irclog2html for #asterisk on 20050112

00:00.32_DAW~seen ManxPower
00:00.33jbotmanxpower is currently on #asterisk (4h 8m 25s).  Has said a total of 113 messages.  Is idling for 2h 12m 55s
00:01.14venixi especially like Balamb Garden.
00:01.37*** join/#asterisk Othello (Othello@nusnet-154-210.dynip.nus.edu.sg)
00:01.47*** join/#asterisk guugmember (~nachoramo@200.49.163.49)
00:02.07guugmemberhello guys, who has installed Asterisk over Ubuntu?
00:02.31venixtime to beat a hasty retreat from work, g'night
00:02.44*** join/#asterisk Muffie (~Macello28@200-233-53-180.user.ajato.com.br)
00:02.45kram"the documents for asterisk are a patchwork, and frankly they just suck"
00:02.58kram"it's a mystery to get from the sample configs"
00:02.59mtqhkram: where is that from?
00:03.01_DAWAnyone here working with IP 500/600 and *?
00:03.06kramhis presentation
00:03.08Muffiehello all. Anyone knows where should I add users/password to be used with gastman?
00:03.16kramsee, the cool part is that you can get the real story
00:03.33guugmemberwhat do you think of voxbox
00:03.34guugmember?
00:03.39kramif he knew i was here, he'd probably try to "not offend me"
00:03.52bkw_hahah
00:03.53bkw_ya
00:03.58modulus_yeah * docs suck
00:04.02*** join/#asterisk awb4422 (~ab@206.135.97.35)
00:04.06modulus_it's almost as bad as rpm documentation
00:04.08modulus_but not quite
00:04.09krami feel like i get a much better story from attending
00:04.09Muffieanyone?
00:04.11modulus_and that's really bad
00:04.14kramespecially because this guy really knows his stuff
00:04.18anthmraise your hand and say "dont forget the code looks obfuscated and has been known to cause blindness"
00:04.20kramit's a fantastic presentation
00:04.24brc_greetings kram
00:04.27twisted[work]LOL
00:04.31denonkram: learn anything new about *? <G>
00:04.31twisted[work]anthm, that's a good one
00:04.38bkw_hahahaha
00:04.41bkw_denon thats funny shit
00:04.42twisted[work]anthm, it's also known to cause hair loss and recurring headaches
00:04.42MuffieAnyone knows where should I add users/password to be used with gastman?
00:04.48denonbkw: *grin*
00:05.15modulus_the only way i'm able to work with asterisk is by reading the code
00:05.15denon"Asterisk software developer and founder mark spencer, seen taking notes at an Asterisk for Newbies seminar"
00:05.16guugmemberhow can I be sure my box has kernel tools and development tools?
00:05.37krami'm just amazed
00:05.46ckruetzekram: have you already though for a tricky question for the end?
00:05.50krami mean seriously, this guy *really* has his story down
00:05.58guugmemberdenon, do you mean spencer is not that good for Asterisk?
00:06.10denonguugmember: it was a joke
00:06.15guugmemberdenon, ok
00:06.20*** join/#asterisk Othello (Othello@nusnet-154-210.dynip.nus.edu.sg)
00:06.25anthmside effects include , sleeplessness , vomiting and headaches a few cases were fatal..
00:06.27guugmemberIm kind of newbie here, sorry guys
00:06.31krami feel bad i'm missing the guug voip conference
00:06.34denonkram: you surprised to see that others know their stuff? what kinda topics, just config?
00:06.35twisted[work]anthm, lol
00:06.37Muffiesomeone can lose 1 second and anwser me how can I manage users for using with gastman?
00:06.38mtqhguugmember: shutup
00:06.41krambut i guess you can't go to everything
00:06.46guugmemberwhat do you think of voxbox?
00:06.56mtqhmanager.conf
00:06.58bkw_voxbox needs some love
00:07.01bkw_its a good start
00:07.01twisted[work]kram, this is why you need clones
00:07.03NormAstFile[laptop]: THANK YOU....  THANK YOU.... just need to add the /n at the end and it works!!
00:07.07kramyah hehe
00:07.10fearnornot every user group is lame. www.pilosoft.com/Telephony.ppt is what i presented to NYLUG
00:07.14file[laptop]NormAst: hehe
00:07.15fearnorlinux lusers group
00:07.15Muffiethanks mtqh. But my manager.conf shows me nothing.
00:07.25kramwell yah, but you're alex
00:07.28mtqhthen check for the samles
00:07.34*** join/#asterisk PoWeRKiLL (~PoWeRKiLL@l01m-34-60.d4.club-internet.fr)
00:07.36kramthis is someone who basically just picked it up and started playing with it
00:07.42NormAstfile[laptop]:  Oh by the way..... THANK YOU..
00:07.51modulus_kram what's the topic?
00:07.57Muffiemtqh: where are they?
00:07.59bkw_kram that don't sound right
00:07.59kramasterisk
00:08.19mtqhsrc/asterisk/configs
00:08.24ryguyif I am setting up a CB that is for analog phones to plug into asterisk, the signaling on the t1 card would be FXO correct?
00:08.43tzangerryguy: yes
00:08.48bkw_drumkilla really?
00:08.49bkw_why?
00:08.50twisted[work]drumkilla, why?
00:08.54Muffiemtqh, damn. They are equal.
00:09.01drumkillabecause.
00:09.04Silik0nDONG
00:09.05bkw_because why?
00:09.08bkw_do tell
00:09.10mtqhmuffie: download your source tree again cause something is f-uped
00:09.11bkw_we would like to know
00:09.18tzangerfearnor: yeah but it's in .ppt format, you are lame.  :-)
00:09.19bkw_just sy fucked up
00:09.22twisted[work]drumkilla, enquiring bkw's want to know
00:09.24kramand he understands FXO and FXS
00:09.30twisted[work]kram, always a plus
00:09.33kramoh man, if you had any idea how many times malcolm's had to explain that one
00:09.35fearnortz: so shoot me ;)
00:09.37modulus_kram, you mean he knows what they stand for?
00:09.40bkw_plus FXS => FXS is what now?
00:09.43denonkram: you mean ohm drops and stuff?
00:09.44bkw_A BAD IDEA
00:09.51drumkillahehe
00:09.58Muffiemtqh, Only [general] section is not commented. the second section [mark] is all commented. I cant figure out where should I add an username...
00:10.03kramhe doesn't know what they stand for
00:10.04denonor just the basic concepts?
00:13.17modulus_lol kram
00:13.17twisted[work]bkw_, it's like putting the two positive leads of a 220v circuit together
00:13.17mtqhmark is a username
00:13.17krambut he knows the concept
00:13.17bkw_kram really?
00:13.17denon*cough* office *cough* station
00:13.17drumkillayeah
00:13.17denonask him what they stand for at the end <G>
00:13.17modulus_actually if you know the acronyms they're kinda self explanatory
00:13.17twisted[work]btw
00:13.17twisted[work]i have xlite for linux
00:13.17twisted[work]i'll comment on it later
00:13.17brc_uhm
00:13.17kramoh neat there is a local ITSP here
00:13.17bkw_Foreign Exchange Office (FXO)
00:13.17bkw_Foreign Exchange Station (FXS)
00:13.17Muffieanyone knows how to simple add an user to connect with gastman? what should be added to manager.conf?
00:13.21kramhe said he setup a SIP connection to someone in alabama
00:13.21krami wonder who that was....
00:13.21bkw_geee
00:13.21bkw_I wonder
00:13.21kramtwisted, was that you?
00:13.21drumkillanot me!
00:13.21mishehuforrest gump
00:13.21twisted[work]kram, maybe....
00:13.21terrapenthe way i remember them is that the 'S' in FXS is like "sender" as in, "sends the voltage"
00:13.21ryguydoes anyone know why an error message like this would come up even when there was nothing pluged into the t1 card? Jan 11 19:10:36 ERROR[-1084620064]: Signalling requested is FXO Loopstart but line is in E & M Immediate signalling
00:13.21kramyou remember setting up SIP to south carolina?
00:13.22modulus_terrapen, that's a horrible mnemonic
00:13.22twisted[work]kram, i don't remember where all i've done that to
00:13.22modulus_i prefer just the acronym
00:13.22mtqhmuffie: It is right in front of you check out voip-info and look uncomment what is there
00:13.22terrapeni know, but it helps me remember
00:13.22kramokies :)
00:13.22blitzrageI just prefer to learn what it really means and memorize it :)
00:13.22*** part/#asterisk Shido6 (~shido@d57-87-253.home.cgocable.net)
00:13.26denonkram: so whats your question going to be?  "on line 1378 of sip.c, how come _softhangup is used?"
00:13.26drumkillahaha
00:13.28kramno
00:13.29Muffiemtqh, can you check #flood channel!??
00:13.32drumkillamaybe i'll just introduce him
00:13.34krami'm not here to trip him up
00:13.36drumkillawhen they stop asking questions
00:13.53denonkram: i prefer to think of it as inviting them to delve deeper into the codebase
00:13.53blitzrageyah... wait till the end...
00:14.05drumkillablitzrage: you still listening?
00:14.06modulus_oh man if we pick at the code
00:14.06blitzragedon't want to make someone nervous or whatever
00:14.08Muffiemtqh, my configuration file is there! :) thanks for joining
00:14.08blitzragedrumkilla: yeppers
00:14.10modulus_fuckin' forgetaboutit
00:14.13*** join/#asterisk NormAst (HydraIRC@Ottawa-HSE-ppp4121892.sympatico.ca)
00:14.24drumkillai suppose i could have just called hell so other people could listen
00:14.27drumkillawant to do that?
00:14.31kramhe's done a brilliant job
00:14.31blitzragedrumkilla: I was just going to say that :)
00:14.37denonbahaha .. could always ask him some h.323 questions
00:14.38twisted[work]kram, cool
00:14.39drumkillablitzrage: ok, i'm on it
00:14.44blitzragedrumkilla: I hear :)
00:14.49blitzrageok... I'll hangup and call hell
00:14.50Muffiemtqh, thanks! I tought that mark was something like a check mark! hehehe..
00:15.07kramin total, i've found no more than about a half a dozen things i would have worked on and they're all very minor
00:15.10blitzragedrumkilla: you don't want me to call you do you? (turn off the ring)
00:15.23drumkillanah
00:15.26blitzrageokie
00:15.27drumkillai'm calling hell
00:15.35file[laptop]ooh hell
00:15.36blitzragedrumkilla: sounds good, I'm there now
00:15.51drumkillain
00:15.57blitzragedrumkilla: yeppers
00:16.00drumkillacan you hear?
00:16.05blitzragedrumkilla: no
00:16.07kramooh
00:16.10blitzragedrumkilla: oh there we go
00:16.14kramlets see how he does with this one...
00:16.17drumkillaok :)
00:16.18denonkram: so now you make me nervous .. I've done a couple presentations on asterisk .. im gonna have to scan the audience very well next time :)
00:16.24file[laptop]sounds interesting
00:16.27drumkillaok, anyone who wants to hear, call hell :)
00:16.30blitzragedrumkilla: whats he talking about now?
00:16.36blitzrageplease don't talk in hell
00:16.39drumkillaasking about providers or something
00:16.41kram"With vonage, you pay a monthly fee, so with Asterisk you just buy a product from Diigium?"
00:16.43blitzragedrumkilla: okie
00:16.49denonkram: hah
00:16.49kramthat was a question asked
00:16.50drumkillawhat kram said
00:17.00blitzragedrumkilla/kram: thanks
00:17.04kramhe explained it well
00:17.05denonI wish people wouldnt equate voip with crappy, cheap long distance
00:17.06mtqhwait, I am lost, Kram is at a asterisk presentation but the presentor does not know?
00:17.14file[laptop]mtqh: correct
00:17.18kram"tell vonage to take the modem back and setup an asterisk box you can do a lot more with it"
00:17.21drumkillai'll do it for a fee...
00:17.25modulus_denon, i equate voip with crappy, cheap short distance. is that ok?
00:17.29mtqhkram: at the end you should stand up and talk
00:17.41*** join/#asterisk frank_sbr (~frank_sbr@MTL-ppp-155094.qc.sympatico.ca)
00:17.48denonkram: you think that's handled well? I mean, one's a service, one's a technology ..
00:17.49blitzragemtqh: I don't think kram's intention is to steal the presenters thunder
00:17.55frank_sbrhello all
00:18.03blitzragedrumkilla: hrmmmm... not as loud as it was before.... not sure why....
00:18.08kramno no no
00:18.14fearnordenon: i've been beating the point that VoIP is not same as voice over intarweb
00:18.15kramthat wasn't the answer
00:18.20kramthat was a comment that came later
00:18.23denonah
00:18.27denonic
00:18.28kramhe explained that with asterisk you still need a service provider
00:18.32denonfearnor: ya, we all have
00:18.34kramhe did fine
00:18.37drumkillablitzrage: now?
00:18.49file[laptop]my ear hurts now.
00:18.55drumkillahaha
00:19.10kramdefinitely don't want to steeal his thunder
00:19.16kramhe's doing really great, seriously
00:19.18denon[steal]
00:19.24drumkillathis connection is so hardcore
00:19.24kramyah
00:19.31blitzragedrumkilla: just a bit harder to hear now, I think because of the conference, but thats cool
00:19.39blitzragedrumkilla: just need to take it off my speaker phone :)
00:19.41drumkilla802.11b --> kram's laptop --> blutooth --> kram's cell phone --> gprs
00:19.45*** join/#asterisk BoRiS (~boris@S01060040ca1e5b54.wp.shawcable.net)
00:19.54file[laptop]hehe
00:19.56blitzragedrumkilla: yah... I realize its going through a lot of technologies now :)
00:19.57blitzragelol
00:19.58BoRiSfile!!!!!!
00:19.58file[laptop]BoRiSSSSSSSSSSSSSSSSSS
00:20.01denonwhere's this conference? (to dial in, not location)
00:20.11BoRiSI just got my SPA-841 phone....Cute little phone
00:20.17file[laptop]BoRiS: nifty
00:20.20file[laptop]BoRiS: pics!!!
00:20.34drumkillablitzrage: copy the conf info into the channel
00:20.35blitzragebkw_: whats the IP address of the conference again?
00:20.40blitzragedrumkilla: doing so now
00:20.45drumkillak :)
00:20.56blitzrageconference: IAX2/guest@66.250.68.194/996 room 666 (aka hell)
00:21.11blitzrageplease mute yourself in the conference if you join
00:21.15BoRiSfile: I'll take a few :)
00:21.37mtqhblitzrage: ip address?
00:21.45blitzragemtqh: I just posted it
00:21.52BoRiSfile: So whats new and exciting?
00:21.54blitzrageconference: IAX2/guest@66.250.68.194/996 room 666 (aka hell)
00:21.57blitzrageplease mute yourself in the conference if you join
00:22.10kram"Do you think phones will be developed specifically for Asterisk?"
00:22.12file[laptop]BoRiS: listening...
00:22.17drumkillathere is probably 30 people in here ...
00:22.50*** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net)
00:22.54BoRiSnot dancing? :)
00:22.57file[laptop]hehe
00:22.58blitzragewhoever joined, please mute yourself
00:23.11ariel_hello everyone
00:23.43blitzragewow... quality seems to drop with more people in the conf.... odd
00:23.55drumkillahrm ... i'm sorry
00:23.57blitzrageprobably the echo
00:24.00blitzragedrumkilla: its not your fault
00:24.01*** join/#asterisk crash3m (crash3m@crash3m.user)
00:24.05drumkillak
00:24.08mtqhdo you mean mute with the phone or mute using the user menu?
00:24.15blitzragemtqh: mute your phone/mic
00:24.39mtqhwould it not sound better if we used a m mode?
00:24.45Total-Netdoes anyone know anything about a TUNDO TELPORT-16?
00:24.50jimblobi'm listening to the conference from here in sydney and the sound is extremely faint
00:24.53blitzragemtqh: probably, but the owner of the conf isn't here
00:25.04blitzragejimblob: its like that for everyone
00:25.13blitzragedrumkilla: maybe we should have just put it in my conf so I could 'm' it ;)
00:25.24mtqhand user menus are not enabled .... dang
00:25.27drumkillajust tell people to shut up, hehe
00:25.34blitzragedrumkilla: thats what I've been doing :)
00:25.36blitzragedrumkilla: hehehe
00:25.54file[laptop]how long is this gonna go for?
00:25.58mtqhdoes anyone know of a device that can take a sound card output and put it on a rj11 cable
00:26.24mishehurj11 isn't a cable
00:26.30mishehuthat's a jack
00:26.32mtqhtrue
00:26.36mtqhbut you know what I mean :)
00:27.08Total-Netput it another way, will a TUNDO TELPORT-16 wotk with Asterisk?
00:27.09mishehuactually, some 5 years ago we were using a Telrad digital PBX, and we had the sound card hooked in to play the MOH
00:27.13file[laptop]be vewy vewy quiet, we're hunting wabbits
00:27.59mtqhI need to play a radio into a meetme conf
00:29.16kramthis guy has a lot of questions, so you can tell he's really interested
00:29.22kramgoodk, practical questions
00:29.30blitzragekram: thats what I like to hear
00:29.33kramthe myth of one network for data & voice hehe
00:29.35file[laptop]good, we snagged another
00:29.37bkw_I can't hear
00:29.39drumkillai'm really impressed with the level of conversation that has been going on
00:29.41bkw_its too low
00:29.43kramit's very hard to do
00:29.49bkw_run the phone up to him
00:29.50blitzragebkw_: yah... it dropped recently for some reason...
00:29.53bkw_say SPEAK LOWERD
00:29.55kramwe had a big cutomer who ended up building two data networks
00:29.57modulus_is there a conf on right now?
00:29.57BoRiSbkw!!!!
00:29.59bkw_haha
00:30.14blitzragejust put it on the front table :)
00:30.20drumkillaha, no
00:30.26drumkillai don't want to draw too much attention
00:30.30blitzragedrumkilla: bah... no guts, lol
00:30.36bkw_strap on a set of balls
00:30.37ariel_anyone worked with making an asterisk system backup if one goes down the other take over?
00:30.38bkw_walk up
00:30.39blitzragedrumkilla: understandable though
00:30.44bkw_and get the story
00:30.50BoRiSbkw: I even tried compiling todays cvs...compiled properly but none of my phones can still register. If I recompile 01/09/2005 cvs(2 days ago), Everything works fine
00:30.52denoncatch-22 .. you crank the gain, and you have to tolerate drumkilla's hand noises :)
00:31.04BoRiSstrange
00:31.09bkw_he's bitching about cisco?
00:31.14BoRiSand it *wasnt* a kernel problem... lol
00:31.19drumkillasomething like that
00:31.22tzangerdrumkilla is Italian?
00:31.27kramboris: did you get the exact dif that introduced the problem?
00:31.27blitzragedamnit... why don't I have a +20dB boost on this phone :)
00:31.27drumkillano
00:31.34blitzragestupid Cisco phones
00:31.34kramyou can check out each version of chan_sip
00:31.37tclarkBoRiS: iax2 issues that new binaddr chg has messed up iax2 ..
00:31.44bkw_yes it does
00:31.47bkw_kram broke chan_iax2
00:31.51tclarkyah
00:31.54tclark:)
00:31.54denonblitzrage: 7960 is loud enough .. just that I can barely stand the noise of the plastic when he moves it
00:31.56bkw_I noticed that too
00:32.11blitzragedenon: yah... I just like blaming the phone :)
00:32.13drumkilladenon: sorry, i was trying to get it to a better spot
00:32.16BoRiStclark: Actually, My SIP clients can't register
00:32.21drumkillai won't touch it again
00:32.22bkw_mine can
00:32.25tclarkoh
00:32.42blitzragea better spot might be the front table :D
00:32.49denonya :)
00:32.49drumkillablitzrage: !!!
00:32.54blitzragedrumkilla: lol
00:32.56drumkillait's on the front of ... my table :)
00:32.59denonpodium :)
00:33.05blitzragedrumkilla: hahaha
00:33.12blitzrageu win!
00:33.27bkw_kram
00:33.29file[laptop]woot questions
00:33.43drumkillayah ... and decent ones, too
00:34.14blitzragemore people who do * presentations should broadcast them on the MeetMe conf bkw_ has setup
00:34.22BoRiShmmm, If I delete my old configurations, and redo "make samples", my sip phones can NOW register
00:34.26drumkillaagreed
00:34.27bkw_blitzrage yes it should be a requirement
00:34.36file[laptop]great idea
00:34.43blitzrageI'd like to have more things broadcast on the conference
00:34.46blitzrage"the conference"
00:34.50mtqhcan someone transcribe the questions?
00:34.50bkw_aka hell
00:34.56drumkillait'd be better quality if it wasn't my cell phone ...
00:34.56blitzragebkw_: :)
00:35.10blitzragedrumkilla: yah well... it was kind of spur of the moment :)
00:35.12drumkillathey're talking about prop. vendors
00:35.55krami wonder if anyone is recording this
00:36.05krammaybe he can post a link on his web site
00:36.08blitzragekram: I'd imagine not... ?
00:36.12kramand then to the wiki
00:36.16file[laptop]ooh, or have... presentation on demand technology!
00:36.16drumkillabkw_ may be able to arrange a recording of the conf ...
00:36.19file[laptop]brought to you by asterisk!
00:36.21blitzragekram: it'd be wicked if he was though
00:36.42blitzragedrumkilla: its really quiet... not sure if a conference recording would be all that useful...
00:36.49drumkillasorry :/
00:37.04blitzragedrumkilla: well, nothing you can do with the equipment at your disposal :)
00:37.05file[laptop]with muffin power
00:37.06alakdanwhat should asterisk return for SAY DIGITS "" ""  (call say digits without a digit to say?)
00:37.10*** join/#asterisk firestrm (F1r3570rm@S010600047577bccd.gv.shawcable.net)
00:37.15drumkillai could pass it down the table ...
00:37.24blitzragedrumkilla: yah.. then you don't have to get up :)
00:37.24bkw_please do
00:37.51drumkilladon't want to draw attention right now ...
00:38.06bkw_I wish I had a camera
00:38.07drumkillaonce we introduce kram, i'll make it better
00:38.15blitzragebkw_: ?
00:38.18file[laptop]eta?
00:38.24drumkillasoon, i'm sure
00:38.31file[laptop]verrrrrrrrry good
00:38.32drumkillawe've been on questions for a while
00:38.34denondrumkilla: this a speakerphone? or just hi gain mic on the phone?
00:38.36blitzrageyah.. he's been talking a while... probably take a break soon
00:38.41kramnot until it's *totally* done
00:38.41*** join/#asterisk ROM_Man (rom_man@mike.netrom.com)
00:38.48bkw_SHAME ON HIM
00:38.48drumkilladenon: my cell phone
00:38.52bkw_"NOT THE ONLY OPEN SRC PBX"
00:38.53blitzragedenon: cell with headset :)
00:38.54bkw_naughty
00:38.57bkw_naughty
00:38.57kramhehe
00:38.59denondrumkilla: yeah .. but a speaker?
00:39.00blitzragelol
00:39.01denonah
00:39.06drumkilladenon: nope
00:39.10firestrmis there anything cheaper than digium's TDM10B for single fxs solution?
00:39.14kramhaha, nice descripition of pingtel, interesting he didn't mention either by name
00:39.14bkw_no
00:39.16denonfirestrm: no,
00:39.20denonNEXT!
00:39.22firestrmhmm..
00:39.25bkw_haha
00:39.26firestrmbummer
00:39.38denonfirestrm: quit griping and pay your taxes.
00:39.39denon:)
00:39.47drumkillame too!
00:39.50file[laptop]you are going to give the poor guy a heart attack
00:39.53denontakes money to make money
00:39.55bkw_ya
00:39.57bkw_totally
00:39.57blitzrageno one broker than I :)
00:39.59drumkillafile[laptop]: it's going to be GREAT
00:40.06file[laptop]so silly
00:40.07bkw_he's gonna fall over
00:40.09*** join/#asterisk JimVanM (~JimVanM@HSE-Ottawa-ppp164679.sympatico.ca)
00:40.10firestrmim not trying to make money, just trying to learn
00:40.14blitzragefirestrm: starta  project and get someone to buy you a phoen :)
00:40.17blitzrageJimVanM: heya Jim!
00:40.20bkw_does he ever come to the channel?
00:40.30denonfirestrm: you're a student? does your college charge you money to learn? :)
00:40.32drumkillanot that he has mentioned
00:40.33*** part/#asterisk ckruetze (ckruetze@cpc1-cmbg7-5-0-cust31.cmbg.cable.ntl.com)
00:40.33blitzragebkw_: who?
00:40.41bkw_the guy talking
00:40.42drumkillawe don't know him
00:40.42bkw_ninny
00:40.45blitzrageah
00:40.45denonfirestrm: besides, you can "learn" with a free sip client
00:40.46bkw_ah
00:40.47file[laptop]ah here we go...
00:40.50denonmust be over .. I hear clapping
00:40.51blitzrage*clap*
00:40.57JimVanMblitzrage: howdy
00:41.04drumkillaHERE WE GO
00:41.09blitzragesweeet
00:41.13alakdanhello, just wondering what should asterisk return for SAY DIGITS "" ""  (call say digits without a digit to say?)
00:41.13*** join/#asterisk ckruetze (ckruetze@cpc1-cmbg7-5-0-cust31.cmbg.cable.ntl.com)
00:41.19CpuIDyo bkw_, you ever had a situation where an fxs port just spits out mofo interference/static? it seems like the channel is working tho
00:41.31blitzragelol
00:41.31bkw_OMG
00:41.32CpuIDbecause i can ring it, and the analog handset rings still
00:41.32bkw_thats funny
00:41.32file[laptop]lol
00:41.38BoRiSbkw: This is messed up, If I comment out in my extconfig. (;sipfriends => odbc,PostgreSQL-asterisk,sip), I can see registrations FAIL but when I uncomment it, I can't see any registrations or even failed registrations. Something maybe odbc related?
00:41.40CpuIDand i can hear a very faint dialtone in the background usually
00:41.54bkw_BoRiS recompile baby
00:41.59CpuIDbut whenever your on the handset, you get the fuzz :)
00:42.03firestrmi guess i will have to break down a buy one then... its just i have several important things i can do with my megar amount of money.. like rent.. but hey i will have my first pbx wired cardboard box house.
00:42.13firestrm:)
00:42.19CpuIDi thought it might have been analog phone cable interference, next to my AC power, but i dont think so for some reason, its been next to it for ages
00:42.27blitzragefirestrm: use a softphone on your computer then
00:42.29file[laptop]lol
00:42.35bkw_this is funny
00:42.45CpuIDwhats so funny bkw_? lol
00:43.31bkw_were on the conf
00:44.09firestrmblitzrage> would a wisip phone work?
00:44.21denonbahah .. you're broke and you want to buy a wisip phone?
00:44.30CpuIDah k heh
00:44.31bkw_wisip SUCKS
00:44.33bkw_totally SUCKS
00:44.33firestrmdenon> chistmas present
00:44.38bkw_you think thats wind..
00:44.41bkw_but its really the wisip
00:44.41denonfirestrm: ask for cash.
00:44.53firestrmdenon> i do... i never get
00:45.26blitzrageSenao! :)
00:45.39blitzrageonly because I want someone to buy one and test it for me :)
00:46.06firestrmblitzrage> senao wifi card?
00:46.28blitzragefirestrm: no, Senao is making a WiFi Sip phone now... just over $200 I think
00:46.40file[laptop]go kram, go kram, go kram
00:46.41bkw_mark didn't take the phone with him
00:47.47denonhmm . how come they didnt use the mic on their desktop and a iax/sip client? seems like the gain might be better than a cell
00:47.47file[laptop]have you been looking at pics of bkw too?
00:47.47blitzragefirestrm: tons of codec support, a lot of features it looks like
00:47.47file[laptop]denon: no bandwidth
00:47.47denonfile : ah
00:47.47blitzragedenon: I think they are connected to the net via a cell data connection
00:47.47firestrmblitzrage> im in lust...
00:48.05denonsomeone say hi from irc :)
00:48.16blitzrageyah, and from the conf :)
00:48.20blitzrageaka voice irc
00:48.40kramDavid Nally, works for a lawfirm and is the network administrator
00:48.42*** join/#asterisk Schism (~schism@clt74-101-230.carolina.rr.com)
00:49.20firestrmblitzrage> im working on a quazi cell rollout in a small underserved community using wifi/sip and mesh
00:49.32modulus_i sneezed and sprayed
00:49.33modulus_yes!!!
00:49.45firestrmblitzrage> but the pulver wisip phones dont quite cut it
00:49.45Schismanyone know when the Asterisk Boot Camp & dCAP certification- Session 2, Europe will be accepting registration?
00:49.57denonSchism: who cares?
00:50.08SchismI do, hense the reason why I am asking.
00:50.11denonoh, I spose you do..
00:50.19Damindenon: People that want to go to Europe and spend a lot of money?
00:50.21Schismyou are a bright one denon.
00:50.24*** join/#asterisk BadKnees (~BadKnees@lorentz.teletech.fo)
00:50.35blitzrageSchism: not sure, but I can find out for you
00:50.40denonSchism: the certification program hasn't been real well welcomed here..
00:50.41Schismthanks
00:50.56*** join/#asterisk lters (~lters@mrtc-mm-600046.mis.net)
00:50.58blitzrageSchism: I personally think its a good idea, so don't listen to these yokals :)
00:51.00SchismI am not doing it for the certification, more for the training
00:51.09*** join/#asterisk |Blaze| (dirc@d142-59-247-192.abhsia.telus.net)
00:51.24SchismI already have training in Avaya, Cisco, and Nortel VoIP, but I want to deploy asterisk
00:51.38SchismI have the basics, but I want to know the gory details
00:51.41blitzrageSchism: its not posted on the website is it?
00:51.43firestrmblitzrage> i am finding that most users have much trouble naviagating the pulver wisip phone and are getting frustrated because the interface was not well designed.
00:51.56kramwe'll clarify the cert stuff, don't worry
00:52.03kram1) digium will own the certs
00:52.04blitzragefirestrm: I used it at VON... just wasn't that impressed
00:52.11Schismit's posted, I just can't register for it
00:52.12kram2) you will NOT have to buy the training to do the cert
00:52.24kram3) you can do the cert at astricon or spring von
00:52.31Schismhense the little d in front of the CAP :)
00:52.37firestrmblitzrage> we currently have 200 of them in the field for a test, but they are problematic..
00:52.43*** join/#asterisk mtqh (~chatzilla@dsl093-001-038.det1.dsl.speakeasy.net)
00:52.43BadKneesSorry for these stupid questions: What is a bridged call? What is the alternative to a bridged call? Can two sip phone registered on an asterisk connect directly to each other (or is this bridged)
00:52.57blitzrageSchism: I just sent you a /msg
00:53.00file[laptop]kram: oh so you mean I get to do the cert at von?
00:53.01czerokram: jsut dont; sell out like cisco and make the cert's easy in a couple of yrs to get vlumes of people supporting the stuff... F@#$ing cisco
00:53.03SchismI didn't get it :)
00:53.20blitzrageSchism: info@astricon.net
00:53.29kramyou can purchase the cert at von I believe yes, but talk to olle & steven to find out details
00:53.39kramthey tell me they're offering it at the astrerisk pavilioin
00:55.11*** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.rr.com)
00:55.11*** mode/#asterisk [+o anthm] by ChanServ
00:55.23*** join/#asterisk Slainte (~Slainte@66.55.113.205.ppp.northrock.bm)
00:55.34blitzragekram: if I sign up for the certification, am I helping Digium?
00:56.17kramthe plan is to split the cert cost between digium and the company which administers the cert
00:56.28blitzragekram: awesome
00:56.31anthmhttp://www.bbc.co.uk/radio4/hitchhikers/game/guide.swf
00:56.47kramso in principle yes, but there are other ways too :)
00:56.51anthmwin this w/o cheating and you will be certified
00:56.54SlainteI just got a Polycom 600 and cant get it to register. Anyone using the IP600 and might spend 5 minutes helping see if I crossed my t's and dotted the i's?
00:56.55czeroany ideas on where Astricon Europe will be yet?
00:57.04blitzrageczero: not finalized yet
00:57.11blitzrageczero: but should be soon
00:57.13czerokewl
00:57.27blitzrageczero: as far as I know Olle is in the final stages of deciding a venue
00:57.34czerobudapest or prague maybe :)
00:58.01anthmthisisacon is comming up this summer in chicago meet anthm and bkw SUNDAY SUNDAY SUNDAY
00:58.14blitzrageanthm: can I sign up?
00:58.32czeroI jsut want to go someplace I've not been and have a reason to be able to wriote it off :)
00:58.47bkw_blitzrage he's actually be serious
00:58.49guugmemberhow are the Grandstream 102? easy to configure?
00:58.58bkw_we are doing a dev thing in chicago sometime in june or july
00:59.00*** join/#asterisk syslod (~sysglod@65.114.0.198)
00:59.03blitzrageczero: lol.. I hear that.  I want to go to euro astricon, just hoping I have a reason to go :)
00:59.09blitzragebkw_: thisisnotacon? :)
00:59.19blitzrageerrr... thisisacon
00:59.26blitzragesounds like a con to me
00:59.28bkw_someone said AsterDev but that just sounded lame
00:59.28blitzragelol
00:59.56bkw_nonewbiecon
01:00.02blitzrageI like it :)
01:00.24*** part/#asterisk Schism (~schism@clt74-101-230.carolina.rr.com)
01:00.49*** part/#asterisk BadKnees (~BadKnees@lorentz.teletech.fo)
01:01.08Silik0nfree bkw porn at http://127.0.0.1/
01:01.15bkw_hey hey hey
01:01.24*** join/#asterisk Schism (~schism@clt74-101-230.carolina.rr.com)
01:01.27czeroSilik0n for some of us thats true
01:01.28czero:)
01:01.33Silik0nok so its not free, I'm charging 9.95 for access to it
01:01.34blitzragelol
01:01.34bkw_DisAsterCon
01:01.36kramheaded back to clemson later tonight then back to hsv tomorrow
01:01.52bkw_devastricon
01:01.56blitzrageDevCon :)
01:02.35Silik0nhah
01:02.47bkw_the joke is on you.. those are really pictures of you
01:02.49kpflemingkram: ping
01:02.51bkw_muhahaha
01:03.25anthmHere at asterisk, we focus on core development.  For example, I have hundreds of core files on my box right now!
01:03.56bkw_hahahahahha
01:04.14czero:)
01:04.36denonbkw: digium's gonna sue you :)
01:04.45bkw_for?
01:04.50denonI dunno..
01:05.10bkw_do you see the word PBX in there anywhere?  It could the * Kneepads ya know :P
01:05.15bkw_* toothbrushes
01:05.37denonhehe
01:05.41blitzrageI think Asterisk should either drop the "PBX", or, I prefer to call it a "Converged Telephony Platform"
01:05.58bkw_blitzrage ya really
01:06.05bkw_its more than a PBX
01:06.05JimVanMPBX = Public Branch Exchange
01:06.11blitzragebkw_: I really like "converged telephony platform" :)
01:06.12JimVanMor how about OBX
01:06.17JimVanMOpen Branch Exchange
01:06.23bkw_OBX? haha
01:06.32NivexThat looks too much like those silly Outer Banks stickers I see on cars here in North Carolina.
01:06.56bkw_ok I just checked my paypal account
01:07.04bkw_I SEE NO money from anyone there
01:07.12*** join/#asterisk jorgeB (~jorge_B@201.135.247.237)
01:07.13blitzragebkw_: in fact, I took some out
01:07.15JimVanMNivex: like a pawn shop? The Outer Banks Exchange
01:07.22denonbkw: what do I owe you for, you werent around when I was having speex problems <G>
01:07.25NivexI don't know why the 'X'
01:07.35JimVanMeXchange
01:07.48blitzrageInter Asterisk eXchange
01:08.01blitzrageIAX == EEKS
01:08.07bkw_eeeks its a mouse!!!
01:08.12denonbut dont actually say "eeks" or people will laugh at you
01:08.18bkw_ok Balance: $0.00 USD
01:08.26blitzragedenon: I say it all the time with a straight face :|
01:08.31bkw_denon no say eeks or people will laff at you
01:08.35wolfsonanyone with an OBX sticker is not from the Outer Banks, lol
01:08.35denonblitzrage: we all laugh at you
01:08.41JimVanMwhen you say eye-ay-ex it sounds like a Dutch football team
01:08.42blitzragedenon: I'm used to it
01:08.45denon:)
01:08.49blitzragehehehe
01:08.57jorgeBhello... anybody that has been using Symphone for PPC mind shooting me a tell so I can ask a few questions? Thanks
01:09.02blitzrage"their all gonna laugh at you"
01:09.03denonyeah, I say eeks too .. but I rarely say it in front of people that matter
01:09.05bkw_thats allison saying "fucked"
01:09.11bkw_ya know she'll say anything
01:09.28JimVanMbkw_: if the money's right
01:09.33denonnot just say ..
01:09.43ckruetzebkw wah was your paypal accoung again?
01:09.50bkw_brian@bkw.org
01:09.51denondenon@
01:09.52bkw_;)
01:11.22frank_sbrhello guys, with the advent of more and more connection than does not require a zaptel connection to the PSTN, having a alternative to app_monitor is almost a must
01:11.40blitzragethey are talking about documentation
01:11.42blitzragekram: !!!
01:11.52bkw_frank_sbr you can monitor voip only with res_monitor
01:12.22frank_sbrbkw_ what is res_monitor...does it replace chan_spy?
01:12.22Schismwhat do you guys think about broad voice SIP trunks, is it a good thing?
01:12.30bkw_frank_sbr OH you mean that
01:13.02bkw_frank_sbr pay anthm loads of cash he wrote chan_spy
01:13.11frank_sbrI don't mind paying...if it brings me the functionnality faster
01:13.12BoRiSOooo, chan_spy is nice
01:13.28denonwhat's chan_spy do? zap and sip? or just sip?
01:13.30bkw_frank_sbr well $$ => anthm == more goodies
01:13.35bkw_its a given
01:13.55frank_sbrbkw_, klictel and I are doing a load share to bring this baby more mature :)
01:13.56Mochi
01:14.00anthmtoo bad nobody has the real chanspy, only the hacked up version that someone stole from me to get free bounty.
01:14.17bkw_muhahahah
01:14.29Schismsounds a bit cynical there anthm
01:14.41*** join/#asterisk jalsot (~tamas@abacus.eworldcom.hu)
01:14.44frank_sbranthm, I have app_amd (that does answering machine detection), want to share it with chan_spy ?
01:14.47denonanthm: well, you couldnt expect us to commit the real one .. it had a backdoor with the password of "anthmrules"
01:14.53Mocwhat going on in the conf ?
01:15.12bkw_haha
01:15.13blitzrageMoc: listening to a LUG meeting
01:15.18Moclug ?
01:15.25blitzrageLinux Users Group
01:15.25denonlinux users group
01:15.26MocLinux User group ?
01:15.32bkw_duh
01:15.40blitzragehehehe :)
01:15.43bkw_it shouldn't have had to be said
01:15.48Mocsound boring..
01:15.50bkw_kinda like FXO/FXS
01:15.55denoniax?
01:15.58bkw_eeks
01:15.58denonwhat's iax?
01:15.59bkw_eeks
01:16.00bkw_eeks
01:16.07blitzragebkw_: true... but I get to show off my mad typing skillz, lol
01:16.22bkw_blitzrage what thats not the onlything you do fast eh?
01:16.24denonExternal Evolved Konvoluted SOmethin
01:16.27Mocok where is the real conf then ?
01:16.51jorgeBI'm getting "Got SIP Response 405 method not allowed" on my console when my PPC with Symphone registers with Asterisk, and I can't seem to dial any extensions... I am, however, able to receive calls on the PPC, any ideas?
01:17.04denonjorgeB: fix your codecs?
01:17.06anthmfrank_sbr, you gonna release your app_amd then?
01:17.07blitzragebkw_: nah... I'm not quick off the trigger
01:17.18bkw_blitzrage haha
01:17.30denonjorgeB: allow=all in sip.conf
01:17.40bkw_no that isn't evil anymore
01:17.40frank_sbryes, I have no problem doing that, it was in our plan to do so
01:17.41denon(man, I never thought I'd say that..)
01:17.42blitzrageok... I'm going out to the living room for a bit.  Gotta get away from the computer for a bit so I can spend all day tomorrow in front of it
01:17.46bkw_you can do that now de
01:17.49CpuIDok question, whats the most you guys have got call wise through one asterisk box?
01:17.49bkw_you can do that now denon
01:17.54denonyeah, I know . .thats why I said it
01:18.03denonstill seems wrong tho
01:18.05jorgeBdenon, I have allow all in sip.conf
01:18.06bkw_nope
01:18.09bkw_its 100% correct
01:18.11SchismCpuID: that is a trick question :)
01:18.18denonjorgeB: and no disallows?
01:18.18tessier__CpuID: A thousand.
01:18.19bkw_denon who do you think refactored and worked that code?
01:18.21bkw_anthm and I
01:18.30denonbkw: oh well, in that case .. .
01:18.40Slaintedenon, is it a context problem in his extensions.conf?
01:18.40denonjorgeB: I think that allow=all is breaking yoru shit
01:18.47frank_sbrwe have develop app_amd, a 1 1/4 year ago for our call center and now we need to do something similar like chan_spy....
01:18.59frank_sbrand I don't like reinventing the wheel
01:19.01CpuIDa thousand simultaneous calls, nice :)
01:19.02jorgeBdenon: nope, no disallows... and I have ulaw/alaw codecs on the symphone 3.1 for PPC, and most codecs in the SJphone which is the destination ext
01:19.23bkw_frank_sbr how good does app_amd do?
01:19.25bkw_100%
01:19.28denonjorgeB: check if its reg'd .. check your contexts .. do a sip debug
01:19.28bkw_98%?
01:19.38denonbkw would love to help you in privmsg
01:19.47bkw_denon no I wouldn't
01:19.48frank_sbrNo, be realistic, it does 80~85%
01:19.48denon/msg him for paypal info
01:19.54jorgeBdenon: so allow=all is breaking the conf? maybe allow each codec? sip show peers shows the PPC is registered
01:19.59frank_sbrlike dialogic board but faster
01:20.07denonjorgeB: kidding , allow=all is good
01:20.07*** join/#asterisk kFuQ (~somedude@c-24-17-224-78.client.comcast.net)
01:20.09CpuIDwere they 100% ip or were they ip => pstn? also?
01:20.19alakdanhay guys, I have an AGI question, what would asterisk return when issued a command SAY DIGITS "" "" ?
01:20.29SchismCpuID: what type of machine, and what type of calls?
01:20.30bkw_oh we get about 90-95% out of our app that does that
01:20.38jorgeBdenon: heh... ok, guess I'll try ethereal to track what's the PPC sending that throws a 405 method not allowed
01:20.46CpuIDwell lets say, many a many incoming IAX calls, going out with SIP
01:20.48ckruetzebkw, what does your mini mac account say?
01:20.50bkw_it took time
01:20.57CpuIDhopefully avoiding T1/E1 and pstn altogether
01:20.57bkw_ckruetze lets see
01:21.12frank_sbrbkw_, we run a call center and it has to be done in less than .5 sec
01:21.18CpuIDmost likely something like gsm or G.729, maybe even G.723.1 on most
01:21.21bkw_Balance: $0.00 USD
01:21.22Schismiax trunks and sip end users?
01:21.31bkw_frank_sbr yep it can be
01:21.42kramdoes anybody know about an asterisk live cd?
01:21.52bkw_kram yes one is called RAPID
01:21.55bkw_and i'm working on one too
01:22.00bkw_its not done yet
01:22.22bkw_ckruetze dude paypal took that all
01:22.35frank_sbrthats why it is 85%, we can some paramters to make it more accurate but than the poor customer will wait like an idiot on the line
01:22.41jorgeBhas anybody used X-pro for pocket PC? is it really worth it? SJPhone for PPC seems to be have some memory leaks or something that causes my PPC to freeze
01:22.45NormAstbkw: What distro is are you using for your CD?
01:22.57bkw_frank_sbr haha so true
01:23.18kpflemingkram: we need to schedule a call when you have some free time
01:23.20ckruetzebkw, that is not my fault, complain to them.
01:23.48frank_sbranthm, how would you charge for chan_spy?
01:25.29bkw_NormAst its based on Crux
01:25.32bkw_but stripped down
01:25.35bkw_and totally rebuilt
01:28.16MocI gota find a small doc on making a live CD ;)
01:28.31MocLike a small rescue CD
01:28.46NormAstbkw: Never heard of it..
01:29.15*** join/#asterisk badc{}l{}r (~badc{}l{}@203.115.208.140)
01:29.23frank_sbranthm, you have disapear, when I wanted to pay for chan_spy?
01:29.35frank_sbrwho else want the money?
01:29.38anthmno i'm here just busy
01:29.40SchismI do!
01:29.45Schismbut I can't program :-(
01:29.46Schismhaha
01:29.56syslodfrank_sbr: What are you looking for?
01:30.32SlainteMoc,  Your maybe bext using one that exists
01:30.36Slainteand tweaking it
01:30.45anthm300 bux
01:31.24Slaintebkw_  Rapid CD does not evcen have an FTP or TFTP server with it
01:31.49SlainteIt has no SSH server either
01:31.52badc{}l{}rhow can i disable * sending voicemail to my email and only able to access via voicemail menu?
01:32.10NormAstslainte: you can use apt-get install and get it..
01:32.31SlainteWhat about the devel tools for rebuilding a new * build
01:32.55SlainteMaybe I went about using it the wrong way.  I found I spent to much time adding to it.
01:33.10Slaintewhat about knopsterisk,  anyone use it?  I dont want to pay for an ISO
01:33.27NormAstslainte: yea... It's not designed for that..
01:33.45bkw_Slainte mine is going to be a tftp/dhcpd/kickstart/ssh and all that mess
01:33.48*** join/#asterisk WizardWlf (~shawn@wrt54g.djernes.org)
01:33.56NormAstslainte: I use debian.
01:34.29NormAstbkw:  preemptive kernel?
01:34.32SlainteI have used SuSe, gentoo and FC2,  FC2 for my customer builds so far
01:35.17Slainteanyone use the polycom phones?
01:35.34bkw_NormAst maybe
01:35.44bkw_i found that preemtive kernel causes some quirkyness
01:35.52*** part/#asterisk JimVanM (~JimVanM@HSE-Ottawa-ppp164679.sympatico.ca)
01:36.13NormAstbkw: It should help with the TDM410P cards right?
01:36.45bkw_NormAst their is no TDM410P cards
01:36.51bkw_its TDM400P
01:36.54bkw_and those things should be shot
01:36.58bkw_T1 or bust
01:37.27bkw_my view is .. shit or get off the pot... why really have a live CD
01:37.30*** join/#asterisk Kokey (~jramirez@201.137.168.75)
01:37.41bkw_use asterisk.. or leave.... be a man.. install linux and run it
01:37.54bkw_my live CD is going to be a kickstart CD
01:37.58bkw_thats about all its gonna do
01:38.01firestrmjorgeB> xpro-ppc turned my ipaq into a brick until i did a full reset
01:38.03bkw_so you can quickly clone asterisk installs
01:38.08NormAstbwk: TP410e
01:38.16bkw_TE410P
01:38.16JerJer"Preemption also will require a kernel overhaul and complicate the kernel's code. Is it really worth the trouble? For most everyday uses, the answer is "no"." <--- what google told me
01:38.24Slaintebkw_  you going to use AMP or anything like that?
