)
23:13.03*** join/#asterisk puzzled_ (~patrick@puzzled.xs4all.nl)
23:13.47BeirdoOK, unborked my zaptel config too
23:13.48Beirdooops
23:14.03Beirdowhy did I turn off call progress detection?
23:14.17Beirdoand why does that bork DTMF detection?
23:16.58*** join/#asterisk mjmac (~mjmac@mjmac.active.supporter.pdpc)
23:18.08*** join/#asterisk adval40 (~adval40@83-98-240-43.teleadsl.nl)
23:19.08mjmacis trying to use a TDM400P installed in a soekris 4801 going to be more work/annoyance than it's worth?  i know the standard case doesn't accommodate the card.  any other issues that people have run into?  (yes i'm also googling)
23:20.07*** join/#asterisk Corndawg_ (whoisit@c-66-176-249-51.hsd1.fl.comcast.net)
23:21.44mjmaci've been running * on a frankenbox in my basement for over a year, but the box is less than reliable and i'm looking for a cheap replacement.  might also just ebay something like an optiplex.
23:22.45*** join/#asterisk SarahEmm (~sarahemm_@MTL-ppp-159032.qc.sympatico.ca)
23:22.51SarahEmmew.
23:23.39Beirdoheh
23:23.41natheI love the word "frankenbox"
23:23.49Beirdowelcome to a montreal IP, SarahEmm
23:24.18mjmacnathe: complete with cobwebs
23:24.23BeirdoOH, BTW, you are evil still :)
23:24.24*** join/#asterisk Micc (~dotirc@c-24-18-35-120.hsd1.wa.comcast.net)
23:25.21blitzrageSarahEmm: in mount royal eh?
23:25.54SarahEmmlol
23:25.55SarahEmmevidently
23:25.59SarahEmmhihi nathe
23:26.51*** join/#asterisk bzbw (~wlwzhang@ip67-153-142-109.z142-153-67.customer.algx.net)
23:28.19SarahEmmmew :(
23:29.57mcreedjrAre there any free IAX providers besides FWD?
23:30.16mcreedjrI just want to play around with IAX trunking :)
23:30.35sig-yes.
23:30.35Nuggetso set up two boxes.
23:31.08mcreedjrsig-: I just the question should've implied I wanted an answer too :)
23:31.21mcreedjrNugget: I just might... I just might.
23:32.02sig-"so set up two boxes" :P
23:32.30mcreedjrsig-: Thanks...
23:32.33sig-or use voipjet, with 0.25$ :>
23:32.41sig-voipbuster..
23:32.47sig-and many others.
23:32.57sig-just look at wiki, it`s really so hard?
23:33.40mcreedjrsig-: Hey now... you were doing so good without flaming the n00b. Alright.. I look at the wiki more before I ask "dumb" questions. lol
23:35.31*** join/#asterisk outtolunc (outtolunc@adsl-69-110-52-25.dsl.pltn13.pacbell.net)
23:36.21Drukenwiki == less stupid questions
23:36.35Drukennot nessessarily less questions... just less stupid ones
23:36.44mcreedjrsig-: Hrrmmm "VOIP Service Providers Residential". Who'da thunked it :)
23:36.47*** join/#asterisk Ariel_ (~Ariel@dsl-20-177.cofs.net)
23:37.01opus_please post your question to pastebin so you don't bother the people trying to idle in the channel thanks
23:37.02mcreedjrThe wiki is sweet
23:37.07cjsig-: can you point him to the right page on the wiki?
23:37.16cjmcreedjr: yeah, we love the wiki here :)
23:37.25mcreedjrcj: I found it thanks
23:37.25cjmcreedjr: all you need to do is read it from cover to cover :)
23:37.38mcreedjropus_: Are you directing that at me?
23:37.43Drukenthe wiki has covers?