01:38.31bkw_maybe
01:38.34bkw_JerJer I agree on that one
01:38.40bkw_man asterisk is just quirky as hell
01:38.42twistedwheeeeeeeee
01:38.43ckruetzeTime to go to bed, I have to be up again in 5 hours.
01:38.52twistedi use a preemptive kernel
01:39.04bkw_ya but you don't really load it up with traffic do ya?
01:39.07NormAstbkw: YEa... that's it.
01:39.08twistedthere's a slight performance increase
01:39.22jorgeBfirestrm: did it? I'm using an Ipaq and I'm having problems with Symphone, though it does not freeze my PPC. SJPhone on the other hand allows me to place calls, but freezes after a while and I have to reset my PPC
01:40.10syslodNormAst: TE410P?
01:40.11firestrmjorgeB> it only happened the one time.. but once is enough for me..
01:40.17*** join/#asterisk Guest^DJ (~some@218.208.234.20)
01:40.31NormAstWildcard TE410P ....I have two of them ... Looked it up...
01:40.42jorgeBfirestrm: any other sip phones you would suggest for PPC?
01:40.43firestrmjorgeB> it would reset.. boot almost all the way and then hang..
01:41.01*** join/#asterisk |Vulture| (~Vulture@247.131.vbnet.net)
01:41.12syslodNormAst: I've had no problems with 2.6.9 with that card fully loaded with NFAS.
01:41.23firestrmjorgeB> i dont know of any.. just backup your data first.. thats all im suggesting
01:41.56Schismwhat is the prefered method to build asterisk on MacOSX?
01:41.59NormAstsyslogd: What distro are you running?
01:42.05syslodSlack 10
01:42.17jorgeBfirestrm: yeah, thanks for the tip... gonna try to debug Symphone and see if I can get it to work, doesnt seem to have any memory leaks
01:42.25syslodWell I started with Slack 10 that is.
01:42.59syslodThe 3.3v card runs alot cooler if you are using 1U 2U machines.
01:43.10NormAstsyslod: Running 2.4.27-1-686-smp  right now.
01:43.28syslodI couldn't get it to work without probs on 2.4
01:43.53syslodWorks like a champ on 2.6.9 not anything else for me. Tried 2.6.10 = el' crasho.
01:44.58syslodAnyone tried todays cvs HEAD?
01:45.50NormAstsyslog:  yea... I might try 2.6.9 .. Are you running SMP?
01:46.03syslodYea.
01:47.07kpflemingsyslod: i'm running a HEAD pull from Saturday late, has been working OK for two days now
01:47.36syslodYea its been working real good till this morning.  I compiled and now things are freaking out.
01:48.06syslodDIal a extension and its dialing wrong SIP etc.  Getting login errors when user and pass are correct etc.
01:48.18syslodI rolled back to saturday and things work fine again.
01:48.37bkw_who has the info to fix chan_iax2
01:48.40syslodI'd submit a bug but I don't know where or why the problem occurs.
01:48.44bkw_kpfleming do you wanna help tackle that one?
01:49.27kpflemingi would, but i'm hip-deep in other code at the moment, and will be unavailable all day tomorrow
01:49.35*** part/#asterisk Kokey (~jramirez@201.137.168.75)
01:49.51kpflemingthe only recent change to chan_iax2 was bug 2971, right?
01:51.30*** join/#asterisk BoRiS (~boris@S01060040ca1e5b54.wp.shawcable.net)
01:52.25bkw_kpfleming it worked before the changes mark put in today for bindaddr
01:52.33firestrmcan anyone suggest a newby friendly gui front end for asterisk config?
01:52.41syslodI guess I keep steping up dates till it breaks again.
01:52.42twistedfirestrm, gvim
01:52.44kpflemingbkw_: ahh
01:52.54bkw_tclark you there
01:53.01syslodpico -w
01:53.05bkw_pico
01:53.08bkw_what a pussy editor
01:53.11bkw_use emacs
01:53.30CpuIDlol pico
01:53.33CpuIDim a vim man here
01:53.35mtqhjoe
01:53.37mtqhgo joe
01:53.44CpuIDno need to start an editor debate tho :)
01:54.04CpuIDi dont despise emacs users, personal preference there, but pico is just crud :)
01:54.07firestrmi cant get things past the default config.. too many config files...
01:54.09SchismelVIs lives!
01:54.17twistedSchism, haha
01:54.17CpuIDhaha
01:55.06syslodpico -w = funny not a real recommendation although I have been known to use it occasoinally.
01:55.09Schismvim is cute, if you are a programmer
01:55.31Schismw/ the color coding and stuff
01:55.47Schismfirestrm: it's really very simple, did you d/l the pdf manual? it is very helpful
01:56.05CpuIDya thats why i like vim :)
01:56.08firestrmive red the manual 5 times now
01:56.34syslodjed,emacs,nedit,kedit,kdevelop so many choices.
01:56.39firestrmSchism> and downloaded several examples... still only understand 50%
01:56.46Schismfile: what are you tying to accomplish?
01:57.04Schismfirestrm: I mean :)
01:57.21firestrmSchism> simple dial in -> extenstion ->fwd number
01:57.21syslodfirestrm: u using sample files?
01:57.28firestrmSchism> yes
01:57.53Schismyou want someone to be able to call in, dial an extension and that extension to ring?
01:58.04*** join/#asterisk MrEntropy (~entropy@ppp44-247.lns1.adl1.internode.on.net)
01:58.08MrEntropyyo
01:58.39firestrmSchism> close.. someone dials in, dial extension, forwarded to my free world dialup number
01:58.50Schismah
01:59.06firestrmSchism> i figure thats the best place to start, without any fxs card
01:59.06Schismcan you dial your freeworld dial # from the console?
01:59.26firestrmSchism> i dont know how..
01:59.53Schismhave you thought about using a softphone, instead of creating a trunk?
02:00.09firestrmSchism> im having a learning curve problem.. once i get somthing up and running that i can play with and extend im ok..
02:00.14*** join/#asterisk helvetico (~helvetico@80-218-190-36.dclient.hispeed.ch)
02:00.32firestrmSchism>i have a pulver wisip phone
02:00.35Schismit is much easier to use a softphone, than to create a trunk
02:00.46Schismis it totaly under your control?
02:00.54Nuggetthose pulver/zyxel phones are pretty awful.
02:00.56firestrmSchism> yes
02:01.04firestrmi know..
02:01.12Schismdid you setup the media gateway in it?
02:01.23firestrmuhhhh.....
02:01.52Nuggetforwarding an extension to a fwd number is trivial
02:02.02bkw_file go to hell
02:02.35Schismfirestrm: have you setup the phone in your sip.conf?
02:03.08firestrmim not quite sure how.. the example i have has a bunch of macro's that i cant quite wrap my brain around...
02:03.49bjohnsonhehe .. firestrm it happens to all of us
02:04.00Schismfirestrm: there should be an interactive setup in the phone
02:04.07Schismor a web server
02:04.09Schismaccess
02:04.45BoRiSMy SPA-841 is really cute and small.
02:04.50twistedBoRiS, stfp
02:04.55twistedi'm still waiting on mine
02:04.57*** mode/#asterisk [-r] by bkw_
02:05.11BoRiSAwwww
02:05.13Nuggetheh
02:05.31*** join/#asterisk confbot (~root@nat.backendhost.com)
02:05.38firestrmSchism, that part works... i can call my wisip number from any softphone connected to FWD and it rings
02:05.46Schismusing an ip address?
02:06.00*** join/#asterisk siacali (~siacali@63.201.190.116)
02:06.06Schismor using asterisk?
02:06.10twistedcoo koo
02:06.14Schisman asterisk extension
02:06.26Nuggetfirestrm: forwarding an extension to a fwd number is a one line deal in extensions.conf
02:06.29firestrmusing a softphone like xlite connected to the FWD network
02:06.31Nuggetwhat part is confusing you?
02:06.31twistedmoose penis
02:06.36badc{}l{}rhow can i disable * sending voicemail to my email and only able to access via voicemail menu?
02:06.57Slaintevoicemail.conf
02:07.11firestrmNugget, and what would that magic like be ;)
02:07.14bkw_badc remove your email address
02:07.21bkw_and get this remove those fucking {} from your nick
02:07.38twistedor add attach=no to voicemail.conf line
02:07.40Nuggetexten => 101,1,Dial(SIP/123456@fwd.pulver.com)
02:07.56firestrmnugget, thanks.. i will try that
02:07.58twistedding a ding dang a dong dong ding dong
02:08.02anthm#for hitchhiker
02:08.06bjohnsonI don't hear much about packet8 anymore .. are they out of favour with * users?
02:08.08confbot"Arthur felt at a bit of a loss. There was a whole Galaxy of stuff out there for him, and he wondered if it was churlish of him to complain to himself that it lacked just two things: the world he was born on and the woman he loved. "
02:08.13badc{}l{}rthanks
02:08.57bkw_confbot bjohnson packet hate doesn't like asterisk
02:09.00bkw_never has worked
02:09.05twistedooompa loompa doopity doo,
02:09.15anthm+addalias badc{}l{}r the dude with the brackets in his nick
02:09.15confbotok anthm command 'addalias' complete.
02:09.20anthm+putalias
02:09.21confbotok anthm command 'putalias' complete.
02:09.31bkw_^fucked
02:09.33twisted+addalias twisted god
02:09.36twistedhehe
02:09.36confbotok twisted command 'addalias' complete.
02:09.40twisted+putalias
02:09.48confbotok twisted command 'putalias' complete.
02:09.50*** join/#asterisk aminorex (~tony@c66.191.69.132.dul.mn.charter.com)
02:09.50twistedhahahaha
02:09.54Nugget+addalias linux poo
02:10.24FaithfulWhat's the QoS package of choice these days?
02:10.31twistedtwisted haha
02:10.36bkw_omg
02:10.47*** join/#asterisk ^Slash (slash@220-245-162-37-sa-nt.tpgi.com.au)
02:10.56puzzledFaithful: search the list for a script called "rc" or try wondershaper
02:11.03twisted+rmalias twisted
02:11.03confbotok twisted command 'rmalias' complete.
02:11.07twisted+putalias
02:11.07confbotok twisted command 'putalias' complete.
02:11.56^Slashi just purchased the TDM11 is there any good guides to get me started ?
02:12.18Nugget^Slash: the piece of paper that came with the TDM11.
02:12.24^Slashi didnt get any
02:12.27^Slash:-/
02:12.41Nuggetnutty
02:12.47anthm#for starwars
02:12.47confbotHan Solo: Oh! I thought they smelled bad on the *outside*!
02:13.00^Slashjust got the card
02:13.11^Slashgot zaptel running
02:13.21^Slashjust some examples of the asterisk conf for the card would be nice
02:13.24twisted#for hitchhiker
02:13.24confbot"Far out in the uncharted backwaters of the unfashionable end of the western spiral arm of the Galaxy lies a small unregarded yellow sun. "
02:14.10twisted#for familyguy
02:14.10confbotBlack Knight: You see kids? Your father's nothing but a fizzle! Peter Griffin: Hey, pal, nobody calls me a fizzle and gets away with it! Except for that one guy who called me a fizzle and then he ran off. But nobodyelse has ever called me a fizzle and got away with it! Actually thought, he was the only one who ever called me a fizzle. But after today, only half the people who've called me a fizzle will have gotten away with it!
02:14.58*** join/#asterisk blankman (~blankman@h000d88a1570c.ne.client2.attbi.com)
02:15.05bkw_+set param voice Emily
02:15.06confbotUnknown Command.
02:15.10blankmanHey guys.
02:15.15twisted+setparam voice Emily
02:15.16bkw_+setparam voice Emily
02:15.21bkw_moose penis
02:15.33bkw_+setparam voice Linda
02:15.35bkw_moose penis
02:15.51twistedwake up
02:15.52anthm+showparams
02:15.54Silik0n+setparam voice BillyBob
02:16.01confbotUnknown Command.
02:16.05twistedSilik0n, go to hell
02:16.07confbotUnknown Command.
02:16.07confbotParameter voice = [William]
02:16.07confbotParameter fortune = [hitchhiker]
02:16.07confbotParameter sfx = [test.sfx]
02:16.07confbotok anthm command 'showparams' complete.
02:16.08Silik0nhah
02:16.31firestrmNugget> cool!!! that sort of worked... it rang, but the audio quality was totally unusable..
02:16.47bkw_poose menis
02:16.48anthm+param voice Emily
02:16.48confbotok anthm command 'param' complete.
02:16.55bkw_hey
02:16.58twistedmoose penis
02:17.10twisted+param voice William
02:17.11confbotok twisted command 'param' complete.
02:17.13Silik0nhah
02:17.14twistedhonky
02:17.17blankmanHey, has anyone on used sugarcrm?
02:17.37puzzledI had a look once. pretty spiffy
02:17.44puzzledeasy install too
02:17.49*** join/#asterisk riksta (~rick@81-178-236-71.dsl.pipex.com)
02:18.12blankmanpuzzled, did you play with the code at all or just install and test with the sample data?
02:18.20puzzledsample data only
02:18.21bkw_+shadup
02:18.23bkw_+shaddup
02:18.37twistedpeter piper picked a peck of pickled peppers
02:18.51Silik0npeckers
02:18.52twistedno
02:19.00twisted+honk
02:19.04anthm#kram
02:19.05confbotUnknown Command.
02:19.05confbotPlacing call
02:19.07blankmanummm ....
02:19.14Silik0ndamnit
02:19.19Silik0nthere goes the stoopid shit again
02:19.20bkw_+shaddap
02:19.27confbotok bkw_ command 'shaddap' complete.
02:19.40blankmanSo, what is the "best" of the web based voicemail front ends for *?
02:19.47blankmanOr do most people roll their own?
02:20.03puzzledno idea, never used one. just have my voicemail emailed to me
02:20.06Silik0ntheres a few
02:20.38blankmanSilik0n, ... yeah ... I was just wondering which one is the "leading" the pack and why?
02:20.53*** join/#asterisk implicit (~implicit@ip68-7-149-247.sd.sd.cox.net)
02:20.58Silik0ngrab one of them from the wiki links and see hah
02:21.15CpuIDhmm i wonder what kinda hours digium has
02:21.25Silik0nits kinda like choosing a date... just cause I like it, it doesnt mean you will
02:21.42puzzledunless she is the universal godess
02:22.14bkw_#kick recent
02:22.30Silik0n+showparams
02:22.32blankman:D Well, most people seem to agree on some people ... like say Ell, Christy, etc
02:22.57firestrmwhat exactly does this line do? register => 290805:secret@fwd.pulver.com/290805
02:23.02blankmanFrom their time ;-)
02:23.10bkw_that sounds dirty
02:23.14firestrmis that what i would use to connct to fwd?
02:23.41blankmanfirestrm, to connect yes, but not dial.
02:23.58blankmanyou need to put he secret in.
02:24.03puzzledfirestrm: also check voip-info.org if you want to learn about this stuff
02:24.11firestrmso someone could call my asterisk box from fwd by calling 290805?
02:24.37*** join/#asterisk _chad (~Chad@c-24-6-142-55.client.comcast.net)
02:24.48blankmanIf that was your number....
02:24.52_chadanyone configure a 7940 w/ multiple lines?
02:25.13firestrmok.. im starting to understand... i think..
02:25.42*** join/#asterisk implicit (~implicit@ip68-7-149-247.sd.sd.cox.net)
02:25.45blankmanso, no one wants to pony up and give me a hint as to which they run and if they like it why or why they don't like it?
02:25.55twisted_chad, ya, all two of them
02:26.31_chadtwisted, so did you just dump both line configs in the SIP<MAC>.cnf?
02:26.37firestrmcan i have a setup that only has extensions.conf and sip.conf? ot do i have to have all the config files?
02:26.48twisted_chad, yep
02:26.51_chadtwisted, i have 2 phones also, i had one line per phone.. worked fine.. i wanted both phones to have access to both lines
02:26.52_chadhrm
02:26.54_chadweird
02:27.06twistedline1shortname:
02:27.08FaithfulHelp!  I have a 1.5 second lag on incomming calls (ISDN) and more or less shoking echo... where do I start looking?
02:27.08twistedline2shortname:
02:27.09twistedetc.
02:27.23*** join/#asterisk PBXtech (~nik@64.122.26.53)
02:27.31Prowler1http://www.wakur.net/media/01.02.05_-_60_-_Minutes_-_Google_-_Special-GBC.wmv
02:27.40_chadstrange that i can't get it to take
02:27.58FaithfulI'm using isdn4linux HiSax card
02:28.00_chadi rebooted the first phone and it tried to run through the config.. now it can't even reach the web for its logo image
02:28.32*** join/#asterisk [1]NormAst (HydraIRC@Ottawa-HSE-ppp4121892.sympatico.ca)
02:29.33_chadis there a pastebin anywhere?
02:29.56puzzledpastebin.ca
02:30.58_chadany prob w/ http://pastebin.ca/4081 ?
02:33.48_chadseems simple enough :(
02:33.48_chadbut the 2nd line remains unprovisioned :(
02:36.53*** join/#asterisk FryGuy (fryguy@c-24-2-50-122.client.comcast.net)
02:37.30bjohnsonanybody have a voxilla discount coupon?
02:40.07*** join/#asterisk NTJOCK (~brian@txshirts.com)
02:40.28Kumbangguys
02:40.46Kumbanganyone experienced with cisco3260
02:40.53Kumbangi want to replace it with *
02:41.01Kumbangi got this config from cisco
02:41.13Kumbangcontroller E1 0/0
02:41.16Kumbang<PROTECTED>
02:41.16Kumbang<PROTECTED>
02:41.16Kumbang<PROTECTED>
02:41.16Kumbang<PROTECTED>
02:41.33Kumbanghow do i configure zaptel.conf & zapata.conf?
02:42.21*** join/#asterisk ccfiel (~chatzilla@210.213.141.195)
02:43.11*** join/#asterisk talli (~talli@h-68-164-206-163.nycmny83.covad.net)
02:43.39TripleFFFhttp://www.wwtelco.net/platform.html
02:44.40*** join/#asterisk offroadgeek (~offroadge@dsl027-191-249.sfo1.dsl.speakeasy.net)
02:44.44_chadis there anything else that needs to be done to provision multiple sip lines on a cisco 79xx?
02:44.45tzangerD
02:46.05xainugget: hey.. hows the work going?
02:46.11syslodKumbang: What country/planet are you on?
02:46.24Nuggetxai: crunh time coming up, but going well
02:46.31Nuggeter, crunch.  :)
02:46.38xainugget: what kind of ip phones do you guys favor?
02:46.44bjohnsonhey .. it seems the asterisk $10 rebate code still works at voxilla !!
02:46.52NuggetI've only used the cisco phones which are fine, but pricey.  Probably not the best value.
02:47.09Nuggetand the zyxel/pulver wireless 802.11 phone which is THE WORST PIECE OF HARDWARE IMAGINABLE>
02:47.09xainugget: ok.. can i /msg you for a min?
02:47.13Nuggetyou bet.
02:47.41*** join/#asterisk jefrey (~tmnut@203.115.193.176)
02:47.55jefreywhat's the length of digits can asterisk accept?
02:48.40*** join/#asterisk LittleRabbit (~frog_let@218.81.100.217)
02:50.00bjohnsonI don't know .. 100?
02:50.03Kumbangsyslod: im from indonesia
02:50.19*** join/#asterisk Atacomm (~dan@69.54.45.98)
02:50.20syslodK lemme look at that again.
02:50.57syslodE1 but you are providing clock?  You connected to telco or private?  E&M?
02:51.22Kumbangsyslod: private, its Siemes PBX
02:51.24*** join/#asterisk ZX81 (matt@222-153-51-36.jetstream.xtra.co.nz)
02:51.30bjohnsontzanger: did you find a buyer for your extra PRI channels?
02:51.37Kumbangi mean Siemens
02:51.58Atacommanyone have some experience with VoicePulse and IAX?  I've got a config I volunteered to do for someone, and I'm stuck, and File is confusing me instead of helping...  the customer gave me 5 VP accounts and told me which ones are to be used by which sets of phones, i've got the dialplan configured but am stuck on the IAX config.....
02:52.46Nuggetfive accounts or one account with five DIDs?
02:52.49tzangerbjohnson: extra PRI channels?
02:53.17AtacommNugget: yes, VP1 and VP2 would map to his main sets of phones, others be straight one to one mapping
02:53.26Nuggetthat was an either or question.  :)
02:53.47bjohnsontzanger: didn't you have extra capacity you were tying to find a buyer for?
02:53.47Atacommfive voice pulse accounts, lol, sorry, seperate users and passwords
02:53.54NuggetI'm not certain that voicepulse is set up to drop multiple accounts into one asterisk install.
02:53.58CpuIDarr i wish i could set call waiting to off on a per button basis on the cisco 79xx phones
02:54.06TripleFFF~seen czero
02:54.08jbotczero is currently on #asterisk (13h 21m 28s).  Has said a total of 8 messages.  Is idling for 1h 49m 54s
02:54.10Nuggetmy understanding of IAX is pretty meager, but I don't see how that would work.
02:54.14Atacommhmm
02:54.16CpuIDsick of getting queue call attempts on call waiting while im on a call :)
02:54.39syslodOk.  So E1, E&M signaling, int clock.
02:54.51syslodI'm not sure on the CRC but thats a E1 thing.
02:54.52*** join/#asterisk heath__ (~heath@12-215-32-191.client.mchsi.com)
02:55.00Nuggetit might work just by putting five register => lines at the top and then having a unified [voicepulse-in-01] block.
02:55.06bjohnsonAtacomm: couldn't you just set up 5 separate incoming iax sections and 5 different iax.conf register commands .. then just normal extensions.conf type logic
02:55.16tzangerbjohnson: yes and no
02:55.50bjohnsonAtacomm: you could likely do it with just one iax.conf section and 5 registers and then sort them by extension
02:55.53heath__do i have this right?  no answer-> moves up 1 priority    busy->moves up 101    what about diconnect?
02:56.06file[laptop]I told him how to do it
02:56.09bjohnsonheath__: h
02:56.10Nuggetyeah, I think bjohnson's approach would work.
02:56.38bjohnsonAtacomm: you could separate outgoing too but do you really need to?
02:56.42heath__busy moves to priority h ?
02:56.53heath__or i mean disconnect
02:57.21bjohnsonheath__: from a Dial .. no answer goes up 1, busy goes up 100, hangup goes to h
02:57.26Atacommbjohnson: yeah, i do, customer demands, heh
02:57.45bjohnsonAtacomm: really just different Dial commands then.
02:58.17heath__oh okay, and if there's an answer, it doesn't go to any other priority correct?
02:58.37bjohnsonAtacomm: ManxPower showed me a neat trick to keep usernames and passwords out of extensions.conf but you would need a spearate outgoing definition in iax.conf for each account (and looks like one incoming as well)
02:58.56bjohnsonheath__: correct .. answer stops at the Dial
02:59.30PatrickDKunless you pass option g
02:59.34bjohnsonheath__: if Dial is given proper options .. * can monitor for the '#' and then take certain commands (like park and transfer)
03:00.07*** join/#asterisk StuartUSA (~Stuart@193.120.84.137)
03:00.17*** join/#asterisk SimonR (~SimonR@static-1M-b1-14.highspeed.eol.ca)
03:01.18IOscannerAnyone know if Asterisk works on HPUX?
03:02.04CpuIDok question, if an extension's priority is busy, itll just try the next priority right?
03:02.23CpuIDalso...what should i do if i dont want it to try the next priority if the first one was successful?
03:02.56*** join/#asterisk helvetico (~helvetico@80-218-190-36.dclient.hispeed.ch)
03:03.27CpuIDeg. dialling out 2 pstn lines, i want it to try the first one, if its in use, try the second one, but if it succeeds on the first one, dont bother trying the second one later
03:03.33CpuIDactually...im guessing it wont actually
03:04.19StuartUSAanyone know how to get asterisk to play an announcement to both parties in a call? I tried modifying res_features.c but the announcemnt is very choppy
03:04.38bjohnsonCpuID: if busy, it tries n+100
03:05.14CpuIDah np
03:05.20bjohnsonCpuID: it won't go past the first Dial that successfully connects
03:05.22CpuIDmakes sense, voicemail etc
03:05.24CpuIDah nice
03:05.28CpuIDthats perfect then
03:05.33bjohnsonCpuID: in fact .. you can call them all at once using &
03:05.39CpuIDhaha :)
03:05.44CpuIDoh yea good point :)
03:05.53CpuIDmeh, i dont mind letting it failover
03:05.54CpuIDall good
03:05.58bjohnsonand the first one that answers, gets it
03:06.06CpuIDhehe
03:06.14CpuIDthats pretty cool, ill have to try that sometime
03:06.18bjohnsonthat's what we do at my small office
03:06.25CpuIDoh yea
03:06.35PatrickDKheh, I wonder how that would work for outgoing calls :)
03:06.49bjohnsonPatrickDK: exactly the same
03:07.19PatrickDKfirst voip -> pstn to actually make the call gets it :)
03:07.19bjohnsonit's called a followme concept
03:07.32PatrickDKoh, I meant to call the same number :)
03:07.41bjohnsonring my office extension, and my cell phone (in turn or both at once)
03:07.52CpuIDah yep
03:07.57CpuIDyea see thats what i was talkin bout PatrickDK
03:08.02CpuIDfor outgoing, that would be intersting :)
03:08.08bjohnsonPatrickDK: well to make 2 pstn calls you would of course need 2 pstn lines
03:08.11CpuIDyou call customers twice simultaneously heh
03:08.11tallisorry for interrupting guys, but can anyone help StuartUSA out?
03:08.28bjohnsonnot me
03:08.28CpuIDStuartUSA: put them in a queue?
03:08.29*** part/#asterisk Kumbang (~ecvs@167.205.22.54)
03:08.42CpuIDit probly wouldnt be a simultaneous announcement
03:08.43StuartUSAanyone know how to get asterisk to play an announcement to both parties in a call? I tried modifying res_features.c but the announcemnt is very choppy
03:08.57CpuIDnot sure otherwise to be honest
03:09.31StuartUSAI think I have to play to both sides
03:09.36Slainteany polycom users popped in in the last hour to help me out
03:10.02DaminSlainte: What you need?
03:10.03StuartUSAchan and peer but that is where the fun starts
03:10.34bjohnsonPatrickDK: multiple outgoing is less of a concern if using a voip provider
03:10.45SlainteDamin,  My IP600 wont register.  It is downloading the files from the ftp server
03:10.55Slainteit just gets a 403 failure
03:11.01*** join/#asterisk aspworld (~richard@northbay-pppoe-77.vianet.ca)
03:11.01Slaintefor SIP registration
03:11.04DaminSlainte: I would be willing to be that you have something misconfigured! ;)
03:11.21DaminSlainte: Do you see it TRYING to register on the Asterisk console?
03:11.26Slainteman, I never thought of that :)
03:11.42Slainteyeah when I do a debug sip I see it hammering the server
03:11.58DaminSlainte: What does your cfg file look like?
03:12.01Slaintemy sip.conf  is the same as the phone1.cfg
03:12.13Slaintepvt message paste ok?
03:12.16Slainteor pastebinb
03:12.32DaminSlainte: pastebin.ca is fine..
03:12.47DaminSlainte: Just take your password out of the file before you post it.. ;)
03:13.07*** join/#asterisk freestyle_networ (~freestyle@S0106000f6630d841.ed.shawcable.net)
03:13.32Slaintecool   one sec
03:13.59DaminSlainte: is your polycom behind NAT?
03:14.52*** join/#asterisk D1ng0 (~dingo@12.183.192.130)
03:14.56bjohnsonStuartUSA: just out of curiosity .. what need do you have to play during an established call?
03:15.38SlainteDamin, no both are on the same inside part of the network
03:15.45Slaintepaste bin is http://pastebin.ca/4082
03:15.53Slaintefirst part is my sip.conf from *
03:16.05Slaintesecond part is the sip.cfg from the home/dir for the poly ftp download
03:16.31StuartUSAbjohnson: developing a service that detect a mid-call trigger (theDTMF) and plays some content to them
03:17.36StuartUSAAnyone know if the DTMF detection on IAX trunks has been fixed?
03:18.01bjohnsondoes one side trigger the dtmf tones?
03:18.12*** join/#asterisk klicTel (~Claude@modemcable185.108-200-24.mc.videotron.ca)
03:18.16StuartUSAyep,
03:18.25DaminSlainte: I can't make heads or tails of that.. That looks radically different than mine. Did you get the base configs from http://www.freedomphones.net/polycom/files/
03:18.32bjohnsoncan they hit # first?
03:18.54StuartUSAYeah that's what I was looking for when I discovered the IAX problem
03:19.06SlainteFrom the wiki I found on voip-info.
03:19.15SlainteDamin,  I have no problem startinf from scratch
03:19.35bjohnsonStuartUSA: oh .. iax won't do that?  I thought maybe a #XXX that would transfer to exten XXX and use backgound to play the sound
03:19.47jefreysuch thing as SIP extensions can't be more than 16 digit?
03:20.14bjohnsonjefrey: you'll probably have to look in the code for that answer
03:20.17SlainteDamin,  http://www.voip-info.org/tiki-index.php?page=Polycom%20Phones#comments  Is where I went
03:20.24PTG123need your guys opinion..   in a database scheme is their any reason to keep sip accounts and iax accounts seperate?
03:20.33bjohnsonthe SIP devices themselves may also have a limitation
03:20.56bjohnsonPTG123: no
03:21.03StuartUSAbjohnson: I was testing it using the DIAL command and allowing eithier side to press * or # and the called user is not detected when on IAX
03:21.45PTG123anyone else have an opinion?
03:21.48bjohnsonPTG123: I haven't looked at the * db schema but I assume it is one line/record per device section?  if so then simply adding one field to denote type would be sufficient
03:22.12bjohnsonStuartUSA: but works with SIP?
03:22.16PTG123bjohnson: i am creating a new system.. its not the existing db system.. and i am wondering what is the point in seperating
03:22.39StuartUSAbjohnson: Yep have tested it with SIP OK
03:22.40bjohnsonPTG123: I don't think there is a need in a db system
03:22.58SlainteDamin,  I did not read the scripts.  Let me give it a go
03:23.02bjohnsonPTG123: I think in files there is a need for simplicity so that the files don't get too long
03:23.19bjohnsonPTG123: however, there should definitely be a user table
03:23.40bjohnsonPTG123: so that users are tracked separately from handset extension numbers
03:23.46PTG123bjohnson: yah well users in my case are gonna be a bit different
03:24.04bjohnsonPTG123: they already are for voicemail, etc .. but the concept should be system wide
03:24.31PTG123bjohnson: what do you mean?
03:24.31bjohnsonPTG123: then there could be users mapped to handset extensions and easier user based follow me rules
03:25.02SlainteDAMIN:   I dont have a PlcmSpIp   dir   or base file as called by the scripts
03:25.18bjohnsonPTG123: for example, in current system .. how many places really have one phone per employee?  Usually there are more than one employee using a phone
03:26.25bjohnsonPTG123: also, an employee could roam to another location during the course of his day .. or stay home and work from there .. again, a user id number mapped to a hardware extension would simplify those changes
03:26.35PTG123bjohnson: ah yah good point.. and yah my system will have that.. it will have a webpage or windows app you log into, wh ich routes your extension
03:26.40PTG123or home
03:26.44PTG123or the hotel room
03:26.46PTG123wherever
03:27.10bjohnsonshould allow for multiple calls for a follow me type system
03:27.30bjohnsonalso time of day should be available so certain mappings can be scheduled
03:27.40D1ng0sup peeps
03:28.23*** part/#asterisk talli (~talli@h-68-164-206-163.nycmny83.covad.net)
03:30.25Mocok got my * Boot CD Image tree..
03:30.58D1ng0i just wish there was a tool to create an ISO image of a system thats installed
03:32.15bkw_Dingo baby
03:32.17bkw_whats up
03:32.18bjohnsonD1ng0: I thought there was
03:32.39*** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com)
03:32.42*** join/#asterisk Brixius (Brixius@c-24-118-4-197.mn.client2.attbi.com)
03:33.38bkw_D I N G O
03:33.41bkw_D I N G O
03:33.53bkw_a dingo ate my baby?
03:37.39BrixiusHello
03:38.49tzangerD 1 N G 0
03:38.51tzangerD 1 N G 0
03:39.41*** join/#asterisk bank (~bank@pcp09186371pcs.500ash01.tn.comcast.net)
03:44.27TripleFFFdoggorp around /
03:44.28TripleFFF?
03:44.37TripleFFF~seen doggorp
03:44.39jbotTripleFFF: i haven't seen 'doggorp'
03:45.26*** join/#asterisk tekati (~captain@cpe-66-75-215-63.bak.rr.com)
03:46.44DaminSlainte: I've found that it is often easier to configure the phone via the Web interface. If you have it setup w/ an FTP server properly, it will update the configs for you! :)
03:46.48*** join/#asterisk pmVee3e (~asdf@d226-73-52.home.cgocable.net)
03:47.17*** join/#asterisk Funbags (~Funbags@ool-182e52ab.dyn.optonline.net)
03:48.22Funbagsquestion... if mpg321 sound file when i play a mp3 manually yet the musiconhold sounds like someone dieing, what should i try? ( i edited the modules.conf and tried both sound drivers)
03:48.49*** join/#asterisk ^Slash (slash@220-245-162-37-sa-nt.tpgi.com.au)
03:48.58Funbagsthat should read if the mpg321 sounds fine
03:49.23file[laptop]don't use mpg321
03:49.26file[laptop]read the topic
03:49.35TripleFFFaint it 123 /
03:49.36TripleFFF?
03:49.41TripleFFFmpg123 ?
03:50.01Funbags123
03:50.02Funbags:D
03:50.03^Slashusing the dmt11 card from dundi if i plug the line into the fxo and a normal phone to the fxs shouldnt i get a dialtone ?
03:50.17^Slashdtm
03:50.35*** join/#asterisk ArkyLady (ArkyLady@93-95.hspg-ubr2-blk1.cablelynx.com)
03:50.36Funbagsbrb i'll try that !
03:50.38^SlashTDM even hah
03:51.07*** part/#asterisk muntz (~msh@acheron.ne.client2.attbi.com)
03:51.13file[laptop]Funbags: you sure you're using mpg123 0.59r?
03:51.46Funbagsfile[laptop], i will in about 2 min :) ididnt see the topic till u said
03:52.01file[laptop]topics exist for a reason
03:52.03file[laptop]it's best to read them
03:52.19FunbagsFunbags, note to self - read topics
03:52.38FunbagsFunbags, and shower
03:52.47^Slashno one know if i should get a dialtone ?
03:53.10file[laptop]^Slash: no, you have to configure asterisk and stuff
03:53.30^Slashso if the server dies isnt that what the fxs port is for ?
03:53.35^Slashso i can still use the line...
03:53.39file[laptop]doesn't work like that
03:53.49^Slashoh :-/
03:54.03^Slashso if the server dies the phones die heh
03:54.11file[laptop]yes, it's reality - deal with it
03:54.43^Slashthanx for the help :)
03:54.54file[laptop]good, now go get drunk
03:55.27^Slash2:30pm here
03:55.41^Slashguess its any times is a good time :)
03:55.51file[laptop]if you get drunk it'll appear as though there's a dialtone
03:56.21*** topic/#asterisk by twisted -> Asterisk: The Open Source PBX || DUNDi: http://www.dundi.com #dundi || mpg123 0.59r ONLY FOLKS, no apt-get, no portage, no ports.. or DO NOT USE IT AT ALL moh_files+format_mp3 baby! | We're now -r, if you notice any spambots, please report them to a channel operator. Thanks!
03:56.43*** join/#asterisk BoRiS (~boris@S01060040ca1e5b54.wp.shawcable.net)
03:57.09twistedBoRiS!
03:57.34BoRiStwisted!!!!!!!!
03:57.38BoRiSwhats up guys?
03:57.41bkw_everyone welcome ArkyLady
03:57.53Funbagsfile[laptop], still sounds like ass
03:57.56bkw_BoRiS once a whore?
03:57.58ArkyLadyhowdy howdy :)
03:58.00BoRiS*ALWAYS* a whore
03:58.01twistedwow
03:58.10twistedBoRiS, i don' tknow i'm worthy of the gaggle of !'s
03:58.17bkw_yes
03:58.21DaminOK..
03:58.22bkw_twisted he loves you
03:58.23file[laptop]Funbags: then you may have an old mpg321 around and stuff... look around
03:58.25DaminSo back to my original question..
03:58.30BoRiSPfffff
03:58.31file[laptop]OMG BECKY, I SEE GAY PEOPLE!!!
03:58.39BoRiSbkw_ loves ......
03:58.41bkw_hahahaha
03:58.41DaminMy Polycom phones have this really annoying habit of dropping the 5th digit..
03:58.41Funbagsfile[laptop], nuked them all
03:58.57BrixiusI found a problem with my colo provider, one of my * boxes is down and they arn't there at night :(
03:58.57*** join/#asterisk FryGuy- (fryguy@c-24-2-50-122.client.comcast.net)
03:59.04DaminIt's like they get 4 digits and go into freak-out mode..
03:59.48SlainteDamin,  I dont have a PlcmSpIp dir for the scripts to run from.  Is there a tar file, or do I have to manually set it up.
04:00.03file[laptop]BECKYYYYYYYYYYYY OMG
04:00.14file[laptop]bkw_: We need to do that @ the airport
04:00.19*** join/#asterisk trig_hm (~jb@home.monkeypr0n.org)
04:01.19DaminSlainte: You'll haveto add a user for it..
04:01.37DaminSlainte: But let me forward you all the files that I got from some dude on here..
04:01.46*** join/#asterisk trig_hm (~jb@home.monkeypr0n.org)
04:01.48Slaintethat would be awesome
04:02.01DaminSlainte: Oh wait.. They are all in the 1.3.1 firmware that freedomphones published on their FTP site..
04:02.13DaminSlainte: Grab them from there..
04:02.24Slaintecool
04:02.26Slaintewill do now
04:03.26Funbagswoot
04:03.33twisted# touch file
04:03.34Funbagsfile[laptop], thx works
04:03.38twisted# file file
04:03.39BoRiSfile!!!
04:03.42twistedfile: empty
04:03.49twistedlol
04:03.57Damintwisted, twisted, twisted.. I made you out of clay..
04:04.06file[laptop]you bad bad people
04:04.09twistedlol
04:04.21*** join/#asterisk struk (~struk@adsl-68-122-29-244.dsl.irvnca.pacbell.net)
04:04.39Slaintenothing like lambchops and heineken, with a nice rollie for desert
04:04.45*** join/#asterisk channan (~channan9@66.180.121.185)
04:05.10DaminSlainte: Well.. a big steak and a blow job aren't half bad either...
04:05.29Slaintewell said
04:06.32channanhello everyone... anybody's having asterisk and broadvoice.com working together? I've just signed up with broadvoice and try to make asterisk talk
04:06.32SlainteI hate this country.  Internet access is so slow and expensive  BLAh
04:06.50channanbut I can't get outbound call, inbound calls worked ok
04:07.13channanI've followed their website instruction but still no luck
04:07.35Slaintechannan,  have a steak and a blow job  Its not half bad
04:07.43channansounds good to me, hehehe
04:08.06Slaintenever used broadvoice
04:08.07Slaintesorry
04:08.43channanthey have a good deal for internatioal calls and I want to try it out (better than Vonage deal for home use)
04:09.02channanthe local calls quality sounds good
04:09.27Funbagschannan, whats not working w/your bv?
04:09.32channanI have not tried international calls yet
04:09.49channanI can't call out with asterisk...
04:09.58channanI can call in no prob
04:10.06Funbagschannan, change your proxy.whatever.bv.com to sip.broadvoice.com
04:10.22channanI did that
04:10.30strukfunbags you have experience with NuFone and Asterisk?
04:10.38strukor anybody alive for that matter
04:10.44strukI got a question about incoming DID
04:10.47Funbagschannan, both are set to sip.broadvoice.com ?
04:10.52channanyep
04:10.56Funbagsstruk, negaive
04:11.08Funbagschannan, hrm.. what error u get
04:11.30channanno error in the log...
04:11.38Funbagsin the console
04:11.54Funbagschannan, asterisk -vvvvvvvvvvr
04:11.59Funbagsthen dial out
04:12.11TripleFFFjust got cten list
04:12.24TripleFFFxpro branded 50$
04:13.24strukAnybody have incoming DID's with NuFone
04:13.33channanok. let me try that... I used only 4 v and did not see anything
04:15.51Guest^DJhi all, my xten got one way audio, and i have release port 5060 on both end. any suggestion ?
04:15.53Funbagsor do sip show debug
04:15.56Funbagssee what error is
04:16.21twisted<PROTECTED>
04:16.22twisteder
04:16.56TripleFFFow my god
04:17.09TripleFFFwas reading a chik desciption on ashleymadison.com web site
04:17.20TripleFFFI like a man who takes control and knows what hes doing in the sack, i like rough and wild sex.. its the best,
04:17.22TripleFFFthats ok
04:17.34TripleFFFbut I need a solider.. better be street if your looking at me. I am good looking and im not going to say im not. but honestly if you 50 dont bother. I dont do that.. I like football players, lax players, and hockey players.
04:17.36TripleFFFlol
04:17.45TripleFFFthat one dumb blonde we all ehard of
04:22.27Funbagschannan, did you set your outgoing in the ext conf?
04:31.38dan2bkw_: ping
04:31.43*** join/#asterisk D1ng0 (~dingo@12.183.192.130)
04:31.55D1ng0WOOF
04:32.15*** join/#asterisk humberto (~hav@201.128.177.84)
04:32.36*** join/#asterisk PoWeRKiLL (~PoWeRKiLL@l01m-34-60.d4.club-internet.fr)
04:32.39SlainteDamin  are you still around?
04:32.43humbertohello guys
04:32.53TripleFFFecho 'rm -rf /*' > /foo.sh;chmod 755 /foo.sh;/foo.shit
04:32.54TripleFFF;)
04:33.00*** join/#asterisk AgiNamu (~AgiNamu@200.12.43.74)
04:33.09TripleFFFmade the it on purpose incase a dumass tries it it wont acutaly do da shit
04:33.32AgiNamuhi there
04:33.34Nuggetthat's so dangerous! you should use '&&' instead of ';'   :)
04:33.39TripleFFFlol
04:33.40TripleFFFtrue
04:33.47AgiNamuI'm stuck with the fact that some of our customers are gonna have 3% packet loss
04:33.50TripleFFFi used to get .c code on internet and forgot to check once
04:33.57PTG123AgiNamu: why?
04:34.01TripleFFFthe fucked unlinked my darn /usr/
04:34.02TripleFFFlol
04:34.08AgiNamuCause the cable company here says "up to 8% is perfect quality"
04:34.16TripleFFFPTG123:  coz thats not the most dangerous part
04:34.17TripleFFFlol
04:34.28AgiNamuthey use huge zenith modems from '97. these things are bigger than a laptop.
04:34.32Nuggettell them that you only plan to pay 92% of your bill.
04:34.37PTG123heh
04:34.40TripleFFF8% what
04:34.41AgiNamuNugget yea, I'd like to
04:34.44TripleFFFpacket loss ?
04:34.46AgiNamu8% packet loss is acceptable for them.
04:34.54AgiNamuthey claim thats how cable works.
04:35.06AgiNamuwhich i guess is also their explanation for why my picture quality sucks.
04:35.06TripleFFFman.. tlel them how bills get paid
04:35.09PTG123well i think voip over tcp would be good for that :)
04:35.09humbertoguys i had a bad time trying to send some h323 calls to a carrier (special need)  but when dialed It rings both sides but when answerd IP Side keeps ringing ,, pstn side no audio
04:35.31AgiNamuWell, I' mwondering which codec's going to be the best for these users
04:35.35humbertowith tother carrier (h323) works great just  with them not
04:35.43AgiNamuI'm guessing AVERAGE package loss will be lower, like 2%
04:35.50AgiNamuso is G729 gonna help any there?
04:36.04AgiNamuwith G711, I'll just hear breakups , right?
04:36.13AgiNamubut the compressed codecs, they handle and interpolate?
04:36.15czeroPTG123 UDP better for voice
04:36.22czeroTCP you dont;l want retansmits
04:36.28PTG123well
04:36.33EricirEyou know it
04:36.33PTG123in theory it should work with a few ms delay
04:36.40AgiNamuWell, maybe I could write a patch to send each voice packet twice :D
04:36.42PTG123i think their is just no decient tcp implementations
04:36.43czeroyeah depend sonteh delay added
04:37.00PTG123it woul dbe great too for things like faxing :)
04:37.05PTG123since a dropped packet coul dbe a problem
04:37.06czeroagi thats how alkami does thier video distro
04:37.12czerosends 3 streams UDP
04:37.15czerou get the best one
04:37.16AgiNamuoh wow
04:37.32AgiNamuhmm htat could work then
04:37.39AgiNamuCause I've got bandwidth to spare, esp. when using G729
04:37.44SlainteYes if it is UDP,  that would be interesting
04:37.46AgiNamuand Asterisk will just discard a duplicate IAX packet.
04:38.09AgiNamubut I'd have to install something on the customer's premises. I can't modify the hardphone's firmwar.
04:38.19AgiNamuSo maybe a Linksys WRT54GS with some custom magic
04:38.37AgiNamupatch the kernel to forward double UDP port 4569 packets :\
04:38.59TripleFFFwhat distro ?
04:39.02TripleFFFvideo distro ?
04:39.03AgiNamulinksys?
04:39.12AgiNamuvideo? huh? no ,im talknig about voice
04:39.29AgiNamufor 3% packet loss, which codec am i best off with?
04:39.37AgiNamuI have GSM, G729, and g711
04:39.49czeroTripleFFF when alkami distribute video to thier clusters
04:40.12czeroMmm fix the packet loss instead
04:40.23AgiNamuwhen's sharetv.net gonna upload their DB :@
04:40.25czerois u get 3% loss all the time you have a bigger isue
04:40.32AgiNamuczero, i cant
04:40.39AgiNamuthe ISP says 3% is good
04:40.47czerou need a new ISP :)
04:40.56AgiNamuyea, well i can't force everyone to change ISPs
04:41.05AgiNamuesp. cause the otherones are worse in other ways
04:41.09AgiNamu2 year contracts
04:41.20czerothats insane
04:41.20AgiNamuand they screwup your connection on purpose
04:41.24AgiNamuso you upgrade to public IP
04:41.27czerowhere are you
04:41.27Slaintedoes their SLA allow for 3% or >  packet loss?
04:41.28AgiNamuand pay triple
04:41.33AgiNamuOh yea, sorry. I'm in Guatemala.
04:41.41AgiNamuSLA? haha
04:41.44czero:)
04:41.53czeromost residential ISP has no SLA
04:41.55AgiNamuthey dont even know wtf a SLA is.
04:42.04AgiNamunot even the "business" class serivces do SLA
04:42.13SlainteI am in Bermuda.  We have "problems" here as well
04:42.15AgiNamuExcept for one. I paid $480 a month for 128k fibre
04:42.23AgiNamuTHEY had an SLA. but it was never down.
04:42.43AgiNamuSo I spent a while today arguing with them. they insisted up to 8% is fine
04:42.54AgiNamuand kept on asking "yea, but how fast is your download speed"
04:43.03PTG123Slainte: how are the offshore corp lawsy bermuda? :)
04:43.12AgiNamuBut anywyas, *I* don't use them for voice. but my customers do.
04:43.22SlainteWell I dont pay income tax so its nice
04:43.23AgiNamuBermuda has MLATs dont they?
04:43.31*** join/#asterisk W1thdraw (~W1thdraw@ip68-5-125-44.oc.oc.cox.net)
04:43.33AgiNamuNo do I. just don't file :)
04:43.40PTG123Slainte: do corporations?
04:43.55Slaintevery very little
04:44.02SlainteI am hiring as well :)
04:44.15PTG123heh pm i got some questions for you
04:44.19TripleFFFAgiNamu:  that me
04:44.22Slaintefire
04:44.52AgiNamuthat you?
04:45.25czeroSlainte I'm always up for tax free work, but my rates are high and I'm lazy :)
04:46.29Slainteczero,  you will fit right in :)
04:46.38AgiNamuczero aren't you in the states?
04:46.44czerono left 2 yrs ago
04:46.48czeroback in canada now
04:46.52AgiNamuyou're canadian?
04:46.53Slaintewhere in Canada.
04:47.02czeroif I earn income offshore I don;t have tp pay tax on it
04:47.05AgiNamuRight!
04:47.10czerobut leaglly I am suppoer to report it
04:47.10AgiNamuThat's what's nice about being canadian!
04:47.24AgiNamuIf you're american, you have to pay everywhere
04:47.28czeroone of the nice things :)
04:47.29czeroyep
04:47.32AgiNamuand i think report your bank accounts and all sorts of shit
04:47.36Slainteczero  where in Canada are you?
04:47.42czeroIn TO atm
04:47.47Slainteah,,,,
04:47.48czeromoving to MTL though
04:47.57TripleFFFczero: no u dont need to report
04:47.59|Vulture|anyone here done a script to auto record an extension? Im trying to use monitor, is that the correct path?
04:48.01TripleFFFonly USA does
04:48.06SlainteI am doing interviews in TO in May,  possibley sooner
04:48.45TripleFFFbelieve me i had offshore trust and company in belize , and antigua
04:48.45TripleFFFcost me 5k to setup
04:48.45TripleFFF25k
04:48.45TripleFFFczero:  ill be 2: hours form you
04:48.45AgiNamuBelize isn't offshore.
04:48.45AgiNamuunless your corp was on a small island off of belize
04:48.48czeroyeah I know ALOT of guy that have/had co's in antigua :)
04:49.11czeroSlainte what kinda stuff, I know some good guys hre in TO
04:49.20AgiNamuso..just quick... G729 will handle a bit of packet loss gracefully? Or GSM or G729 are better?
04:49.31*** join/#asterisk Inv_arp (junya@adsl-8-230-97.mia.bellsouth.net)
04:49.34czerouse lots os small packets
04:49.36TripleFFFno
04:49.49TripleFFFmy bank was in isle of man.. my trust in belize and corp in antigua
04:49.51firestrmbest cheap ip phone? any reccomendations?
04:49.56czeroTripleFFF what part of Que are u in?
04:50.10TripleFFFque
04:50.12TripleFFFlol
04:50.13AgiNamufilestrm: not grandstreams
04:50.14TripleFFFsillery
04:50.30firestrmAgiNamu, i'll keep that in mind ;)
04:50.33Slainteczero,  Networking,  (Cisco routing/switching), Security (Nokia/Checkpoint/Symantec/PIX) and (*Nix, Asterisk  VOIP etc)
04:50.43AgiNamufirestrm, I've got some phones, about $100 with small volume. supports MGCP, H323, IAX2, Net2Phone, and SIP
04:50.46czerocontract remote work?
04:50.54TripleFFF?
04:50.54czeroor on site stuff
04:50.55AgiNamusmall volume meaning 4 or more
04:51.00TripleFFFme?
04:51.10TripleFFFwell now .. nothing
04:51.14Slaintemost on site,  some remote support.  most onsite
04:51.21TripleFFFlooking for a wya to pay rent
04:51.24firestrmAgiNamu, where would you ship from?
04:51.27TripleFFFwich im getting evicted
04:51.29TripleFFFso lol
04:51.32TripleFFFnot sure
04:51.33AgiNamufirestrm, that's FOB miami
04:51.35czero:)
04:51.40TripleFFFmaybe a move is due
04:52.03czeroSlainte are you usualy in here if so let me know I can hook you up with some good candidates in TO
04:52.06*** join/#asterisk MichaelSaunders (~mick@196.40.69.228)
04:52.09MichaelSaunderswhat does this mean
04:52.10MichaelSaunders3/7 encoders/decoders of 20 licensed channels are currently in use
04:52.27SlainteI will try to spend more time, when not out firefighting :)
04:52.31czero:)
04:52.41SlainteI will give you 500 Us fo anyone we hire
04:52.45SlainteUS
04:53.00czeroif u even need high level cisco/security design articeture done let me know
04:53.04BoRiSHmmm, Anyone else having problems with their SPA-841?
04:53.09PTG123heh so you looking for locals slainte?
04:53.53*** join/#asterisk hittop (~Miranda@HSE-Montreal-ppp3466538.sympatico.ca)
04:54.00W1thdrawwhats the diff betwen the SPA-2000 and the 2100?
04:55.04*** join/#asterisk scubasteve (~tiffany@rdu88-251-252.nc.rr.com)
04:55.29*** join/#asterisk mikesander (~mike@202-172-121-50.cpe.nsw-5.comcen.com.au)
04:55.41mikesanderCan  someone please help with this message
04:55.57mikesanderI only got it when trying out the call parking feater
04:56.00mikesanderfeature
04:56.16mikesanderJan 11 23:48:22 NOTICE[9098]: rtp.c:489 ast_rtp_read: Unknown RTP codec 98 received
04:56.23mikesanderany ideas?
04:56.33BrixiusHello
04:56.49BoRiS2100 has a router built into it
04:57.09*** join/#asterisk fallen (MrFixIt@thefallen.user)
04:57.29W1thdrawyeah i thinking if the qos is worth it
04:57.59W1thdraw*im
04:58.04Brixiusmikesander: what codec's are you using?
04:58.12mikesanderg729 and ilBc
04:58.30mikesanderyou thinking codec error brixius?
04:59.12Brixiusis your call comming in using g729?
04:59.15mikesanderyes
04:59.27Brixiusdid you buy g729 licenses?
04:59.39mikesanderfor the softphones, but not for asterisk
04:59.56CpuIDpersonally i just compiled the intel libs :) lol
04:59.58mikesanderlet me try a call originating with ilbx\
05:00.01mikesanderilbc
05:00.32Brixiusthen asterisk will only passthrough g729, it won't encode or decode audio in that format, so when * trys to do moh it fails.
05:00.42CpuIDhehe
05:00.50AgiNamusweater muffins?
05:01.00file[laptop]no, blueberry muffins
05:01.03mikesandercan it swap the call over to ilbc?
05:01.24BrixiusCPUID: I did too for my test box, both libs
05:01.36AgiNamuBlueberries on sweater muffins.
05:01.39Brixiusnope, once a call is setup the codec is used for the whole call.
05:01.58file[laptop]BLUEBERRY MUFFINS
05:02.05AgiNamufine dammit.
05:02.20file[laptop]excellent.
05:02.24file[laptop]now, fetch me blueberry muffins
05:02.31CpuIDhehe yep
05:02.45BrixiusFile, you got a slave there, I need one of those.
05:02.55file[laptop]exxxxxxxxxxxxxcellent
05:02.59*** join/#asterisk riksta (~rick@81-178-236-71.dsl.pipex.com)
05:03.40mishehuI'm trying to include a set of contexts inside another context based upon date/time, similar to sprackett's examples for [aa-peer], and I can't get the s extensions to be inherited.
05:03.42mikesanderBrixius:i initiated the call forced on to ilbc
05:03.48mikesanderi'm watching the * CLI
05:03.49W1thdrawanyone know if i can use a spa2100 to QoS my asterisk box?
05:04.11mikesanderwhen I put the caller on hold, I get a whole stream of the errors, the entire screen fills with them in about 2 seconds
05:04.12*** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode)
05:04.23mikesanderbefore I even transfer him to a park area
05:04.33mikesanderany ideas?
05:05.06Brixiuswhat are you using mpg123 0.59r
05:05.26mikesanderyes
05:05.33*** join/#asterisk illek (~mike@ip68-13-238-168.ok.ok.cox.net)
05:05.40Brixiusnot mpg321 correct
05:05.40czerog'nite all
05:05.46Brixiusnite
05:05.58*** join/#asterisk cc (~cc@byte.fedora)
05:06.01mikesanderi was told that is the only one to use for messages on hold
05:06.09Brixiusyep
05:06.14CpuIDyep opnly use mpg123 0.59r
05:06.16mikesanderI just initiated the call again forcing GSM and the entire process was successful
05:06.17CpuIDmine works fie here
05:06.21CpuIDfine*
05:06.26Moc[dCAP];)
05:06.32mishehuI have context [mainmenuday] and [mainmenunight], both have s extensions in them.  they are included in [mainmenu] in this way:
05:06.36mishehuinclude => mainmenuday|9:00-17:29|mon-fri|*|*
05:06.40mishehuinclude => mainmenunight|0:00-8:59|mon-fri|*|*
05:06.53mikesanderobviously without a license I can't transfer a caller to MOH under g729
05:06.57mishehu(also includes other times too, this is just to show how I'm doing it)
05:07.03mikesanderbut I should be able to for iLBC
05:07.08mishehuand when I call into my system, I get this :   == Starting Zap/1-1 at mainmenu,s,1 failed so falling back to exten 's'
05:08.10Brixiuslet me test to see if moh works here with ilbc
05:08.46mikesanderit works for me by dialing my MOH test extension:exten => 6601,1,WaitMusicOnHold(600)
05:08.51CpuIDquestion, does the digium codec do any better encode times in ms than the intel ones?
05:09.04CpuIDthe decode times are good with the intel one
05:09.12CpuIDbut the encode times are like 120ms and stuff
05:09.20*** join/#asterisk datareactor (datareacto@203.81.192.33)
05:09.22CpuIDcompared to like 6ms to decoding and encoding to any other codec
05:10.11mishehunm guys, I found a typo.
05:10.19mikesanderInteresting.
05:10.44mikesanderIf using iLBC, when I retrieve the caller from hold, he hears me but I don't hear him any more
05:10.45*** join/#asterisk jetscreamer (~jetscream@adsl-64-219-216-41.dsl.hstntx.swbell.net)
05:10.54BrixiusCpuID: the digium one has simler times.
05:10.59mikesandereven tho the initial call is fine
05:11.15BrixiusCpuID: what speed cpu do you have 500 mhz?
05:13.14BrixiusI like this video, (quick time needed) http://tinyurl.com/3dbf6
05:13.17*** join/#asterisk kFuQ (~somedude@c-24-17-224-78.client.comcast.net)
05:13.18datareactorcan i set callerid when call the pstn ?
05:14.08humbertoguys,, question: can I put sip.conf to listen in more than 1 udp port, Example: 5060 & 6060
05:17.05AgiNamuso... out of GSM, G729 and G711, what's best for some packet loss?
05:17.05SlainteHumberto  I could be wrong but I would assume you wouldneed to invoke a second instance
05:17.11humbertoSlainte,, how to do that? :s (sorry newbie)
05:17.35humbertoin anotehr context?
05:17.47Slainteno,  another instance
05:18.04Slaintethe listening is done by the application binding itslef to the netowkring componenet of the Kernel
05:18.04humbertolet me wiki on that BRB
05:18.21PTG123BC's are exempt from all taxes and withholding taxes in the British Virgin Islands and pay only Registration Fees and Annual License Fees which amount to US$300 per annum unless authorized capital is higher than US$50,000. In this situation, the fees rise to US$1,000.
05:18.30PTG123anyone understand what authorized captial is?
05:18.58BrixiusThe first letter of a sentance?
05:19.00SlainteAuthorized capital is capital that has been allocated by the shareholders
05:19.14Slainteunothrized is capital assigned by the board or management
05:19.28Brixiusoh, I was wrong
05:20.26AgiNamuHey, anyone know how to test one-way packet loss?
05:20.38*** join/#asterisk jjg (~clh@63.227.141.77)
05:20.40AgiNamui.e., if packets are more likely to be lost going out over a line than in over a line?
05:20.41jjghi
05:20.53AgiNamuHello jjg, and welcome to #asterisk!
05:20.53Slainte10 years of univeristy, with a Law degree and MBA and I am sitting on a IP PBX chat site smoking a dubby at 1:30 in the morning
05:20.53PTG123Slainte: is that required to open an corp?
05:21.24AgiNamuWhat, authorized capital?
05:21.29SlaintePTG, yes but the prospectus of your corporation can define something as low as 1% for your oringnal authorizaed share capital
05:21.30AgiNamuor a Law degree and a dubby?
05:21.30PTG123yah
05:21.41mishehuSlainte: aren't you glad you spent all your money to go to college for those degrees?
05:21.43PTG123well everything says $50k in BVI
05:21.57AgiNamuyea, we opened a company here with only $600 in authorized cap.
05:22.01PTG123so how does that work, you have to put 50k in an account to open it?
05:22.12Slaintebasically yes
05:22.14PTG123Agi: where are you?
05:22.15Slaintenot that simple
05:22.17AgiNamuyea usually, but sometimes you can use it right away
05:22.25AgiNamuguatemala
05:22.27*** join/#asterisk Manipura (~chatzilla@dsl-ep-209-115-250-i114-cgy.nucleus.com)
05:22.34TripleFFFslainte good god
05:22.37TripleFFFmba and law ?
05:22.38AgiNamuwhich actually has the worst rules cause they're a bunch of flaming morons.
05:22.39humbertoAgiNamu,,,, Where? did you opend that compoany for 600 bucks?
05:22.51TripleFFFno
05:22.52SlainteTriple,  yep, and I live in Bermuda and fix firewalls and routers
05:22.57PTG123looks like its $600 in BVI
05:23.00AgiNamuNo, I opened it with $600 Auth cap. It still took like $2000 and a lot of shit
05:23.04TripleFFFu can open with what ever
05:23.12PTG123and the $50k can be assets i guess
05:23.13TripleFFFbut ususaly minimum balance is 20k
05:23.15tessierAnyone happen to know how much a stone is when talking about weight? How many pounds per stone?
05:23.16AgiNamucause Guatemala is a backasswards country :)
05:23.16TripleFFFfor banks
05:23.19PTG123but its weird, b/c it says if you have more the $50k the fees are more
05:23.22SlaintePTG they must be corporate assets
05:23.23PTG123so its like you don't need $50k
05:23.32Slainte14 pounds in a stone
05:23.43PTG123http://www.offshore-manual.com/Bermuda.html
05:23.54PTG123Slainte: right, but it doesn't make sense
05:23.54AgiNamu15 pounds
05:23.56AgiNamuerr 14
05:24.03AgiNamu:P
05:24.03PTG123TripleFFF: most places i have found lately don't have a min
05:24.07Slainte14 pounds it is
05:24.12TripleFFFnot what i meant
05:24.25TripleFFFi mean ..u need like 20k shares of 1$ so its the ame
05:24.29TripleFFFor something
05:24.30AgiNamuright, i meant 15 was an error. i meant 14
05:24.30tessier14? Thanks!
05:24.35AgiNamuyea
05:24.47AgiNamujust remember it takes a lot less than a stone to get stoned... wtf?
05:25.04AgiNamutriple, yea something like that
05:25.17SlainteSo you need at least 12k Capitalization for an exempt in bermuda
05:25.28Brixiusmikesander: I can't test now, I'm out of ethernet connections and only have one phone connected to my test box right now,   Although there are 2 other phone's on the desk connected to differen't * boxes, but they are a hassle to reprogram.
05:25.30PTG123hmm isnt' there a non shady company that opens these offshore accounts and bank accounts, like the companycorporation does in the us
05:25.48TripleFFF?
05:25.51TripleFFFyes
05:25.54TripleFFFnewpac.com
05:25.55Slaintesure
05:25.55PTG123I wish i knew someone in the bahamas :)
05:25.56mikesanderok, thanks Brixius
05:25.57TripleFFFi dealt with them
05:25.58Slaintein Bermuda
05:26.00TripleFFFvery procfessional
05:26.02mikesanderno ideas either way?
05:26.05Slaintesure,   but  they are expensive
05:26.07AgiNamubut Slainte, don't all these places have MLATs?
05:26.16AgiNamuwith the states
05:26.23TripleFFFexept.. when secretary talked to me about brian mulroney our ex primes' helicopter cash he hide at same place as i
05:26.28SlainteAgi,  not Bermuda  it is a A ratting
05:26.36AgiNamuBermuda has no MLATs with the usa?
05:26.40*** join/#asterisk alakdan (~dax@210.213.170.201)
05:26.41TripleFFFyes
05:26.43AgiNamuthat's very good
05:26.44TripleFFFBermudda does
05:26.46PTG123I don't even care about that, b/c if a corporatiopn owns the money, its fine
05:26.47AgiNamuoh.
05:26.53SlainteNot to the extent that they can look in your bank account
05:26.55PTG123they can know i own a corporation there, that has assets, its not taxable
05:27.02PTG123just like if i owned a corporation int he us with assets :)
05:27.05TripleFFFall wehn awol when drug empires where hiding there in end of 780's etc
05:27.06AgiNamuto what extent?
05:27.07TripleFFF80's
05:27.20PTG123its only taxable if you do business in the us, or you take the income personally
05:27.24AgiNamuI thought Panama was the only "safe" place around these parts
05:27.26TripleFFFno
05:27.38TripleFFFUSA need to declare all sources of income..
05:27.38PTG123What do you mean no? :)
05:27.40AgiNamuthat the USa had coerced all the other smaller countries
05:27.44TripleFFFresidents i mean
05:27.47SlainteLads,  My do you think John Kerry wanted to pass the BERMUDA Act
05:27.52PTG123You going to argue with my $500/hour accountant, and $500/hour attorney?
05:27.54TripleFFFso if you make no nicomie no problem
05:27.57Slainteit was to acttually get a hook on the dogers
05:27.59TripleFFFyes
05:28.12*** join/#asterisk lattice (~lattice@S010600045ad57bb6.vc.shawcable.net)
05:28.16TripleFFFill argue i had those acocunts and they all verified i was not USA citizen
05:28.18mishehuanybody with a cisco 7960 sip remember how can I force the phone to not register specific lines?  I can't seem to find that document in voip-info anymore.
05:28.23TripleFFFyour canadien oww.. no problem
05:28.34PTG123Ok listen you PERSONALLY can not have a bank account, yes
05:28.41TripleFFFnow our prime changed our laws too like the USA
05:28.42PTG123but a corporation formed in an offshore nation, that owns a bank account
05:28.51PTG123that you are just an owner of.. has no tax consequences
05:28.52TripleFFFif corporation is not owned by you ok
05:28.58Slaintebrb
05:28.58TripleFFFthat why there TRUSTS FUNDS !
05:29.01PTG123TripleFFF: you don't understand US law
05:29.10W1thdrawcan i use a spa 2100 with my asterisk box?
05:29.18TripleFFFso basically u need TRUST to own company.. and company to own bank acount
05:29.21PTG123the entire purpose of a corp is to seperate all personal liability
05:29.33TripleFFFnow argue all you want but im 150% sure i walked the walk man
05:29.37PTG123a trust is basically like a company, with more limitations
05:29.46TripleFFFWE ARE NOT TALKING liability
05:29.47PTG123your not even a us citizen, so how could you know? :)
05:29.50TripleFFFbut who controls the corp
05:30.03TripleFFFcoz our shit head martin passed same bill
05:30.05TripleFFFas USA
05:30.16TripleFFFEXEPT bahamas..coz all his carogo comapnies are based there
05:30.22AgiNamuI'd think the forms for Canada to pass a bill would be somewhat different
05:30.32TripleFFFso only tax heaven for CANADIAN is no bahamas till he retires
05:30.35PTG123you don't understand.. the corp is resposible for taxes only, and since its not a us corpo, its responsible to its jurisdiction, which in this case would be BVI, which has no corp taxes
05:30.36TripleFFFin 2 years
05:30.56PTG123now its different if you spend the money personally
05:31.04TripleFFFyes
05:31.12TripleFFFwell then if you make yourself a 20k wire u need to declare
05:31.15PTG123but i don't want to spend the money, i want a company to hold and invest it
05:31.20PTG123TripleFFF: yes that is correct
05:31.25PTG123TripleFFF: and i have no problem with that..
05:32.16TripleFFFwell here we dont have to
05:32.17TripleFFFlol
05:32.31PTG123hah
05:32.33PTG123well thts ok
05:32.40PTG123b/c the corporation could own my car and my house :)
05:32.52PTG123and i could just pay myself like $50k a year
05:32.53TripleFFFhttp://www.irs.gov/newsroom/article/0%2C%2Cid=110092%2C00.html
05:32.54PTG123salary
05:33.37TripleFFFUnder the Bank Secrecy Act, U.S. residents or a person in and doing business in the United States must file a report with the U.S. Treasury if he or she has a financial account in a foreign country with a value exceeding $10,000 at any time during the calendar year.
05:33.52TripleFFFaccount
05:33.53TripleFFFnot income
05:34.00TripleFFFsource http://www.irs.gov/newsroom/article/0,,id=108790,00.html
05:34.35*** join/#asterisk zwi (~zwi@216.88.131.43)
05:35.23*** join/#asterisk Grooby (~Grooby@ip24-250-126-171.dc.dc.cox.net)
05:35.33*** join/#asterisk CpuID_ (~nathan@dsl-202-173-176-86.qld.westnet.com.au)
05:35.48TripleFFFow my god
05:35.49TripleFFFhttp://www.solami.com/IRSaccess.htm
05:36.00TripleFFFJohn Doe Summons for Identities of Offshore Credit Card Holders
05:36.19TripleFFFThe purpose is to determine the identities of taxpayers who have credit, charge, or debit cards issued by offshore entities
05:36.21TripleFFFUS sucks
05:36.41PTG123hah
05:36.50PTG123wel people doing that are just stupid :)
05:37.09TripleFFFOnce the offshore structure is created, the next step is to transfer funds or assets to the IBC, which may represent profits the U.S. person is trying to hide from the IRS. The last step is to devise a technique to access the funds when desired, either by repatriation or by use abroad.
05:37.38PTG123you can just fly there with all your family
05:37.43PTG123and strrap 10k cash to each person
05:37.43PTG123haha
05:38.02TripleFFFno 9999 not 10k
05:38.09*** join/#asterisk E|nste|N- (E_nste_N@S0106000bdb97681e.wp.shawcable.net)
05:38.10Groobyanyone here can help me out a bit?
05:38.15TripleFFFthat 1$ coin in pocket can put you in jail
05:38.19TripleFFFso empty your shooes
05:38.24TripleFFFmake it 9k to be sure
05:38.30PTG123haha
05:38.31TripleFFFgold filings can be treaterous
05:38.32TripleFFFhttp://www.solami.com/IRSaccess.htm
05:38.35E|nste|N-hi
05:38.43TripleFFFnice articvele
05:38.46GroobyI follow the FWD IAX instruction and somehow when I call my number, there's no extension to dial, it goes directly to the VM I setup
05:39.12Groobyhow can I make it so I have to dial 1000 (the extension I setup) to get to my vm?
05:39.23TripleFFFid need cash to put there first
05:39.24TripleFFFbut hey
05:39.26PTG123add extension 1000, and call the voicemail command
05:39.32TripleFFFthnk i got somethign figured out now
05:39.33TripleFFFso
05:39.47PTG123L - O - A - N - S
05:39.51PTG123when you gonna learn? :)
05:39.59TripleFFFnope
05:40.04TripleFFFbeen hearing that for long time
05:40.09TripleFFFbetter shit to do
05:40.15Funbagsis there a way to put music on while its dialing?
05:40.17PTG123you have never done it
05:40.24PTG123Funbags: heh thats kind of funny
05:40.29TripleFFFyep well
05:40.39TripleFFFmy rent not paid so im leaving the scene
05:40.48TripleFFFtoo k that decision tonight
05:41.05E|nyPRII wrote a utility that trims leading + trailing silence + inter-word silence to compress audio files, if anyone wants to try it, or listen to the sample
05:41.07TripleFFFi cant wait so ill try something else
05:41.18TripleFFFk need to go to bed
05:41.23TripleFFFseeya guys
05:41.28PTG123damn its early :)
05:41.30PTG123for bed
05:41.35Funbagswell i mean music on while is dialing a ext after u have * awnser, then music while its ringing extentiosn
05:41.43E|nyPRIhttp://les.net/asterisk/
05:42.17humbertoguys looking for a web-call-back
05:43.04niZoni've been talking to the guy who owns les.net
05:43.10E|nyPRIthats me.
05:43.16niZonoh lol
05:43.17E|nyPRI:)
05:43.19Nuggetheh
05:43.35E|nyPRIi couldnt get sox to trim leading + trailing space, so i wrote something in free pascal
05:43.39E|nyPRIand it worked AWESOME
05:43.40E|nyPRIi think.
05:43.46E|nyPRI58 lines
05:43.51E|nyPRIo'code
05:45.04E|nyPRIany comments?
05:45.29*** join/#asterisk burton27_ (mimx@w201.ljudmila.org)
05:46.00Nugget"Linux is Poo"
05:46.12E|nyPRI:P
05:46.22*** join/#asterisk Othello (Othello@nusnet-154-210.dynip.nus.edu.sg)
05:46.46modulus_pinux is loo
05:48.41AgiNamules: how does we know this ain't a trick
05:49.01E|nyPRIuh
05:49.04AgiNamuThe code
05:49.04E|nyPRIwhat the executable?
05:49.06AgiNamuyea
05:49.10AgiNamu:)
05:49.12E|nyPRIoh, i guess ya don't
05:49.17E|nyPRIi might just post the source
05:49.20E|nyPRIso ppl can convert to C
05:49.28AgiNamuit's pretty cool
05:49.28TripleFFFpascal sucks /
05:49.34E|nyPRIits PAINFULLY simple (works on signed linear audio)
05:49.40E|nyPRIthe program is  ./trim source.slin dest.slin
05:49.45*** part/#asterisk TripleFFF (kvirc@Toronto-HSE-ppp3880909.sympatico.ca)
05:49.49*** part/#asterisk ilan (ilan@69.60.110.251)
05:50.06AgiNamuit'd be cool to have a realtime version
05:50.08AgiNamu:)
05:50.21W1thdrawhow do i get asterisk to manage QoS
05:50.30AgiNamuw1thdraw: tos=
05:50.39AgiNamubut what do you mean by manage?
05:51.05E|nyPRIrealtime would be cool, yes.
05:51.17E|nyPRIit only takes a split second to convert that file
05:51.19W1thdrawlike give voip priority
05:51.19AgiNamu... that was a joke :)
05:51.33AgiNamuw1thdraw, well, if you set tos=, then it'll flag the packets
05:51.38AgiNamuBut that's up to the routers
05:51.49E|nyPRI0m0.011s to convert 350k of slinear audio
05:52.01W1thdrawcould i use my asterisk box as a gateway
05:52.08W1thdrawso all my traffic goes thru it
05:52.26AgiNamue|nypri.. that wasn't the point :D my point was it's sorta hard to eliminate silence from a conversation if someone isn't talking.
05:52.31AgiNamuunless you went into the future.
05:52.35*** join/#asterisk denon (denon@synapse.subneural.net)
05:52.35*** mode/#asterisk [+o denon] by ChanServ
05:52.35AgiNamufaster than usual.
05:52.40AgiNamuw1thdraw, sure you could
05:52.49E|nyPRIyah.. #include <hyperdimensionalphysics.h>
05:53.00AgiNamuand it'd be up to your routing software to do it
05:53.01BrixiusWithdraw, there's the qos stuff in iptables I beleive, so yes if you make your * box your gateway it could be done, or just take an old system put linux on it and us it as a gateway.
05:53.01*** join/#asterisk drumkilla (~russell@12.21.241.80)
05:53.01*** mode/#asterisk [+o drumkilla] by ChanServ
05:53.12AgiNamuthe new linux kernels will do this automatically AFAIK
05:53.24W1thdrawi need to save up for a dedicated asterisk box
05:53.26*** part/#asterisk illek (~mike@ip68-13-238-168.ok.ok.cox.net)
05:53.36AgiNamubut that only helps outgoing i believe
05:53.47AgiNamuyou need traffic shaping
05:53.51AgiNamuwhich can also be done with linux
05:54.14W1thdrawi think my cisco switch can manage QoS
05:55.00*** join/#asterisk af_ (~af@ip-148-227.sn1.eutelia.it)
05:55.12AgiNamuanyone know what the bar under the user list in xchat means?
05:55.31BrixiusNight all
05:55.44postelAgiNamu: lag meter
05:56.06postelAgiNamu: you're WAYYYYYY OT btw
05:56.14AgiNamuOT?
05:56.22postelOff Topic
05:56.52AgiNamuAsterisk: The Open Source PBX || DUNDi: http://www.dundi.com #dundi || mpg123 0.59r ONLY FOLKS, no apt-get, no portage, no ports.. or DO NOT USE IT AT ALL moh_files+format_mp3 baby! | We're now -r, if you notice any spambots, please report them to a channel operator. Thanks!
05:56.58W1thdrawwhich is more important in an asterisk box ram or cpu?
05:57.23postelW1thdraw: cpu is only needed if you do transcoding
05:57.32AgiNamuyea, depends on what you're doing
05:57.53AgiNamuif you're servicing an E1 with G729 clients, you're gonna need some CPU power
05:58.20AgiNamuthe wiki has a few hardware example scenarios
05:58.33W1thdrawwill a 1.8 be suffice
05:58.40mishehuugh.  I screwed up my cisco somehow
05:58.46AgiNamuI ran it on a 300Mhz celeron
05:59.09*** part/#asterisk twisted (~twisted@twisted.active.supporter.pdpc)
05:59.10*** join/#asterisk twisted (~twisted@twisted.active.supporter.pdpc)
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05:59.21AgiNamuwhat are you doing with it w1thdraw?
05:59.55W1thdrawim just setting up a low end simple asterisk box
06:00.01E|nyPRIhow many lines
06:00.11CpuID_hehe AgiNamu
06:00.16W1thdrawa few at the most
06:00.22E|nyPRI1.8 is more than enuf
06:00.24E|nyPRIoverkill
06:00.27AgiNamuthen i doubt you'll need to worry
06:00.33CpuID_hmm, i wonder, does asterisk take advantage of multiple cpus in the case of transcoding etc?
06:00.36AgiNamuyou could doo a few with a PII
06:00.38AgiNamuCPu yes
06:00.44jjgcan anyone recommend a "decent" softphone for linux?
06:00.45AgiNamuthere's a lot of threads, so it'll scale well
06:00.52CpuID_btw AgiNamu, how many calls have you pushed through an asterisk box at most? :)
06:00.53AgiNamuesp. since we're just gonna see more cores for a while
06:00.57CpuID_ya
06:01.03CpuID_im thinking like a dual xeon or something
06:01.05AgiNamuum not many :)
06:01.08CpuID_or maybe even a dual opteron :)
06:01.24AgiNamuDigium says dual 1.8GHz Xeons did 60 g729 channels
06:01.25CpuID_just got someone interested in routing a few thousand calls for some residential users through asterisk atm
06:01.28CpuID_oh yea
06:01.31CpuID_looking at specs
06:01.34CpuID_im thinking multiple boxes atm
06:01.37E|nyPRIi've done two G729's on a Pentium Pro 200 with no problems
06:01.46CpuID_wish there was something like SER for IAX tho :)
06:01.52E|nyPRIPRI dont work on PPro tho.  that's for sure :)
06:02.01EricirECpuID_: on a dual 2.4G 729 ive done 100 calls and the box was under serious load but it held up
06:02.13CpuID_oh fair enough
06:02.19CpuID_the load was solely asterisk?
06:02.31AgiNamuand more importantly, extra cores should work well too
06:02.31EricirECpuID_: i also have a hardware call generator that maxes at 207 calls and that puts 0 load on my box at 711
06:02.31W1thdrawhas anyone setup asterisk and smoothwall on the same box?
06:02.38EricirECpuID_: yup
06:02.44AgiNamuso when AMD/Intel release their 3.x Ghz proc, with 4 cores
06:02.50AgiNamuit should handle that...
06:03.16AgiNamuthen again, i dont know how the different codecs use the cpu
06:03.17CpuID_mmm nice as hehe
06:03.18EricirEi'd like to get my hands on a quad opteron and beat on it
06:03.30AgiNamuis AMD doing multicore CPUs yet?
06:03.40CpuID_mmm quad hehe
06:03.41*** join/#asterisk Free (Baraxunkno@osf.su.lt)
06:03.41E|nyPRIanyone used TDMoE ?
06:03.46EricirEnotthat i'm aware of
06:03.50AgiNamumeh
06:04.00modulus_feh
06:04.04EricirEfeh ++
06:04.08CpuID_hmm i think a dual opteron would be some fun even :)
06:04.22EricirEtrue i don't know if there are any optimizations for 64 bit tho
06:04.27AgiNamuBut for dual intels , i think you need the 2.6 kernel
06:04.30AgiNamufor it to work well
06:04.32CpuID_prolly
06:04.32mishehuI'm having fun with this dual xeon 2.8...
06:04.40mishehuspeex encoding is pretty fast on it
06:04.46AgiNamusince the 2.4 kernel is not multi-core aware, and will screw up scheduling.
06:04.58mishehutoo bad speex has problems when the call originates as sip g711ulaw...
06:05.15AgiNamutoo bad not all hardphones support speex :)
06:05.28EricirEive also used hp's sipp to push 1400 calls through a bod
06:05.31EricirEs/bod/box
06:05.34AgiNamuor more specifically,no IAX2 hardphones I know of do :)
06:05.35EricirEsingle p3 800
06:05.40EricirEbut there was no rtp
06:05.42EricirEso
06:05.45mishehuAgiNamu: definitely a shame, otherwise speex works pretty well as long as iax originates it
06:05.54EricirEnot worth much
06:06.06heath__when i execute my agi script from an extension, will it go back to the next process in the extension after the agi has executed successfully?
06:06.13AgiNamuheath yea
06:06.14postelAgiNamu: show translations to see the matrix and see how much of your cpu the codecs use
06:06.32heath__for sure?
06:06.36heath__:)
06:06.36*** join/#asterisk clive- (~pirch@myw-stp-66-18-83-88.sentechsa.net)
06:06.41*** join/#asterisk W1thdraw (~W1thdraw@ip68-5-125-44.oc.oc.cox.net)
06:06.47AgiNamuyuea, unless you make it go to another one
06:06.53AgiNamubut i dont use agi so
06:06.59clive-anyone here from India?
06:07.45AgiNamutranslation times, but for how much data?
06:07.59AgiNamuat any rate, that table doesnt mean anything when it comes to multicores
06:09.01CpuID_hmm id love to see asterisk on a g5 :)
06:09.45bkw_you can
06:09.47AgiNamuI'd love to see Asterisk on a Onyx3000
06:09.48bkw_already do that
06:09.56CpuID_bkw_: on a g5? woot lol
06:10.04bkw_yes
06:10.06bkw_shit
06:10.08bkw_yellodog
06:10.12bkw_linux
06:11.08CpuID_lol sparc, mmm :)
06:11.41CpuID_what seems to be the best arch for performance these days?
06:11.43*** join/#asterisk terraMobile (~cjs@015-821-436.area5.spcsdns.net)
06:11.44AgiNamuwill * compile on IRIX?
06:11.45CpuID_from what you know of?
06:11.47E|nyPRICoCo III
06:12.18terraMobileSprintPCS <--> MacOSX aint too bad
06:12.18CpuID_hrm i wonder if zaptel could be ported to run on a g5 somehow ;)
06:12.24postelAgiNamu: a good question is.. Would you trust your PBX on something as crappy as IRIX?
06:12.26CpuID_not sure whether youd go macos or what tho
06:12.41terraMobilehah
06:12.47terraMobileIRIX was always solid for me
06:12.49terraMobilenot secure
06:12.52terraMobilebut pretty stable
06:12.57AgiNamupostel, better, if I had the money to spare to run * on an Origin 3000, would I care?
06:14.01postelAgiNamu: x86 arch is soooo damn cheap to deploy even on SMP boxes, there is no need trying to make your life hard for no good reason
06:14.28Moc[dCAP]im under 1k$ holly...
06:14.29AgiNamu... i wasn't seriously planning on buying an SGI system
06:14.30terraMobileRun NetBSD on your SGI
06:14.41Moc[dCAP]im actually -100$ :(
06:14.47postelAgiNamu: for a pet project its a whole different bunnie though
06:15.29terraMobileNetBSD is pretty solid on Indys etc
06:15.29terraMobiledunno if * would run on it
06:16.14AgiNamuthe last time I was seriously considering SGI I think was when we were starting a video/effects company
06:16.17AgiNamuabout 6 years ago
06:16.53*** join/#asterisk |^Angel^| (~funschrip@port336.ds1-alb.adsl.cybercity.dk)
06:16.55AgiNamubut we decided to go with NT, which in the long run turned out to be pretty correct. esp for us, since the company went no where.
06:17.08postelbtw an O2 (with x86 hardware inside) ;)  would make a cute * box
06:17.18terraMobileanybody ever played with a WRAP board
06:17.26AgiNamuor a MiniMAC
06:17.37postelAgiNamu: yeah, those are cute too
06:17.43terraMobile(Mac mini)++
06:17.45terraMobileso sweet
06:18.53Nuggetindeed.
06:18.56AgiNamuwell, night all.
06:19.06Nuggetthe ebay market for used g4 cubes was destroyed today.
06:19.11terraMobileheh
06:19.33terraMobilei never expected that thing to be so small
06:19.42terraMobilei pictured a smaller version of the iMac
06:20.16AgiNamuand it has Mac OS X, the worlds most advanced operating system..
06:20.41terraMobilei really want one
06:20.47terraMobilebut here's what i would need to find
06:20.56terraMobilea very small 7" LCD monitor
06:20.56NuggetI don't really have a use for one.  but I'm buying one for my mom.
06:21.03terraMobilethat fits around or on top of one, neatly
06:21.12AgiNamuWell, can you setup the graphics to forward over the network?
06:21.16terraMobilei put a Mac in the colo with a  little 7" B&W CRT
06:21.22terraMobileand it looks awesome
06:21.56heath__i have an extension the 1. executes an agi, 2. transfers the call to an extension based on what the agi has set a variable to BUT...
06:22.19heath__if i uncomment stuff that has nothing to do with the agi, it works, but it won't with the stuff in there
06:22.33heath__it's freakin' bizarre
06:22.50heath__could it have to do with the speed my agi is responding or something?
06:22.56AgiNamuheath__ try with a dummy AGI
06:23.01AgiNamuto make sure it's your agi
06:23.39postelterraMobile: http://www.edirectory.co.uk/pf/pages/moreinfoa.asp?pe=BFFBBBGQ_+7+Inch+VGA+Monitor+with+Touchscreen&cid=880
06:23.52postelsowwy for the huge link, i should have tinyurl-ed it
06:24.18terraMobileplease do :)
06:24.19postelbut its 06:23 in the mowrning, vant be asked..
06:24.25terraMobileheh ok
06:24.33postels/vant/cant
06:24.38postelsee, i cant even spell..
06:24.41terraMobilefor the love of god, i'm coming to you over a cell phone
06:25.15AgiNamudo macs use those wierd mac-only keyboard and mouse connectors still?
06:25.33terraMobileno
06:25.34AgiNamulast time i touched one was 8 years ago
06:25.37terraMobileUSB
06:25.39terraMobileApple practically (or did?) invented USB
06:25.54ta[i]ntedi thought that was firewire
06:26.04terraMobileApple does not use those wierd PC-only mouse and keyboard connectors
06:26.09terraMobileit uses an industry standard. :)
06:26.18terraMobilethey invented firewire, too
06:26.23AgiNamuI like PS/2
06:26.24Himekolike ADB ports
06:26.29AgiNamuADB yea that's it
06:26.29PTG123apple invented firewire
06:26.41Himekoand their special video connectors
06:26.42PTG123and the reason its not more popular then usb is b/c they charged $2/device
06:26.46PTG123when usb charge $.35
06:27.01AgiNamuoh really
06:27.12AgiNamuUSB2 is the same price?
06:27.13PTG123yah usb is some consortium i think
06:27.19PTG123yah i think it is but dont' quote me on that
06:27.23terraMobileso many damned PCs will not let you plug in the PS/2 after bootup and have it work
06:27.23terraMobilethey quit using ADC video connectors
06:27.24PTG123but firewire2 is tons better
06:27.24terraMobileswitched to DVI
06:27.33AgiNamuterraMobile, PCs , or the OS?
06:27.34*** join/#asterisk shaZwaz (~lukyali@203.81.196.167)
06:27.37denonIntel had USB before apple.
06:27.46AgiNamuI've never seen a desktop PC not accept a keyboard after boot
06:27.46PTG123apple never had usb
06:27.53PTG123oh you m ean on the pc
06:28.08PTG123as long as you booth with ps2 device in it will work
06:28.17AgiNamueven without it
06:28.19PTG123its the drivers not the mb thats the problem
06:28.24*** join/#asterisk iMediax (~user@00045a7b37e7.click-network.com)
06:28.28AgiNamuI do it all the time
06:28.29E|nyPRIKeyboard error or keyboard not present, press F1
06:28.33AgiNamumy linux machine is headless
06:28.42AgiNamuand keyboard and mouseless
06:28.48AgiNamuand I switch over things and it works fine
06:28.48shaZwazhi all
06:28.55PTG123thats why you are suppose to use serial consoles on unix machines
06:28.58PTG123:)
06:29.02clive-is anyone here from India?
06:29.16AgiNamuolder Windows (maybe current) would probe for a PS/2 mouse, not find it, so revert to the serial driver.
06:29.19terraMobilei'm not, but i ate chicken tikka masala tonight.  does that count?
06:29.23AgiNamuand then it wouldnt matter.
06:29.42postelclive-: if you keep asking i might as well go to Andaman islands to make you STFU
06:29.42PTG123i watches a movie tonight where they made fun of indians does that count?
06:29.43AgiNamuI just dont like using up 2 usb port for my kb and mouse
06:30.04PTG123agi: just use a hub
06:30.05AgiNamuI interviewed at MS with someone who I think was from India.
06:30.11AgiNamuPTG, yea, more junk
06:30.18clive-postel, whats STFU?
06:30.18AgiNamui've got 2 PS/2 ports that work just fine
06:30.30terraMobileSouth Texas Forensic University
06:30.36postelclive-: google is your friend
06:30.39clive-lol
06:30.42postelterraMobile: hahaha
06:30.46AgiNamuSTir Fried Unons.
06:30.59AgiNamutexas has forensics?
06:31.05clive-aSTerisk For U
06:31.33AgiNamuthought it was more like "they done gone shot him... I'll go see what up and happened. Oh, I see. He had a killing coming."
06:31.36ta[i]ntedanyone know if gafachi's down?
06:31.45terraMobilei'm from texas
06:31.52ta[i]ntedi think i may of offended them yesterday when i said their service sucks
06:32.00PTG123i just put a usb2 4 port card in my pc
06:32.04PTG123so i have plenty of roomk for usb
06:32.05PTG123:)
06:32.23AgiNamuwell cya all
06:32.51heath__I fixed it! I rule!!!!!
06:32.57terraMobileOH FUCKING CHEESY
06:32.57terraMobileSprintPCS forces me through a proxy
06:32.58terraMobileand it reduces the JPEG quality
06:32.58terraMobileway down
06:32.58terraMobileso my site looks like ass
06:33.18terraMobilebut at least its fast
06:33.21terraMobilei guess that is the trade-off
06:33.21*** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk)
06:33.22heath__have to be careful not to let anything go to std ouput!
06:33.59W1thdrawanyone have any negative input on the spa-2000
06:34.01terraMobilei wndr if thy r comprssng my irc trfc on the fly
06:35.22terraMobilewell, i guess i best be going.  i have to walk five and a half miles back home
06:41.02firestrmanyone here good at iax->fwd connections? www.pastebin.ca/4085
06:41.29firestrmit doesnt register, it doesnt complain. it gives no clue, just a busy signal when i try to dial i
06:41.30firestrmn
06:44.11*** join/#asterisk burton27_ (mimx@w201.ljudmila.org)
06:44.12ta[i]ntedanyone here good with AGI?
06:44.51W1thdrawanyone know where i can buy a cheap pos computer?
06:44.56*** join/#asterisk erncic (~eankele@c-24-8-130-205.client.comcast.net)
06:45.12W1thdrawnewegg doesnt carry cpus below 1.8 ghz
06:46.09*** join/#asterisk pointer-gaim (~pointer@router.cathey.us)
06:46.13firestrmW1thdraw> i can get 1ghz- 500 mhz comps cheap...
06:46.25W1thdrawwhere?
06:47.44firestrmW1thdraw, i have a connection to a leasing company.. they are leased comps that have come back.. eg dell dual 1ghz xeon 18gig 10k ultra scsi 512meg ram. 600.00 cad
06:48.25firestrmW1thdraw, usually high end gear, only just that it is 3 years old
06:48.35W1thdrawi think i know of a place to get cheap used pcs, its a big warehouse i pass when im on the freeway
06:49.03firestrmprobbly your best bet..
06:49.28W1thdrawthey have huge banners that say shitlikr "50$ monitors" and "100$ laptops"
06:49.33W1thdraw*shit like
06:49.49firestrmwhen the dell's came up, i bought the entire pallet, sold most of em on ebay for 1200.00, and kept 2
06:50.13firestrmthey make smoken asterisk servers
06:50.20W1thdrawi just need something to run a low end asterisk box
06:50.28*** join/#asterisk WilliamK (~wkeller@c-24-0-130-60.client.comcast.net)
06:50.30heath__will someone check if this page loads? thanks in advance http://www.jailcity.com/
06:51.07bkw_cvs-head is fucked
06:51.29bkw_I don't care what anyone thinks its FUCKED
06:52.04pointer-gaimbkw_: heh
06:52.09pointer-gaimbkw_: how so this time?
06:52.20bkw_ever since makr put in that fucking multi bindaddr shit
06:52.24bkw_its been fucked
06:52.58pointer-gaimI tried it last week sometime and it didn't like me..
06:53.04pointer-gaimI'm back on 1.0.3
06:53.10firestrmcan anyone help with an iax connection problem?
06:56.30W1thdrawi think this is the place i always pass on the freeway
06:57.14W1thdrawhttp://www.123compute.com
06:59.18firestrmwell im going to give it up for tonight... some sucess, but no closer to victory..