23:38.42mcreedjrDinner time
23:38.50mcreedjrThanks guys/gals for the help
23:39.06opus_it was a joke
23:39.47SarahEmmlol Druken
23:39.52Drukenopus_: your just an idle lurker
23:40.06DrukenSarahEmm: :P
23:40.12opus_please, use pastebin i'm trying to sleep
23:41.03file[laptop]SarahEmm: KITRICH
23:41.10Drukenhttp://pastebin.ca/583473465383476489374397598709860565954687364859568560956
23:41.42file[laptop]outtolunc: nooooooo
23:42.06Drukeni can officially say i have NEVER fallen asleep infront of my computer
23:42.14file[laptop]I have
23:42.15Drukeni know when to goto bed
23:42.19file[laptop]I have a picture of it too
23:42.38file[laptop]my hands were on my keyboard too...
23:42.46Drukeni used to have a friend who fell asleep in front of his on a nightly basis...
23:42.47file[laptop]I shall find it
23:43.04SarahEmmhi file
23:43.05SarahEmmmeow!
23:43.08file[laptop]hiiiii
23:43.12SarahEmmshi
23:43.14SarahEmmoops
23:43.47file[laptop]http://myworld.q2u.net/images/Photoblog_20050320_SleepingFile.jpg
23:43.58file[laptop]I had the sense to take off my glasses mind you
23:44.35Drukenfile[laptop]: was that in the hotel at astricon ?
23:44.40file[laptop]VON
23:44.56Drukenwasen't VON in toronto ?
23:44.56outtolunchaha
23:44.58file[laptop]and if you go to Cluecon you can see it happen there in real life!
23:45.03file[laptop]Druken: there's MANY VONs
23:45.15outtoluncand no one superglued your fingers to the keyboard while you slept .. damn
23:45.17Drukenahh... i c
23:45.21file[laptop]outtolunc: indeed
23:45.55outtolunci wish i would have gone
23:46.12Drukeni couldn't use my laptop like that...
23:46.22outtolunci do every night
23:46.24file[laptop]I do it all the time
23:46.28outtoluncmore pillows tho
23:46.43file[laptop]two pillows... one feather, one something else
23:46.46Drukendesk and chair for me thanks...
23:46.57Drukenmy back wouldn't thank me to do that
23:47.10outtoluncthe back is fine, it's the neck that gets me
23:47.36Drukenmy back is screwed... unfortunatly... and at such a yonge age too... :(
23:47.43file[laptop]oh frell, House is on at 10... so is Big Brother
23:49.58cjfile[laptop]: tivo to the rescue!
23:50.32file[laptop]ha
23:51.02Drukeni'd like to own a small cable company... i think that'd be cool....
23:51.14SarahEmmooh, house is on tonite.
23:52.12Drukenand house is... ?
23:52.19*** part/#asterisk Cresl1n (~Cresl1n@207.111.174.1)
23:53.19file[laptop]television show
23:53.39SarahEmmwoo and it's a new one
23:53.42SarahEmmlast week was a rertun
23:53.53Drukenreruns blow ass...
23:54.14Drukeni wish they would continue "dead like me" that show rocked
23:54.27blitzragewhen you pass args in AGI like AGI(my_agi.agi,arg1,arg2) those are args passed to the .agi file?
23:54.31cjDruken: yeah, and the chick was hot
23:54.41*** join/#asterisk adval40 (~adval40@83-98-240-43.teleadsl.nl)
23:54.49SarahEmmyes blitzrage
23:54.53blitzrageSarahEmm: thx :)
23:54.59Drukencj: i wouldn't have had to think about it... course, which chick ?
23:55.21blitzrageSarahEmm: for a couple mins I'm like, "how do I access those args from within the AGI", I'm dumb, I just need to pass info from a variable :)
23:56.43*** join/#asterisk JBenden (~JBenden@216.206.239.171)
23:56.46cjDruken: the newly undead one.  And mostly all of the undead chicks, really
23:56.51*** join/#asterisk adval40 (~adval40@83-98-240-43.teleadsl.nl)
23:57.11Drukencj: :)
23:58.18JBendeni've got a question/patch about cdr_addon_mysql and want to know how i bring it up so it gets added to the cvs... can anyone offer any advice?
23:59.55Nuggetstop using mysql.

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