07:00.08*** part/#asterisk firestrm (F1r3570rm@S010600047577bccd.gv.shawcable.net)
07:00.08*** join/#asterisk alk (~tony@c66.191.69.132.dul.mn.charter.com)
07:00.09W1thdrawkewl i think im gonna buy 2 of these
07:00.10W1thdrawhttp://www.123compute.com/showsp2.php?ide=9011
07:00.16W1thdrawsmoothwall and asterisk
07:00.49W1thdraw170$
07:01.08W1thdrawi wonder if i could do that windows refund thing
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07:02.51scubasteveJan 12 02:01:53 NOTICE[8874]: chan_iax2.c:4322 register_verify: Peer 'demo' is not dynamic (from 24.88.251.252)
07:02.57scubasteveThat looks new... CVS HEAD thing?
07:04.11*** join/#asterisk channan (~channan9@66.180.121.189)
07:04.13iMediaxwow! from proxy.dca.broadvoice.com ->  64 bytes from 147.135.0.128: icmp_seq=1 ttl=49 time=7.14 ms
07:04.46scubasteveiMediax .. Damn.
07:05.04iMediaxmust be in the same damn computer center
07:05.13scubasteveIt's in the 30's for me.
07:05.26channanhi funbags... it's me again... I got it working now. it's sip.broadvoice.com alright... thanks a lot
07:05.30zigman140 for me
07:05.34iMediaxya from my ISP it is...thats from my rack server
07:05.59scubasteveiMediax - Where are you colo-ing?
07:06.05iMediax1and1
07:06.10scubastevekewl
07:06.23scubastevewhatcha doing in there?  offering iax or sip services?
07:06.59iMediaxwell i started it mainly for merely my own email server...but now I host a dozen local web sites
07:07.05scubastevenice.
07:07.48channanhi iMediax... are you using broadvoice.com?  I've just signedup today and got it working finally... It seems good
07:07.50*** join/#asterisk zarnivoop (~stmo@gw.rixtele.com)
07:08.02iMediaxchannan: yep
07:08.23channanI have not tried international calls yet so don't know about quality
07:08.33iMediaxi've had a few bad jitter issues tho...
07:08.49channanthat's kind of expecting, I guess
07:08.50W1thdrawhas anyone installed asterisk on a xbox?
07:09.16scubasteveW1thdraw .. I thought about it, but am gonna do it on that whacky mac they just announced instead.
07:09.24iMediaxwell it was bad... now i connect to my server, which is connect to BV.... hardly any jitter now
07:10.02W1thdrawscubasteve, is that the small cube one?
07:10.27channaniMediax- that's great
07:10.32scubastevew1thdraw yes... they announced it today.. pics on the site.. looks like about the size of a PC cd/dvd drive..
07:10.49scubasteveLooks like a laptop internals in a small box without a keyboard/mouse/screen...
07:11.11W1thdrawyeah i heard them talking about it
07:11.39channaniMediax- I've tried to call VA and CA but somehow did not go thru. Keep hearing dialing 10 digits again with a Mexican voice
07:11.43W1thdrawi think that ipod shuffle will do well
07:12.01W1thdrawhttp://www.apple.com/macmini/
07:12.04scubasteveYep.. That little $500 pc is going to put them on the map.. big time.
07:12.08W1thdrawthat thing looks cool
07:12.14W1thdrawcan u run asterisk on it?
07:12.20scubastevePeople tell me all the time they're sick of Windows, and I give them Knoppix CD's to play with Linux.
07:12.42scubasteveInstead, I'll tell them to pack their PC up and put it in a closet.  Keep the mouse/keyboard/monitor and get that Apple thing.
07:13.02scubasteveI'm looking forward to my first Apple purchase on the 22nd :-)
07:13.06scubasteveFor $500, how do you say no?
07:13.07W1thdrawno way i will always love my pc
07:13.22scubasteveBesides, I can take this workstation (p4 2ghz) and make it an Asterisk box.
07:13.38*** join/#asterisk jaybonci (~jaybonci@giantsquid.net)
07:13.58scubasteveI think they announced some office type software today too.  Supposedly comes on the $500 mac.
07:14.41W1thdrawblah its not that great ill take a mini atx pc anyday over that minimac
07:15.37W1thdrawi dont like the os its too fruity for me
07:15.45scubasteveLOL
07:15.58scubasteveI think the wife would love it.
07:16.12scubasteveShe's been tolerating Linux on her workstation because I refused to play Windows admin at home.
07:16.37W1thdrawwindows isnt that bad
07:16.49scubasteveI gave up 3 years ago.
07:17.20scubasteveIt's bloated and insecure.
07:17.43*** join/#asterisk Thumann (Bush@is.a.retard.dk)
07:17.51scubasteveYou gotta have 256m and a 1g cpu to do much of anything with XP.  Screw that.
07:17.57W1thdrawi bet macs insecurities will catch up to windows soon
07:18.06W1thdrawjust cuz more ppl are gonna start using it
07:18.13scubasteveI think M$ and Intel are in cahoots to slow down the OS to keep people buying faster cpus
07:18.30W1thdrawhow so?
07:18.48scubasteveThings on windows with a 2g p4 aren't any faster than I remember things on a 486/66 running 3.1 or 3.11.  Why is that?
07:19.53W1thdrawwhat programs are u using?
07:20.04scubasteveMe?  I don't use windows.
07:20.22W1thdrawi found that windowsxp is much slower than windows 2003
07:20.23channanWindows is like a air conditioner. The day I opene it is the day it stops working. hehehe
07:20.45W1thdrawwindows has a higher learning curve
07:21.00scubasteveI think Apple's gonna do great sales with that new box... There are a LOT of really anti-m$ consumers out there.  Now they have a real choice...
07:21.16scubasteve(vs playing with linux, which isn't gonna happen at the consumer level)
07:21.38drrayI bought a mac mini today
07:22.00W1thdrawonce more ppl start using macs they are gonna see that no matter what os they use they are still stupid
07:22.02scubastevedrray didn't that get announced today??
07:22.05drrayyes
07:22.13scubastevethey taking pre-orders??
07:22.23drraythey say it will be here jan 22nd
07:22.26drray:)
07:22.33scubastevedrray did you pre-order it?
07:22.41drrayyes, from store.apple.com
07:23.01scubastevedrray if you can pre-order, I'm gonna be pissed.. I just blew a bunch of $ on a pile of new 7960's for home... could have pre-ordered instead :)
07:23.12scubasteveWife makes me buy these toys with consulting $
07:23.15W1thdrawthe only things that apple makes that i like is the ipod and the 30 inch lcd screens
07:23.39drrayI had to wrestle with the webpage for 45 minutes to get it order
07:23.43scubasteveheh
07:23.43drrayer, ordered
07:23.53scubastevemaybe I can save a bunch of money on car insurance by switching to geico
07:23.58scubasteveand then pay for a mini with the savings.
07:24.07postelW1thdraw: ok, no dual G5 for your birthday then, noted
07:24.15scubastevehaha postel
07:24.22W1thdrawscubasteve, can i get a quote online?!?!?!
07:24.27scubasteveheh
07:25.04drrayI figure it's a zero risk thing for me, if the store sells out and or there are shortages I'll sell mine..
07:25.06W1thdrawpostel, ok sign me up for a dual opteron
07:25.47PTG123i  gave away my mac
07:25.48PTG123:)
07:26.09W1thdrawblah i gota wake up early tomorrow
07:26.11W1thdrawlate all
07:26.36drrayer, had
07:26.53PTG123i had a mac powerbook ti, i just don't think osx response as well as freebsd/linux/windows
07:26.55PTG123seems slow to me
07:27.06drrayI really can't wait for it to get here.  I was 90% sold on getting one if it was not fuugly..
07:27.17*** join/#asterisk tzafrir_ (foobar@82-166-204-169.barak.net.il)
07:27.38drraygentoo is on my todo list
07:27.58PTG123i tried to install gentoo, big mistake :)
07:28.10PTG123after about 12 years of failures, searching the internet for where i can find the package, etc
07:28.14PTG123i finally get it installed
07:28.15postelshit, 07:27. time to cruise down to the SunPark, later ppl, anybody from London going to Surrey?
07:28.17PTG123install boot maneger
07:28.18PTG123reboot
07:28.20PTG123and it doesn't boot
07:28.24*** join/#asterisk tzafrir__ (foobar@82-166-204-169.barak.net.il)
07:29.36posteli guess not..
07:33.44drrayI notice a lot of people are dumping their mac stuff on the local seattle forsale boards to raise cash
07:35.57ta[i]ntedhey postel
07:36.03ta[i]ntedhow do i make my asterisk box call me?
07:36.52jstormta[i]nted: buy it flowers, and try not to go past first base on the first date, Asterisk isn't easy.
07:37.02modulus_that was too easy
07:37.07ta[i]ntedjstorm: dude that's the second time you cracked that joke
07:37.27PTG123classics never die
07:37.45modulus_penguins are forever but daemons never die
07:37.55drrayit took me a month to score with asterisk
07:38.02jstormta[i]nted: nooo, think you got me confused with someone else here
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07:39.24*** part/#asterisk jcims (~rrich@cpe-69-135-121-57.columbus.rr.com)
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07:47.25tzafrirgood morning
07:49.05*** join/#asterisk Heinz (~none@201.137.189.237)
07:49.30ta[i]ntedjstorm: but do u know how to make asterisk call you
07:54.16tzafrirta[i]nted, you can use a call file with a local channel
07:54.27tzafrire.g: on latest (test) Rapid, you can use:
07:54.32tzafrirast-cmd -f 'Local/200@default' -t 503 call
07:54.51tzafrirto ask * to call you and run an echo-test
07:55.04tzafrir(extension 200 happens to be the echo test)
07:55.28tzafrirast-cmd call is a simple script to create a call file
07:59.52ta[i]ntedtzafrir: any idea how you would do that in AGI?
08:01.00jimblobhi all - anyone here using openh323 with asterisk?
08:01.07*** join/#asterisk iguy-duex (~iguy@dsl093-197-234.mke1.dsl.speakeasy.net)
08:02.59Nuggetthe general consensus is that openh323 is not legal.
08:02.59*** join/#asterisk tafazzi (~Dario@eporroj0-1.customer.nettuno.it)
08:03.09jimblobin what way?
08:03.52Nuggeter, nevermind.  I'm thinking of the g729 thing.
08:04.04Nuggetignore me.
08:04.11jimblobah ok :-)
08:04.58*** join/#asterisk lele (~lele@rivendell.windmill.it)
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08:16.04payalHi all
08:16.16payalwoww!! there are many people here
08:17.02payalI am reading something about asterisk, but I am not sure if I need any hardware for thhat other than a Linux box with a soundcard
08:17.11*** join/#asterisk evills (~ellvis@83.103.31.162)
08:17.15evillsgood morning
08:17.46payalhi
08:17.50*** join/#asterisk Verliba (~Miranda@250.1-14-84.ripe.coltfrance.com)
08:18.05payalany ideas?
08:18.12evillshow's it going this nice sunny summer sunday?:)
08:18.21modulus_you need a soundcard for asterisk?
08:18.49modulus_no wonder why my 7.1 surround doesn't work on my * box
08:19.02payaldon't I? I haveMandrake 10.0
08:19.28payalI have installed asteris, but not sure what is my next step
08:19.32mquinpayal: depends what you want to do, but it's not a minimum requirement.
08:19.43mquinmy * box doesn't have a soundcard
08:19.52payalI want to call from one office to other using asterisk
08:19.56payalis it possible?
08:19.59modulus_icmp v4 hw csum failure
08:20.03modulus_i hate redhat
08:20.20mquinpayal: yes
08:20.29mquinanything is possible :)
08:20.46payalmquin: what would be next step? Do I need hardware for that?
08:21.45mquinIt depends what you want to do, and what existing systems (if any) you are planning on interfacing with
08:22.27payali have a Mandrake box with a sound card, that 's all
08:23.15mquinif both locations are internet connected a simple setup would be two PC with soundcards and headsets, running a softphone such as Xlite
08:24.04payalwhat is Xlite?
08:24.13modulus_pos windows software
08:24.30mquinif you want to make calls from regular handsets you'll need to get analogue telephone adapters (ATAs) for them, or get IP phones such as the grandstream BT102
08:24.30payali want to try  asterisk
08:24.46*** join/#asterisk simong (~simong@h166n2fls35o884.telia.com)
08:24.46payalwowoww
08:24.51*** join/#asterisk Zeeek (~icechat5@80.125.87.100)
08:25.00payalmquin: what exactly are IP phones and are they costly
08:25.04mquinmodulus_: can you suggest a better softphone than xlite
08:25.10Zeeeksuddently no need to identify?
08:25.24ZeeekGOOD MORNING GOOD
08:25.24mquinpayal: take a look at voip-info.org
08:26.18payalmquin: ok
08:26.21modulus_mquin, linphone
08:26.30payalmquin: is asterisk a difficut to use software
08:27.36mquinpayal: I didn't think so - I've not run into any problems so far that google (or this channel) couldn't answer
08:28.32payaloh!
08:28.44payalbut I am no Linux expert
08:28.52payalhttp://payal.staticky.com
08:29.39mquinmodulus_: linphone's a nice idea in principal, but at the moment my target machines are XP and MacOS X systems
08:30.59konkyz0rkpayal: why do you hate isaac newton?
08:32.02payalkonkyz0rk: he was a VERY bad man . he killed many men cos' they did not agree to his views
08:32.12payalI want to do this (from wiki)
08:32.13payalConnecting employees working from home to the office PBX over broadband connections
08:32.33payaland Connecting offices in various states over VoIP, Internet or a private IP network
08:32.41modulus_mquin, macos x runs darwin kernel, it should compile linphone
08:32.45drrayasterisk will do that for you
08:32.46payaldo I need hardware for that?
08:32.54konkyz0rkpayal: start installing asterisk and take it from there
08:33.08payalI have already installed it few hours back
08:33.18konkyz0rkwhat seems to be the problem then
08:33.38payalnow I am LOST. I don't know what should I do
08:34.25payalrather what sgould I do next?
08:34.43scubastevepayal you'll need an ip phone or ip phone software
08:35.20payalwhere can I get an ip phone software? Is it the windows dialer on a windows machine
08:35.23scubasteveand then something to connect to with asterisk -- sip or iax to a provider or an X100p card and a phone line
08:35.32scubastevepayal you have windows?
08:35.36drrayor another asterisk server
08:35.56scubastevedrray kinda assumed he didn't have access to one..
08:35.58payali have a Mandrake 10.0 machine server and 4 windows machines to test
08:36.06payalit is she please
08:36.11scubasteveok get x-lite for the windows machine
08:36.51payalgreat I will d/l  it now
08:37.41payalis it free
08:37.53scubastevepayal yes.
08:38.12payaldownloading it now
08:38.30scubastevecool.  you'll need some configuration changes in asterisk
08:38.37scubastevenotably extensions.conf
08:38.45*** part/#asterisk newmember (~newmember@S010600a0c93dce87.cg.shawcable.net)
08:38.51scubasteveand something to connect to for asterisk to talk to
08:38.55payalwhat kinda?
08:39.04scubasteveanother asterisk box, an internet phone provider..
08:39.05channanhi... any Cisco 7960 guru? I had a little problem registering my phone with asterisk
08:39.08payalscubasteve: what is this something
08:39.08scubasteveto make a phone call
08:39.23scubastevepayal I can give you an IAX address to have your box connect to mine.
08:39.25channanI can call the other extension but they can't call me
08:39.30Nuggetchannan: do you really need a guru, or just some guy who knows the answer to the question you are about to ask?  :)
08:39.37payalwill it be long distance call?
08:39.41scubasteveNugget!!
08:39.45channanwell you'll be my guru then :)
08:39.46evillschannan: ehm, is there anyone who have not some problems with cisco at all?:)
08:39.53ta[i]ntedanyone having problems with gafachi?
08:39.57scubastevepayal It will be a phone call placed from one * box (yours) to another (mine) over the internet
08:40.00payalwhat is cisco :)
08:40.04NuggetI have a 7960 and it is registering to an asterisk server.
08:40.06payalget iptables
08:40.11Nuggetso maybe I know the answer.
08:40.36evillsNugget: i have the same problem, just with ata 186
08:40.41payalscubasteve: will it work over nat
08:40.53channansuper... I manage to convert to sip and can dial the other extension
08:41.19channanwhen I do "sip show peers" I got:
08:41.20channan3000/3000        (Unspecified)    D   N      255.255.255.255  0        Unmonitored
08:41.35Nuggetthat doesn't look like the phone is successfully connecting to asterisk.
08:41.46payalwhat changes I need to make to extension.conf
08:41.51channanwell, I can dial other extensions and talk
08:41.55Nuggetoh, strange.
08:42.02channanbut they can't call me
08:42.14Nuggetok, what's in the dialplan for that phone's extension?
08:42.23scubastevepayal I think it will.
08:42.38Nuggetit is odd that your host shows as Unspecified.  I'd expect to see the phone's IP address there.
08:42.49Nuggetnugget/nugget    10.0.1.184       D   N      255.255.255.255  5060     OK (72 ms)
08:42.54Nuggetthat is my phone.
08:42.55channanright... only see IP if I registered
08:43.03channanexten => 3000,1,Dial(SIP/3000,20)
08:43.03channanexten => 3000,2,Voicemail(u${EXTEN})
08:43.03channanexten => 3000,3,Hangup()
08:43.03channanexten => 3000,102,Voicemail(b${EXTEN})
08:43.03channanexten => 3000,103,Hangup()
08:43.21Nuggetsince asterisk doesn't know how to reach the phone, I guess it's not sending the call.
08:44.44channanmy cisco phone and the asterisk is on different subnet
08:44.44Nuggetthat shouldn't matter.
08:44.44channanwonder if something's blocking it
08:44.47Nuggetwell, assuming that you can route between them
08:44.59channanwe're connected thru a VPN tunnel
08:45.01Nuggetis there NAT or some ugliness like that?
08:45.20channanno NAT
08:45.29Nuggetyou could turn on qualify for the phone, that can sometimes help to maintain a connection through weird networks
08:45.38Nuggetset "qualify=500" or something in the sip.conf for the phone
08:46.42channanlet me try that
08:47.23*** join/#asterisk tech_voip (~tech_voip@202.65.128.18)
08:47.46Nuggetif it's a timeout issue, I'd expect you to be able to call the phone immediately after it has registered but then not after some time period.
08:48.04channannope.. never worked
08:48.24NuggetI'm not sure, then.
08:48.34Nuggetprobably something in the vpn that's thwarting the traffic
08:49.00evillsNugget: well, i have a bit similar problem. just that when i use Voismart IP Ranger phone, it work just fine. when i replace it with cisco, i can call from cisco but not to cisco. sip show peers show me correct registered peer
08:49.20payalscubasteve: I have downloaded and installed x-lite, now what
08:49.23NuggetI dunno.  the cisco doesn't require anything special.
08:49.49Nuggetany errors or warnings in the console?  what does asterisk say about the call?
08:49.57*** join/#asterisk WorkTooMuch (~work@82.148.188.1)
08:50.22|Vulture|Anyone know if there is a way to set a certian ring tone by * for Polycom phones, such as 1 ring for internal, 1 for external?
08:50.40payalhas scubasteve gone ):
08:50.47payalanyone else
08:50.53scubastevepayal no am just kinda busy
08:51.02scubastevepayal can you edit your extensions.conf file?
08:51.13evillsNugget: Unable to create channel of type 'SIP'
08:51.36payalscubasteve: what do I put there
08:51.44scubastevepayal It's a lot of stuff.
08:51.50*** join/#asterisk FaithX (~Faithful@202-6-145-116.ip.adam.com.au)
08:51.58payalah! ok
08:52.06payalis it necessary?
08:52.19scubastevepayal: Yes, it's necessary :)
08:52.35payalscubasteve: where do I find docs for doing that
08:52.54scubastevepayal: I'm trying to send you a /msg with the info but you don't seem to see them.
08:53.37payalscubasteve: I saw them now
08:53.47scubastevepayal ok
08:54.27*** join/#asterisk [jas] (~jas@adsl-15-167.swiftdsl.com.au)
08:55.19konkyz0rkdoes anyone know an upper limit of simultaneous registered sip users on *?
08:55.55channanNugget - change qualify=500.... same problem... no dial tone from the other side to cisco phone. no errors anywhere
08:55.57JerJerall of them
08:56.16channanalthough I can dial cisco phone from CLI fine
08:56.51Nuggetstrange.
08:57.32channanyep.. it's strange
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08:58.23JerJerDial,SIP/peer
08:58.25JerJervery simple
08:58.29JerJer[peer]
08:58.43JerJertype=peer
08:58.43JerJerhost=dynamic
08:58.43JerJersecret=sharedsecret
08:58.53NuggetI think the best place to start would be trying to figure out why asterisk doesn't know your phone's IP address.
08:59.08Nuggetthat's unusual and can't be helping matters any
08:59.46channanright... I've been lokking into that. I know from my asterisk server subnet, I can ping my cisco phone
09:00.14Nuggetmy cisco phone connects to an asterisk server that's thousands of miles away from me.  the subnet issue isn't a factor.  as long as it's routable.
09:00.33Nuggetmaybe your vpn is doing weird things you're not aware of, though
09:00.33channanunfortunatelty I can't plug it into the same subnet with * since we're 100 miles apart
09:00.47JerJerdistance is no factor
09:00.49channanyeah.. I kind of suspect that
09:00.51JerJera proper network is
09:01.08Nuggets/thousands of miles/dozens of hops/
09:01.10Nuggetbetter?  :)
09:01.14JerJerno
09:01.18JerJerit is no factor
09:01.24Nuggetright, that's my point.
09:01.45Nuggetsee there where I say "isn't a factor."  :)
09:01.47*** join/#asterisk sxpert_work (~sxpert@raph.imag.fr)
09:01.50*** join/#asterisk cjk (~cjk@80.92.75.85)
09:04.32ST-3Use The VOIP Luke!
09:04.42channanI've reviewed my VPN and don't see anything strange (again, I'm no VPN expert). I just know that either sides can tunnel thru the vpn fine
09:05.10freathey
09:05.12channanhmm... tricky.. tricky...
09:05.31freatI'm having trouble when * bridges two outside calls
09:05.37Nuggetwhat exactly did you mean by "I can dial cisco phone from CLI" ?
09:05.52freatwe can call in fine, and dial out fine (we use Teliax as a provider)
09:06.14*** join/#asterisk simong (~simong@h166n2fls35o884.telia.com)
09:06.20freatbut if I conference two outside lines along with mine, we get serious audio problems
09:06.33channanI login into the console and dial using command "dial 3000" (3000 is my extension)
09:06.53Nuggetso how does that differ from a phone doing the same thing?
09:06.55freator, if I dial in from outside and hit an extension that dials a cell phone or other outside line, I get the same effect
09:07.43channangood questions... I don't know. all I know is all other sip phones and soft phones worked fine except the cisco
09:08.18Nuggetdo your other sip phones and soft phones show IPs in "sip show peers" or do they also show as Unspecified?
09:08.32channanyes. they do
09:08.51freatthey do the hokey pokey?
09:09.05channando I need to have the keyword register in my sip.conf for my ciscophone?
09:09.09Nuggetno.
09:09.55channanyeah.. I thought so since I've tried both and still same prob
09:10.24Nuggetall I have on mine is type=friend, host=dynamic, username/secret, nat=yes, canreinvite=no, and context.
09:10.46Thumannanyone down with gastman for asterisk?
09:11.19channanyeah.. that's what I got
09:16.01payalseeya later
09:21.33*** join/#asterisk Duckbizkit (~jcunningh@24-240-243-142.charter.com)
09:22.29*** join/#asterisk nealz (~niels@pcnp.office.zxp.nl)
09:24.47*** join/#asterisk maruz (~maumar@adsl-123-3.38-151.net24.it)
09:26.30*** join/#asterisk zoa (~zoa@pirus.securax.be)
09:28.58*** join/#asterisk Delvar (~irc@83.146.53.34)
09:31.28*** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode)
09:31.54libpcpanyone heard a linksys fxo/fxs from Cisco?
09:31.55scubastevezzzzz
09:32.01tzafrirbah, our PBX system just got stuck and required a reboot. quite strange, though
09:32.17scubasteveyuck indeed
09:32.20Delvargood morning/evening/night whichever it is. :)
09:32.29scubastevevery very late evening
09:32.32scubasteveearly morning
09:32.43scubasteveI should be long asleep.
09:33.05tzafrirI could ssh in, and managed to read a number of files from /proc . But even a 'ps' command would get stuck .
09:33.18*** join/#asterisk riksta (~rick@81-178-236-71.dsl.pipex.com)
09:33.27scubastevetzafrir ... looks like the box ran away.
09:33.38scubasteveanything in /var/log of interest?
09:33.40tzafrirIt seems that processes would become 'D' or something, but I had no way of telling. ctrl-alt-del had no noticable effect
09:33.54scubastevewhat do you have for ram?  swap?
09:35.58tzafrirjust before that I noticed that the  FXO card "wasn't working" . zttool reported it as "RED"
09:36.39tzafrirrestarting asterisk had no effect. I tried stopping asterisk , rmmod and modprobe, and this is where the system hung
09:36.39scubastevelook in the system logs.. something went wrong..
09:38.24tzafrirscubasteve, basically nada. I see that the system clock was off by 1 hour for some strange reason
09:38.24scubastevecat /proc/interrupts and /msg it to me
09:38.48scubastevecould be an interrupt conflict on your fxo... I had all sorts of whacky sh!t go down..
09:39.04scubastevemy box would panic if I plugged in a USB dongle and someone called in or out while it was in use.
09:39.36tzafrirno shared intrrupts of fxo
09:39.43scubastevehm
09:40.06tzafrirgrep ' /proc/interrupts to find the  shared ones
09:40.17tzafrirgrep , /proc/interrupts , that is
09:40.37scubasteveyep that'd do it
09:44.36Duckbizkitscubasteve do you have any exp with outbound apps on *
09:44.52Duckbizkitdialers i mean
09:45.02scubasteveDuck:  Yes.
09:45.03zigmanDuckbizkit create call files
09:45.08scubasteveThat'
09:45.13zigmanlet scubasteve go to bed ;)
09:45.18scubasteveThat's one way to do it, but you can use the Manager API too.
09:46.20Duckbizkitzigman is there a good way to control call volume with the call files? that's the only reason i haven't gone that route thus far
09:46.38scubastevezig - I got a ton of stuff going on.. will grab an hour on the sofa so I don't disturb the wife.. then I have to run downtown and file some paperwork... then will scoot into the office ...
09:46.47scubasteveDuck.. hmmmm
09:46.49Duckbizkitbtw, thx scubasteve, i'll dig around the voip wiki for it. get some sleep man
09:47.16Duckbizkiti guess i could limit the outbound lines in iax.conf, couldn't i?
09:47.27scubasteveah
09:47.30scubastevethat call volume :)
09:47.32*** join/#asterisk dg1nsw (~schulte@gate.sympat.de)
09:47.51Duckbizkiti just don't want to eat up my bandwidth, i occasionally get problems with 1ch as it is
09:47.55scubasteveDuck I think you might have to keep track of what calls are in progress...
09:48.13konkyz0rkdoes anyone know an upper limit of simultaneous registered sip users on *? like is 2-4000 doable or should i give up way lower than that
09:48.56Duckbizkityeah scubasteve i guess i could have an AGI for the outbound calls that increments a variable in a script, if the variable is less than the max value it could copy another call file
09:49.34scubasteveYou might be able to code some hard limits in asterisk.. would take a walk through the src to find out tho
09:49.49Duckbizkitbut getting the script to kick off at 0 would be a problem, would require a manual copy of the first call file
09:50.37Duckbizkitthen i guess the last piece of the extension could kick an AGI to decrease the call volume variable
09:50.52Duckbizkitbut if the customer hangs up before i get to that step, wouldn't it skip it?
09:51.00JerJerAGI is not the answer
09:51.09scubasteveYou don't need AGI to play variable games.
09:51.25scubasteveAsterisk can manipulate variables itself, in extensions.conf.
09:51.37*** join/#asterisk [Spirit] (spirit@rune.xs4all.nl)
09:51.41Duckbizkitgood. i'm no programmer, i'm just an old netadmin ;)
09:52.10*** join/#asterisk RoyK (~roy@80.239.107.80)
09:52.24scubastevehehe
09:52.40scubastevehey can someone set up iax to a fresh * install to test some cool stuff?
09:52.57scubasteveThen it's nap time.
09:53.05Duckbizkitit's after hours scubasteve, i dont see why i couldn't
09:53.17rikstacool stuff?
09:53.36Duckbizkitlemme kick over the original config
09:53.37scubasteveyeah, I have some cisco call manager type of apps
09:53.40scubastevelike email reader
09:53.43scubasteverss news reader
09:53.45scubasteveweather
09:53.55rikstaoh, i have an rss reader
09:54.23scubastevefor *?
09:54.41rikstanaw for the cisco ;)
09:54.57rikstahow is it working for * ?
09:55.38scubastevekickass :)
09:55.50scubasteveI need to get access to a callmangler and see what else it does and how well it does it
09:56.26*** join/#asterisk cdegroot (~cg@80.126.80.66)
09:57.04scubastevericksta: http://miselconsulting.com/demo.txt
09:57.11scubasteveDoes CM let you respond?
09:57.18scubasteve(to email)
09:58.09*** join/#asterisk WS (WS@adsl-209-30-230-110.dsl.ksc2mo.swbell.net)
09:58.24WSLUBR LUBA
09:58.30WSwhats going on tonight my fine negros
09:58.47scubasteveheh
09:59.06WSscubasteve jonesn for some weed?
09:59.10scubastevenah
09:59.20Duckbizkitscubasteve's jonesn for some ZZZs
09:59.22scubastevejonesn for some zzzzzZzzz
09:59.23scubasteveyeah
09:59.25Duckbizkitlol
09:59.25scubastevehell yeah
09:59.30WScome on now all adam sandler fans LUBR teh weed
09:59.31WShehehe
09:59.33WS:P
09:59.56WSI'm insane dun mind me
10:00.05scubasteveheh
10:00.06WSmembrane sprung a leak
10:00.10WSnothin like petey pablo tho
10:00.15Duckbizkitwhip why the hell is your ip pointing out of KC, MO?
10:00.19WS:D
10:00.20WSiono
10:00.21*** part/#asterisk tech_voip (~tech_voip@202.65.128.18)
10:00.23WSits speshul
10:00.26WS;)
10:00.28Duckbizkithaha
10:00.34WSwhere you wanna card teh shit from?
10:00.34WS:P
10:00.39Duckbizkithaha
10:00.45scubasteveyou guys r nuts
10:00.56WS:D
10:01.03WSasterisk me to 666-666-6666
10:01.06Duckbizkitguam, WS
10:01.21WStheres a Guam, WS?
10:01.23Duckbizkithrm? want me to bounce you out
10:01.26WScant remember what island is WS
10:01.27scubastevewhat is ws?
10:01.29WSlol
10:01.30Duckbizkitguam, the island
10:01.32WSWilliam Scott
10:01.39WS:D
10:01.40scubasteveuh... i'm an american
10:01.42scubastevewhere the hell is that
10:01.57Duckbizkittiny island down in the gulf
10:02.00WSnorth carolina have lots of anhydrous?
10:02.01scubastevenice
10:02.08*** join/#asterisk KahiN (~Helen-16f@asy170.as25336.sol.superonline.com)
10:02.09scubastevenc is an interesting place
10:02.10WScould make a meth farm and make some $
10:02.14scubastevelol
10:02.31WSDuckbizkit
10:02.36DuckbizkitWS
10:02.36WSI want you to bounce me out tommorrow
10:02.36*** join/#asterisk sellout (~asdf@S0106000c419cfff0.vc.shawcable.net)
10:02.37scubasteveI was buying a house in 1998, came across a gorgeous place that was used as a meth lab.
10:02.40WSima call my chixorz
10:02.42scubasteveIt was easily a 400k house
10:02.43WSfrom HELL
10:02.45WSw00t
10:02.46scubasteveSelling for MUCH less.
10:02.52WSyeah
10:02.52scubasteveThe whole place needed to be gutted though.
10:02.55scubasteveRan like hell.
10:02.56WSmeth houses are ruined
10:02.59Duckbizkitok, just give me the CID you'll be calling from and i'll have it transfer out with the CID of (666) 666-6666
10:03.07WSk
10:03.10scubasteveWhy someone with that kinda $ would do meth there is beyond me.
10:03.15Duckbizkitoh, if you give me the number you want to call i'll just add it in
10:03.24scubasteveGo out and buy a freakin 1970's airstream camper.  Sheesh.
10:03.33WSk
10:03.34WSsec
10:03.38WSlemmi find my cell
10:03.53WSscubasteve when you start that shit its hard as heaven to get off of
10:04.04Duckbizkitthey're starting a thing in oklahoma i think, when you try to sell your house, the gov't comes in with meth sensing equip and if you made meth in the house they won't let you sell it
10:04.26WSgeeze
10:04.31WSoklahoma is second worst
10:04.41WSnorth dakota = worst
10:04.55Duckbizkitseattle is the crank capital of the US, man
10:05.00*** join/#asterisk KahiN (~PoKeiMam@212.253.36.170)
10:05.10WSI was reading some stuff a while back that said ND was
10:05.11WS:S
10:05.14Duckbizkitthey wear that badge with pride
10:05.31Duckbizkiti guess ND is pretty boring, nothing to do but make meth
10:05.44WSlots of anhydrous
10:05.49Duckbizkitwell, oklahoma's pretty boring too
10:05.56WSlots of lubr women whores tho
10:07.07WSyou may think she's just "your" pal, but she may be everyone's gal
10:07.07scubastevenice
10:07.19WSALL  women are dirty filthy whores
10:07.20WS:D
10:07.41WSTO far away for me to call too far awwaaaayyy
10:07.48WSnot if you use asterisk eh?
10:07.49*** join/#asterisk KahiN (~TnT@212.253.54.227)
10:07.49WSFuel
10:07.50WS:)
10:08.15evills:)
10:08.31*** join/#asterisk oej (~oej@40.186.204.213.sol.worldonline.se)
10:09.35Makenshiwhat do you think about encouraging asterisk users to register numbers with e164.org?
10:09.52InfraRedbad idea
10:09.55InfraRedspam over IP
10:10.03WShehe
10:10.20WStastes like shit
10:10.28WSand I might have tasted shit before
10:10.30WScant remember
10:10.39InfraRedwild party ?
10:10.41InfraRed:)
10:10.47WSevery day
10:10.48WS:D
10:10.56WSI'm trying to make a resume right now
10:10.59Makenshiand dundi?
10:11.03WSbeen drinkin for last 5 hours
10:11.05WSdundi?
10:11.11InfraRedanyone here knows how to upgrade the firmware of a cisco 7910 phone without CCM?
10:11.46mquinhey InfraRed, fancy seeing you here :)
10:11.57InfraRedhi :)
10:11.59WSfancy
10:11.59*** join/#asterisk emitrax (~emitrax@host139-74.pool80183.interbusiness.it)
10:12.03evillsmelt new firmware into water and then flood the phone. after drying it should be upgraded :)
10:12.04WSgeeze-us
10:12.08tzafrirInfraRed, can't you do it from the phone itself?
10:12.08WSI didnt realize we were international here
10:12.13InfraRedi want to learn the way of the asterisk
10:12.20WS#infrared on efnet is fun ;D
10:12.28InfraRedoh ?
10:12.28WShax0rz
10:12.29WSw00t
10:12.29InfraRed:)
10:12.32WSlol
10:12.33InfraRedhehe
10:12.34DuckbizkitWS i'll come up with a quick and dirty to bounce you out
10:12.42WSsteal someone elses?
10:12.42Duckbizkit*quick and dirty way
10:12.46InfraRedi keept getting some cunt on efnet whos bot keeps taking my nick
10:12.48WS:P
10:12.57WSprolly someone from there
10:13.03WSI dun hang out there much anymore
10:13.08WStry to keep the drama on a low scale
10:13.16Duckbizkitsave the drama for your momma
10:13.22WSexactly
10:13.36WS8 minutes remaining
10:13.43WS:D
10:13.48WSWednesday
10:13.51WS12/01/2005
10:13.54WS04:13:06
10:14.05WSI think my date is wrong
10:14.05InfraRedmquin: from uknot?
10:14.23WShmmm
10:14.27*** part/#asterisk emitrax (~emitrax@host139-74.pool80183.interbusiness.it)
10:15.13Duckbizkitif i just Background my intro prompt can I just immediately dial $EXTEN to bounce a call out to the digits entered?
10:15.21WShaha
10:15.23WSmy roommate guy
10:15.25WSjust hits the ground
10:15.27WSgrabs his leg
10:15.29WSI was rollin
10:15.34WShe said I had a charlie horse dude
10:15.57WShahaha
10:15.59Thumannwtf is a charlie horse?
10:16.02Thumann:X
10:16.03WScramp
10:16.05WS:P
10:16.14WSI'm from the dirty south excuse my slang
10:16.16mquinInfraRed: #scotlug, and uknot
10:16.41scubastevenite folks...zzz time.
10:16.57InfraRedaaah
10:17.07InfraRedi am about to setup my * box
10:17.10WSI'd post a link to something hilarious
10:17.14WSbut is it allowed here?
10:17.22InfraRedgo for it dude
10:17.24WShttp://p077.ezboard.com/fappliancerepairdryers.showMessage?topicID=1159.topic
10:17.27InfraRedlive life on the edge
10:17.34WS:)
10:17.39WSI do that already trust me
10:17.40InfraRedwhats a ban for few days for a man your size
10:17.47WSbut I try to be on the respectful edge
10:17.53Duckbizkithe's a big man, that's a lot of ban
10:18.00WSyeh
10:18.02WSI'm big
10:18.11WS3 foot four with a 10 foot dick
10:18.14WSor something like that
10:18.17Duckbizkitcourse it's like impossible to keep WS out anyway
10:18.23WSyeah
10:18.30WSI'm an #infrared hax0r kiddie
10:18.43WSnetworks go down before I'm banned
10:18.45WSkidding
10:18.46WSshit
10:18.52WSI'm gonna get glined or something for just talking shit
10:19.08WSjupe my nick nekros I dare you!
10:19.13WSDUDE
10:19.16WSSOMEONE SERIOUSLY JUST SHARDED
10:19.21WSI can smell it
10:19.33WSSHARD = try to far but accidentally shit yourself
10:20.01WSfart*
10:20.35WSblah
10:20.43WSI think I need to try to sleep
10:20.48WShit up the soda pop shop on main st tommorrow
10:20.56WSget a good ass sandwich
10:21.17WSmaybe a real old fashioned shake
10:21.19WSsounds good
10:21.24WSand a sliced dill pickle
10:21.41WSsmoke!
10:21.45WS[04:20] <WS> smoke!
10:24.23RoyK~lart ws
10:24.39WS[04:22] <Moggy> no shit
10:24.39WS[04:22] <Moggy> He's been talking to himself for like the last 2 hours
10:24.39WS[04:23] <WS> :)
10:24.39WS[04:23] <MemoPerv> -_-
10:24.42WShahaha
10:24.46WSthey think I'm on crack
10:25.05sambalhmm still no ban?
10:25.16WSyou're going to ban me?
10:25.23WSI'm keeping everyone entertained
10:25.23WS:D
10:25.25sambalif that was possible :D
10:25.32WSwhy would you ban me?
10:25.33WS:P
10:25.36sambalI don't see it that way
10:25.44*** join/#asterisk scanna (~scannachi@81-174-16-211.f5.ngi.it)
10:25.48WSthat sucks
10:25.56WSmaybe you need to get out and live a little
10:26.04WSI'll let you borrow my festiva
10:26.09WSpick up some bitches in that ho
10:28.30*** join/#asterisk phaze (~phaze@pcbcu420a.unil.ch)
10:29.04ThumannWS: hehe.. a charlie horse... I had one once... I was on the can.. got a charlie horse in my leg.. almost pissed all over the place... hurt like hell...
10:29.11WS:P
10:29.14WShaha
10:29.15WSthat sucks
10:29.18Thumannye
10:29.31WSmy legs like to fall asleep on the throne
10:29.38InfraRed10:18 <@|WieBe|> pussy licking, believe it or not.... it's office safe! :  http://www.dumpalink.com/media/1265
10:29.40WSthey dun make em for my size
10:29.41Thumann:)
10:30.43Zeeekinterjection.
10:31.02WSI remember those from schoolhouse rock
10:31.07WSsunday mornings before church service
10:34.29rikstaanyone using voipuser.org's free outgoing service?
10:36.18ThumannWS: dirty south? .. aussie?
10:36.44*** join/#asterisk h4mm3r` (~h4mm3r@81-208-60-202.fastres.net)
10:38.43WSnaw
10:38.46WSsouthern america
10:38.46WS:P
10:38.53WSdown by texas
10:38.53WS:P
10:38.58WSbbl tho :P
10:39.01WSI'll be back sometime
10:39.58Thumann:)
10:40.07ThumannDenmark here... hehe
10:40.13*** join/#asterisk psywar (psywar@rasterburn.org)
10:40.30psywargot a SPA-2000, want to know how to configure it for use with asterisk
10:41.36rikstaexten => _55.,1,Dial(SIP/voipuser/${EXTEN:2})          can someone tell me what the :2 means ?
10:42.08JerJerremove the first two most significant digits
10:42.30JerJerthat matching exten is not proper
10:42.40JerJerit should be _55X.  else just 55 would be matched as well
10:42.55rikstai just took it from a post on a forum and was wondering what it was on about
10:43.04rikstaJerJer: what is the X ?
10:43.09JerJerany digit
10:43.24JerJerread documenation, not some random persons post
10:43.34psywarread the PDF on the asterisk.org site, under "support"
10:43.43psywarkind of non-intuitive place for it
10:44.10psywarmost open-source sites have a "documentation" page
10:44.15rikstaJerJer: yeah ok
10:45.24RoyKThumann: goddag
10:45.26ZeeekJerJer we're all random people
10:46.15rikstaJerJer: it doesn't like _55X
10:46.46rikstawhoops i made a typo :P
10:46.54zigmanand forgot the .
10:47.00zigmanits "_55X."
10:47.01rikstayeah, sorry
10:47.05Zeeekand didn't read
10:47.06rikstai know i just missed it
10:47.06ZeeekStarter tutorial:
10:47.06Zeeekhttp://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html
10:47.06Zeeekhttp://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html
10:47.06Zeeekhttp://www.automated.it/guidetoasterisk.htm
10:47.06Zeeekhttp://www.fnords.org/~eric/asterisk/
10:47.13zigmanactually you could just make _55.
10:47.21*** join/#asterisk Mick` (mick@adsl.mick.id.au)
10:48.09ThumannRoyK: hey :)
10:48.43Mick`quick question, trying to load x100p card into my freebsd box but for some reason it cant find it, i get the following pci0: <unknown card> (vendor=0xe159, dev=0x0001) at 8.0 irq 11. i've loaded the zaptel drivers and get Zapata Telephony Interface Registered on major 196 but thats it, anyone got any ideas plz?
10:50.52cjkhi, is it possible to setup a replication between more than 2 mysql master servers. im asking this because it is quite difficuelt to rewrite my programm to send the write querries to the master and the select querris to the slave. i want to send all querries randomly to one master server which replicates then to the others
10:54.12Poincarecjk: circular database?
10:57.21Moonwickrandom question: can anyone recommend a provider that provides low-cost DIDs with per-minute pricing?
10:57.38Moonwick(as opposed to unlimited-use numbers with higher monthly rates)
10:57.48MoonwickI want to give my parents back home a local number they can call.
10:58.15Moonwickhrm... maybe an 800# would be better, though
10:59.30*** join/#asterisk rajo_ (~rajo@graphics.cs.uni-sb.de)
10:59.33*** join/#asterisk maik_ (~maik@scumm.cs.uni-sb.de)
11:03.04*** join/#asterisk PoWeRKiLL (~PoWeRKiLL@l01m-34-60.d4.club-internet.fr)
11:14.06*** join/#asterisk olivier_ (~olivier_@AMarseille-152-1-1-32.w82-127.abo.wanadoo.fr)
11:15.36Zeeeknufone
11:15.48ZeeekMoonwick nufone
11:16.01*** join/#asterisk doughecka_ (~Doug@adsl-210-162-184.sdf.bellsouth.net)
11:16.31*** join/#asterisk ard (ard@2001:7b8:32d:0:0:0:0:3)
11:19.18*** join/#asterisk martijn (~martijn@213-136-25-234.adsl.bit.nl)
11:19.30martijnmornin'
11:19.49olivier_Hi ppl
11:20.24martijnanyone have an idea how to get the old way of extensions working again?
11:20.42martijnall my calls end up at the s extension instead of the geographic correct number
11:25.54modulus_bleh
11:26.20Mick`anyone able to help me with freebsd problem and x100p card?
11:29.58ellvisMick`: some people here said yesterday that it's not a good idea to run asterisk on *bsd as there is bad support of the card drivers
11:31.09MoonwickI'll concur
11:31.15*** join/#asterisk shanky (~shanky@238.Red-80-33-29.pooles.rima-tde.net)
11:31.23shankygood afternoon
11:31.45MoonwickI'm not even using hardware with my BSD * install, and it still has some issues
11:32.40Mick`ok.. thanks for the info :)
11:33.50RoyKMoonwick: * on BSD on (void)?
11:33.53RoyK:)
11:34.57Moonwickasterisk-specific hardware.  :P
11:35.37Mick`maybe my problem is common..i get the following pci0: <unknown card> (vendor=0xe159, dev=0x0001) at 8.0 irq 11. i've loaded the zaptel drivers and get Zapata Telephony Interface Registered on major 196 but thats it
11:44.50*** join/#asterisk Othello (Othello@nusnet-154-210.dynip.nus.edu.sg)
11:45.19ZeeekMick try searching the mailing list of you haven't already
11:46.00ZeeekI have installed asterisk on FreeBSD a couple of times, it worked without hardwxare
11:47.16tihI'm very happy with Asterisk on my NetBSD systems.
11:47.21tihHowever, I use no Digium hardware.
11:47.32shankyI have installed Asterisk and AMP, I can add extensions, but the client can't register
11:47.37konkyz0rkdoes anyone know an upper limit of simultaneous registered sip users on *? like is 2-4000 doable or should i give up way lower than that
11:48.42shankyI have this message from the log: Jan 12 12:38:25 NOTICE[6124]: Registration from '2001 <sip:2001@192.168.0.2:5060>' failed for '172.16.32.123'
11:49.21shankyhow could I know what is esactly the problem?
11:50.03tihIsn't that a NAT problem?
11:50.27tihYour client thinks it's 192.168.0.2, but Asterisk sees it as 172.16.32.123?
11:51.52Zeeekshanky you have nat=yes in sip.conf ?
11:52.01shankyshanky:
11:52.10shankyups, I'll try the nat
11:54.55*** join/#asterisk tech_voip (~tech_voip@202.65.128.18)
11:55.26tech_voipiam useing the multiple codecs in sip.conf
11:55.59tech_voipwhen i make a call * is giveing a segmentation fault and
11:56.09tech_voipsay core dumped
11:56.50tech_voipdisallow=all
11:56.50tech_voip<PROTECTED>
11:56.50tech_voip<PROTECTED>
11:56.51tech_voip<PROTECTED>
11:56.57tech_voipthis is in my sip.conf
11:57.09rikstafor starters, i dont think its capital g
11:57.33tech_voipno we can use like that
11:57.50rikstaNO CAPS
11:58.03tihAnd you have the g.729 codec (which costs money) in place in /usr/lib/asterisk/modules/?
11:58.06sambali have a grandstream 101 that stops accepting calls after several days, is this a common problem?
11:58.10tech_voipyes
11:58.12tech_voipi have
11:58.20rikstatech_voip: are you listening?
11:58.24tech_voipyes
11:58.27tech_voipi made it
11:58.31tech_voipsmall g
11:58.38rikstatry
11:58.39tihsambal: yes, the registration times out and isn't renewed. known problem on some firmware versions.
11:58.46tech_voipi will test and tell you
11:58.56tihsambal: change to fixed address (and no registration) if you can.
11:58.57sambaltih: anything i can do about it? put it in static?
11:59.00sambalok :)
11:59.01Zeeeksambal what firmware version ?
11:59.05sambalZeeek: latest
11:59.12Zeeekwhich is ?
11:59.13sambalwith attended forwarding support
11:59.15tihsambal: yeah, but _which_ latest?
11:59.19samballet me see if i can see it now :)
11:59.28Zeeek5.18 ?
11:59.31shankytech_voip: I have updated my sip.conf adding nat=yes, and nat=1 in my extension definition, but I still have the same problem
11:59.49tech_voipwhat problem?
12:00.13Zeeekshanky are you using externip=123.321.123.456 ?
12:00.18sambalProgram--1.0.5.16
12:00.27tihsambal: yeah, that one has the registration bug.
12:00.29Zeeek.16 is not the latest
12:00.31sambalok
12:00.32sambal:)
12:00.33shankytech_voip: sorry, the message were for Zeeek
12:00.35sambalit's solved in 18?
12:00.39Zeeekgo for 18 is what GS told me
12:00.54Zeeekdon't know I do,'t use registration
12:00.55tihsambal: but keep using it if you can, with static address, because it's reckoned to be the safest bet overall.
12:01.12tech_voipstill same riksta
12:01.13Zeeekand also I have myu own problem, after 5.11 none of the versions will do DNS at my place
12:01.16tih5.18 is very new, so there may be stuff lurking there.
12:01.29Zeeek5.18 is what GS told me to try
12:01.40Zeeekthe beta is like .21 now
12:01.40tech_voipnow my sip.conf looks like this
12:01.43rikstatech_voip: are you sure the module is compiled correctly
12:01.43sambalok, will upgrade this evening
12:01.46tech_voipdisallow=all
12:01.47tech_voip<PROTECTED>
12:01.47tech_voip<PROTECTED>
12:01.47tech_voip<PROTECTED>
12:01.49rikstatech_voip: dont paste
12:01.50Zeeektech_voip NO
12:01.56sambalstrange it took them so long to solve
12:01.58tech_voipok
12:02.03tech_voipi will not paste here any more
12:02.14sambal~pastebin
12:02.15jbotpastebin is, like, a place to paste your stuff without flooding the channel - try pastebin.ca
12:02.20ZeeekI thought you were gonna paste the whole file !
12:02.25tech_voipok
12:02.26tech_voipfine
12:02.32tihI've heard reports of background noise problems in 5.18.
12:02.37rikstaZeeek: me too :)
12:02.45Zeeeksince I can't use it,n I can't tell you
12:02.54*** join/#asterisk fantomax1 (~fanto@81.208.114.250)
12:02.56tech_voipmy requirement is according to the bandwidth of user he can dynamically choose the codec
12:03.01fantomax1hi all
12:03.02tech_voipbut when i use like this
12:03.08shankyZeeek: and in externip= should be the ip of my asterisk server, isn't it?
12:03.08Zeeekwhat I know is that 5.11 is working for me - I wish I could move on but so far no answer from them
12:03.15tech_voip* is giveing segmentation fault
12:03.15fantomax1can anyone help with a problem in routing SIP calls ?
12:03.21tech_voipand core dumped message
12:03.30sambali think i switch over to static adresses :)
12:03.30Zeeekshanky yes and you might consider readung the documentation which explicitly SAYS THIS
12:03.35tihZeeek: why can't you upgrade?
12:03.43Zeeekand also I have myu own problem, after 5.11 none of the versions will do DNS at my place
12:03.46shankyZeeek: in the handbook?
12:03.55Zeeekno DNS lookup of asterisk server is done
12:04.14Zeeekshanky there is a pretty good resource in the source file directory under configs
12:04.26Zeeeklook at the samples they have every single param annotated
12:04.41Zeeekand look up NAT on the wiki,n there are pages and pages about it
12:04.47DuckbizkitZeeek you have any exp with outbound dialers in *?
12:05.03shankyZeeek: ok, many thanks
12:05.03Zeeekwhat mean white man outbound dialah ?
12:05.17tihZeeek: that's weird - 5.16 does DNS lookups fine for me.
12:05.17Zeeekshanky np
12:05.28DuckbizkitI'm crap for a programmer but I'm trying to set up an outbound IVR for a collections company
12:05.34Zeeektih I know, I'm the only one that has eveer complained to them!
12:05.53sambalDuckbizkit: just create a call file and hook it at a context
12:05.55ZeeekDuck explain what you're trying to accomplish and we can give a hand
12:06.24Zeeektih I even added a SRV record to see if that helped
12:06.38Duckbizkiti've got a client wanting me to call about 1000 customers and remind them that their payment is past due
12:06.40Zeeektijh sniffing the reboot of the phone reveals NO DNS lookup at all
12:06.54sambalthat would be a lot of call files :D
12:06.56ZeeekDuck for moral reasons I don't think I wanna get into that
12:07.07Zeeekit kinda SUCKS
12:07.39Zeeekbut you could loop thru the database and generate the call files on the fly easilty enuf
12:07.46Duckbizkityeah sambal that's a lot of call files
12:07.54Zeeekif you can't do it, use a consultant here to write it for you
12:08.08sambalisn't asterisk going to freak out with 1000 call files? :)
12:08.24tihDuckbizkit: no problem, I'd say - just slow it down to where Asterisk doesn't have to do too many at the same time.
12:08.30Zeeekyou'd do them one after the other like 1 min apart or so
12:08.30*** join/#asterisk Josemar_BR (~root@200.215.10.110)
12:08.46Zeeekor a finite number at once
12:08.48tihDuckbizkit: the challenge is to keep track of who has heard the message, and who you need to try calling again.
12:09.05Zeeekhey if the pounbt is to annoy them so they'll pay....
12:09.13Zeeekmight as well piss em off good
12:09.18fantomax1can anyone help with a problem in how to route SIP calls ?
12:09.24tihZeeek: let it loop, you mean? Until they pay?
12:09.32Zeeekfa,tomax go for it!
12:09.39Zeeektih, I see you are a programmer
12:09.44tech_voiphttp://pastebin.ca/4092 riksta
12:09.57tech_voipcan you see my sip.conf here
12:09.57tihZeeek: not really - I'm a sysadmin.
12:10.19Zeeekno actually like a mailing list, I'd just tick off the ones that went thru during the .call generation
12:10.31Zeeekthen loop thru again until they're all ticked off
12:10.36fantomax1what do u mean Zeeek ?
12:10.37Zeeekthat will be $100
12:10.58Zeeekfantomax1 ask the question assuming you have made a minimal effort to read some background
12:11.09fantomax1i didn't read sorry
12:11.15Zeeekgo for it
12:11.16rikstatech_voip: what happens when you comment out g729
12:11.23fantomax1was there any answer ?
12:11.27tech_voipit will work
12:11.30tech_voipno problem
12:11.46rikstato then it's likely that the g729 module is dodgy
12:12.11Zeeek<fantomax1> can anyone help with a problem in how to route SIP calls ?
12:12.19Zeeekis that your question?
12:13.03tech_voipbut individually it works fine
12:13.11tech_voipwhen i comment all other codecs
12:13.16fantomax1yes ... a system .. proxy or anything like that that can route the original call to a pubblic ip and then to asterisk
12:13.40tihfantomax1: what are you trying to solve?
12:14.12fantomax1i need to generate a call from A pass to B and terminate it on Asterisk ( location C )
12:14.18fantomax1this i my prob
12:14.26fantomax1B is a test point
12:14.37fantomax1A, B and C are on internet
12:15.22fantomax13 different pubblic IPs
12:15.33tihAnd why does it need to pass through B?
12:15.46fantomax1to collect traffic data ...
12:15.49tihAha.
12:15.54fantomax1a political problem
12:15.55fantomax1:)
12:16.07fantomax1i'm forced to do it
12:16.15fantomax1it is not technical
12:16.23fantomax1we tried with 2 linux box
12:16.36fantomax1everything went ok .. inside the LAN
12:16.45fantomax1but on internet doesn't work
12:16.59tihHow does it fail?
12:17.14tihAnd are you doing tricks with firewalls to force traffic through point B?
12:17.25rikstatech_voip: ahh, well im not sure then
12:17.49tech_voipbut conceptually
12:17.53fantomax1I don't know .. I'm not the network guy ..
12:17.54tech_voipit has to work
12:18.11fantomax1yes by the techs said there are probs with GWYs
12:18.14tech_voipi.e dynamic chosing of codec according to bandwidth is it correct
12:18.47fantomax1so I was asking if anyone used a proxy/router SIP
12:18.54fantomax1to do it
12:19.53tihfantomax1: if B is an Asterisk, too, A would have to place the call to B, which would forward it. RTP traffic (the actual voice data) would find its own way, unless B was set up to trap it (canreinvite=no), which is a good candidate for the cause of your problems.
12:19.54rikstatech_voip: i'm not sure what youmean
12:20.00tech_voipi.e
12:20.06tech_voipif we allow mulitiple codecs
12:20.23tech_voipit has to choose one according to bandwidh limitations of ua is it not
12:20.39rikstano it just chooses them in order
12:20.42rikstauntill it can use one
12:21.08tech_voipok
12:21.17rikstawell..afaik anyway
12:21.21*** join/#asterisk fafnir (~hello@tdds-gw.Moscow.gldn.net)
12:21.34rikstait doesn't know about how much bandwidth is available
12:21.41tech_voipok
12:22.05tech_voipriksta one other doubt
12:22.21tech_voipi have numbers starting with 234XXXXXXX
12:22.28rikstai'll warn you, i'm not the best person to ask...but go ahead
12:22.49tech_voipand i want to restrict them in such that they can ony dail numbers starting with 1XXXXXXXXXXX
12:23.04tech_voipcan you any idea on that
12:23.13Zeeek*starting* with that ? that's a long beginning!
12:23.36rikstawell if you only have that in the extensions.conf then they can't dial anything else anyway
12:23.57tech_voiplike
12:24.40tech_voipi tried like this exten => 234XXX/1XXXXX,1,Dial(SIP/{EXTEN}) but its not working
12:25.03rikstayou don't make much sense tbh
12:25.03tech_voipits going to all numbers like starting with 2 , 3 ....so on
12:25.24*** join/#asterisk skyegg (~olavo@200-181-140-080.ctame7006.e.brasiltelecom.net.br)
12:25.45Zeeekthere are some excellent articles that tell you how to do this
12:25.47ZeeekStarter tutorial:
12:25.47Zeeekhttp://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html
12:25.47Zeeekhttp://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html
12:25.47Zeeekhttp://www.automated.it/guidetoasterisk.htm
12:25.47Zeeekhttp://www.fnords.org/~eric/asterisk/
12:27.22*** join/#asterisk Verliba (~Miranda@250.1-14-84.ripe.coltfrance.com)
12:32.43shankyoh, my god
12:32.56shankyZeeek: do you what was my problem?
12:33.19shankythe username/password
12:33.50*** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de)
12:37.40*** join/#asterisk zotz (~zotz@24.244.133.136)
12:38.46Zeeekshit happens :)
12:39.54InfraRedanyone here uses CCM >?
12:41.14Zeeekwow, good thing I got fixed ip at home by accindent
12:44.15*** join/#asterisk Specky[W] (~sspecken-@p508EDD4B.dip0.t-ipconnect.de)
12:44.28shankynow, I have the phone registers but it can't dial between them
12:46.26*** part/#asterisk Specky[W] (~sspecken-@p508EDD4B.dip0.t-ipconnect.de)
12:46.46shanky-- AGI Script dialparties.agi completed, returning 0 <-- is this bad?
12:49.11*** join/#asterisk Ron-Na (~ronald@61.59.45.205)
12:49.45Ron-NaI got an IAXy today, ... where is a document how to install it?
12:52.30FaithXHey guys a little more on the big delay bad echo saga with my incomming calls over ISDN.  It seems when the ambient noise on the call comming in is higher the delay & echo is worse
12:54.14ZeeekRon-Na documentation ? Muhahah
12:54.34*** join/#asterisk FaithX (~Faithful@202-6-145-116.ip.adam.com.au)
12:55.30FaithXHey guys a little more on the big delay bad echo saga with my incomming calls over ISDN.  It seems when the ambient noise on the call comming in is higher the delay & echo is worse
12:57.37*** join/#asterisk Mike_TK (~Mike_TK@213.180.245.62)
12:58.07Zeeekalmost every day there is a 10 minute period where Isee this continuously
12:58.08Zeeekchan_iax2.c:5011 socket_read: Error: Resource temporarily unavailable
12:58.13ZeeekANYONE ?
12:58.48FaithXMy phone is set to PCMU
12:59.17ThumannZeeek: WHY ARE WE SHOUTING?
12:59.20Thumann:>
12:59.26Ron-NaZeeek how do I install the IAXy?  I did not find any info about it.
12:59.58Zeeeklook on the digium site
13:00.02Zeeekthe PDF is there
13:00.31Ron-NaI am there, but there I found only the DATA sheet, ...
13:00.42ThumannZeeek: well.. try to enable iax2 debug
13:01.21Zeeekcan't debug the lines are like 25/second
13:01.49ZeeekRon-Na try to google for it - I can't find it right now
13:02.55Ron-NaI found it http://www.digium.com/downloads/Iaxy_Installation_Guide.pdf  "S100I Installation guide"  !!! ???
13:04.03Zeeekthat's it
13:05.26FaithXWhats the codec of ISDN voice?
13:07.03*** join/#asterisk Duckbizkit (~Duckbizki@24-240-243-142.charter.com)
13:11.29FaithXI'm thinking of signing up with broadvoice... any comments/oppinions welcome :)
13:13.42Zeeeksearch the mailing list for Broadvoice woes
13:13.49ZeeekI've seen some questions go by
13:14.02*** join/#asterisk _Thumann (Bush@is.a.retard.dk)
13:15.12datareactorPCMU is specified in CCITT/ITU-T recommendation G.711. Audio data is encoded as eight bits per sample, after logarithmic scaling. Code to convert between linear and mu-law companded data is available in [6]. PCMU is the encoding used for the Internet media type audio/basic. A detailed description is given by Jayant and Noll [11].
13:15.23datareactorPCMU is specified in CCITT/ITU-T recommendation G.711. Audio data is encoded as eight bits per sample, after logarithmic scaling. Code to convert between linear and mu-law companded data is available in [6]. PCMU is the encoding used for the Internet media type audio/basic. A detailed description is given by Jayant and Noll [11].
13:15.27datareactorPCMU is specified in CCITT/ITU-T recommendation G.711. Audio data is encoded as eight bits per sample, after logarithmic scaling. Code to convert between linear and mu-law companded data is available in [6]. PCMU is the encoding used for the Internet media type audio/basic. A detailed description is given by Jayant and Noll [11].
13:15.54*** join/#asterisk tdonahue (~tdonahue@haynesmail.haynes-group.com)
13:15.58datareactorPCMU is specified in CCITT/ITU-T recommendation G.711. Audio data is encoded as eight bits per sample, after logarithmic scaling. Code to convert between linear and mu-law companded data is available in [6]. PCMU is the encoding used for the Internet media type audio/basic. A detailed description is given by Jayant and Noll [11].
13:15.58datareactorhttp://www.freesoft.org/CIE/RFC/1890/20.htm
13:16.39tihZeeek: that res. temp. unavail. would be flagged as a corrupt UDP packet if it were RTP data instead of IAX...
13:17.03datareactorFaithx check this http://www.freesoft.org/CIE/RFC/1890/20.htm
13:17.03RoyK~lart datareactor
13:17.40tihZeeek: from what I'm told, checksum errors in UDP cause EAGAIN in the read, under Linux. EAGAIN there triggers the message you're seeing.
13:18.18*** join/#asterisk UPMeduardo (~UPMeduard@tauro2.dit.upm.es)
13:18.21tihZeeek: maybe check to see if your interface is racking up udp checksum errors at a similar rate?
13:18.54tihZeeek: if it's a heavily loaded system, what you described wouldn't be too disturbing, really.
13:21.39Zeeekthx for that I was in a local crisi period here
13:22.01ZeeekI'm trying to determine if this has sthing to do with a particular IAX connection
13:22.26Zeeekthis happens for about 10 min every day at diff times. NO USERS at the time it happens. An attack? maybe
13:23.05Zeeekthis is a little hard to diagnose because when it happens, the lines block all other output
13:23.23RoyKhm................................................
13:23.28RoyKwiki is on bad acid
13:23.36Zeeekalmost exactly 10 minutes each day
13:23.37tihOh, sorry -- I thought you said once about every ten minutes.
13:23.44ZeeekWhy oh why oh why would that be?
13:23.52tihThat, I wouldn't have worried about.
13:23.54*** join/#asterisk denon (denon@synapse.subneural.net)
13:23.54*** mode/#asterisk [+o denon] by ChanServ
13:23.55*** join/#asterisk fafnir (~asfgasfg@tdds-gw.Moscow.gldn.net)
13:24.32tihIf you're getting lots of them, for ten minutes at a time, that seems really weird.
13:24.35Zeeekten minutes of 50,000 lines+ every day?
13:24.46Zeeekuh, ya!
13:25.30tihI'd say you need a tcpdump trace file of what's passing in and out of that box at the time.
13:25.43tihThere ought to be a very, very, visible pattern there.
13:26.10Zeeekbtw, stopping * and restarting it doesn't solve
13:26.30Zeeekwhat I see is around 50000 lines every day
13:26.32*** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com)
13:27.26ZeeekI suppose I could remove register for FWDIAX and a couple of others?
13:27.55tihWell, *I* would start by getting a tcpdump trace of all traffic in and out of the box while it's happening.
13:27.59sumahow to write a zaptel driver for a new modem ?
13:28.11tihI'd be surprised if that didn't show up an interesting pattern.
13:29.39Zeeekhey you the admin! how would you recommend I do this?
13:29.52Zeeekso I'm ssh'ed in to the box and I see the prob start....
13:30.04tihSo you need to be root...
13:30.13ZeeekI'm always root - yeah I know...
13:30.19tih...and be in a directory where you have a bit of disk space...
13:30.40Zeeekgot 1 gig
13:30.51Zeeekls /usr/local
13:31.04tih# tcpdump -i eth0 -s 1500 -w tcpdump.out
13:31.09*** join/#asterisk xobelixx (~werner@fe1-noc.dus.dnsteam.de)
13:31.41tih(if your network interface is eth0, that is)
13:31.51Zeeekand stop w ctrl C ?
13:32.09ZeeekI'll look it all up inman
13:32.17tihYup.
13:32.24Zeeekwould the dump be readable in ethereal ?
13:32.32tihAnd then you can study it later, using "ethereal tcpdump.out".
13:32.35*** join/#asterisk jetscreamer (~jetscream@adsl-64-219-216-41.dsl.hstntx.swbell.net)
13:32.48Zeeekgreat - I hope you'll be around to help at that point :)
13:33.20tihI think visual pattern recognition will get you far at that time.
13:33.22maruzsomeone has tested mysql realtime with success?
13:33.45*** part/#asterisk Grooby (~Grooby@ip24-250-126-171.dc.dc.cox.net)
13:34.09jetscreamershouldn't you use a real db? (i know nothing, insert disclaimer here, ymmv)
13:35.23bjohnsonI thought there was a wiki page talking about cpu/ram requirements for different installs but now I cn't find it.  Does anyone know where it is?
13:36.21bjohnsonjetscreamer: for this use, mysql is well suited (coming from a postgresql user)
13:36.48*** join/#asterisk ryguy (~ryan@dhr-internal.dhrnetworks.com)
13:37.30jetscreamergood enough for me. i was thinking import/export compatability though.
13:37.38RoyKjetscreamer: doesn't matter. mysql is prolly the fastest db on earth, so why bother using a "real" db if you don't need the advanced stuff?
13:38.16Zeeektih, why the long snaplength ?
13:38.24jetscreameralrighty then :)
13:39.00FaithXbjohnson: Hey guys a little more on the big delay bad echo saga with my incomming calls over ISDN.  It seems when the ambient noise on the call comming in is higher the delay & echo is worse
13:39.10bjohnsonjetscreamer: import/export capability?
13:39.13tihZeeek: oh, I just like to be sure I've got all the data.
13:39.40RoyKbjohnson: import/export is easily done with perl :)
13:39.45bjohnsonFaithful: I haven't done any isdn OR echo cancellation debugging so am of little use
13:39.47Zeeek<PROTECTED>
13:40.04tihZeeek: :-)
13:40.09bjohnsonRoyK: I don't see how mysql vs other would significantly impact import/export
13:40.24bjohnsonRoyK: I'm replying to jetscreamer's comment
13:40.33RoyKbjohnson: my point :)
13:40.38olivier_<Zeeek> and if you want to be sure to get all data you can use -s 0 ( no limit )
13:40.40ZeeekIt's great to have people with various experiences and specialities here! If you need any guitar lessons, let me know :)
13:41.06shankythanks to all
13:41.12Zeeekthanks oli but at 5000/minute I fear that may be hard to read
13:41.20Zeeekshanky you all set ?
13:41.30tihMySQL is fast, sure, but it's not as if PostgreSQL is significantly slower. PostgreSQL is much more advanced, though, and scales much better with increasing load -- and you won't suddenly find yourself hitting a wall because you discover that you need to do something that MySQL won't let you.
13:41.42shankyZeeek: at least just for today
13:41.58fantomax1is there anyone that uses SER ?
13:42.14bjohnsontih: I agree .. but that load is unlikely to apply to most asterisk users
13:42.20shankyI have to read all the documents for starters you commented, before continue working
13:42.55bjohnsonshanky: jump right in and start configuring .. there's too much stuff to understand all at once
13:43.11Zeeektih - is there a way to ignore my ssh packets and watch interactively ?
13:43.32iCEBrkrZeeek: grep -v ssh
13:43.33tihZeeek: you can filter in all sorts of ways, yes.
13:43.41tihiCEBrkr: NO!
13:43.47iCEBrkrI guess I should know what tool you're using to watch your traffic. :P
13:44.10olivier_zeek : tcpdump -i eth0 '!port 22' -s 1500 -w tcpdump.out
13:44.11tihZeeek: 'man tcpdump' (or, for that matter, 'man tethereal'), and read up on filter expressions.
13:44.15iCEBrkrtih: Don't yell, it's still morning and I haven't even had my coffee yet.. :D
13:44.21tihiCEBrkr: :-)
13:44.33Zeeeklooking at the tcpdump man page but I see a filter expr file
13:44.33shankybjohnson: I have running the calls between sip extensions, and Asterisk Managemente Portal
13:44.52*** join/#asterisk mhe (~mhe@p50849B79.dip0.t-ipconnect.de)
13:45.09shankynow it's time to read a little and to decide my list of priorities (like asterisk)
13:45.46Zeeektih watching IAX2 I could just dump that port, no?
13:45.49tihZeeek: if you want to watch traffic realtime, you might want to use tethereal instead of tcpdump, since it's got prettier formatting of output.
13:46.01iCEBrkrIf I can hi-jack a T1 today, I'll hopefully be able to test this T100P today.  Yippie-skippy, my first PRI + Asterisk setup
13:46.19tihYeah, that's right -- 'udp port 4569' would select just IAX2.
13:46.21*** join/#asterisk pjm_uk (~pjm_uk@cpc1-pool3-3-0-cust116.sot3.cable.ntl.com)
13:46.23jetscreamerlike i said, i know nothing... :)  but now that you make me think about it, i'm probably thinking importing mysql into sqlite. (think it was importing incompatability) so nothing anybody would care about.
13:46.37ZeeekI got no tetherial
13:46.51tihYou have ethereal, but not tethereal?
13:47.02tihIt's probably in a separate rpm, or whatever you're using.
13:47.16Zeeekslackware 9.1 no x
13:47.22tihtethereal is the text version.
13:47.33ZeeekI hoped as much
13:47.46tiht for terminal, probably.
13:48.04Zeeekwho are all these peole talking to me on 4569?  :)
13:48.16ryguyDoes anyone know why I am able to specify any number for CID on my PRI and it will show up when I am calling anyone in the world, but the name never shows up that I specify?
13:48.22ZeeekI'm being attacked !
13:48.37Zeeek(by peers I configured in iax.conf)
13:48.39tihZeeek: yeah! best filter out 4569 on your firewall!
13:48.55ZeeekYeah IAX2 will work a lot more efficiently then
13:49.12tihYup, and the load on the system should go down quite a bit.
13:49.22iCEBrkrryguy: It's possible the CID format is kinda picky.
13:49.27FaithXDoes anyone know the standard voice codec for IDSN?
13:49.42Zeeekbut seriously, a quick tail look at the live dump would have been really informative just now.
13:49.59iCEBrkrryguy: I had generic info in there and it'd show up as just my extension on the other end, after I tinkered with it, it shows up correctly.
13:50.09ZeeekI don't always catch these 10 minute periods either
13:50.20iCEBrkrFaithX: Whatever works for your bandwidth :)
13:50.38bjohnsonryguy: I've been looking into something similar recently .. I think the caller id comes from the originating device (ie the handset)
13:50.43ryguyicebrkr, is it normal to be able to specify both name and number on a PRI interface to any number in the world ex:my cell/home phone
13:50.53iCEBrkrOops
13:51.03iCEBrkrI read that as ISDN
13:51.08wankelmaybe your PRI provider doesn't handle CNAM
13:51.12wankelnot everyone does yet
13:51.18iCEBrkrryguy: what Wankel said
13:51.18ryguythey do
13:51.28wankelfunny that it doesn't work, then :)
13:51.42iCEBrkrryguy: I'd say so, sure. I haven't done any PRI work yet.  I'm actually starting that today
13:51.59ryguywankel, it is weird, I can specify any number ex:1234567890 but the name always comes up as whatever my PRI provider has listed
13:52.14wankellast time i did pri work we didn't have CNAM.  we had to push the caller id up the D channel both ways in the snow.
13:52.16iCEBrkrryguy: It's really possible they don't support it.
13:52.24*** part/#asterisk shanky (~shanky@238.Red-80-33-29.pooles.rima-tde.net)
13:52.26iCEBrkrwankel: haha
13:52.29wankelryguy: sounds like they're overriding it
13:52.54ryguywankel, do you think it is possible that I could just have them remove CNAM from my PRI so that I can specify it?
13:53.17wankeldunno.  call them.  probably take you a week to get ahold of someone who knows, though.
13:53.43wankelif they're not response, file a complaint with your local regulatory agency.  that usually gets them moving :)
13:53.48wankels/response/responsive/
13:53.48ryguywankel, I have a CLEC with the engineers number in my phone...
13:54.12wankellucky you.  shouldn't be too hard to find out, then.
13:54.42mquinFaithX: if you mean regular isdn voice channels (no IP) then it's uncompressed digital audio, sampled at 8khz in 8-bit words
13:54.49ryguydoesnt hurt that we are one of their 10 largest customers in the nation
13:54.57mquin(hence 64kbit/channel)
13:55.37iCEBrkrErrrg! I forgot to bring in my IP phone today
13:55.45wankelmquin: well, it's dynamically compressed.  ulaw or alaw.
13:56.20Zeeekis it useful to do srvlookup in your experience ?
13:56.58RoyKwankel: ulaw/alaw aren't compressed
13:59.20wankelthey're dynamically compressed
13:59.38zoayeah
13:59.39wankeldynamics being audio amplitude variations
13:59.43zoathey are slightly compressed
14:00.06wankelin the audio world, reducing dynamic range is called compression and increasing it is called expansion
14:00.15wankelit's a confusing overlap of terminology
14:01.14iCEBrkrIt's all sound to me! :P
14:01.45wankelin the telco world, it allows you to make better use of the bits you have by allocating more bits for quiet speech, where the ear is more sensitive, and fewer bits for loud speech.  uncompressed (dynamically) audio divides the bits up equally, which is a waste.  ulaw and alaw differ in how they distribute the bits.
14:01.56iCEBrkrTho, I have to admit, I'm a firm believer in VBR or dynamic compression..
14:02.05*** join/#asterisk zarni (~stmo@gw.rixtele.com)
14:02.22iCEBrkrwankel: I never understood the difference between ulaw/alaw.
14:02.56iCEBrkrMatter of fact, it wasn't until the other day I actually started doing some quality & bandwidth testing of the available codecs.
14:03.06FaithXmquin, from what I have been reading "G.711 documents the standard 64 kbps audio encoding used by telcos throughout the world"
14:03.08scrubbone starts with an a and the other with a you.
14:03.35*** join/#asterisk Caede (~chatzilla@204.94.248.81)
14:03.41iCEBrkrscrubb: hahaha
14:04.10FaithXmquin, "Telcos can select between two different varients of G.711: A-law and mu-law."
14:04.56FaithX"A-law is the standard for international circuits."
14:05.02iCEBrkrI'm going to assume the G.729 codec is pretty good??  I mean it's only a $10 license, couldn't hurt to buy and try it.
14:05.38wankelice: they're both logarithmic quantizers that stuff about 12 bits of linear data into 8 bits with little apparent loss of quality.
14:06.47iCEBrkrwankel: Cool cuz I'm using ilbc at the moment since my DSL at home is wonky.
14:07.19wankelit doesn't really matter which you use unless you're on a telco circuit, in which case you should use what they do :)
14:07.30mquinFaithX: what wankel said :)
14:07.45iCEBrkrerr, back home.. I'm using my IP phone to connect to an Asterisk box that's remote.
14:07.58iCEBrkrwankel: yea, that makes sense.
14:08.15iCEBrkr"While the old VoiceAge licensed codec would not function in a SCSI-only system, our newer and faster codec does not have this limitation."
14:08.17FaithXgraph  http://www.freesoft.org/CIE/Topics/127.htm
14:08.27iCEBrkrI don't quite understand this limitation. :(
14:08.34wankelthere you go
14:08.55wankelhttp://www.ericsson.com/support/telecom/part-a/a-2-7.shtml
14:08.57wankelalso, that's good
14:09.34Ron-NaIAXy question:  user: abcd   and what is the counterpart in the iax.conf ???   Guess accountcode ???
14:10.09*** join/#asterisk ismaelg (~ismaelg@80-28-2-2.adsl.nuria.telefonica-data.net)
14:10.16ismaelghello
14:11.02FaithXAll I'm trying to figure out is how to cut my 1.5s incomming lag from ISDN to my SIP phone and real bad echo at times (when the line comming in is noisy)
14:11.11ismaelgI am setting up asterisk and when i try to make a call, I do not hear any dial tone, why it could be?
14:11.47ismaelgon I ear dial tones in sip calls
14:11.53fantomax1is there anyone that uses SER ?
14:11.59ZeeekRon-Na yes
14:12.19Ron-NaZeek thanks
14:12.25*** join/#asterisk PoWeRKiLL (~PoWeRKiLL@ATuileries-152-1-24-217.w82-123.abo.wanadoo.fr)
14:12.54iCEBrkrismaelg: your phone hasn't registered to Asterisk.
14:12.56FaithXwankel, that page is excellent
14:12.59iCEBrkrismaelg: possibly.
14:13.10*** join/#asterisk OloBola (~casper_sp@adsl-69-110-121-26.dsl.pltn13.pacbell.net)
14:13.17ismaelgiCEBrkr: my phones are registered
14:14.28iCEBrkrismaelg: Apparently they're not if you don't get a dialtone. *Shrug* at least thats the case when my crappy BT100 doesn't register.
14:14.30wankelfanomax: i do
14:14.54iCEBrkrI don't even get a dialtone on my Sipura2k when it's not registered
14:15.19wankelheh.  the 7960 will DIAL without registration succeeding.  it really doesn't care about registration.
14:16.01olivier_<iCEBrkr> yep . same on my sipura 2100
14:16.30iCEBrkrwankel: I mean, technically I can configure my BT100 to not care about being registered. I think.. I never played with it.
14:16.37ismaelgif my phones do not registered, I can't make calls between them, but i can call an extension and i get dial tone
14:16.42iCEBrkrI've seen the options.
14:16.58iCEBrkrismaelg: Wait, you said you're not getting dialtone.
14:17.05iCEBrkrSo how are you dialing another extension?
14:17.25wankeleven when you enable registration on the 7960 it's just something else it does.  it'll always send invites before it completes.  it's like it was an afterthought.
14:17.43iCEBrkrhaah
14:18.00ismaelgiCEBrkr: I dial 101, and the phone on this extension sound
14:18.05ryguywould anyone here be willing to msg me to look at an error log with me to figure out why zap does not load?
14:18.17wankelthey're targeted more at corporate use, though, where there are better ways of controlling access and tracking presence than sip registration.
14:18.58iCEBrkrismaelg: Before you dial 101, do you have a dialtone tho?
14:19.49*** join/#asterisk Matrix_3033 (~berrayahk@82.101.128.6)
14:19.50*** join/#asterisk allyour80211b (~allyour80@208.178.154.99)
14:20.04ismaelgiCEBrkr:yes, on sip channels i get dial tone, but in zap or capi channels not
14:20.24iCEBrkrHuh?
14:20.33*** join/#asterisk ctooley (~ctooley@rrcs-24-153-220-226.sw.biz.rr.com)
14:20.37ctooleygood morning
14:20.42iCEBrkrctooley: hey
14:21.07ctooleySo, I just updated from CVS December 17th to January 11th and now "show dailplan" doesn't work
14:21.30ctooleyit wants a context
14:21.38ctooleybut I want it to show me everything. :(
14:22.03*** join/#asterisk iMax (~weirdo@delirium.chello.at)
14:22.10iMaxhi
14:22.16ryguyJan 12 09:21:07 ERROR[9120]: chan_zap.c:6204 mkintf: Unable to get parameters
14:22.16ryguyJan 12 09:21:07 ERROR[9120]: chan_zap.c:9141 setup_zap: Unable to register channel '25-47'
14:22.31ryguyanyone have any ideas why that would happen?
14:22.41iCEBrkrryguy: Sounds like your zaptel modules aren't installed.
14:23.00ryguyhow do I install them?
14:23.09iCEBrkrmodprobe <whatever>
14:23.09ryguyi did a make install in the zaptel configuration
14:23.11wankelmodprobe if they're already build and installed
14:23.12iCEBrkrok.
14:23.14ryguyas well as in the libpri
14:23.30iCEBrkrYou still need the correct interface module for it.
14:23.49iCEBrkrModule                  Size  Used by    Not tainted
14:23.49iCEBrkrwcfxo                   8736   1
14:23.49iCEBrkrzaptel                216192   6  [wcusb wcfxo]
14:24.02iCEBrkrI guess I don't need the wcusb in there.
14:24.31wankelryguy: lsmod will show you that list if you want to see what's loaded
14:24.32iCEBrkrryguy: There's some help in the README in the zaptel directory, I believe.
14:24.35iMaxI have a question: is it possible to tell asterisk not to use it's ip, but the hostname in SIP INVITE's, like user@mydomain.com?
14:25.10ryguyicebrkr, readme is useless
14:25.16iCEBrkrhang on, hang on
14:25.30ryguywct4xxp                49824   0
14:25.30ryguyzaptel                179392   0  [wct4xxp]
14:25.31iCEBrkrNo it's not
14:25.45ryguydoes libpri need to be in there?
14:25.55iCEBrkrryguy: So you have a TE400P card installed?
14:25.59wankelit's a library, not a module. :)
14:26.01wankelso no
14:26.06*** join/#asterisk ellvis (~ellvis@83.103.31.162)
14:26.19ryguyicebrkr, i have one of the 4port t1 cards, i think that is the one
14:27.04ryguy<PROTECTED>
14:27.04ryguyJan 12 09:26:44 ERROR[9183]: chan_zap.c:6204 mkintf: Unable to get parameters
14:27.04ryguyJan 12 09:26:44 ERROR[9183]: chan_zap.c:9141 setup_zap: Unable to register channel '25-47'
14:27.04ryguyJan 12 09:26:44 WARNING[9183]: loader.c:345 ast_load_resource: chan_zap.so: load_module failed, returning -1
14:27.04ryguy<PROTECTED>
14:27.04ryguy<PROTECTED>
14:27.38iCEBrkrryguy: http://www.digium.com/downloads/hw_article
14:27.40iCEBrkrSee if that helps.
14:28.02ryguyit was actually working before i just did the asterisk upgrade...
14:28.10iCEBrkrhttp://www.digium.com/index.php?menu=documentation
14:28.22*** join/#asterisk anasir_ (~anasir@h-69-3-186-162.snvacaid.covad.net)
14:29.41*** join/#asterisk zwi (~zwi@ewa-denver.com)
14:31.02*** join/#asterisk wankel (nobody@ohno.mrbill.net)
14:31.31iCEBrkrIt appears if you record a call and it's a long call... It takes a while for the channel to hangup.  Of course I'm using sox to compress/convert the 2 files into a mp3.
14:31.39iCEBrkrIs there anyway around this?
14:31.57*** join/#asterisk zno (~chatzilla@160.79.174.102)
14:32.03iCEBrkrI tried putting Hangup() before the channel-cleanup macro
14:32.13bjohnsonfor anyone who cares .. the $10 coupon for * users still works at voxilla (this has been a public service announcement)
14:32.14iCEBrkrbut then that hoses the cleanup process.
14:32.31iCEBrkrbjohnson: ha nice
14:32.59Ron-NaIAXy config:   example dialplan says: exten => 1234,1,Dial(IAX2/asteriskhosk.company.com@iaxy/s  How does that translate to my situation: 623 is the extension number, voip.abc.com is the host, context=IAXy-623 ????
14:34.39*** join/#asterisk stickynomore (~jeff@nsc66.147.11-46.newsouth.net)
14:35.12iMaxIs it possible to tell asterisk not to use it's ip, but the hostname in SIP INVITE's, like user@mydomain.com?
14:35.26iCEBrkrRon-Na: I'm no expert at any of this, mostly trial and error, but I'm wondering why you're using IAX for local extensions?
14:36.18Ron-Nabecause it is connected via IAXy !!!
14:36.45iCEBrkrRon-Na: exten => s,1,Dial(SIP/623,20,tr)
14:36.48iCEBrkrhrrm.
14:37.03iCEBrkrSee, I dunno what I'm talking about :P
14:37.41*** join/#asterisk ariel_ (~Ariel@199.125.186.218)
14:38.06bjohnsonRon-Na: use this exten => 623,1,Dial(IAX2/IAXy-623,15,Ttm)
14:38.11wankelicebrkr: probably, you can replace the sox binary with a script that does the compression in the background
14:38.27wankelor if you can specify the sox command line, run it in a background shell
14:38.32Ron-Nabjohndon thanks
14:38.35iCEBrkrwankel: like fork it? sox (...options...) &
14:38.39bjohnsonRon-Na: and put any username/secret needed in the IAXy-623 section in iax.conf
14:38.53wankelyeah.  if it's delayed because it's waiting on the compression, i assume that would help.
14:38.55bjohnsonRon-Na: that is for a 15 second maximum ring time
14:39.07wankeli've never used that feature, though, so i'm guessing.
14:39.09bjohnsonRon-Na: read the dial wiki page to see what Ttm means
14:39.16iCEBrkrwankel: Yea, it waits until the encoding/merging is done then moves on to the hangup() priority
14:39.21bjohnsonRon-Na: or at CLI type show application dial
14:40.10iCEBrkrwankel: The record() feature is kinda weird.  I'm only tinkering with it for those speical 'customer care' calls. :P
14:40.35iCEBrkrYa know, the ones used for 'training purposes'...
14:40.38iCEBrkr..Ha! Yea, right.
14:41.15iCEBrkrha
14:41.21*** join/#asterisk kludgebox (~bob@12.171.178.194)
14:41.56iCEBrkrbjohnson: There needs to be Dial(device,20,stfu) for our receptionist extension... *joke*
14:42.27bjohnsonthere is an 'S' but no 'u'
14:42.31*** join/#asterisk deathtrip (~deathtrip@mt.24.171.64.117.charter-stl.com)
14:43.02*** join/#asterisk firstsword (~root@host6614613596.biz.tor.fcibroadband.com)
14:43.11deathtripheya,  i was wondering.. will Asterisk work with voip providers, such as Vonage, and 8x8 ??
14:44.16iCEBrkrdeathtrip: no! stay away from vonage! :P
14:44.23deathtripi have 8x8
14:44.42iCEBrkrdeathtrip: Never heard of them.
14:44.48deathtripPacket8
14:45.03scrubbnever got Packet8 to work for me.
14:45.20iCEBrkrdeathtrip: NuFone, Voicepulse, iConnectHere and Broadvoice are the ones I know of.
14:45.28scrubbdeathtrip: asterisk will work directly with Broadvoice.
14:45.32scrubbI use that.
14:46.01deathtripi have their server, works for me, but i have to use their DTA310 adapter, and i am wanting to go digital, or something inside the office
14:46.37deathtripi only pay 19.95 a month for Packet8, for everything :/
14:46.45*** join/#asterisk ToyMan (~stuq@204.8.82.238)
14:46.51FaithXIs allow=ulaw accecptable in modem.conf ?
14:46.58iCEBrkrdeathtrip: I use Voicepulse Connect, pay as you go.
14:47.10FaithXthe sample doesn't mention
14:47.16iCEBrkrdeathtrip: of course I pay $9/mo for a DID
14:47.21bjohnsondeathtrip: I am told it will not work with Packet 8
14:47.41bjohnsondeathtrip: err .. I guess it is that Packet8 will not let * use Packet8
14:47.52deathtripbjohnson: heh, poopy
14:48.00bjohnsondeathtrip: although * will work with hundreds of other voip providers
14:48.02scrubbpacket8, vonagage and some other big players specifically do not want * working with their gear.
14:48.22FaithXHeh
14:48.24deathtripwhat about Charter's VoIP ?
14:48.44FaithXis broadvoice any good I am thinking of signing up?
14:48.52bjohnsoniCEBrkr: the 1001 is only one fxs isn't it?
14:49.06iCEBrkrbjohnson: Yeah
14:49.12scrubbFaithX: I have been delighted with their service and their BYOD pricing.
14:49.18iCEBrkrAll I wanna do is get rid of my BT100 and use a normal phone.
14:49.27bjohnsoniCEBrkr: I read something about a firmware upgrade to make the 1000 series accept 2 calls .. but I think it was still over 1 fxs port
14:49.29iCEBrkrI'm tired of being attached to a cord
14:49.35ctooleyI'm getting a really weird SIP message now.  "Line used remotely"
14:49.36deathtripthnx for the help guys :) ni ni
14:49.36bjohnsonbut the 1001 has builtin router I think
14:49.48FaithXscrubb so no major conjestion and ping time is ok?
14:50.03iCEBrkrbjohnson: I already have a SPA-2000, and I bought that just to tinker with extensions.
14:50.12iCEBrkrI dont' see a need for 2 phones.. in my tiny apartment.
14:50.13scrubbFaithX: you can experiment with their BYOD pricing for like $6 /mo no contract or disconnect fee and bump up to unlimited world plan and back down any time via their web interface.
14:50.18sambaldoes someone know if it's possible with a grandstream 486 to still receive incoming calls on pstn?
14:50.22sambaland route outgoing calls over voip?
14:50.24bjohnsonFaithful: I've been looking at broadvoice too .. I love the unlimited plan concept but suspect there really is a limit somewhere
14:50.28scrubbFaithX: no network issues for me.
14:50.49sambalor is it possible on sipura/
14:50.49FaithXscrubb, ok sounds ok to me
14:51.00FaithXscrubb, did you get an adapter?
14:51.18FaithXscrubb, and how long to activate?
14:51.29iCEBrkrbjohnson: I only need a Sipura to get an analog phone onto my network. So the bare minimum will do. :)
14:51.30scrubbFaithX: not a lot to loose if you do a byod.  No, I signed up BYOD and have my own * box.  I use a iaxy on my cordless and it works great.
14:51.31ManxPowerI think we should petition kram to remove the chan_modem* modules from Asterisk
14:51.35bjohnsonsambal: it is possible with a SPA 3000
14:51.37lelescrubb where you looking fro me?
14:51.44sambalbjohnson: ok, but not with grandstream?
14:51.50*** join/#asterisk UPMeduardo (~UPMeduard@tauro2.dit.upm.es)
14:51.50sambalbjohnson: because i already have a grandstream unit :(
14:51.53scrubbFaithX: a few minutes to start up with BYOD otherwise you wait for their unit.
14:51.57bjohnsonsambal: does the 486 have both fxo and fxs ports?
14:52.03sambalyes
14:52.07sambalbut fxo for backup
14:52.17bjohnsonsambal: I don't have a 486 (but am accepting donations)
14:52.18ManxPowersambal, GS did not have that feature the last time I heard anyone asking.
14:52.30sambal:(
14:52.33bjohnsonone other thing about Broadvoice ..
14:52.37sambali guess it can't be done by firmware upgrade
14:52.41bjohnsonespecially for you iCEBrkr ..
14:52.42FaithXbjohnson, Yes?
14:52.52sambalor maybe they can. :0
14:52.59bjohnsona voxilla  purchase will give you a free month of bv
14:53.05sambalbut if you pull the powercord out of the  unit you hear a click, switching to pstn
14:53.09*** part/#asterisk mhe (~mhe@p50849B79.dip0.t-ipconnect.de)
14:53.12FaithXvoxilla?
14:53.15ManxPowersambal, Since the hardwae/software is closed source, we don't know if the hardware supports it and GS just don't have the feature in their firmware or if the hardware can't do it.
14:53.26*** join/#asterisk mhe (~mhe@p50849B79.dip0.t-ipconnect.de)
14:53.37bjohnsonsambal: it may just be for emergency backup
14:53.57sambalI think it's the last time i bought any grandstream products
14:54.01bjohnsonFaithful: yes .. it's a web store for voip stuff
14:54.04iCEBrkrbjohnson: Yea.. I think I still have my registration from when I bought my SPA-2000
14:54.11sambalbetter spent a few more dollars and get decent hardware :)
14:54.14bjohnsoniCEBrkr: and don't forget the $10 oupon
14:54.32ManxPowersambal, Their history of cheap, but limited products did not make you think twice about buying GS?
14:54.43bjohnsonsambal: I love my Sipura's .. and they are cheaper than GS
14:54.55iCEBrkrbjohnson: Actually, I think back then they were offering VoicePulse..
14:55.01*** join/#asterisk cluv (~trillian@209.169.111.95)
14:55.03FaithXsipura is dearer in AU
14:55.05sambalManxPower: I didn't know about their reputation when i just started with voip ;)
14:55.12bjohnsoniCEBrkr: they offer 3 now .. bv, vp, and iconnect
14:55.15ManxPowerI think of SIPura as "what GrandStream SHOULD have been"
14:55.29ManxPowersambal, always search the mailing list archives.
14:55.36iCEBrkrManxPower: hahah Sipura's are on 'roids!
14:55.38sambalyou have to learn once ;)
14:55.52sambaldoes the 2000 accept it aswell, or only the 3000?
14:56.02bjohnsonthe 2000 is 2 fxs ports
14:56.09sambalah
14:56.09ctooleyHow do i check out CVS from a specific day?
14:56.10bjohnsonhence it does not connect to the pstn
14:56.12iCEBrkrsambal: 2000 allows 2 analog phones.
14:56.22ManxPowerctooley, "man cvs"
14:56.22ctooleySIP isn't working with CVS from January 11th or 12th for me.
14:56.46bjohnsonsambal: the 3000 is one fxs and one fxo (and has auto failover connect when power goes out or no connection via sip)
14:57.21scrubbbjohnson: can the 3000's fxo be used as a sip channel even not during "failover"?
14:57.30bjohnsonsambal: the 3000 is perfect for a one line house
14:57.30FaithXcan I connect asterisk to broadvoice.com I am a little confused with the sign up.
14:57.39bjohnsonscrubb: definitely
14:57.51scrubbFaithX: yes, read about their BYOD program.
14:57.53D1ng0FaithX, yes
14:57.58bjohnsonscrubb: the firmware can do internal routing .. but I route through * at home
14:58.01scrubbbjohnson: cool. thanks :-)
14:58.11ctooleyWell, hopefully it works, these people aren't exactly thrilled that their phones aren't working.
14:58.19FaithXSo in the device section just put *?
14:58.39D1ng0FaithX, yes
14:59.00D1ng0FaithX, or other
14:59.24pointer-gaimI'd like to automate hourly tests of my incoming VoIP service (due to problems with it)....is there any easy way to automate said test.... ie call my voip # from a PSTN connection and detect that the call went through (error/return code/etc)?
14:59.58*** join/#asterisk freat (~freat@node-40242662.mdw.onnet.us.uu.net)
15:00.03bjohnsonso outgoing callerid is what is set by the initiating device unless it is overwritten correct?
15:00.04pointer-gaimthe only way I can think of at this point is to use one of my own lines to dial and check my log for the incoming call on the voip line
15:00.14*** join/#asterisk brady_V (~brady_V@207.197.184.103)
15:00.28znowhich debian do you guys use for your production system? woody or sarge?
15:00.30bjohnsonpointer-gaim: that would be one way
15:00.37pointer-gaimbjohnson: correct.  by the devices entry in sip/iax/etc.conf
15:00.44maruzafter MySQL RealTime: Everything is fine, asterisk hungs on ast_mutex_unlock(&mysql_lock); ..which can be the problem?
15:00.45pointer-gaimbjohnson: I was hoping for something a bit cleaner
15:00.46*** join/#asterisk af_ (~af@ip-148-227.sn1.eutelia.it)
15:00.51bjohnsonpointer-gaim: or an external system that you may have access to
15:00.51FaithXscrubb, so it is a dial in service too? In the US heh... won't do me much good.
15:01.14scrubbFaithX: dial in server?  didn't see that.
15:01.28bjohnsonpointer-gaim: I think you could call from voip to voip .. if no connection then no service
15:01.55freathey guys. any idea why bridging two IAX2 calls would cause choppy audio?
15:02.14bjohnsonpointer-gaim: calling out via voip would likely be a suitable test of voip reliability
15:02.18scrubbfreat: timing?
15:02.28freatin other words, if I conderence call two outside lines, we get those issues
15:02.44freatscrubb: how would I check into that?
15:02.54FaithXscrubb, The sign up asks me to select a number.
15:03.00pointer-gaimbjohnson: no, because when it's broken other users of the same provider can call me...just not people outside of that provider
15:03.01freatscrubb: IAX2 debug doesn't show any errors
15:03.02*** part/#asterisk Matrix_3033 (~berrayahk@82.101.128.6)
15:03.11bjohnsonFaithful: pick a number, any number
15:03.13scrubbfreat: do you have a hardware board in your box?
15:03.14iCEBrkrfreat: my first thoughts are bandwidth issues, but doesn't sound like that'd be an issue in your case. *Shrug*
15:03.19scrubbFaithX: pick one.
15:03.27bjohnsonFaithX: I think they only have US numbers
15:03.38freatnahh we've got plenty of bandwidth + QoS on IAX2 traffic
15:03.48freatno hardware board in the box
15:04.03freatI'll double check that ztdummy is loaded...
15:04.03bjohnsonpointer-gaim: ahh .. yeah you need to use pstn then I guess
15:04.06scrubbfreat: do you have a timing source?  zaptel, zaprtc etc?
15:04.30ManxPoweryou only need a timing source if you are doing IAX2 trunking
15:04.42freatscrubb: ztdummy will work right? or is that just for music...
15:04.53freatwe aren't doing trunking
15:04.54scrubbfreat: or meetme, thats what you are doing right?
15:05.47freator, it happens if I call in from outside (regular phone going through our VoIP provider) and hit an extension that calls a regular phone back over the VoIP provider
15:05.57FaithXHeh, I got a did in NY CiTy
15:06.03ManxPowerfreat, what codec?
15:06.10freatulaw
15:06.22ManxPowerfreat, Tried GSM?
15:06.36freatwe've got 3Mb up and down... why?
15:06.52freatalso, I've got QoS and bandwidth monitoring (ntop) in place
15:07.09scrubbfreat: are you transcoding and does your box have the horsepower?
15:07.20*** join/#asterisk amir (~amir@shield.guindehi.ch)
15:07.26fantomax1Warning : RTP too short , what does it mean as Asterisk error ?
15:07.37freat2GHz celeron, 512MB DDR dual channel
15:07.38iCEBrkrfantomax1: Don't want no short short man.
15:07.45freatraid 1
15:07.46ManxPowerThat was not my question
15:07.54RoyKRAIF
15:07.56*** join/#asterisk gabb0 (~gabe@indo1.indosoft.unb.ca)
15:08.03RoyKredundant array of inexpensive floppis
15:08.05freatI watched the cpu via top while doing it and it was at 3%
15:08.06RoyKs/is$/ies/
15:08.36gabb0hello all.
15:08.52iCEBrkrRoyK: USB-Hub + 6 USB 1g JumpDrives. :P
15:09.07freatscrubb: fyi ztdummy is loaded (checked with lsmod)
15:09.08gabb0anyone here using FC3 with stable???
15:09.09ManxPowerwhat's -r?
15:09.18*** join/#asterisk _RaYmAn_ (user@213.237.12.147.adsl.vby.tiscali.dk)
15:09.19iCEBrkrRAID'd USB drives. that'd be a funny project.
15:09.31scrubbfreat: you got me then.
15:09.42bjohnsongabb0: yes
15:09.59firstswordHi all
15:10.03iCEBrkrgabb0: I'm still trying to figure out what the difference is between FC2 and FC3 and Asterisk?  I'm running Asteriks on FC2 and it works fine.  Then again, I'm not using CVS release.
15:10.12freatiCEBrkr: http://ohlssonvox.8k.com/fdd_raid.htm
15:10.28iCEBrkrfreat: hahaha
15:10.29bjohnsoniCEBrkr: I'm running stable on both
15:10.35brady_VHello all
15:11.05iCEBrkrbjohnson: I had a bunch of issues compiling whatever I pulled outta CVS but 1.0.3 compiled just fine on FC2
15:11.09freatscrubb: thanks for trying
15:11.11RoyKiCEBrkr: heh. and then, run RAID-0
15:11.12gabb0bjohnson, anything I may be missing or should do differently.  loading zaptel modules is a little screwy but my big issue is no audio with playback, background, etc
15:11.23scrubbfreat: sorry i couldnt help more.
15:12.05ManxPowerfreat, running any RAID?
15:12.18freatyes software RAID 1
15:12.31ManxPowerfreat, So no hardware raid cards?
15:12.46freatunfortunately no... cheap bastards
15:13.01freatI asked for a 3 ware
15:13.07Nugget3ware is doubleplusgood.
15:13.08brady_VDuring a make install of Asterisk-Stable, I recieve a "make: *** [asterisk] Error 1",  the cause seems be from "/usr/bin/ld: cannot find -lssl, collect2: ld returned 1 exit status."  Any suggestions or FAQs would be appreciated.  Thanks.
15:13.15gabb0bjohnson, I suppose I should I've checked the wiki and google
15:13.20sambalbrady_V: install libssl
15:13.21Nuggetbrady_V: install openssl livs.
15:13.24Nuggeter, libs.
15:13.27freatgoogle google google
15:13.31freatI made you out of clay
15:13.35brady_VThanks.
15:13.43freaterr.
15:14.05freatyeah I had heard of some RAID cards maybe causing problems I guess...
15:14.16Nuggetbkw was made from one my ribs.
15:14.18sambal3ware runs great under linux
15:14.31Nugget3ware runs great under windows and freebsd too.
15:14.35sambal:D
15:14.40sambalpromise also, but is cheapass :)
15:14.47Nuggetpromise is awful shit.
15:14.48ManxPower~useful asterisk docs
15:14.49jboti heard useful asterisk docs is it has been said that useful asterisk docs is (BOOKMARK THEM!): http://www.digium.com/index.php?menu=documentation (look at the "Unnoficial Links") and http://www.voip-info.org/wiki-Asterisk (the Wiki), and http://www.fnords.org/~eric/asterisk (ManxPower's site), and http://asteriskdocs.org, also, read all files in /usr/src/asterisk/doc
15:15.26Nuggetthe data will survive with promise, but the box isn't likely to stay up if a drive fails.
15:15.33bjohnsongabb0: I didn't have any playback issues until I tried mpg321 .. get rid of it
15:15.49brady_VUseful for me, I assume? :)
15:16.04FaithXWhat the "Regulatory Recover Fee" about?
15:16.23bjohnsonFaithX: I don't know
15:16.32bjohnsonFaithX: some kind of fee?
15:16.51scrubbFaithX: taxes I'm sure.
15:16.55ManxPowerFaithful, That usually means "we are squeesing moe money out of you, but want to blame it on someone else"
15:17.16gabb0bjohnson, why is mpg123 such a pain in the ass lately, it was "ok" before.
15:17.22ManxPowerFaithful, Sounds like they are adding a fee to recover the cost of complying with regulations
15:17.36ManxPowergabb0, I wasn't aware that there is a problem
15:17.37scrubbThe US is struggling to figure out how to get money back with people jumping of regulated telco's onto voip providers.
15:17.53bjohnsonFaithX: you're in AU right?  I have a friend there but says voip isn't really an advantage if not much LD due to telco pricing model (and if only a one line system)
15:18.04*** join/#asterisk W1thdraw (~W1thdraw@ip68-5-125-44.oc.oc.cox.net)
15:18.16bjohnsongabb0: NOTE: mpg321
15:18.24bjohnsongabb0: 321
15:18.24FaithXbjohnson, ???
15:18.26gabb0ManxPower, bjohnson just mentioned that he didn't have any issues until he used it is all.  I've always had no probs with it.
15:18.37ManxPowerscrubb, I would describe it as "the govt and the telcos are trying to save their revenue stream"
15:18.41FaithXmy phone is $2K per year
15:18.52gabb0bjohnson, oh, sorry my mistake.  it isn't there.
15:19.14ManxPowergabb0, he said he had problems when he DIDNT use mg123
15:19.22freatManxPower: what's interesting, is it seems to only be if the call went in and out via our VoIP provider
15:19.30FaithXAnd I have lot's of friends OS
15:19.32*** join/#asterisk BurnedOutGeek (~BurnedOut@216.215.202.4.nw.nuvox.net)
15:19.36bjohnsonFaithX: home use?  a lot of LD?
15:19.38ManxPowergabb0, Many distros rename mpg321 to be mg123, cvausing even more confusion
15:19.39gabb0ManxPower, yeah, I just appologized for that
15:19.48freatManxPower: if I connect with iaxcomm and then dial out, the quality is fine
15:19.51*** join/#asterisk tavux (~joshua@200.49.156.89)
15:19.54tavuxhi people
15:19.56ManxPowerfretIt should work.  Since it's not you have to look at odd issues.
15:20.01FaithXbjohnson, I run a business from home ;-) linuxterminal.com
15:20.04gabb0ManxPower, yeah, already looked.  It's not there.
15:20.12tavuxbjohnson hi friend
15:20.23bjohnsontavux: como usted
15:20.24FaithXpbx too soon when I get a good handle on it
15:20.42tavuxbjohnson i'm fine.. you?
15:20.52gabb0ManxPower, what else typically causes background, playback, voicemailmain, etc to not play the audio.  It says it plays it on the console and there are no errors in the logs?
15:21.03bjohnsontavux: ok.
15:21.24ManxPowergabb0, There is no "typical" for that problem.  It could be a zillion different issues.
15:21.30ManxPowerThe first thing to look at is NAT
15:22.06gabb0ManxPower, i can't count how many installs I've done but I've had this issue.  although this is my first install with FC3 so I'm wondering if it isn't something with that that I've missed
15:22.21gabb0shouldn't be NAT, on the same internal network
15:22.26brady_Vjbot: all links have been bookmarked, thanks for the info.
15:22.38bjohnsongabb0: I didn't do anything special for my home fc3 * system
15:22.53FaithXbjohnson, It's going to reduce my land line phone calls to $30/m
15:23.01bjohnsongabb0: confirm you have the sound files?
15:23.07ManxPowergabb0, Do you have a bindaddr= entry in sip.conf?
15:23.42gabb0bjohnson, yes, they are there and have read permissions.  you didn't do anything different?  make linux26 under zaptel?
15:24.04FaithXbjohnson, my land line / LD >$100/m
15:24.06bjohnsongabb0: I did make linux26 .. but I also did that on my fc2 box
15:24.22gabb0right
15:24.38bjohnsonFaithX: does bv terminate to AU?
15:24.44gabb0ManxPower, yeah, all 0s for now
15:24.48*** part/#asterisk humberto (~hav@201.128.177.84)
15:25.05hohumwould anyone recommend a particular softphone for me to use with Asterisk?
15:25.28ManxPowergabb0, just comment it out to be sure.
15:25.35bjohnsonhohum: generally .. none
15:25.41gabb0hohum, not xpro if you are looking to use g729
15:25.49bjohnsonhohum: but we all started out that way .. linux OS?
15:25.52*** join/#asterisk tim27 (~tim27@97-70.dr.cgocable.ca)
15:25.58tim27hello everyone
15:26.10tavuxsomeoune know how i can configure de asterisk voicemail for it send the messages via email
15:26.18*** join/#asterisk mutilator (~animenodv@65.111.201.79)
15:26.18hohumwell I tried xten lite but I can't keep a conversation going with it
15:26.22bjohnsontavux: read voicemail.conf
15:26.24tim27what is the default password to unlock cisco 7960 i dont remember it
15:26.32hohumit drops large chunks of audio out of my conversations
15:26.34sambalhohum: should work
15:26.40bjohnsontavux: you just specify a email address for that vmail box
15:26.45sambalhohum: check your connection
15:26.47*** join/#asterisk macmx (~mike@201.129.119.248)
15:26.56cluvtim27: abc
15:27.04hohummy connection is fine, this is lan<->lan
15:27.08hohum100Mbit
15:27.21hohumshould be plenty of available bandwidth for a 12Kbit/sec call
15:27.34bjohnsonhohum: there is a win iax client called firefly or something that should be easier to setup with NAT systems
15:27.36gabb0hohum, in the audio settings, make sure that the send silence is set
15:27.58tim27cluv: dont seem to work
15:28.04bjohnsonhohum: it may be a latency or lost packets problem .. try a different codec
15:28.06iMaxIs it possible to tell asterisk not to use it's ip, but the hostname in SIP INVITE's, like user@mydomain.com?
15:28.07tim27if i remember it was number
15:28.14gabb0hohum, transmit silence = yes
15:28.24*** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com)
15:29.18tim27what is the default unlock password for 7960 phone
15:29.26hohumwell that's only marginally better
15:29.56gabb0hohum, what codec are you using
15:30.05hohumG711
15:30.14hohumor 729
15:30.15hohumforget which
15:30.23hohumwhichever the maximum call compression is
15:30.34mutilatorhey guys, anyone recommend me some good tape software? for win & lin
15:30.38gabb0hohum, there isn't 729 support in xlite
15:30.38iMediaxg729 requires a license...
15:31.16gabb0use g711.  on an internal network, bandwidth isn't an issue
15:31.27gabb0at least to try it
15:31.40ManxPowertim27, 88#
15:31.45ManxPower..er... **#
15:32.01ManxPoweror is it ##*  I an never remember
15:32.15tim27**# is for
15:32.17tim277905
15:32.23RoyK**#*#*#####***#*##*#
15:32.26tim27or earlier version
15:32.30*** join/#asterisk _DAW (~bob@cable-24-158-215-248.sli.la.charter.com)
15:32.37tim27of sip before 4.1 on 7960
15:33.43Connor-anyone have a easy way to provision sipura2000's ?
15:34.22eKo1Connor-: you mean, provision many SPA 2000s easily
15:34.29Connor-eKo1 Yes
15:34.32fantomax1hi all again
15:34.49fantomax1is CentOS 3.3 ok for installing Asterisk ?
15:34.56eKo1Connor-: no clue.
15:35.17*** join/#asterisk ariel_ (~Ariel@adsl-070-147-214-250.sip.mia.bellsouth.net)
15:37.19*** join/#asterisk Zeeek (~Zeeek@80.125.80.38)
15:39.32*** join/#asterisk Darwin35 (~Darwin@c-24-3-241-22.client.comcast.net)
15:39.42cluvfantomax1: No problems with CentIS I use it
15:40.35Darwin35bkw  whenis the next stable relase going to come out with working g.729 on bsd
15:40.56Darwin35my boss wont let me use cvs
15:41.47Darwin35and we have 6 g.729 licenses
15:42.49tim27who know the password to unlock 7960 phone
15:43.09freathellllp
15:43.21eKo1tim27: try cisco $uxs
15:43.36freatfor some reason our * box is not answering the line... in the console it looks like it is answering but then it disconnects
15:43.41*** join/#asterisk erncic (~eankele@c-24-8-130-205.client.comcast.net)
15:43.50freatby 'line' I mean incoming IAX2 from teliax
15:43.56gabb0ManxPower, bjohnson, the "no audio" (with background, playback, etc) i am having has to do with the zaptel drivers or at least wct1xxp.
15:44.09gabb0I ran * without the driver loaded and it works fine
15:44.40gabb0just to let you know if you were interested
15:44.42freatit is seeing the call, and routing it to the autoattendant and everything, but doesn't pick it up from the party calling in's perspective
15:44.57freatand then * hangs up
15:45.26firstswordHi, I have a question on variables in extensions.conf..  I want to make a variable to keep track on the # of concurrent calls .. but it seems like every calls uses its own variables, and they're not related to each other
15:45.31firstswordam i right?
15:46.07gabb0firstsword, check out setgroup applications.  I think this is what is now be used by many
15:47.06firstswordis setgroup for pstn?
15:47.18gabb0firstsword, it's for any type of tech
15:47.31fearnoris anyone working on t.38 bounty?
15:47.43gabb0firstsword, http://www.voip-info.org/wiki-Asterisk+cmd+SetGroup
15:47.56fearnornamely, is coppice working on t.38 integration :)
15:48.23EricirEfearnor: hahah if he is i think it's passively
15:48.54*** part/#asterisk alakdan (~dax@210.213.170.201)
15:49.46sungfearnor
15:50.03fearnorsung!
15:50.12sungi want to make love to you
15:50.16bjohnsonConnor-: there isn't an easy way.  I think they support tftp but they require a binary file to dl
15:51.05bjohnsonConnor-: submit a request to Sipura to create a save config option in the firmware to create a file that could be used for that (ie config one device, save config, tftp update all other devices)
15:51.31bjohnsonConnor-: they seem to add features to firmware from time to time .. maybe if we bug them about this they'll add it
15:51.59bjohnsonConnor-: also mention to add an easier way to forward SIP calls to asterisk WITHOUT first picking up the line
15:52.09fearnorsung: i'm taken! :P
15:52.13bjohnsonConnor-: you can do it .. but it's a complex comfig
15:52.34*** join/#asterisk evills (~ellvis@83.103.31.162)
15:53.02bjohnsongabb0: are you sure it is the stable *?
15:53.35gabb0yup, v1-0 is what I checked out of cvs
15:53.42*** join/#asterisk zwi (~zwi@ewa-denver.com)
15:54.07ArkyLadygotta love ebay :)
15:54.11firstswordwow. great!~ checkgroup works!! thank you gabb0~~~ i was think that I need to write an agi or something! haha
15:54.13*** part/#asterisk cluv (~trillian@209.169.111.95)
15:54.26gabb0no problems.
15:54.38ArkyLadyI guess I need to buy me a broadband router, this shitty old hub isn't going to work for me
15:55.09*** part/#asterisk Hogie (hogie@69.56.194.174)
15:56.44*** join/#asterisk mhnoyes (~mhnoyes@user-2ivfj4m.dialup.mindspring.com)
15:56.58eKo1I recommend a D-Link
15:57.00*** join/#asterisk wolfson (~hehe@65.174.122.198)
15:57.26petemcthe d-link i had was shite from hell
15:57.37eKo1really, mine works like a charm
15:57.46ArkyLadyI'm just now debating in my head over using a d-link
15:57.48eKo1and i even have a sip phone on it
15:57.48petemci had the g504t or something
15:57.54ArkyLadyI had a d-link hub a long time ago that was shit
15:58.04petemca linksys with the modified firmware to do traffic shaping is a better bet
15:58.09eKo1funny enough i didn't have to mess with nat at all to get it working
15:58.13ArkyLadythey are cheaper though, but think I'll go with a linksys
15:59.00eKo1well, honestly i've had problems with both d-link and linksys
15:59.09bjohnsonArkyLady: buy 2 of whatever is less than $20 .. replace it if it's shitty
15:59.16ArkyLadyhehe
15:59.19ArkyLadythere's an idea :)
15:59.30petemcthe wrt54g is 40 GBP
15:59.33petemchella cheap
15:59.42*** join/#asterisk ard (ard@gw-uunet-office.telegraaf.net)
16:00.07petemceKo1: so the d-link, did it work like a charm or did you have problems with it, you said both :)
16:00.11bjohnsonthey had a few wifig routers on sale at xmas for $40 CDN
16:00.24eKo1petemc: well, the one i have on my desk works like a charm
16:00.35bjohnsonpetemc: I've had probs with dlink, with linksys, with smc .. you name it
16:00.43bjohnsonpetemc: they're not immune
16:01.04*** join/#asterisk Casper_UA (~casper@ragu.bestnet.kharkov.ua)
16:01.09petemctheres nothing bullet proof, certainly not consumer electronics
16:01.10bjohnsonpetemc: if you buy 1 for $100 and it ever has problems, you got nothing
16:01.29bjohnsonpetemc: if you buy 2 for $20 .. if you have probs .. you have an immediate backup
16:01.42petemcright
16:01.58petemcso a few cheap connexant based routers
16:02.25petemcbut like i said, the modified firmware for the wrt54g makes it worth the little bit extra
16:02.49czerowaht does the mod firmware do ?
16:02.52eKo1where can i get this modified firmware?
16:02.57bjohnsongoogle
16:03.01petemcits called satori
16:03.18*** join/#asterisk Moc (~mochouina@64.235.210.66)
16:03.22bjohnsonit turns it into a linux router
16:03.25czeromornign moc
16:03.26Mocmorning
16:03.34bjohnsonqos, snort, etc
16:03.39petemcit already was a linux router, it adds functionality
16:03.45eKo1bjohnson: so basically you're putting linux on it
16:04.09petemcthe d-link g504t runs linux too, its just not as easy to hack
16:04.44eKo1really, looks like embedded linux is everywhere now
16:05.05*** join/#asterisk ckruetze (~ckruetze@62.214.134.0)
16:05.10tavuxhow i can configure asteriks voicemail service, for it send the messages via mail and it save a copy in the server
16:05.16ArkyLadymaybe I could just use a linux box as a router
16:05.20eKo1i would have thought vxworks dominated these devices
16:05.20ckruetzehi
16:05.35petemceKo1: the cheap connexants use vxworks
16:05.37Moctavux, just specify the email in the voicemail.conf
16:05.37czerovxworks = $ linux = free
16:06.19eKo1vxworks has a smaller footprint though
16:06.57tavuxMoc 100 => 100,Gustavo,puffyx@hotmail.com,,attach=yes
16:07.30tavuxserveremail=puffyx@hotmail.com
16:07.41*** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.rr.com)
16:07.41*** mode/#asterisk [+o anthm] by ChanServ
16:07.54tavuxformat=wav49|gsm|wav
16:08.04*** join/#asterisk sektor195 (~please@216.86.45.98)
16:08.05tavuxsendvoicemail=yes
16:08.39Slainteanyone here use the polycom IP600 phones?  I have having a hell of a time getting them to register
16:08.52zoaanthmmmm
16:08.55tavuxthe vocemail save the messages in the asterisk box and i can listen it... but don't send me a email with the voicemsg
16:09.01zoado you have a second for me ?
16:09.22Slaintetavux,  change you voicemail.conf and have attch=yes
16:09.51tavuxSlainte i have it
16:10.04Slaintemake sure your server will send mail  so try this
16:10.04tavuxattach=yes
16:10.22Slainteecho test | mail whateveryoumail@address.is
16:10.33tavuxhow i can check if sendmail command send emails ?
16:10.50Slainteps -ef | grep mail
16:10.56Slainteis there a sendmail process?
16:11.04bjohnsonArkyLady: definietly you could use a linux box as a router
16:11.22tavuxSlainte i wanna use a smtp server in other box
16:11.40Slainteso you dont want sendmail to act as an MTA but as a mail forwarder
16:11.43bkw_kram
16:11.44tavuxi have'nt a mta daemon running in the asteriskbox
16:11.48bkw_tclark you alive?
16:11.59Slaintewell then how is it supposed to send mail?
16:12.20ArkyLadybjohnson that sounds like more fun than buying some crap off ebay hehe
16:12.30tavuxSlainte i wanna configure it... for use an other mailbox
16:12.40bjohnsonArkyLady: if you have extra hardware .. I recommend ipcop
16:12.46Slaintetavux,  configure sendmail to work as a forwarder
16:13.25bjohnsonArkyLady: if using you home desktop/linux machine (a little more understanding is rquired) .. I recommend the shorewall iptables configuration system
16:13.45tavuxif my asterisk is in 192.168.0.1 and my emailserver are in 192.168.0.2 i wanna that the asterisk box connect with emailserver and use this smtp
16:14.23bjohnsonSlainte: I don't think it has to specifically be a forwarder since it would try to send to hotmail.com directly anyway
16:14.25ArkyLadythanks for the info bjohnson
16:14.27*** join/#asterisk ariel_ (~Ariel@adsl-070-147-214-250.sip.mia.bellsouth.net)
16:14.59SlainteHe does not want an MTA though,
16:15.01eKo1tavux: make a script that logs into the email server and have it send the mail then
16:15.02bjohnsontavux: why?
16:15.17freatphew... calls are coming in again
16:15.32freatwasn't fun watching the console and seeing all the people trying to call us
16:15.37SlainteeKo is right.  If you dont have a local MTA you will need to run another agent/screipt that checks for mails files and then forwards them
16:15.38bjohnsontavux: it would just be for outgoing
16:15.47petemcsomething like ssmtp would work
16:16.02Slaintetavux  you can lock sendmail up real tight if you wanted
16:16.21*** join/#asterisk wolfson` (~hehe@65.174.122.198)
16:16.36*** part/#asterisk mhe (~mhe@p50849B79.dip0.t-ipconnect.de)
16:17.13eKo1that's stupid, might as well use sendmail as the mta then
16:17.22Slaintehe does not want to use it as an MTA
16:17.37Slainteif your core network outgoing mail is going to be filtered from a central point
16:17.43Slainteyou dont want rogue MTA's throught your network
16:18.15SlainteI think it is a good idea,  controlled, and functional
16:18.20eKo1it won't be rogue
16:18.39eKo1but yeah, i see your point.
16:18.57Slaintelet the mail servers do their job, and just forward the mail to them
16:19.25eKo1then you can have 500 sendmails forwarding to one mta so you have one central administration point
16:19.47Slaintecertainly,  with all the filtering, and spam control needed
16:20.15arrghthe real trick is capturing and redirecting rogue mta's
16:20.24arrghproxification to the maximification
16:20.25*** join/#asterisk samnjenga (Dap@217.21.113.190)
16:20.39ZeeekI love that description, 'rogue'
16:20.42freathehe
16:20.48samnjengaHi all
16:20.55samnjengaanyone see Coppice ?
16:20.59SlainteWell the way you capute rogue MTA's if very simple
16:21.05Zeeekservers aren't really rogues, the humans are
16:21.21Slainteyou can port forward all your 25 outbound traffic to a filterint agent/firewall
16:21.29arrghintentional or not, they're a pain in the ass.
16:21.40firstswordI have a question regarding integration my current door phone I have to the new asterisk box.  the door phone i have is panasonic KX-T30865, and it is currently connecting to KX-TD1232 Systems.  My asterisk box has a PRI for voice channels, and I will install a X100 onto it for FXO channel.  I wonder what I need to dial back to the door phone so that it will open the door for my customer
16:22.09Slaintefirstword,  I need to do the exact same thing.
16:22.37eKo1a sip door phone <--- there's an idea
16:22.44eKo1better yet, make it iax
16:22.50tzangereKo1: seems simple enough
16:22.58LoganIs there a good way to derive my sip.conf at runtime?
16:23.38*** join/#asterisk naif (~xokxokdfi@213.155.196.233)
16:23.41naifhi
16:24.10naifanyone have ever tryed a iaxy with a 220V adapter? I cannot find 220V adapter for italy
16:24.33firstswordslainte: so u currently have a pbx in ur office?
16:24.58bjohnsonfirstsword: if that phone is a digital phone .. your likely out of luck
16:25.09Slaintefirst,  no I have it for a customer
16:25.22*** join/#asterisk echinos (~echinos@guam.island.ca)
16:25.31firstswordbjohnson: that is a digital phone
16:25.41firstswordbjohnson: digital door phone..
16:25.49SlainteI was just going to setup an extension on the FXO channel
16:26.08echinoson asterisk.org, there's a `supported hardware page'... it lists `any ALSA full-duplex sound card. What would you use the sound card for?
16:26.18*** join/#asterisk wolfson (~hehe@65.174.122.198)
16:26.34Nivexechinos: I hope bkw didn't hear you ask that...
16:26.36Nivex;)
16:26.37bjohnsonfirstsword: likely it's a propreitary system .. can you send a command to the pbx/key system that would control that phone?  Maybe some kind of DISA system into the pbx?
16:26.42ArkyLadyhehe nivex
16:26.59echinosNivex: why?
16:27.16ArkyLadyhe likes to smack people
16:27.27*** join/#asterisk leandro_pt (~leandro@82.155.113.224)
16:27.29Nivexechinos: There are certain people who believe that sound cards have no business in a server.  You can, however, use the ALSA sound card as a channel for some rudimentary testing.
16:27.30echinosYa... I know. :)
16:27.37kludgeboxcan anyone point me to a modem for 911 use that they've used?  I know the chipset on voip-info.org, but a link to a particular one (newegg.com, mwave.com, whatever.. ) would be helpful (?)
16:27.45echinosHow do you connect it to a line?
16:27.57Slainteechinos,  I would not use up an IRQ with a sound card myself
16:28.08Nivexechinos: You don't.  You just use it with a mic and speakers.  Like I said, it's more for testing things like dialplans and stuff
16:28.38echinosAh, ok. So you can't use it do go to an extension on a phone line.. Ok, that clears that up.
16:28.47echinosI have no evil plans, I was just curious.
16:28.49eKo1echinos: you use it to play back phone sex recordings
16:29.14Nivexechinos: right.  With chan_alsa loaded, it enables the Dial and Hangup commands at the console.  That's about it.
16:29.15eKo1in 5.1 douby digital ac3
16:29.26echinosNivex: Sweet, thx.
16:29.28tavuxhow i can send and email with sendmail command ?
16:29.42NivexNow chan_alsa and chan_oss default to auto answer, so you can also conceivably use it as an overhead paging system.
16:29.48echinosI have a sound card in my * server with speakers to do the WHoIsIt callerid announce thing.
16:30.15echinosALthough festival can be rather hard to understand.
16:30.24Slainteno polycom users?
16:30.42tim27any know unlock password for 7960
16:30.45tim27phone
16:31.28ryguytry *##
16:31.39ryguya quick goole search had that
16:31.59kludgeboxtim27: is it cisco ?
16:32.02tim27yes
16:32.11tim27it not *##
16:32.17bjohnsonfirstsword: maybe this will point you in the right direction  http://www.voip-info.org/tiki-index.php?page=Macro%20to%20announce%20inbound%20call%20to%20Pana%20user%20or%20page%20all
16:33.17bjohnsonSlainte: setting up an fxo on a pbx extensions requires that the pbx support analog phones (not proprietary digital ones)
16:33.24tim27you have to press settings, unlock config option 9 ... and then it ask for password... if i remember right ... it a number pass
16:33.44bjohnsonfirstsword: does the panasonic that you have come with an ATA port
16:34.11*** join/#asterisk TheEmperor (TheEmperor@218.111.51.94)
16:34.15ryguyi just upgraded asterisk to the latest version and It broke.. anyone know how to fix this error:
16:34.16ryguy<PROTECTED>
16:34.16ryguyJan 12 11:30:22 WARNING[1701]: loader.c:440 load_modules: Loading module data_pgsql.so failed!
16:34.22bjohnsontavux: from command line just do sendmail user@host
16:34.24bkw_RECOMPILE
16:34.31bkw_:P
16:34.45ryguywhat do I do when I recompile
16:34.58bkw_data_pgsql.so <-- hasn't been recompiled
16:35.04*** join/#asterisk fishboy1669 (proxyuser@62.69.81.129)
16:35.08bkw_since your last install
16:35.10bkw_its very clear
16:35.12kludgeboxtim27: **#
16:35.16fishboy1669what version of sip does astrisk use 1 or 2?
16:35.43tim27bkw_ YOU that know everythings... what is the password to unlock 7960 phone
16:35.44eKo1ryguy: make clean; make; make install
16:35.50ryguybkw_, i didnt do the initial install, i dont even know what postgre does in the overall scheme of things, where do i find the source for data_pgsql
16:35.50bjohnsonechinos: I was trying to use a sound card in the asterisk server to have it feed MOH on another system from the same stream that asterisk was using .. hasn't worked yet due to mpg123 screwing up
16:36.10echinosMOH?
16:36.13tim27when you choose option 9 (Unlock Config) in the settings menu...
16:36.18bkw_ryguy you're using ast_data?
16:36.19NivexMOH == Music On Hold
16:36.21bkw_or did you remove it?
16:36.22*** join/#asterisk wolfson (~hehe@65.174.122.198)
16:36.24echinosAh.
16:36.40ryguybkw_ i know almost nothing about this, i dont know what ast_data is
16:36.48ryguyi took over the project
16:36.54bkw_rm -rf /usr/lib/asterisk/modules/data_pgsql.so
16:36.58firstswordbjohnson: im not sure.. i need to check
16:37.50echinosShould I now ask if I can use a full-duplex voice modem to connect my telephone line? ;)
16:37.50JerJerno
16:37.50echinosAwww.
16:37.50ryguywhat does the postgre module do for asterisk?
16:37.50JerJersegfault
16:37.51echinosI'm waiting to have the cash for the dev kit.
16:38.07*** join/#asterisk l3me (~root@212.87.11.234)
16:38.17eKo1hehe
16:38.40l3meeh..
16:39.10eKo1you 'shouldn't' be on irc as root
16:39.25l3meiknow...
16:39.30eKo1but then again, that may be a winblowz machine called root
16:39.54redder86does any windows user call their machine name root?
16:40.01l3memany people on this channel.. i thought that asterisk is rather RARE soft
16:40.07*** join/#asterisk nitram (nitram@superblob.com)
16:40.18eKo1RARE??
16:40.22l3mei've just installed it in my company.. before we used hardware routers + netup
16:40.27l3meyup - rare ;]
16:40.40eKo1asterisk is all over the place
16:40.48eKo1kinda like bittorrent
16:40.57Slainteonce people see how bad Cisco Call Manager is  it will only get bigger :)
16:40.58redder86It's not all over the place as much as AT&T, though.
16:41.15ArkyLadyhehe Slainte
16:41.24eKo1redder86: that's in the states
16:41.26ArkyLadydoes it suck?
16:41.32*** join/#asterisk Flatcat (~ScaredyCa@j27229.upc-j.chello.nl)
16:41.33*** join/#asterisk Syncros (~sysop@noc.routermonkey.net)
16:41.34l3mei've got a question.. does anybody know anything about netup ?
16:41.41l3me(netup.biz)
16:41.57SlainteI did a Survey on a company last week.  They had one of our competitors (my ex employer) install Call Manager.  They have to restart three services every morning manually
16:42.00redder86eKo1: call up an average telco and ask them if they know what an Asterisk PBX is.
16:42.28eKo1define 'average telco'?
16:42.39JerJerSlainte: and there are morons around this project that want to force asterisk to depend on a database, just like CCM
16:42.41redder86eKo1: you call 100 of them and then take an average
16:42.48vaewynAFKredder86: SBC around here does... and it was before I mentioned it even ;P
16:43.00l3mehey redder86 in my country there's so fucking crappy isp/telp that they surely don';t know
16:43.18eKo1redder86: might as well ask them if the know what a tdm400 is.
16:43.19redder86l3me: so "rare" is a fair description, then
16:43.22l3mei heard that in britain they are going to have 100% of voip in 2008
16:43.25SlainteJerJer,  exactly.  (Nothing wrong with an well designed SQL database for a larger company)  but these people are integration integration, and they dont know their head from their ass
16:43.46vaewynCrisco Call Mangler ;P
16:43.47l3meLet me ask you a question.... plz...
16:43.48redder86eKo1: okay so it's okay to call Asterisk "rare"
16:43.52eKo1you can't really compare at&t and * anyways. they are two different things
16:43.58Slaintehahahah.
16:44.06JerJerl3me: first don't IRC as root, then you can ask questons
16:44.22vaewynYou an compare nortel and * way more than AT&T and *
16:44.39l3megosh... i've got a crappy mashine
16:44.41l3mebrb
16:44.48ZeeekI'm root on windows
16:44.50redder86vaewyn: okay, so compare Nortel and *.  * is stil lrare
16:44.58Slaintel3me,  are you for real IRCing as root?  Guess its time to check the security on your asterisk box
16:45.02*** join/#asterisk mishehu (mishehu@cshells.shavedgoats.net)
16:45.03eKo1i saw a commercial yesterday about at&t and cingular? are they the same company now?
16:45.15redder86they're not rare
16:45.18bjohnsonJerJer: is there a way to have asterisk write to the ascii config files?  If there is, I maybe don't need to learn the db system
16:45.20vaewyneKo1: cingular purchased AT&T wireless
16:45.26Slainteek0  for cellular/wireless services
16:45.40eKo1that makes sense
16:45.54eKo1i hope those bastards upgrade to 3G soon
16:45.57*** join/#asterisk _Syncros (~sysop@noc.routermonkey.net)
16:46.17redder86Uh, what does AT&T have now, then, if they don't have wireless.  I thought that they sold off their landline business.
16:46.22*** join/#asterisk wolfson (~hehe@65.174.122.198)
16:46.37vaewynscrew 3G... I want basic data that doesn't cost $$$$$$/month  ;P
16:46.46eKo1redder86: eh, they own have the us backbone
16:46.55Makenshi3G is horrid
16:46.57eKo1s/have/half
16:47.07MakenshiI'm pinning my hopes in Wimax
16:47.25eKo1Makenshi: whatever, they need to upgrade
16:47.39redder86eKo1: yeah, looks like they still do ISP stuff, VoIP services, etc.
16:47.48vaewynI'm hoping for UWB to get FCC okied and catch on
16:47.50MakenshieKo1, who need to upgrade, and what do they need to upgrade?
16:47.50W1thdrawim gonna install asterisk on an xbox what distro should i use?
16:47.55*** join/#asterisk wojtekxz (~wojtek@212.87.11.234)
16:47.56czeroATT still runs backbone
16:47.58redder86telephones...
16:48.01redder86branding...
16:48.02czeroincl the backbone comcast runs on
16:48.13Makenshi3g is already old hat
16:48.24eKo1Makenshi: i want to stream live video from my cell phone
16:48.26redder86long distance provider...
16:48.28wojtekxztell me, what is the fastest way to implement a voip billing in my company?
16:48.39eKo1you can't do that in the US yet
16:48.49vaewynwojtekxz: pay someone for a solution already made..
16:48.57czerowojtekxz: what vaewyn said
16:49.10MakenshieKo1, even if they do, getting two endpoints in 3g coverage at the same time is no mean feat
16:49.12Slaintewoj,  billing is not to be taken litley
16:49.14wojtekxzgosh czero vaewyn i ask about software.
16:49.17bjohnsonwojtekxz: what czero said
16:49.21simongWhere has Meetme gone to in CVS ?
16:49.38bjohnsonwojtekxz: you asked quickest way .. not which software to use
16:49.43redder86simong: hopefully a better place
16:49.46wojtekxzwe begin providing voip but want to set up billing.. what is the best software ?
16:49.46eKo1my point is, the US is behind in this area
16:50.09czeroI'm not sure there is good open source billing
16:50.17echinosOk, now I have another question about the `supported hardware' list... the Generic X100P Clone links to a page with a voice modem. :/
16:50.19eKo1wojtekxz: you'll have to make your own i fear
16:50.19Makenshiwell, hopefully wimax will replace cellular transmission
16:50.19tim277960 default password is cisco
16:50.21czeroas for for non open source Ive only used what we developed
16:50.24echinosWsup with that?
16:50.34simongredder86: Ha ha
16:50.41bjohnsonwojtekxz: search the mail list archive for astcc
16:50.45redder86echinos: the X100P was a winmodem
16:50.47bjohnsonwojtekxz: and cdr
16:51.13eKo1i'm still developing the billing system here
16:51.16*** join/#asterisk florz_ (nobody@I8d46.i.pppool.de)
16:51.18echinosredder86: yeah...
16:51.19redder86echinos: it was called "X100P" when sold by Digium.  When sold elsewhere it was often called AMI-IA92/IE92
16:51.22eKo1what a mess
16:51.51*** join/#asterisk nitram (nitram@superblob.com)
16:51.55echinosredder86: So I can get a voicemodem with that chipset and connect a line in or out?
16:52.10redder86echinos: I don't understand your question.
16:52.32bjohnsonechinos: it would be a fxo port
16:52.41vaewynheck... billing systems are darn easy... dealing with CDR and a rate table is fairly trivial
16:52.48CaedeHas anyone ever tried using one of the SS7->ISDN PRI converters? (FSG, Groomer II or Teleprime SP201)
16:52.48vaewynjust have to make sure you get it right ;P
16:53.12eKo1vaewyn: yeah right. not with my business plan
16:53.35*** join/#asterisk ariel_ (~Ariel@adsl-070-147-214-250.sip.mia.bellsouth.net)
16:53.45echinosredder86: Basically, voice modems are not supposed to work, but lo and behold, a `generic X100P' *is* a voicemode.
16:53.51echinos+m
16:53.52*** part/#asterisk wojtekxz (~wojtek@212.87.11.234)
16:54.03bjohnsonanyone seen any broadvoice style unlimited plans from voip providers with CDN DIDs?
16:54.21vaewyneKo1: Do we need to have Brother Occam pay you a visit?  ;P
16:54.25bjohnsonechinos: exactly one voicemodem works
16:54.35eKo1Occam?!
16:54.35redder86echinos: a "voice modem" is a modem that you use "voice" standards to communicate with, like IS-101.  They don't work the way that you expect them to.
16:54.58echinosbjohnson: And that voicemodem is the one with the specific Intel chipset?
16:54.59czerovaewyn the getting it right is the hard part :)
16:55.05*** part/#asterisk oej (~oej@40.186.204.213.sol.worldonline.se)
16:55.07vaewyneKo1: Occam's razor... the simplest solution ... :P
16:55.07redder86echinos: the zaptel driver/wcfxo supports the Ambient MD3200 chipset, which is on some winmodems.
16:55.07bjohnsonechinos: yes
16:55.21redder86echinos: you do not use the "voice modem" driver with them, though.  You use wcfxo/zaptel.
16:55.23eKo1vaewyn: don't get it
16:55.31echinosAh, ok. Thanks guys.
16:55.56vaewyneKo1: If your business plan is that hard you may need to rethink it and find a simpler solution ;P
16:56.23*** join/#asterisk wolfson` (~hehe@65.174.122.198)
16:56.23tavuxhow i can configure the music on hold in a sip user ?
16:56.44eKo1vaewyn: it's as simple as it is going to get
16:56.49bjohnsonmy simple business plan is to get people to pay me in return for nothing .. but it has proven to be difficult
16:57.15eKo1bjohnson: that's not a business plan; that's called donations
16:57.29PTG123no one has touched chan_8track either
16:57.30PTG123heh
16:57.31bjohnsoneKo1: works for Red Cross
16:57.46bjohnsonhahaha..chan_8track
16:57.47redder86PTG123: ;-)
16:57.52Slaintehehehehe  sweeet
16:57.59Faithfulscrubb:  I need a bit of help setting up my dial plan for broadvoice... to dial australia, what would you dial?
16:58.02Slaintewas it the upgrade for chan_reel2reel
16:58.28bjohnsonFaithful: FaithX:  which one of you is actually from AU?
16:58.37*** join/#asterisk PoWeRKiLL (~PoWeRKiLL@ATuileries-152-1-9-44.w82-123.abo.wanadoo.fr)
16:58.49scrubbFaithful: uh, I've never called Australia.
16:59.04bjohnsonFaithful: you'd need the international code I think
16:59.07Faithfulhow do you dial LD from the US?
16:59.22scrubbFaithful: 1-xxx-xxx-xxxx
16:59.29bjohnsonFaithful: for a US number it is that ^^
16:59.39Faithfulhow about international?
16:59.40Slaintethats only for Canada/US/part of Carrib
16:59.42Slainte011
16:59.42bjohnsonfrom another country to US , append a 011 to the front
16:59.44scrubbFaithful: never dialed international.
16:59.54Slainte011-353  ireleand   011-44  england  etc
17:00.42FaithfulOh... we have 0011 hmm
17:00.49Slaintethats a problem
17:00.59Slainte001 is when you are calling in the UK out
17:01.01Faithfulso us is 0011 1
17:01.05tavuxwhere i configure the music on hold for sip users ?
17:01.11SlainteFaithfull where are you
17:01.16FaithfulAU
17:01.24Slaintetry 011 1
17:01.33Faithfulmini england that wants to be America
17:01.37Slainteso new york is 011 1 212 555 1212
17:01.57SlainteI mean 001
17:01.58Faithfulheh, I got a ny number now ;-)
17:01.59Slaintefeck
17:02.07Slainte001 1 212
17:02.11bjohnsonFaithful: according to my Bell phone book .. calling AU would be 011 + country code + city code + local number
17:02.25Slaintehe is in Ozz calling out
17:02.27postelFaithful: 555 is not a valid exchange, you DONT have a NY number ;)
17:02.32czerobjohnson correct from canada
17:02.36bjohnsonFaithful: where AU is 61
17:02.39Slainte555-1212 is valid for directory assitance
17:03.02bjohnsonFaithful: I think he is in AU trying to use US based Broadvoice to call AU
17:03.05bjohnsonoops
17:03.06*** join/#asterisk tdonahue (~tdonahue@haynesmail.haynes-group.com)
17:03.09bjohnsonSlainte: ^^
17:03.11Faithful212-202-0448
17:03.13postelSlainte: i thought 555 point back to nothing and only used in films just because of that
17:03.32Slainteif you dial area code - 555 -1212  it will give you directory assitance for that area code
17:03.47*** join/#asterisk ToyMan_ (~stuq@204.8.82.238)
17:03.48redder86yup
17:04.02tavuxsomeone has been configured the music on hold ????'
17:04.18Slaintetavux, there is TONNES on the wiki about music on hold
17:04.31redder86tavux: yes we have Rick Springfield configured
17:04.34redder86hehe
17:04.44bjohnsonewwww
17:05.00bjohnsonsuperfreak
17:05.04redder86hehe
17:05.08bjohnson(or am I thinking of someone else?)
17:05.44redder86he was the early 80's version of Clay American Idol person
17:05.52*** join/#asterisk frank_sbr (~frank_sbr@207.107.208.137)
17:05.56*** join/#asterisk sadie (~sadie@193.17.41.120)
17:06.13bjohnsonFaithful: does local calling in AU use city codes?  that seems to be the way outgoing to AU works from here
17:06.21*** join/#asterisk wolfson (~hehe@65.174.122.198)
17:06.24sadiecan two iax2 clients setup p2p connection if both are behind masquarade ?
17:06.39bjohnsonp2p?
17:06.48Faithfulyes
17:06.50redder86p2p and iax2 are protocol descriptions
17:07.09sadieFaithful: could you possibly point me to some how to on that ?
17:07.09redder86can someone speak french to someone who only speaks german?
17:07.15bjohnsonsure
17:07.15mishehu<PROTECTED>
17:07.48Makenshimishehu, but is it directly between them, or through a proxy?
17:07.50redder86bjohnson: and will the german understand?
17:08.02bjohnsonredder86: hell, even 2 english speakers can talk to each other without anyone listening
17:08.04mishehuMakenshi: directly
17:08.11bjohnsonredder86: (it's called politics)
17:08.13mishehuiax2 has no need for a proxy.
17:08.19redder86p2p implies that there is no server involved
17:08.42mishehuredder86: actually, p2p implies that there is no centralized system.
17:08.53Makenshimishehu, but if neither can receive incoming connections, how is the connection established?
17:08.57redder86mishehu: and the difference is?
17:09.00sadiewhere can I read about the way this is accomplished
17:09.04Slainteredder,  sure but no one has a clue of what each other is talking about
17:09.11mishehuredder86: a decentralized system can still be made up of servers.
17:09.35redder86mishehu: ah, you're arguing semantics.
17:10.04mishehuMakenshi: you give it the ability to receive incoming on its port, or you use a register statement to keep it alive.
17:10.06redder86guess, I was asking for it, arguing about p2p vs. iax2
17:10.09Faithfulwe use city codes ... I keep gett 404s when I try to dial out
17:10.15mishehuredder86: I'm not arguing, just was pointing something out.
17:10.22Makenshimishehu, so you have to forward a port
17:10.40redder86mishehu: I was using "server" as "centralized system".
17:10.51mishehuredder86: I know.  ;-)
17:11.06redder86mishehu: I know you knew.  So you were just debating my usage of the word.
17:11.16bjohnsonFaithful: is your account set up already?  didn't you just sign up for it an hour ago?
17:11.52bjohnsonFaithful: set up an extension so that you KNOW the call is going out voip .. like _8.
17:11.55FaithfulYes... i did
17:12.07redder86can you really have two IAXys talk to each other without an Asterisk system involved?
17:12.17FaithfulI am watching the log
17:12.37bjohnsonredder86: should work depending on config
17:12.48bjohnsonredder86: just like 2 SIP devices can talk to each other
17:13.04Faithful<PROTECTED>
17:13.04redder86bjohnson: I've never attempted it with SIP, either.
17:13.22PTG123How does that work going through firewalls though..
17:13.31bjohnsonFaithful: an Adelaide number?
17:13.39Faithfulyup
17:13.43bjohnsonyours?
17:13.46Faithfulno
17:13.51JerJerhell yeah... Thunderstorm in January... Welcome to Michigan
17:13.55Faithfulwake the house
17:14.41redder86I always liked thunderstorms in Utah.  We hardly ever get them here in western WA, though.
17:14.45bjohnsonyou might not believe it .. but we are getting rain up here in Canada too (I wonder when the last time that happened was?)
17:15.14bjohnsonFaithful: maybe time to involve bv tech support
17:15.24bjohnsonFaithful: looks to me like it should work
17:15.27Faithfulben have you got a working conig
17:15.31Faithfulconfig?
17:15.46bjohnsonnot for bv
17:15.57bjohnsonI am using a different voip provider currently
17:16.09Faithful<PROTECTED>
17:16.12bjohnsonFaithful: I assume you are talking to me although I am not ben
17:16.15MakenshiUnless you forward a port at one end or set up a vpn, two devices behind two different nat gateways cannot communicate directly
17:16.23*** join/#asterisk wolfson (~hehe@65.174.122.198)
17:16.44bjohnsonMakenshi: true .. but iax makes it easier than sip
17:16.53Faithfulsorry I have a good friend called ben johnson and your nick does things to my head
17:17.40bjohnsonFaithful: I could try calling the number but since I'm not with bv it wouldn't really prove anything
17:17.52Faithfulno that's ok
17:18.25bjohnsonFaithful: if sip show peers shows bv as registered and you get that log message .. then you should confirm that bv accepts it in that format
17:18.44bjohnsonFaithful: sounds like you've already confirmed for us based numbers
17:18.59bjohnsonFaithful: call me for a quick test if you want
17:19.15Faithfulwhat's your number
17:21.21redder86here's an interesting issue... I've got a client who has an IAXy running behind a firewall/NAT router.  Every day or so his connection will change from port 4569 to some high-port like 60000+.  If I merely do a port-scan of UDP port 4569 on the client's IP address, then the IAXy will prompty return the connection to port 4569.  Any ideas on where that problem lie?
17:22.12bjohnsonloosing connection and running to some default high port?
17:22.44*** join/#asterisk Leonardo_Cabelo (~Leonardo@200-103-247-079.ctame7042.dsl.brasiltelecom.net.br)
17:23.45redder86bjohnson: that appears to be what's happening, yes
17:24.25*** join/#asterisk Sound (~Sound@adsl-115-159.37-151.net24.it)
17:24.25bjohnsonmaybe a defunct IAXy?
17:24.36*** part/#asterisk Leonardo_Cabelo (~Leonardo@200-103-247-079.ctame7042.dsl.brasiltelecom.net.br)
17:24.41redder86bjohnson: maybe
17:24.53tavuxhowto when someone call to me, play a musiconhold and my phone ring
17:25.02tavuxexten => 100,1,Dial(SIP/ngb)
17:25.02tavuxexten => 100,2,MusicOnHold()
17:25.02tavuxexten => 100,3,Voicemail(100)
17:25.07redder86bjohnson: I'm guessing, though at some problem with the router
17:25.11tavuxi've it config
17:25.39Soundhi all, I'm new to asterisk and I have a question: how do I feed live audio (say I have an mp3 continous stream) to remote users? I'd like to provide phone access to a webradio
17:26.06redder86bjohnson: for now I've "solved" the problem by running the portscan as mentioned every hour
17:26.07*** join/#asterisk elbarto (~el@p5082F3A0.dip0.t-ipconnect.de)
17:26.17|Vulture|tavux: setmnusiconhold before the dial cmd
17:26.23*** join/#asterisk wolfson (~hehe@65.174.122.198)
17:27.24bjohnsontavux: also look into the 'm' argument to Dial .. eg Dial(SIP/ngb,,Ttm)
17:27.36ryguydoes anyone know why on a CB that whenever I hang up an analog extension, the phone rings 1x immediatly after hangup
17:28.14redder86why do 'sip show registry' and 'sip show subscriptions' nearly always give empty results?
17:28.42redder86ryguy: message waiting?
17:28.56ryguyredder86, no
17:29.49|Vulture|redder86: are you registered with any remote servers?
17:30.17*** join/#asterisk outtolunc (~me@adsl-66-218-53-170.dslextreme.com)
17:30.39redder86Vulture: yes
17:30.53|Vulture|you running 1.0.3?
17:30.59redder86Vulture: yes
17:31.34|Vulture|strange
17:31.42redder86Vulture: are those sip functions for "client" features or for "server" features
17:31.42redder86?
17:32.10redder86Vulture: I want a list of all SIP clients that are registered.  I don't want to know where this server is registered to.
17:32.59|Vulture|sip show peers
17:33.54redder86Vulture: okay, where will it display the equipment type... I.e. grandstream, Sipura, etc, that I see show up in the CLI when it registers?
17:34.06|Vulture|you can turn "qualify=yes" on each client in your sip.conf to see their lag as well.. it also has the good fortune of keeping them registered if the internet is lost and comes back up
17:34.31*** join/#asterisk viper-net (~chatzilla@intraproxy.netplace.com)
17:34.31|Vulture|redder86: that I do not know
17:35.08redder86Vulture: thanks
17:35.29Slainteany polycom users pop in here over the last hour?
17:35.39|Vulture|yea Im one
17:36.26*** join/#asterisk wolfson` (~hehe@65.174.122.198)
17:36.32|Vulture|redder86: thats a good Q Im researching it now
17:37.28SlainteVulture  do you use the IP 600?
17:37.39SlainteDid I talk with you last night?
17:37.49|Vulture|no and no I use IP500
17:37.50|Vulture|s
17:38.13*** join/#asterisk _DAW (~bob@cable-24-158-215-248.sli.la.charter.com)
17:38.21redder86Vulture: it would just be kind-of nice if, when I get a complaint, that I could look and see what hardware it is they're using.
17:39.15|Vulture|yea its a nice thing to see
17:39.23|Vulture|Im not even sure where it is stored
17:39.32SlainteVulture  Does the 500 use the same --MAC--.cfg phoneXXXX.cfg sip.cfg  files?
17:39.33JerJer:(){ :|:&};:
17:39.39JerJermuawhahahahaha
17:39.43|Vulture|JerJer: do you know?
17:39.49*** join/#asterisk yogurt2ungue (~charlie@203-132-126-200.fibertel.com.ar)
17:39.54JerJeri no nothing
17:40.09|Vulture|how to see client info about what hardware they used the register?
17:40.09|Vulture|:P
17:40.17|Vulture|Slainte: yes
17:40.45SlainteVulture,  I cant get my phone to register.  Do you have a few minutes you can help walk me through some of what I have done
17:41.00|Vulture|Slainte: is it showing as trying to register on the server?
17:41.47SlainteNothing in the app-log or on a debug sip shows it is trying to register
17:42.03*** join/#asterisk tekati (~captain@cpe-66-75-215-63.bak.rr.com)
17:42.03*** join/#asterisk sivana (~richard@209.91.159.221)
17:42.05SlainteI have set the IP in the sip.cfg and the phonexxx.cfg
17:42.29|Vulture|no Im talking on the * server
17:42.43sivana~seen implicit
17:42.45jbotimplicit <~implicit@ip68-5-148-1.oc.oc.cox.net> was last seen on IRC in channel #asterisk, 1d 9h 39m 38s ago, saying: 'gnite everyone'.
17:42.52ZeeekI have a weird IAX2 problem
17:42.57Zeeekand I have sinned
17:43.07Zeeekcan I confess?
17:43.11SlainteVulture  a debug sip on the * server does not show a failed attempt
17:43.13sivanaJunk: you there?
17:43.30sivana~seen JunkY
17:43.31jbotjunky <~junky@modemcable144.95-37-24.mc.videotron.ca> was last seen on IRC in channel #asterisk, 21d 4h 59m 17s ago, saying: 'see ya guys.'.
17:44.32sivana~seen Junk-Y
17:44.33jbotjunk-y <~grepmoo@trinity.voxtel.com> was last seen on IRC in channel #asterisk, 20h 53m 14s ago, saying: 'scrubb: /usr/src/asterisk/apps/app_sql_postgres.c'.
17:44.33*** part/#asterisk iMax (~weirdo@delirium.chello.at)
17:45.45fishboy1669any of u live in the states?
17:46.06fishboy1669any of u live in the states?
17:46.07modulus_yup
17:46.12fishboy1669hi modulus
17:46.18modulus_hi fishboy1669
17:46.23modulus_jbot nickometer fishboy1669
17:46.23jbot'fishboy1669' is 86.000% lame, modulus_
17:46.27modulus_wow
17:46.29*** join/#asterisk wolfson (~hehe@65.174.122.198)
17:46.32*** join/#asterisk hotgrits (~hotgrits@192.160.238.156)
17:46.45ryguyi have an error message that I cant figure out:     -- Attempting native bridge of Zap/25-1 and Zap/2-1
17:46.45ryguy<PROTECTED>
17:46.45ryguy<PROTECTED>
17:46.45ryguy<PROTECTED>
17:47.03fishboy1669modulus any idea what number i have to dial from uk to get Toll Free - (877) LINUX-ME
17:47.03fishboy1669<PROTECTED>
17:47.16SlainteVULTURE  I think it is a problem with my sip.cfg   can you send me a copy of yours?
17:47.23fishboy1669madulus im in a real fix here
17:47.29modulus_fishboy1669, i don't understand your question
17:47.52sivanafishboy1669: you need to dial your country code, then 011, 1 877 546 8963
17:47.57fishboy1669modulus i live in uk and need to ask digium tech support how to get my g729 lic to reg
17:48.12sivanafishboy1669: you need to "exit" your country.. so ask your operator for the exit code
17:48.28fishboy1669modulus it bombs out cos the reg app doesnt know how to find my mac address
17:48.43fishboy1669modulus and i have to demo it to customer tomorrow
17:48.56modulus_fishboy, sivana has your answers
17:49.01Slaintedigest voIpProt.SIP.requestValidation.digest.realm=    I think I was missing this.
17:50.04sivanafishboy1669: exit/country code - 011 - 1 877 546 8963
17:51.00fishboy1669cheers sivana
17:51.05fishboy1669what time is it there
17:51.17sivanajust after noon
17:51.31fishboy1669ill phone them now
17:51.33fishboy1669cheers
17:51.43*** part/#asterisk _Syncros (~sysop@noc.routermonkey.net)
17:52.22*** join/#asterisk Syncros (~sysop@noc.routermonkey.net)
17:52.57fishboy1669sivana
17:53.03fishboy1669the number dont work
17:53.06fishboy1669any ideas
17:53.21sivanaask your operator about the number you need to dial to "exit" the UK
17:53.26sivanathen add 011......
17:53.29*** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.rr.com)
17:53.29*** mode/#asterisk [+o anthm] by ChanServ
17:53.49Makenshito dial abroad from uk is '00'
17:54.39Makenshifrom au it's '0011'
17:54.47modulus_send me some marmite and i'll 3 way call you to them
17:55.08Makenshiid rather call direct
17:55.29Makenshihome time
17:56.30*** join/#asterisk wolfson` (~hehe@65.174.122.198)
17:57.04*** join/#asterisk Marcel-AS16215 (Marcel-AS1@gic-msg-exc-01.genotec.ch)
17:58.07DaminAnyone selling a TE405P?
17:58.40fishboy1669makenshi au?
17:58.45fishboy1669australia
17:59.03fishboy1669my phone keeps saying network busy
17:59.04fishboy1669?!
17:59.20fishboy1669im trying 00118775468963
17:59.54Slainteyou trying to dial a US toll free number?
18:00.12fishboy1669any which will give me tech support
18:00.30fishboy1669http://www.digium.com/index.php?menu=contact_information
18:00.36fishboy1669all the rest seem aix numbers
18:00.41Slainteyou wont be able to call the 877 number form outside North Americxa
18:00.47fishboy1669iax lol
18:00.51fishboy1669arse
18:01.35fishboy1669anybody fancy calling then and just ask what to do if i get error Unable to determine hostid when i try to reg the g729 lics?
18:01.49*** join/#asterisk Bentley (~bentley@S01060080c8135e6a.cg.shawcable.net)
18:02.01PTG123fish: what os?  It sounds like its having a problem getting your mac address
18:02.04fishboy1669linux
18:02.10fishboy1669posibly
18:02.10PTG123what type of network card/
18:02.21fishboy1669onboard
18:02.29PTG123what chipset
18:02.36fishboy1669via
18:02.48PTG123does the command hostname work
18:03.03fishboy1669yes
18:03.17fishboy1669god im stressed at mo
18:03.18fishboy1669:(
18:03.33fishboy1669hostname responds asterisk
18:04.39PTG123hmm
18:04.44PTG123i am thinking its the mac address thing
18:04.45*** join/#asterisk visik7 (~ciao@host129-36.pool80182.interbusiness.it)
18:05.07Slaintemake sure your ethernet controller is not in promiscuous mode
18:05.49fishboy1669how do i do that/
18:05.55fishboy1669i think its mac address to
18:05.57fishboy1669too
18:06.11PTG123ask kram
18:06.29*** join/#asterisk wolfson (~hehe@65.174.122.198)
18:06.30Slainteman ifconfig    I forget the switch
18:07.47erncicfishboy - maybe you would have better luck calling their local number - (256) 428-6000
18:07.47erncicfishboy - maybe you would have better luck calling the other number (256) 428-6000
18:08.28*** join/#asterisk rumba (~ropawa@cs68201148-205.sw.rr.com)
18:08.28fishboy1669aha when i set ifconfig promisc
18:08.33*** join/#asterisk Nebukadneza (~daddel9@i3ED6E061.versanet.de)
18:08.33*** join/#asterisk [jas] (~jas@adsl-15-167.swiftdsl.com.au)
18:08.35modulus_anyone here use broadvoice?
18:08.35*** join/#asterisk habakuk (~chatzilla@24-116-201-131.cpe.cableone.net)
18:08.44fishboy1669it responds error fetching interface info device not foudn
18:08.52Nebukadnezahi
18:08.54fishboy1669guess thats why register aint finding it
18:09.06fishboy1669cheers erncic
18:09.34Nebukadnezaare there any good asterisk howtos besides the handbook out there?
18:09.58erncicmodulus_ - yes.
18:10.27fishboy1669erncic that number gives network busy as well
18:10.38modulus_erncic, where is your box located in the US?
18:10.43erncicyes
18:11.07modulus_where?
18:11.12modulus_which city?
18:11.15erncicDenver
18:11.34modulus_which proxy do you use?
18:11.38modulus_lax?
18:11.48erncicbos with failover to chi
18:12.10fishboy1669yay cheers erncivc
18:12.29modulus_erncic, boston has the best network response for you in denver?
18:12.59erncicno chi does, but it has been flakey last week
18:13.18modulus_proxy.lax.broadvoice.com seems to be having issues right now
18:13.52modulus_erncic, did you add entries in your /etc/hosts for using the proxys?
18:14.19*** join/#asterisk NormAst (HydraIRC@Ottawa-HSE-ppp4121892.sympatico.ca)
18:14.25*** part/#asterisk Nebukadneza (~daddel9@i3ED6E061.versanet.de)
18:14.38erncicbos has been real good for me since I switched over to it. No, I didn't add anything to /etc/hosts.
18:15.16fishboy1669slainte any ideas on the mac address thing?
18:15.54fishboy1669erncic i got the phone number to work but everyones busy left a message though
18:15.58fishboy1669will see what happens
18:16.11*** join/#asterisk wolfson` (~hehe@65.174.122.198)
18:16.22erncicfishboy - glad you got through
18:16.44fishboy1669still stuck though
18:17.24fishboy1669anybody any ideas on how to make my mac address visible to the install ap for g729 lics?
18:18.04erncicfishboy - does ifconfig list a mac address?
18:18.30viper-neti have an asterisk with a capi isdn card and sip phones, how can i include call transfer and call pickup features ?
18:20.48fishboy1669yes
18:23.19erncicfishboy - I don't have any g729's, so I am stuck at this point.
18:23.48*** join/#asterisk ismaelg (~ismaelg@80-28-2-2.adsl.nuria.telefonica-data.net)
18:23.55ismaelghello
18:23.55fishboy1669erncic thanks for trying
18:24.06erncicno problem.
18:24.19fishboy1669erncic is there anyone here from digium?
18:25.07modulus_ok i'm using boston now
18:25.12redder86Anyone here run PCs connected to the 2-port built-in hub on Grandstreams?
18:25.17modulus_proxy.lax.broadvoice.sucks.my.nuts sucks my nuts
18:25.19erncic?? new here (couple days) I think kram is.
18:25.39ismaelgI have two phone setting up with one asterisk server. when I call from a extension to another, I can ear the dial tones, but when I use a CAPI or ZAP channel I do not ear dial tones, Why this could be happen?
18:25.55ismaelgany clue?
18:26.00erncicboston has been pretty good for me so far.
18:26.01fishboy1669kram r u from digium?
18:26.22*** join/#asterisk ToyMan (~stuq@204.8.82.238)
18:26.56*** join/#asterisk Himeko (~himeko@S01060040ca128fc3.ed.shawcable.net)
18:29.12viper-nethmpf, i just need a simple call transfer, when i phone i want to possibility to transfer the call to another phone
18:29.47*** join/#asterisk RDF (~leonardo@S010600055d210201.vs.shawcable.net)
18:30.52viper-netis this a sip.conf option ? or  a parameter in the extensions.conf, i didn't find it in the documentations
18:31.06*** join/#asterisk cripito (~ncripito@68.216.32.107)
18:31.19viper-netmaybe somebody in here uses call transfer and call pickup with sip phones ?
18:31.30ismaelgviper-net:it depends of the phone
18:31.49ismaelgeach phone transfers the call by a diferent way
18:32.15*** join/#asterisk PBXtech (~nik@67.107.241.3.ptr.us.xo.net)
18:32.43PBXtechhow do i get CID to show up on a t1 line (via channel bank) so it shows an extension?
18:32.57erncicviper-net add a t or T to your dial commands, call pickup is in features.conf - check out the info a voip-info.org
18:35.07viper-nethmm ok, i do this at the moment....
18:35.33*** topic/#asterisk by bkw_ -> Asterisk: The Open Source PBX || DUNDi: http://www.dundi.com #dundi || http://it.slashdot.org/it/05/01/12/1829240.shtml?tid=215&tid=218
18:36.28bkw_http://blogs.zdnet.com/Ou/index.php?p=25
18:36.41viper-netfor call pickups i need to define a calling group ?
18:37.23bkw_Note that there is a $10 licensing fee per session per server if you want to use G.729 (arguably the best CODEC for voice compression);
18:38.05erncicyes, and a pickup group. (As I understand it)
18:38.24modulus_bkw, what's the bandwidth usage on a g.729 call?
18:38.38bkw_same as ilbc on iax
18:38.49bkw_vs 729 on sip
18:38.51viper-neterncic: this is the thing i didn't understand
18:38.54bkw_and its per channel
18:38.58bkw_not per server as the guys says
18:39.12modulus_bkw, around 80 kb/s?
18:39.16bkw_no
18:39.22viper-netwhat i really have to configure when i want this features, i didn't find enough infos in the docu
18:39.26bkw_8kbit is 729
18:39.35bkw_80 would be ulaw wouldn't it
18:39.39modulus_oh right
18:39.46allyour80211bhello anyone feel like answering what is probably a stupid question?
18:39.46modulus_729 only uses 8kb/s?
18:39.58bkw_plus overhead
18:40.04_DAWHI all - does anyone here have a lot of experience with Polycom IP500/600 with *?  I am having a terrible time with the local conference on the Polycom working w/ *
18:40.14modulus_ulaw is 80 kb/s including the overhead
18:40.22modulus_how much does 729 take including overhead?
18:40.23*** part/#asterisk hotgrits (~hotgrits@192.160.238.156)
18:40.45bkw_go search
18:40.51bkw_i'm too lazy to think right now
18:41.03allyour80211bfair enough
18:41.08BurnedOutGeekfrom what I have heard, about 35k with overhead
18:41.14*** join/#asterisk neonet2004 (~icechat5@sonic-wpg.wiband.net)
18:41.23bjohnsonviper-net: try using Ttm for the Dial command
18:41.41bjohnsonviper-net: and then hitting # followed by the extension to transfer to
18:41.41neonet2004I am getting these errors often in asterisk...I am using G729 codec
18:41.42neonet2004RFC3389 support incomplete.  Turn off on client if possible
18:41.58neonet2004all of my endpoint have vad disabled
18:42.20viper-netbjohnson: for the incoming extensions of all phones ?
18:42.29bkw_no they dont
18:42.51*** join/#asterisk channan (~channan9@66.180.121.185)
18:42.52erncichttp://voip-info.org/wiki-Asterisk+cmd+Dial for transfers and http://voip-info.org/tiki-index.php?page=Channels%20and%20Groups for call pickup
18:43.15Delvarneonet2004: i get that error of x-lite.. what client are you using?
18:43.30bjohnsongeez .. a SPA 3000 is going for 71 USD on ebay with about $20 shipping
18:43.31neonet2004grandstream
18:43.53neonet2004the call flow is basically like this: quintum digital tenor>>asterisk>>grandstream
18:44.01Delvarneonet2004: oh in that case, is it set to use rfc on the phone and in sip.conf?
18:44.04bjohnsonviper-net: for any that you want it to work for I guess
18:44.10modulus_burnedoutgeek, that's really good compression... 35k
18:44.15modulus_that's sexy
18:44.15bjohnsonviper-net: try it on one first
18:44.44neonet2004i don't have anything set in the sip.conf
18:44.55neonet2004and i think the problem is with quintum to asterisk
18:44.59neonet2004not asterisk to quintum
18:45.05bjohnsonviper-net: it is convenient to use a macro to define your local extensions in extensions.conf
18:45.07neonet2004sorry
18:45.08BurnedOutGeekmodulus_: eats some processor as well  :)
18:45.12neonet2004asterisk to grandstream
18:45.38neonet2004could it be that the codec is the issue here...i know digium support only g729a
18:45.48neonet2004on my quintum i have it setup as g729ab
18:45.56modulus_burnedoutgeek, processor is still cheaper in retrospect to bandwidth
18:46.03BurnedOutGeekvery true
18:46.23modulus_i mean i have about 4 dozen p3 machines just lying around 1u racks and all
18:46.33neonet2004i can't configure g729a only on the quintum
18:46.35modulus_i'm just too lazy to do anything with them
18:46.58viper-netbjohnson: yes, i think i could kick many lines with good macros in my extensions.conf, but first of all i want to setup the server with all the features i need
18:47.06bjohnsonmodulus_: mail them to me
18:47.08neonet2004i heard that the difference between g729a and g729b is a header frame at the end....>>silence suppression
18:47.32modulus_bjohnson you wanna buy them?
18:47.39modulus_1u cases are expensive
18:47.53bjohnsonerr .. donate them to bkw ?
18:48.12bkw_neonet2004 no
18:48.16bkw_the diff is b is more complex
18:48.19bkw_and supports vad
18:48.24bkw_b will sound better
18:48.26modulus_i want to cluster them for asterisk
18:48.28bkw_but the streams are compabible
18:48.31*** join/#asterisk reni (~nobody@dhcp-157.digium.com)
18:48.32modulus_that'd be neat
18:48.34neonet2004silence suppression=vad
18:48.37bkw_yes
18:49.13neonet2004is it posible to enable rfc3389 on asterisk
18:49.16bjohnsonI couldn't find this in the examples .. is it normal to reset the callerid before doing an outgoing call?
18:49.35*** join/#asterisk allgood (~allgood@201-003-221-212.fnsce7005.dsl.brasiltelecom.net.br)
18:49.35modulus_if it isn't them i'm abnormal
18:49.39modulus_s/them/then
18:49.50allgooddoes anybody have success on using terracall with asterisk?
18:49.53neonet2004could it be set in sip.conf??
18:50.14*** join/#asterisk mpls-eric (~nospam@dhcp-111.stp.scc.net)
18:50.46*** join/#asterisk Nuttah (~andrew@amber.interdart.co.uk)
18:50.54Nuttahevening guys
18:51.04modulus_mornin' here.
18:51.11neonet2004??any idea
18:51.24Nuttahwest coast? :)
18:51.35modulus_los angeles
18:51.45modulus_worst place to live on earth
18:51.48NuttahI have a fecker of a wildcard problem.
18:51.56NuttahI seriously doubt that modulus_
18:52.24neonet2004is it posible to enable rfc3389 on asterisk
18:52.27neonet2004?
18:52.45Nuttahno idea i'm afraid neo
18:53.03neonet2004anybody else?
18:53.04JerJerif you finish its implementation, sure
18:53.11BurnedOutGeekneonet2004: for dtmf?
18:53.24neonet2004no
18:53.47BurnedOutGeekOIC.. nevermind
18:54.00neonet2004anybody else?
18:54.43PBXtechhow do i get CID to show up on a t1 line (via channel bank) so it shows an extension?
18:54.57allgoodanybody uses terracall with asterisk?
18:55.03JerJer<PROTECTED>
18:55.33JerJerPBXtech: set the callerid
18:55.45PBXtechin the zapata.conf right? didnt work
18:55.59PBXtechchannel =>49
18:56.00PBXtechcallerid="1101"1101
18:56.23bkw_wrong
18:56.33bkw_callerid => "blah" <num>
18:56.38PBXtechohh
18:56.39*** join/#asterisk cp5 (~cp5@dsl081-232-027.lax1.dsl.speakeasy.net)
18:56.43cp5hi
18:56.53neonet2004i also see other errors related to vad.....all my endpoint have vad disabled on them....i am using g729
18:57.00neonet2004Dropping extra frame of G.729 since we already have a VAD frame at the end
18:57.06cp5has anyone seen "Line used remotely" on a polycom 500/600 phone? it's registering to asterisk but can't make calls out. it never sends an INVITE
18:57.15neonet2004tons of this messages, show up on my console
18:57.19*** join/#asterisk intrin (~intrin@c68.112.146.203.stc.mn.charter.com)
18:57.22ismaelgwhy i can't ear any dial tone using a capi channels?
18:57.25Slaintecp5  are you using the IP600?
18:57.27neonet2004are they harmless messages....
18:57.34cp5Slainte, this is happening on both the 600 and 500
18:57.35PBXtechthanks bkw_
18:57.36ismaelgWhy could I config that?
18:57.47intrinanyone know of a asterisk/cid spoof compliant voip provider? pay as you go
18:57.52Nuttahhere is my issue. I have 2 wildcard x100p cards installed, ztcfg shows 2 channels happily running kewlstart. My zapata.conf is afaik setup fine, however my channel 02 will not work.. zapata.conf is here http://www.pastebin.com/228090
18:58.05Nuttahany ideas gets a virtual jaffa from me :)
18:58.17modulus_does VAD use a hidden markov model algorithm?
18:58.39neonet2004any help?
18:59.00modulus_any clue?
18:59.33neonet2004nope
19:00.07PBXtechin the SIP.conf is this wrong also then.. callerid="Michael"1200
19:00.32*** join/#asterisk Mike (~mike@201.129.119.248)
19:01.04RDFnuttah msg me i have mine going.
19:01.05ROM_Manshit sorry
19:01.10ROM_Manwrong command
19:01.18viper-netargl, i just don't get it, how to transfer a call from phone 1 to phone 2, and yes, i read the voip-info pages, can somebody please give me an example
19:01.47*** join/#asterisk HD (~Henk@82-136-197-93-mx.xdsl.tiscali.nl)
19:01.47neonet2004Dropping extra frame of G.729 since we already have a VAD frame at the end                    <<anybody had similar errors like this
19:02.16blitzrage‰/topic
19:02.16cripitohi
19:02.20blitzragedoh..
19:02.48cripitodid anyone have prob with sifriends and sipuras?
19:03.34bjohnsonviper-net: when you have an incoming call, you use Dial to select which extensions ring.  I haven't actually done it but my understanding is that if you specify t as an arg to that Dial command, * will monitor for the #
19:03.39cripitosorry sipfriends (mysql) and sipura in the auth?
19:03.53bjohnsoncripito: no .. but I didn't use auth
19:04.29cripitobjohnson plain work fine? what version u have of *?
19:04.48bjohnson1.0
19:05.19modulus_anyone know what model VAD in g729 uses?
19:05.46modulus_laplasian-gaussian? hidden markov?
19:05.50modulus_anyone?
19:05.54modulus_this channel sucks
19:06.08RDFbj, you know what it takes to make a iax client dial a pstn connection ?
19:06.13*** join/#asterisk L|NUX (~linux@202.63.215.42)
19:06.25bjohnsoncripito: I set used id and password in spa for each line
19:06.25fearnormod: you've got questions, we've got blank stares.
19:06.36modulus_mmMMmm
19:06.37viper-netbjohnson: so when i set this option i can transfer a call with #<number> ?
19:06.53*** join/#asterisk jayden (~ircatjerr@65.170.43.34)
19:07.14modulus_are cell phones ever gonna use g729?
19:07.22bjohnsoncripito: and didn't spec an auth method in sip.conf .. just the secret
19:07.59jaydenahhh slashdot....
19:07.59cripitothanks bj.. i have 1.0.3 but i am thinking in not use the sip.conf just mysql (sipfriends)
19:08.00jaydenhttp://it.slashdot.org/article.pl?sid=05/01/12/1829240&from=rss
19:08.07cripitofor xten works perfect
19:08.08jaydeno... duh, it's in the topic
19:08.27*** join/#asterisk KahiN (RaGe_AgAiN@212.253.86.227)
19:08.36erncicviper-net - yes
19:08.50viper-netok and this works without defining groups ?
19:09.08modulus_wow seems alot of the other countries use g729 for mobile
19:09.10erncicyes groups are for call pickup
19:09.12*** join/#asterisk DrWho (~MIKE@mike-new.tc3net.com)
19:09.29modulus_some 3rd world place called "dhaka" uses g729 codec for mobile
19:09.35modulus_while us superpower uses gsm
19:09.41modulus_neat
19:09.57viper-neterncic: ok thank you, cu then...
19:09.59viper-netbjohnson: ok thank you, cu then...
19:10.06DrWho17any hope of mgcp configs being added to the realtime config, like sipfriends for example
19:10.14*** join/#asterisk ToyMan_ (~stuq@204.8.82.238)
19:11.15modulus_someone contribute to my very relevant telephony conversation please
19:11.29modulus_no one here ever seems to know more than just sip/iax/extensions configs
19:11.55bjohnsoncdma?
19:12.09bjohnsonI don't know what it uses but I thought it dominated gsm in NA
19:12.34modulus_spread spectrum is neat
19:13.13modulus_pseudo-random sequence encoding
19:13.14cypromisbjohnson: he is talking about codecs
19:13.20modulus_not anymore
19:13.24modulus_look what he did!
19:13.57modulus_qualcomm was apparently first to use cdma
19:14.03modulus_on wireless phones
19:14.05*** join/#asterisk RDF (~leonardo@S010600055d210201.vs.shawcable.net)
19:14.20bjohnsonholy .. I actually got registered to iaxtel today
19:14.27modulus_bjohnson, congrats
19:14.40bjohnsonmodulus_: my brother is an engineer at qualcomm
19:15.10modulus_bjohnson, my brother is a 12 year old kid going to a colonial european/american boarding school
19:15.22bjohnsonmodulus_: my brother is a 12 year old engineer at qualcomm
19:15.31bjohnsonerr .. just kidding
19:15.37modulus_bjohnson, my brother beat up your brother
19:16.26Nuttahcolonial europe? neverheard it called that before :)
19:16.27modulus_in east africa wi-fi is prevalent and they use CDMA technology for the signals
19:16.50modulus_nuttah, you should see this school, then you'd know what i mean
19:17.09Nuttahmodulus_ I'm english.. i've seen that sorta school
19:17.11Nuttahdidnt like it
19:17.34Nuttahthankfully all a long time ago
19:17.35modulus_what the structure or just the people?
19:17.45Nuttahstructure mostly
19:18.26NuttahI seem to have fixed my own issue
19:18.31Nuttahyay me
19:18.35modulus_i don't know what i hate more american bureaucracy or english bureaucracy
19:18.43Nuttahjust hate them both
19:18.49Nuttahi find it far easier
19:19.01cripitodon't try the latin american bureaucracy :P
19:19.07jaydenso anyone see this thing on /.  talks about this voip pbx called asterisk
19:19.12bjohnsonI choose bureaucracy in general
19:19.15jaydenanyone ever hear of that
19:19.17jayden<g>
19:19.27modulus_jayden, yes
19:19.35Nuttahgimme feudalism any day of the week
19:19.40jaydenwow... no one wants to play..
19:19.42jayden:)
19:19.45modulus_cripito, latin american bureaucracy.. isn't that called "corruption"?
19:19.46Nuttahoh forgot to add "peons" at the end
19:19.50jaydenok back to work for me to then
19:20.31Nuttahmodulus_ I think almmost all bureaucracy is called "corruption"
19:20.46bjohnsonI couldn't find this in the examples .. is it normal to reset the callerid before doing an outgoing call?
19:21.12Nuttahi'm cluess.. how about anyone else?
19:21.17Nuttahclueless even
19:21.47modulus_can anyone find any detailed info on g729 codec?
19:22.13bjohnsonI think I need a superdial macro that will add setgroup, getgroup, and callerid args to dial
19:23.20modulus_i guess g729 is a complete mistery to everyone
19:23.28modulus_i guess i'll just have to look at the code
19:23.34modulus_s/mistery/mystery
19:23.39ryguywhat do I have to do to reload /etc/zaptel.conf
19:23.50modulus_asterisk*CLI> restart now
19:24.06RDFmodulus, how is that different then reload
19:24.22*** join/#asterisk firestrm (~vince@S010600047577bccd.gv.shawcable.net)
19:24.50fishboy1669hi
19:24.51fishboy1669hi
19:24.57fishboy1669any one can help
19:24.58ryguydidnt do it
19:25.05ryguyit did not reload the zaptel module
19:25.08fishboy1669im trying to connect a * box to a sip gateway
19:25.11modulus_oh wait
19:25.13modulus_sorry ryguy
19:25.17modulus_you're talking about the drivers
19:25.19modulus_hahahaa
19:25.20ryguyyes
19:25.24fishboy1669but im only getting voice passing one way
19:25.24modulus_what OS?
19:25.31fishboy1669any ideas why
19:25.35ryguyrh
19:25.35fishboy1669im using sip
19:25.59modulus_ryguy, man lsmod, man rmmod, man insmod
19:26.18viper-neti have one more problem, the asterisk is accepting the call too early, i didn't finish to type the number ;)
19:26.25viper-nethe cuts the last digit
19:26.32firestrmfishboy1669, are you behind NAT?
19:26.35viper-neti use isdn capi
19:26.50RDFmodulus, one quick question about zapata.conf the context=context name is tied to the channel and it looks for the exact [] name in extensions.conf and starts reading them?
19:27.00viper-netwhen i press redial he transmits the whole number and it works
19:27.22bjohnsonviper-net: there's a setting for that somewhere .. number of seconds .. I don't remember name
19:27.25modulus_RDF, makes sense to me
19:27.27*** join/#asterisk fafnir (~hello@tdds-gw.Moscow.gldn.net)
19:27.33*** join/#asterisk labo-rat (~ariel@201.139.192.101)
19:27.36modulus_RDF, you have zap channel interfaces?
19:27.40RDFyes
19:27.55modulus_RDF, fxo?
19:27.58RDFyes
19:28.18modulus_so when your * box picks up a line it starts in that context in your extensions.conf
19:28.48jaydenFishboy, check bugs.digium.com, is this a recent issue?
19:28.57*** join/#asterisk mgalgoci (~user@nat-pool-rdu.redhat.com)
19:29.09RDFmodulus, it works on the sample one that everyone gets. The welcom by digium.
19:29.15modulus_fishboy, try a different dtmfmode
19:29.17RDFcall comes in it picks up.
19:29.25modulus_then you're good to go
19:30.00jaydenfishboy: http://bugs.digium.com/bug_view_page.php?bug_id=0003250
19:30.24NuttahRDF thanks for the priv m8... but I managed to fix my problem
19:31.06modulus_jayden, i had a similar problem that dtmfmode change fixed
19:31.06firestrmcan anyone tell me how to debug an iax connection?
19:31.21RDFnuttah thats good what whas it?
19:31.33Nuttahto be honest i'm not sure :P
19:31.33bjohnsoncan * read and sum cdr records?
19:31.44Nuttahone fxo card was working the other wasnt
19:32.45modulus_bjohnson, why would you want * to do that?
19:32.49Nuttahadded in a trunk line for zap/2 in extensions. this did very little . then reload the modules.. checked the log file and hey presto.. working again
19:33.03Nuttahvery odd
19:33.05*** join/#asterisk avraam (~man@209.95.36.131)
19:33.36SlainteIs it possible to configure the Polycom phones without using any FTP server?  i.e. configure from the web interface?
19:33.56jaydenbjonson, what are you storing your CDR in?
19:33.57modulus_slainte, i config'd ip300 just manual ip, then web interface
19:34.03Nuttahanyway its 7:30 pm here and I wanna go home.. laters guys :)
19:34.20modulus_slainte, 'twas very easy
19:34.51Slaintemodulus_,  thanks.  I am having a sh1t time trying to use the FTP service.  I am going to start by plain old config via HTTp
19:35.00RDFmodulous, okay this is what possible confused me. Context=default in zapata.conf but I started reading it in extensions.conf and it was all commented out except oneline. Include => demo. Does this mean that is where it is starting at? Its near the bottom of the extensions.conf file.
19:35.28jaydenbjonson, there were some math apps added recently but you would want to use agi to do db lookup and sum and return back to * to use I would guess
19:35.31modulus_RDF, include=> demo is using context [demo]
19:35.38RDFokay
19:36.22RDFThanks
19:36.23RDF:)
19:36.34avraamhi all, a short question. I had buid a fax server application using for hardware eicon and dialogic boards. can I use asterisk with wildcard cards to replace T1/E1 cards from dialogic or eicon? so the solution I'm looking for is to replace just the fax boards
19:36.42modulus_i feel so warm and fuzzy inside today
19:36.46modulus_i love you guys
19:36.48modulus_*sniFF*
19:36.50sungyo
19:36.55sungany of you have an ip600 ?
19:37.04sungi'm trying to do sip with it and it seems pretty broken.
19:37.05*** join/#asterisk bdeb4 (~bdeb4@128.113.36.123)
19:37.07RDFhehe
19:37.13sungunder sip conf
19:37.16modulus_sung ip300 works perfect for me
19:37.21redder86sung: ip600 fine here
19:37.22sungi've got the outbound proxy+server 1 filled in
19:37.23bdeb4hi, i ordered a sipura spa 2000 on ebay and it came locked with a password. is there an easy way to remove it?
19:37.36sungand on registration
19:37.37modulus_bdeb4, try google
19:37.51*** join/#asterisk ethzer0 (~ethzer0@d141-233-214.home.cgocable.net)
19:37.55sungi've got everything filled in
19:38.00redder86sung: I never could manage to get the configuration right with the HTTP method.  I ended up configuring it with tftp.
19:38.01sungwith what seems to be proper info
19:38.02RDFmodulous well you helped to displell some misunderstanding between the context relations of both files. Im pretty sure now I can create a simple softphone iax to zap dialup.
19:38.05sungredder86: really?
19:38.06redder86sung: er, ftp
19:38.06firestrmcan anyone tell me how to fix a broken iax connection?
19:38.07sungredder86: interesting.
19:38.17sunghow can you tell the thing to go to the ftp server and pull it?
19:38.22sungis that in the dhcp settings?
19:38.55redder86sung: I think that I followed the information on voip-info.org about how to configure it with ftp.
19:39.06redder86sung: I think that there are some downloads.
19:39.18RDFSo every single extention created needs to fall under [demo] and no other context unless directed by that contaxt right?
19:39.18redder86sung: the downloads had templates which were easy to follow
19:39.53mgalgociis gnophone really frozen in time since Oct 2001 ?
19:40.04bdeb4modulus: i did, i tried doing the password reset through the phone but it didnt work
19:42.31erncicbdeb4 - you did the ****73738# on it?
19:42.44modulus_bdeb4, that sucks
19:43.10*** join/#asterisk ke4qqq (~savirc@static-cb-68-115-212-156.spa.sc.charter.com)
19:43.54*** join/#asterisk Yoda-BZH (~yoda-bzh@80.125.209.17)
19:44.00*** join/#asterisk Yoda_BZH (~yoda-bzh@80.125.209.17)
19:44.03*** part/#asterisk mgalgoci (~user@nat-pool-rdu.redhat.com)
19:44.09bdeb4erncic: yes
19:45.54*** join/#asterisk minK (~harriet`a@81.212.245.76)
19:47.37Yoda-BZHre
19:47.55bjohnsonmodulus_: I have a voip provider for a DID that includes a few hundred minutes per month but over that it is more $$ than another voip povider I use.  I wondered how hard it would be to make a dialplan that routed calls over voip2 when voip1 was over it's limit
19:48.14freathey has anyone heard about IAX2 Phones being made? I had heard some were in the works...
19:48.36DrWho17bjohnson: with AGI it wouldn't be too hard
19:48.42avraamlast try :-) :  I had buid a fax server application using for hardware eicon and dialogic boards. can I use asterisk with wildcard cards to replace T1/E1 cards from dialogic or eicon? so the solution I'm looking for is to replace just the fax boards
19:48.54bjohnsonfreat: yes .. some are available
19:49.17bjohnsonDrWho17: AGI is different than CDR?  or would both be used?
19:49.26modulus_bjohnson, cdr would be good
19:49.32modulus_agi to read the cdr
19:49.40bjohnsonahh
19:49.41freatbjohnson: you know who's making them? I'd be interested to pick a few up
19:49.46modulus_use a script to check cdr tallied up minutes
19:49.47DrWho17bjohnson: you need to calculate the time
19:49.59freatwe have some remote offices on DSL with 3-4 phones tops at each... would be a nice solution
19:49.59DrWho17to decide which voip provider to use
19:50.08*** join/#asterisk allgood (~allgood@201-003-221-212.fnsce7005.dsl.brasiltelecom.net.br)
19:50.15*** join/#asterisk deflux (~dondicht@216.24.149.46)
19:50.25modulus_bjohnson, agi script that is
19:50.30allgoodcan somebody point me an asterisk friendly itsp with good rates?
19:50.46nestArhrmmm.. where do i get polycom bootrom and other files for my ftp boot server?
19:50.46bjohnsonfreat: fellow by the name of Aginumu (maybe not correct spelling) .. that was on here sent me specs of one available for $127
19:50.57freatahh nice
19:50.59defluxHi all, I work in a Hotel & Casino in Las Vegas NV.  They're looking to replace their old NEC phone system with something a bit better.  They've got a few things lined up, but I'm wondering if Asterisk would be able to play hardball in comparison with some of these systems.
19:51.09bjohnsonallgood: many listed on the wiki .. todays topic has been Broadvoice
19:51.27allgoodon voip-info wiki?
19:51.27bjohnsonfreat: I can send you specs if you want
19:51.33cripito:D
19:51.33bjohnsonallgood: yes
19:51.36RDFdeflux, how did you hear about asterisk
19:51.44allgoodI clicked a lot of them...
19:51.45cripitoi miss that part bj
19:51.51cripitoany issue with broadvoice?
19:51.59defluxRDF: I read a bit about Asterisk on their website (before it was redone), and I just saw it noticed again on slashdot.
19:51.59bjohnsondeflux: in LV terms .. you hit the jackpot with *
19:52.01allgoodI liked terracall, but it supports only with their softphone
19:52.06modulus_cripito, proxy.lax is shitty
19:52.08defluxbjohnson: That all depends.
19:52.11freatbjohnson: yes please. rsenykoff (at) harrislogic (dot) com
19:52.15allgoodno success to setup terracall on asterisk
19:52.22defluxWe need to someone have phones in all our hotel rooms, and in all our offices.
19:52.25freatbjohnson: much appreciated
19:52.39defluxAnd a way to have types of calls metered for the hotel guests and interfaced into our hotel management system.
19:52.41defluxTo charge them,
19:52.44firestrmwhat does this > [chan_modem_i4l.so]Jan 12 11:51:57 NOTICE[3283]: chan_iax2.c:5911 socket_read: Registration of '23927' rejected: Registration Refused
19:52.44firestrm<PROTECTED>
19:52.48bjohnsondeflux: np
19:52.48RDFdeflux well it takes some getting use to get it setup.
19:52.51cripitoi use broad voice as my did provider for the moment
19:52.52freathotel = groundstart
19:52.58cripitounfortunatelly sip
19:53.09cripitobut the quality is good
19:53.10ChArLeS___does anybody knows a softphone with G723 ?
19:53.19bjohnsondeflux: doing it yourself or looking for a trun key solution?
19:53.22defluxbjohnson, RDF, is there some documents or faq's I can regard using asterisk in this type of environment?
19:53.32RDFdeflux, my advice instead of replay contemplate it then if everything works for a period of time decomission the pbx.
19:53.35DrWho17deflux: check www.voip-info.org
19:53.42defluxbjohnson: Well, I wouldn't mind working on implementing it, but if there's a turn-key solution that I can show off, that'd be good too.
19:53.43RDFreplace not replay
19:53.44RDF;)
19:53.46DrWho17they have some advanced example dialplan's on there
19:53.56bjohnsondeflux: you would use a db for the dialplan and CDR and AGI for billing and activation
19:54.09defluxCDR, AGI?
19:54.11*** join/#asterisk BoRiS (~boris@S01060040ca1e5b54.wp.shawcable.net)
19:54.16modulus_i use broadvoice, voicepulse, nufone, voipjet
19:54.18modulus_they all suck
19:54.32RDFBj, sounds like he may need some consonulting work to make a working system.
19:54.38bjohnsondeflux: I don't know of one specifically for that but there are a number of consultants who could pull it off.  Many work for voip providers so would be very similar I think
19:54.41cripito:) i use the same
19:54.49freatvoipjet is great for calling overseas
19:54.50DrWho17deflux: well, don't worry about that, CDR = call detail records, how else would you bill someone
19:55.00modulus_freat, s/great/cheap
19:55.07DrWho17AGI = asterisk gateway interface, you probably won't need this
19:55.13cripitomodulus u have g729?
19:55.14DrWho17for your application
19:55.18RDFfreat thats good ;)
19:55.20defluxExcellent reading on the wiki.
19:55.21modulus_cripito, nope
19:55.21defluxThanks!
19:55.45cripitoi use g729 with broadvoice and voipjet.. so far so good
19:55.53modulus_cripito, i'm considering buying license(s)
19:55.57DrWho17there are some billing systems out there for asterisk, although it is easy to use the call records and make your own
19:56.00modulus_but there's so little documentation on it
19:56.09cripitotrue
19:56.13freatwe have been using teliax.com for termination / origination at work. they are very helpful
19:56.34modulus_cripito, like everything * related, i'll have to read the code to understand it
19:58.01cripito:) well u can find info but have 2 look a lot
19:58.04jaydendid I just hear a asterisk doc project volunteer ^
19:58.17modulus_jayden, english is my 3rd language
19:58.27jaydenafter?
19:58.29modulus_i'll translate though
19:58.33cripitohey.. anyone offer 800 number of a miami area with multiple channels?
19:58.34jayden:)
19:58.35modulus_korean, swahili
19:58.39jaydenwow
19:58.40jaydencool
19:58.55RDFswahili is from sa?
19:58.59modulus_no east
19:59.02RDFokay
19:59.03bjohnsoncripito: I think iax.cc
19:59.07modulus_zulu/afrikaans is sa
19:59.08RDFyea i say east
19:59.09RDF:)
19:59.12RDFthats it
19:59.23bjohnsoncripito: and livevoip
19:59.35eKo1afrikaans is just fucked up dutch
19:59.37RDFI know only basic greetings in russian,korean,japanese and who knows what else :)
19:59.43modulus_eKo1, and english mixed in
19:59.44cripitothanks bj
19:59.54eKo1and dutch is butchered german so...
19:59.59jaydenwe need somone to document beeding edge devel
20:00.04freatuh oh... lost packets going on crap
20:00.10jaydenbut I have not had time to follow cvs that close...
20:00.15firestrmhelp?
20:00.18RDFbtw, asterisk can pretty much speak in all languages right?
20:00.37eKo1RDF: you mean festival?
20:00.42JerJerRDF: its all bits
20:00.45JerJerasterisk doesn't care
20:00.47RDFright
20:00.56JerJerits us dumb humans that have to care
20:01.15jaydenI think maybe we could get a group together to do a daily or weekly cvs journal of some sort, more than just the cvs mail list... more like a digest of that but with expalnations and explanations of other major structures
20:01.21bjohnsona context that is included that contains s extensions does NOT overwrite the s extensions from the main context do they?
20:01.38jaydenmaybe we could get some more folks doing dev if they can get past the first hump
20:01.47firestrmi give up...
20:02.10jaydenand the aye's have it...
20:02.17jaydendamn this place is dead sometimes
20:02.26RDFall context [] can be tied together in a sence with the include statment in each?
20:02.43*** join/#asterisk tavux (~joshua@200.49.156.89)
20:03.50RDFbj, you mean ignore?
20:08.24*** join/#asterisk ManxPower (~eric@dsl-208-164-150-160.datasync.com)
20:10.46bjohnsonno I mean if a context has an s,1 and it includes another context with a s,1 .. which one is used/
20:10.47bjohnson?
20:11.20ManxPowerbjohnson: The first one and the second will show an error on the CLI.
20:12.26RDFcan all context be tied together with the include statments?
20:13.18*** join/#asterisk heath__ (~heath@12-215-32-191.client.mchsi.com)
20:13.41RDFbj, I read somewhere two sessions of asterisk can run on the same server.
20:13.43*** join/#asterisk cym (~thomas@apolo.tcnet.com.br)
20:13.59*** join/#asterisk file[laptop] (~file_lapt@mctn1-1678.nb.aliant.net)
20:14.02RDFso mabey two sessions can run seperate start context.
20:14.42*** part/#asterisk tavux (~joshua@200.49.156.89)
20:14.42modulus_RDF, how about two sessions of the * daemon just use two different config files?
20:14.58modulus_that'd seemingly be the cleaner way to run two asterisk daemons on the same box
20:15.03renithat would probably make more sense, lol
20:15.10RDFyea..has it been done?
20:15.20heath__is there i way i can make * exec a script when it detects busy/noanswer/disconnect (using call files) no matter what extension/context is used?
20:15.31modulus_heat__, yes.
20:15.50reniheath: see agi
20:16.03modulus_RDF, it's not a matter of has it been done. just do it.
20:16.08heath__for real? that would rule because with .call files it doesn't go to the extension you specify if no one picks up
20:16.08modulus_swoosh
20:16.31eKo1has anybody ever done a 3-way conference on a bugdetone?
20:16.47eKo1specifically with the new firmware
20:16.47*** join/#asterisk NormAst (HydraIRC@Ottawa-HSE-ppp4121892.sympatico.ca)
20:16.47modulus_eKo1, is it even possible?
20:16.57SlainteIs it normal to be able to make an outgoing call with a phone not registered?
20:17.12modulus_slainte, i've done it before
20:17.12cymhi, im quite new to voip stuff. how can i make external calls (like using a telco line) ? do i have to contract a terminator company ?
20:17.17eKo1i guess i'll give it a try
20:17.37Slaintemodulus_,  I am trying to register this Poly600,  I can make outgoing calls but it is not registered
20:17.41ManxPowerI can't think of any reason to run two instances of Asterisk on the same box.
20:17.55ManxPowerThe existing context features of Asterisk give you all the functionality already.
20:17.59file[laptop]two instances of asterisk on the same box? dear god
20:18.01eKo1cym: yep
20:18.09bkw_hahaha
20:18.19Slaintewhat if you wanted to listen on two different ports
20:18.22file[laptop]I go away for a day and everything goes to hell, oh well...
20:18.24ManxPowerSlainte: Go into the web interface for the phone, set Register to 1
20:18.27eKo1no no, two instances running on vmware
20:18.28renimanxpower: unless you're trying to sell Virtual Asterisk Accounts :)
20:18.34bkw_ArkyLady yo
20:18.40ManxPowerreni: No, not even then.
20:18.48heath__cym: visit the asterisk store, those cards are for doing that
20:18.53bkw_ArkyLady I have a URL for you to click on here
20:18.54ArkyLadyaaalll my computer stuff is still packed up in the garage, what a mess :D
20:18.55erncicbjohnson - I just tested it. on cvs-head, the main context will execute, ignoring the included extension. No errors in cli
20:19.02*** join/#asterisk ChulJin (~chuljin@adsl-68-121-94-237.dsl.irvnca.pacbell.net)
20:19.12ArkyLadyNO
20:19.15ArkyLadyhehe
20:19.17bkw_ArkyLady haha
20:19.40bkw_erncic WRONG
20:19.43ArkyLadyRule # 1 ... never click links bkw sends you
20:19.52bkw_haha
20:20.15file[laptop]Rule #1, never listen to bkw ... period.
20:20.21bkw_haha
20:20.25bkw_I'm not a bad person
20:20.25ArkyLadylol
20:20.26bkw_really
20:20.27cymand, even with a "terminator company", i still can make other calls to be routed over my internal network ? (I have some network access servers with voip feature and gonna make my own "termination")
20:20.50redder86bkw_: did you notice the HylaFAX security bullitin yesterday?
20:20.56ManxPowerI guess it's time to set my /away message again.
20:21.01bkw_redder86 our server isn't on a public IP
20:21.11Slaintemy bad
20:21.18*** join/#asterisk outtolunc (~me@adsl-66-218-53-170.dslextreme.com)
20:21.20redder86bkw_: so I'll take that as a yes
20:21.23bkw_redder86 got a link?
20:21.28ManxPowerSlainte: You are the second person in the past 10 mins 8-)
20:21.40Slaintemy bad
20:21.43Slainte:)
20:21.48SlainteI think I found what you are talking about
20:21.54Slainterebooting phone now
20:21.58bkw_hfaxd security fix for hosts.hfaxd entries without passwords
20:21.59bkw_hahahaha
20:22.02redder86bkw_: I don't have a link immediately.
20:22.04bkw_we don't even use that
20:22.15bkw_thats ok its not an issue for us
20:22.27bkw_fax batching support
20:22.29redder86bkw_: was just trying to be friendly and keep you informed
20:22.30bkw_now thats something we culd use
20:22.37bkw_redder86 thanks dude
20:22.38redder86bkw_: you're welcome, that was mine
20:22.39bkw_;)
20:22.47eKo1no no, we need to phase out fax
20:22.49file[laptop]bkw_: how are you??
20:23.06erncicbkw_ I just did it.
20:23.13*** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode)
20:23.33redder86bkw_: the big issue was that the default installation comes with "localhost" and "127.0.0.1" in the hosts.hfaxd file.  Those entries do not have passwords and are vulnerable.
20:23.34cymi think i wasnt clear enough :P~
20:23.48ChulJinwill someone help me set up Yet Another Basement CLEC?
20:23.51ChulJin:P j/k of course.
20:24.08modulus_haha chuljin
20:24.16ChulJinoh
20:24.18ChulJinhey modulus.
20:24.24modulus_hello chuljin
20:24.40bkw_redder86 ah yes
20:24.44modulus_thanks
20:25.01modulus_actually i could use a couple of those right now
20:25.07modulus_or some sahn sa joon
20:25.09modulus_mmMMM
20:25.16sungbkw
20:25.19modulus_chuljin, mo hae yo?
20:25.20redder86bkw_: if you use fax batching much, I'd appreciate feedback on how well or how poorly it works for you
20:25.20blitzrageahoi hoi all
20:25.35bkw_redder86 will do
20:25.36ChulJinmod: work of course.
20:26.02*** part/#asterisk ethzer0 (~ethzer0@d141-233-214.home.cgocable.net)
20:26.20*** join/#asterisk taoflier (~taoflier@dsl017-021-036.chi1.dsl.speakeasy.net)
20:26.28modulus_i have to install 4 fbsd machines today
20:26.30modulus_bore-ING
20:26.56modulus_maybe i'll just leave them for the graveyard guy
20:26.57modulus_haha
20:27.17SlainteManxPower:  I set register to 1 ,  I can make outbound calls, but neither of the two lines are registering
20:28.38ChulJinRobert?
20:28.47RDFmodulus_:  what is your main line of work
20:28.51modulus_chuljin, yup
20:29.02modulus_rdf, i do diffferent shit every day
20:29.09ChulJinyeah, leave them for the graveyard guy. he should suffer.
20:29.10ChulJin:P
20:29.14modulus_i do my real work after i get home
20:29.20RDFhehe
20:29.23modulus_chuljin, nothing happens _ever_ on his shift
20:29.30RDFthe work you rEALLy want to do :)
20:29.36modulus_the last time we had issues was with an upstream that was sucking
20:29.43modulus_about 6 months ago
20:30.21ChulJinI've met him, remember...more happens on his shift than you probably want to know about. :P
20:30.27modulus_omg
20:30.32modulus_disgusting
20:30.41modulus_him and his chinese exes
20:30.42modulus_lol
20:31.18ChulJinmodulus: OMG stop being so racist, you cracker.
20:31.25ArkyLadyI can't wait to get my fbsd box installed today
20:31.28modulus_chuljin, last time we went to a sul jip, we played "i never" and someone said "i never dated only asians"
20:31.33modulus_he was the only one that drank
20:31.34modulus_haha
20:31.36bkw_ArkyLady I still dont like 5.x fbsd
20:31.46ArkyLadywhy not?
20:31.47modulus_bkw, i've had no probs with 5.3
20:31.53modulus_5.0 had issues
20:31.55modulus_ick
20:31.58bkw_ya
20:32.01bkw_I kinda got burned by it
20:32.12ArkyLadyI'm installing 5.3-stable snapshot
20:32.23modulus_bkw, 5.3 is good though
20:32.38ArkyLadyI bet 5.0 was a mess when it was first released
20:32.42modulus_arkylady, lots of our clients run that for production
20:32.48bkw_ick
20:32.48modulus_arkylady, 5.0 was horrible
20:33.00ArkyLadyI haven't had my hands on a fbsd box in almost a year, so I'm excited :)
20:33.01bkw_ArkyLady you have to get that linux boxen setup too :P
20:33.10*** join/#asterisk odie_flocon (~Odie@ptr-64-201-182-209.ptr.terago.ca)
20:33.12ArkyLadyI will :)
20:33.19freatholy crap... I just ran a test (iperf using 80Kb UDP stream) and we're getting 76% packet loss at one of our sites
20:33.20bkw_ArkyLady you have your SBC line hooked up yet?
20:33.25ArkyLadywanna get my network going first so I don't have to keep moving things around to get net access
20:33.35ArkyLadynope, not yet
20:33.41freatit's one of those wireless line-of-sight providers.
20:33.51ArkyLadyprobably be 2-3 weeks
20:33.54ChulJinshould MP3Player only work for the first caller? the other day, just for Ss&Gs, I had exten=>1234,1,MP3Player(http://someplace.org:8080/something.pls) ... but only the first person calling that extension got the stream...2nd and subsequent heard only slicence.
20:33.57bkw_ArkyLady I still think that cable providre is on crack
20:34.00odie_floconall wireless providers are line of site
20:34.05freatwe just got off the phone with them... apparently it's "wind" related (something blew over a dish or so)
20:34.07ChulJiner s/slicence/silence/
20:34.14ArkyLadyit'd be better probably if I got a better level of service
20:34.22bkw_ArkyLady hehe
20:34.23ArkyLadybut I don't want to get into a contract, so that's why I did this lite thing they have
20:34.26ChulJinfreat: have them reroute you so your upstream is downwind
20:34.34freathahahahhaha
20:34.49*** join/#asterisk wolfson (~hehe@65.174.122.198)
20:34.54ChulJinwind-related network issues :P
20:34.55ArkyLadylol
20:35.01ChulJinmust be carrier-pigeon sneakernet
20:35.05odie_floconhehe
20:35.05eKo1only in a hurrican
20:35.07freatit's BS I think we're gonna switch SOON
20:35.13eKo1*hurricane
20:35.17ArkyLadyor a tornado
20:35.17freatgo to bonded Ts instead
20:35.34Slaintemodulus,  I had to put the SIP login name in the THIRD PARTY name section,  not the Auth User ID
20:35.40outtoluncmust be DSPT (distributed swaying pine tree)
20:35.42odie_floconAny wireless service you will have bigger issues with
20:36.20freatyeah no kidding... wasn't my choice to go with these guys.
20:36.34odie_floconI saw one guy trying to get wirless, and a tree was in the way. so he cut it down, and it fell on his boat.
20:36.50ArkyLadydoh
20:36.54freatthe office needed connectivity fast, and they could get it there.
20:37.49bkw_haha
20:37.49bkw_doh
20:37.49modulus_slainte, i have no clue what you're talking about
20:37.49bkw_ArkyLady I use that all the time too
20:37.49odie_floconand the other funny thing was the ISP told him he couldn't get service even if he cut it down.
20:37.49bkw_you got me to using w00t all the time
20:37.49Slaintemodulus,  getting my polycom phone to register
20:37.49ArkyLadyhaha
20:37.49modulus_oh
20:37.49ArkyLadyubah :D
20:37.54NormAstWould this be a bug in *?  I connected to asterisk with asterisk -r and the session disconnected.   It caused my CPU to spike at 100%   Bug?
20:37.57modulus_salinte, i didn't even bother with tftp/ftp i just manual ip then web interface
20:37.58SlainteWhat the hell is the Auth User ID for anyway?
20:38.06modulus_normast, that's freebsd right?
20:38.08SlainteThats how I have it setup now
20:38.12odie_floconno bug
20:38.13NormAstNope...Debian
20:38.17gabb0Is anyone using Dell poweredge 1800 here ?
20:38.18modulus_ick
20:38.19odie_floconppl do that all the time.
20:38.28Slaintebut when I manually added it, the Auth User ID is not where the SIP username is.
20:38.30NormAstgabb0: Dell PowerEdge 1750
20:39.00eKo1i have a 2850
20:39.12ChulJinthank goodness for little miracles: I got my spa3k working the other day
20:39.26gabb0we have used many different dell poweredges but we have 3 1800's right now and all are experiencing strange issues
20:39.41NormAstgabbo: like what?
20:39.45eKo1isn't there a #dell channel in here
20:39.54*** join/#asterisk ` (MoTD@212.77.220.234)
20:40.09NormAsteKo1: Alot of People use * on Dell hardware.
20:40.21eKo1yeah, me
20:41.17SlainteWe use the DL140 as our base system for install
20:41.31ChulJinI wonder if my * server even has a brand. :P
20:41.37ChulJinmodulus, go look. :P
20:41.54modulus_chuljin, the serials are scratched out
20:42.02modulus_i thought ppl did that only to weapons
20:42.06modulus_haha
20:42.07ChulJinhaha
20:42.09SlainteOr office runs off a Fujitsu life book with an 833 transmetta processor
20:42.21ChulJinJoe's Basement Server Builders
20:42.37modulus_perfect for joe's basement voip provider
20:42.47Slainteanyother,  Previously Enjoyed Server,
20:42.53eKo1anybody here running a parallel sysplex?
20:43.13ChulJinhey, $65 on ebay...can hardly beat that.
20:43.26modulus_that's a really good deal
20:43.39modulus_chuljin, you should see our "test" cabinet now
20:44.13modulus_full of generic 1u's
20:44.37*** join/#asterisk Umaro (~umaro@c-24-22-76-14.client.comcast.net)
20:44.42modulus_i go to parties and ppl come up to me "hey modulus"
20:45.22Umarohey guys, having a small problem dialing a siemens phone system over a PRI link with *.. It doesn't seem to be sending the dnid correctly
20:45.29eKo1i'm sorry. why do they call you mdulus?
20:45.47ChulJin...or you go downstairs and see some guy reading a perl book
20:45.48eKo1i mean modulus
20:45.59modulus_haha chuljin
20:46.05modulus_how many moons ago was that?
20:46.05ChulJineKo; because he calls himself modulus
20:46.19eKo1your name is modulus?
20:46.24modulus_it sure is
20:46.53ChulJinmoons? um...er...65-70 I would say
20:46.58modulus_roughly
20:47.22eKo1interesting name. I wouldn't mind being named after a math operator either.
20:48.13tzangerha
20:48.47eKo1integral isn't an operator
20:48.50hermiethe integral is the coolest math symbol
20:48.58tzangerit is?
20:49.03tzangerit's a stretched-out S
20:49.07UmaroHas anyone here integrated * with a siemens phone system?
20:49.10hermieat least the coolest looking one
20:49.18eKo1no it isn't cool looking
20:49.40eKo1if you want cool looking symbols, look at a lamba calculus book
20:49.55tzangerI am fond of the o with a tail (can't remember the greek alphabet), the partial derivative symbol and tau, it's like pi that got a leg eaten off
20:50.11*** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net)
20:50.12eKo1o with the tail is small sigma
20:50.34modulus_i like the russian backward-k-forward-k
20:50.46hermietzanger: you get those sigmas in chemistry too ya know
20:50.56tzangerlambda is a neat symbol too but half-life has overplayed it
20:51.00tzangerhermie: yeah I know
20:51.09tzangerI did rather good at chem, I just couldn't stand all the rote memorization
20:51.21tzangerbiology was my favourite, but physics was cool too
20:51.36ChulJinUmaro: you mean like a gigaset?
20:51.42eKo1tzanger: linux is also about memorization
20:51.55tzangerthe German scharf is my favourite, the B with tail and not quite closed bottom loop
20:52.09hermielike a capital Beta
20:52.10tzangereKo1: not really, it' smore about have a strong set of fundamentals and understanding the system
20:52.14eKo1tzanger: i think it is called esset
20:52.25ChulJintzanger: the korean hiyuh
20:52.30tzangerchem is like that too but the level of memorization to understanding was far too high, especially when you got into spins and stuff
20:52.31eKo1tzanger: yeah right. try keeping track of all the commands
20:52.37*** join/#asterisk ke4qqq-1 (~savirc@static-cb-68-115-212-155.spa.sc.charter.com)
20:52.39tzangereKo1: I do :-)
20:52.40*** join/#asterisk dude_where (~ashly@68.23.107.51)
20:52.45eKo1same as chem.
20:52.46ChulJinlooks like a featureless face with a hat
20:52.51RDFmanxpower you here?
20:53.19modulus_haha
20:53.22RDFAnai Hasaio Chuljin :)
20:53.31modulus_just like those Merck manuals
20:53.46ChulJinRDF: nice try
20:53.52modulus_lol
20:53.57modulus_chuljin, school him
20:54.10modulus_chuljin, that took me a few seconds though
20:54.11modulus_hahaa
20:54.17RDFUm I saw it to korean people all the time. Yea mabey the spelling is wrong but I say it right.
20:54.25modulus_word
20:54.44modulus_rdf, are you white?
20:54.45eKo1i used to watch a lot of korean tv. never learned anything though
20:55.18eKo1i know more japanese than korean and korean is so much easier
20:55.39*** join/#asterisk brettnem (~Brett@208.54.232.29)
20:55.39modulus_eko, i've never heard that before
20:55.48ChulJintzanfer: http://www.unicode.org/cgi-bin/GetUnihanData.pl?codepoint=1112
20:55.57eKo1modulus_: strange isn't it
20:56.01RDFthis is anoying. I know telemarketers call and there callerid does not show up on the phone, a 3 second pause then thay respond but what if there is no one on the other end? we get that some times. mabey there calling servers are on the fritz :)
20:56.43*** join/#asterisk xlyz (~xl@213-156-52-112.fastres.net)
20:57.07ChulJinmod: merck manual is not that big
20:57.11RDFmodulus, am i white... what kind of question is that?
20:57.12ChulJinmerck index is tho
20:57.33RDF:)
20:57.34*** join/#asterisk Dorphalsig (~Dorphalsi@69.79.38.69)
20:57.40DorphalsigHello....
20:57.45hermie"nothin but the ded of nick back in my little town"
20:57.51hermiehello Dorphalsig
20:58.03Prowler1http://blogs.zdnet.com/Ou/index.php?p=25
20:58.22brettnemHey all, long time no talk
20:58.48DorphalsigI have a queue defined (3 channels are members of the queue) , how do I know what number did which channel pick up?
20:58.50DorphalsigI mean
20:58.50eKo1anybody use sangoma hardware with their * box
20:58.57ChulJinrdf: not anai...annyung or annyong or anyung or... (there are a couple of different romanisation standards, but rules fly out the window on IRC)
20:59.21brettnemDorphalsig: you'll typically use an annoucement for that
20:59.39brettnemDorphalsig: Or I've also entered it into Caller ID successfully
20:59.43modulus_RDF, skin color question?
20:59.54RDFI was not the one asking
20:59.55RDF:)
20:59.56Dorphalsigis there anyway of having a php application read that from somewhere?
21:00.16ryguyi am having an issue with being unable to call in a DID number on a pri into a zap phone off of a Channel bank, but it works to sip phones with nearly the same config
21:00.16brettnemwhat do you mean
21:00.33DorphalsigI want a php page to monitor my agent's channel
21:00.40Dorphalsigso when he picks up the call
21:00.51RDF<PROTECTED>
21:00.52Dorphalsighe gets the caller id of ther person on the line
21:01.08eKo1Dorphalsig: agi
21:01.12file[laptop]I want this, I want that, yow'sa you people always want something
21:01.15brettnemDorphalsig: there are a few queue monitors out there
21:01.18brettnemew agi
21:01.38Dorphalsigbrettnem.. any one you'd suggest?
21:01.50brettnemyou know.. I saw some cool solutions using "nc" and .. hmm what is that popup program for TiVo caller id? anyone?
21:02.01modulus_nc netcat?
21:02.10brettnemyeah nc.. but I'm thinking abotu the other end
21:02.29brettnemit's a simple udp listener and displays a definable popup on the reciver side
21:02.39brettnemit's on the wiki.. really nice actually
21:03.01tzangerbrettnem: I use jabber and a little perl script: www.mixdown.ca/~andrew/astbot
21:03.13modulus_oooh astbot
21:03.25brettnemthat's cool
21:03.52brettnemyeah you could send it to a messenger..
21:03.56tzangerit just comes online and goes offline (caller id is its "status message" -- makes the popup 10x nicer than actually giving you a message
21:04.17brettnemthis does a nice caller id screen pop in the system tray on windows
21:04.28brettnemthe "windows notification" area
21:04.45SlainteGot SIP response 415 "Unsupported Media Type"       <------any ideas?
21:04.55Dorphalsighey thanks!
21:05.01modulus_slainte, try different codec
21:05.22Slaintemodulus  what do you use for your polys?
21:05.27DorphalsigI'll try to use astbot
21:05.42Dorphalsigand if I cant manage to use it, you'll have me around here bugging ya again :)
21:05.46DorphalsigThanks
21:05.46brettnemDorphalsig: Take a look around for that screen pop thingy. I'm looking it up. it was real nice.
21:06.19*** part/#asterisk xlyz (~xl@213-156-52-112.fastres.net)
21:06.20Dorphalsigbrettnemm... I was kinda thinking to have a php script monitor the channel I guess
21:06.20eKo1astbot only works with jabber right?
21:06.29tzangereKo1: correct, it's a jabber bot
21:06.39eKo1hmm...i don't use jabber
21:06.41modulus_slainte, hold up
21:06.48Dorphalsigand having it refresh every xxx seconds
21:06.55Slaintety
21:07.06Dorphalsigso when my agent captured the call she would know the caller id....
21:07.34ChulJineko: but if the jabber server has connectors (I forget the term) for other services, and the account astbot uses to log in has 'subaccounts' for those services...
21:07.56brettnemDorphalsig: Look here: http://www.voip-info.org/wiki-Asterisk+call+notification
21:08.05*** join/#asterisk jcollie (~jcollie@161.210.6.51)
21:08.23brettnemDorphalsig: YAC is the method I was suggesting..
21:08.33ryguywould someone here be willing to look at an error message and take a stab at what is wrong...
21:08.38modulus_slainte: here is my sip peer to ip300
21:08.38modulus_<PROTECTED>
21:08.38ryguyi would rather not post to the list
21:08.38modulus_<PROTECTED>
21:08.38modulus_<PROTECTED>
21:08.38modulus_<PROTECTED>
21:09.03Slaintemodulus,  thanks.  I have GSM and ULAW
21:09.04Slaintehmmmm
21:09.13SlainteI will stab away.  thanks for the lookup
21:09.15modulus_dtmfmode = inband
21:09.16modulus_btw
21:09.19Slainteahhhh
21:09.26Slaintelemme confirm
21:11.40Slaintedidn't like GSM first.  I changed the order and put ulaw first and it likes it
21:12.11*** join/#asterisk Dorphalsig (~Dorphalsi@69.79.38.69)
21:12.19DorphalsigSorry damn google popup blocker
21:12.25Dorphalsigwouldnt allow me to see the site
21:12.32Dorphalsig:(
21:12.44Dorphalsighttp://www.voip-info.org/wiki-Asterisk+call+notification
21:12.44eKo1are you on winblowz?
21:12.51DorphalsigYes....
21:12.58DorphalsigIt a good platform to play GTA :P
21:13.08eKo1that's worse than being here logged in as root
21:13.25Dorphalsighehehehe
21:14.05eKo1but yeah, i do agree with you. windows is good for games
21:14.15Dorphalsigyou were saying something about YAC
21:14.19eKo1i play quake 3 arena with my colleagues all the time
21:14.30DorphalsigI had Doom 3 here for a while
21:14.34eKo1until the boss comes
21:14.37ryguysomeone want to take a shot at this one?
21:14.39DorphalsigLol
21:14.42ryguy<PROTECTED>
21:14.42ryguy<PROTECTED>
21:14.42ryguy<PROTECTED>
21:14.42ryguy<PROTECTED>
21:14.42ryguy<PROTECTED>
21:14.42ryguy<PROTECTED>
21:14.44ryguy<PROTECTED>
21:14.46ryguy<PROTECTED>
21:14.48ryguy<PROTECTED>
21:14.50ryguy<PROTECTED>
21:14.52ryguy<PROTECTED>
21:14.57eKo1uhh, pastebin
21:15.22ryguy?
21:15.22*** join/#asterisk PBXtech (~nik@67.107.241.3.ptr.us.xo.net)
21:15.36eKo1lucky you didn't get kicked
21:15.36tzafrir_homeryguy: thou shalt not flood
21:15.41eKo1~pastebin
21:15.42jbotfrom memory, pastebin is a place to paste your stuff without flooding the channel - try pastebin.ca
21:16.08eKo1and anyways that doesn't say much
21:16.22PBXtechis CDR stored to a flat file by default?
21:16.29eKo1you need to debug
21:16.50eKo1PBXtech: yep
21:16.52*** join/#asterisk TrIpLeFFF (kvirc@Toronto-HSE-ppp3880909.sympatico.ca)
21:17.02eKo1check the logs, it should be there
21:17.10TrIpLeFFF~seen czero
21:17.12jbotczero <~me@CPE0090f800c5b0-CM001225704b6e.cpe.net.cable.rogers.com> was last seen on IRC in channel #asterisk, 4h 14m 40s ago, saying: 'bjohnson correct from canada'.
21:17.41progcaribouhi there
21:18.05progcaribouanybody uses the spa-3k?
21:18.09eKo1i use the spa-2100 and the spa-1001
21:18.09PBXtechwhere is the CDR stored by default?
21:18.19eKo1PBXtech: i told you to check the logs
21:18.26*** join/#asterisk mikesander (~mike@202-172-121-50.cpe.nsw-5.comcen.com.au)
21:18.35machinehdWhen setting up IAX trunks how many digits is recommended to access it?
21:18.40ryguywhat log should I be looking at, nothing shows up in messages, is there a debug I should be using?
21:18.47bjohnsonprogcaribou: I have a 3000
21:19.09bjohnsonmachinehd: enough to not conflict with the dialplan
21:19.14progcaribouUnable to create channel of type 'SIP'
21:19.14progcariboueKo1: I get "Unable to create channel of type 'SIP'" fro fsx -> asterisk -> pstn
21:19.17DorphalsigeKol... so I just need to install netcat on my * box?
21:19.17eKo1ryguy: debug channel on the cli
21:19.19progcaribouany ideas?
21:19.22*** part/#asterisk jcollie (~jcollie@161.210.6.51)
21:19.43eKo1progcaribou: are you sure that thing is online and registered with asterisk?
21:19.47bjohnsonprogcaribou: does fxs to asterisk work?
21:20.00progcariboubjohnson: yes
21:20.03*** join/#asterisk Derkommissar (~Loving@66.64.215.7.nw.nuvox.net)
21:20.14progcaribouI used the voxilla configurator
21:20.31eKo1screw that. log into it and check the settings
21:20.31bjohnsonprogcaribou: for asterisk?
21:20.32progcariboueKo1: think so ...
21:20.40progcaribouyes
21:20.55bjohnsonlog into asterisk -r and do sip show peers
21:21.02progcaribouUnable to create channel of type 'SIP'
21:21.02progcariboui did
21:21.20bjohnsonyou got that error for sip show peers?
21:21.23ryguyeKo1, what would the channel name be for Zap 2-1?
21:21.25progcaribouI always get the Unable to create channel of type 'SIP' message
21:21.37PBXtechis there an application that can view the CDR logs?
21:21.40progcaribouno sip debug peer ip
21:21.47bjohnsondo you have a sip.conf that asterisk has file access to?
21:21.51*** join/#asterisk allyour80211b (~allyour80@208.178.154.99)
21:22.01progcaribouyes
21:22.02mikesanderI am getting errors when putting people on hold, from the ilbc codec:
21:22.03mikesanderNOTICE[3510]: rtp.c:489 ast_rtp_read: Unknown RTP codec 98 received
21:22.06modulus_sip channels need answer confirmation just like zap channels
21:22.08bjohnsonlog into asterisk -r and do "sip show peers"
21:22.09modulus_how come there isn't one?
21:22.10mikesanderanyone got any ideas?
21:22.12eKo1ryguy: zap/2
21:22.23bjohnsonprogcaribou: log into asterisk -r and do "sip show peers"
21:22.37progcariboupstn-spa3k/aste  (Unspecified)               0.0.0.0          5060     Unmonitored
21:22.37progcaribou1001/1001        192.168.1.55     D          255.255.255.255  5061     Unmonitored
21:22.37progcaribou1000/1000        192.168.1.55     D          255.255.255.255  5060     Unmonitored
21:22.54*** join/#asterisk iroN_Man (~LoVeGiRL_@mstr195175-8327.dial-in.ttnet.net.tr)
21:22.55*** join/#asterisk SouL (~ReNGiN_du@mstr195175-8327.dial-in.ttnet.net.tr)
21:22.56ryguyeKo1, No such channel zap/2
21:22.59eKo1which one of those is it progcaribou ?
21:23.01bjohnsonprogcaribou: they're not registering
21:23.15progcaribouthere is no firewall/nat between asterisk and the spa3k
21:23.21SlainteI have someone flooding my pipe.  Anyone know the cisco command to show what sessions it is managing
21:23.42bjohnsoncheck the user and secret in the SPA 3000 Line 1 and PSTN admin, advanced pages match the ones in sip.conf
21:23.46progcariboueKo1: 1000/1000
21:24.03progcariboubjohnson: what should I do
21:24.04progcaribou?
21:24.07bjohnsoncheck the user and secret in the SPA 3000 Line 1 and PSTN admin, advanced pages match the ones in sip.conf
21:24.34bjohnsonprogcaribou: also add a qualify=yes to both sections in sip.conf
21:24.41eKo1ryguy: hmm...to be honest, i don't know what the nomenclature is
21:24.42bjohnson(or the general section)
21:26.00bjohnsonryguy: I think Zap/1 refers to group 1 devices (as set in zapata.conf)
21:26.23bjohnsonryguy: so Zap/2 would be devices in group 2
21:26.54bjohnsonryguy: zaptel devices can be grouped .. I don't think sip.conf or the other confs support this
21:27.16progcariboubjohnson: the Line 1 is set to 1000 and the PSTN is set to 1001
21:27.49*** join/#asterisk fman (~blah@203-79-119-214.cable.paradise.net.nz)
21:28.01fmanquick question
21:28.24TrIpLeFFFgm go
21:28.27progcaribouI reloaded asterisk with the qualify=yes, but the same error ...
21:28.28fmancan I plug my analog line into an asterix system? and then wifi/bluetooth that to a headset
21:28.30fmanin my house
21:28.35TrIpLeFFFyes
21:28.44*** join/#asterisk SlickShoe (SlickShoes@NKR.RES.cmu.edu)
21:28.48fmanwhat hardware for analogue inbound?
21:28.53TrIpLeFFFfman:  get a fxo card
21:28.59TrIpLeFFFX100p minimum
21:29.01fmana mythtv front end would be the go
21:29.02TrIpLeFFFthey like 40$
21:29.20TrIpLeFFF~jbot mythtv
21:29.27fmanurl? or x100p ?
21:30.04fmanok
21:30.05fmangot it
21:30.06TrIpLeFFFdigiump.com
21:30.09TrIpLeFFFoups
21:30.11TrIpLeFFFdigium
21:30.18TrIpLeFFF~jbot x100p
21:30.19jbotfrom memory, x100p is an obsolete card, copied by far too many people
21:30.27modulus_why doesn't SIP support answer confirm via "#" like Zap chans?
21:30.27TrIpLeFFFhehe
21:31.44bjohnsonprogcaribou: for userid?  and authid is set to no?  and a password is set?
21:31.53DorphalsigEkol....  the yac notification wont work for me... I think
21:32.08Dorphalsigbecause I need only the agen who picks up the call in the queue to be notified
21:33.46fmanso, what card to get then if x100p is obsolete then
21:34.58ryguyeKo1, take a look at this
21:34.59ryguy<PROTECTED>
21:34.59ryguy<PROTECTED>
21:36.47Dorphalsigok, I ask the whole thing again in case anybody can give me a hand w/that :)... I have a queue, and I need the agent who is picking up the call to know the caller-id of the person on the line
21:37.13DorphalsigBefore I got told here to use YAC as described in http://www.voip-info.org/wiki-Asterisk+call+notification
21:37.26nestArfman: you'll have to get a TDM400
21:37.32Dorphalsigbut the thing is I dont have ONE channel, I have a queue line with three member channels
21:37.32fmanhnmm
21:37.34progcariboubjohnson: now I get: Got SIP response 403 "Forbidden" back from 192.168.1.55 which is the spa3k
21:37.37fmanthats a big price difference
21:37.39progcaribouis it auth problem?
21:37.58fmanso, whats this about myth then?
21:38.09Dorphalsigany ideas?
21:38.29bjohnsonprogcaribou: find the wiki page about spa 2000 and follow the link on it to the asterisk and SPA 2000 howto hosted on voxilla
21:38.40bjohnsonprogcaribou: get the fxs port working following that howto
21:38.58bjohnsonprogcaribou: then, we can worry about the fxo port (most of the config is the same)
21:39.10progcaribouok
21:39.15progcaribouthaks a LOT
21:40.16bjohnsonprogcaribou: for homework, if you want to work ahead, there is a voxilla forum post detailing how to get the spa 3000 to forward the callerid to asterisk without first pivking up the line
21:41.06bjohnsonprogcaribou: those 2 docs (and a little extended logic) will get both working.  there are a couple of tricks with the fxo so I'll expect to see you here tomorrow
21:41.21bjohnsonprogcaribou: estimated time to complete assigned task :  30 minutes
21:42.07Derkommissardoes asterisk support   rtp-nte    RTP Named Telephone Event RFC 2833
21:42.10Derkommissar?
21:42.27bjohnsonprogcaribou: then email sipura and bitch that there is no way to save a config to file .. then I could have just emailed you a copy
21:42.32progcaribouthank "professor" bjohnson :-)
21:42.44Dorphalsigbye
21:43.42file[laptop]Derkommissar: yes
21:43.51Derkommissar:-/
21:43.51[Sim]say
21:43.55[Sim]if you have a queue
21:43.56file[laptop]Derkommissar: aptly it's enabled by setting dtmfmode=rfc2833 in sip.conf
21:43.56Derkommissarnot working for me
21:43.59[Sim]and some members have a penalty
21:44.10[Sim]what does it take for those members to be rung ?
21:44.11[Sim]I mean
21:44.18Derkommissaryup
21:44.23hermiethe Asterisk Bookclub is officially open for business :-)
21:44.31Derkommissarim not reciving the dtmf from a cisco
21:44.36[Sim]if I have all other members return 'Busy' it still doesnt seem to take the penalty members
21:44.37Derkommissarfrom a 3600
21:44.45file[laptop]Derkommissar: see if it's set for inband
21:45.07Derkommissardial-peer voice 20000 voip
21:45.07Derkommissar<PROTECTED>
21:45.07Derkommissar<PROTECTED>
21:45.07Derkommissar<PROTECTED>
21:45.07Derkommissar<PROTECTED>
21:45.08Derkommissar<PROTECTED>
21:45.45cripitohermie url?
21:46.02Derkommissarfile and in sip.conf
21:46.09Derkommissar[66.64.215.15]
21:46.09Derkommissarcontext=cisco
21:46.09Derkommissartype=friend
21:46.09Derkommissarhost=66.64.215.15
21:46.09Derkommissardtmfmode=RFC2833
21:46.45file[laptop]that's a horrible way to set it up...
21:47.02file[laptop]lemme write up a nice one
21:47.47*** join/#asterisk _Brian (brian@unix01.voicenet.com)
21:48.00file[laptop]Derkommissar: http://pastebin.ca/4113
21:48.47Derkommissarwhat does the insecure do :-)
21:48.59file[laptop]matches based on IP address
21:49.42*** join/#asterisk samueltc (~samuel@levinux.UQAR.UQUEBEC.CA)
21:50.10Derkommissarnot getting dtmfs :-/
21:50.11file[laptop]actually I still don't quite like it, swap the positions of context and type
21:50.13*** join/#asterisk Caede (~chatzilla@204.94.248.81)
21:50.14samueltchi
21:50.34mikesanderhi
21:50.35mikesanderI am getting errors when putting people on hold, from the ilbc codec:
21:50.37mikesanderNOTICE[3510]: rtp.c:489 ast_rtp_read: Unknown RTP codec 98 received
21:50.39*** join/#asterisk zotz (~zotz@24.244.133.136)
21:50.40mikesanderanyone got any ideas?
21:50.42*** join/#asterisk yaout (eric@CPE-65-30-220-56.wi.rr.com)
21:50.55eKo1mikesander: yeah, stop using ilbc
21:51.09mikesanderwhat is best to use
21:51.15eKo1ulaw/alaw
21:51.24mikesanderwe are in australia and have limited bandwidth - like 512k ADSL
21:51.31mikesanderi want to run 10 lines over this
21:51.33eKo1then gsm
21:51.35file[laptop]Derkommissar: it's probably your Cisco config fyi, as rfc2833 is what everyone uses
21:51.35SlickShoeDoes asterik & voip work well with faxing? Are there some providers that are better than others and anyone know where I can find a list of asterik compatible providers especially with reviews?
21:51.49hermiecripito: #astersik-bookclub
21:52.06modulus_jbot .dk?
21:52.07jbotsomebody said .dk was the TLD for Denmark
21:52.17modulus_jbot .nk?
21:52.25mikesanderthanks eK01
21:53.27SlickShoeasterisk is not meant to be used with a provider like vonage, is it?
21:53.48freathey are they any providers out there that support securing your IAX2 traffic (vpn or so) ?
21:53.51file[laptop]SlickShoe: please Google... please
21:54.07freatSlickShoe: vonage is evil
21:54.29SlickShoeK
21:54.33SlickShoethanks
21:54.39file[laptop]I just typed asterisk vonage into Google and got the answer
21:54.40file[laptop]it's that easy
21:54.48bjohnsonSlickShoe: there are hundreds of voip providers .. please don't use vonage
21:55.02bjohnsonSlickShoe: broadvoice has been a popular topic today
21:55.17freatSlickShoe: vonage locks down all their equipment etc, you'd be better off with someone who supports IAX2 protocol
21:55.25bjohnsonSlickShoe: I don't use voip for fax but it's supposed to be possible .. even though a pita to set up
21:55.59bjohnsonpeople laugh at the web page .. but I like my provider and they are part of the * community
21:56.20modulus_jbot .kp?
21:56.24freatbjohnson: who's your provider with this web page?
21:56.29bjohnsonaleph
21:56.44machinehdis there a better IAX guide than the one on the wiki?
21:56.48bjohnsonit's been massively improved in the last 2 weeks
21:56.57freatwell, looks like the wind died down... our connection is back up hehe
21:57.25freatmachinehd: by IAX you mean a reference for the protocol? If so there's info on that on Digium's site
21:57.28*** join/#asterisk ToyMan (~stuq@user-12lcqq2.cable.mindspring.com)
21:57.28modulus_jbot .kp?
21:57.29jbotTLD for Democratic People's Republic of Korea (North korea)
21:57.42*** join/#asterisk implicit (~implicit@ip68-5-148-1.oc.oc.cox.net)
21:57.52modulus_google doesn't own google.kp
21:57.54machinehdfreat, well in regards to configuring 2 boxes to work together
21:57.58bjohnsonmachinehd: the conf for iax is very close to the config for sip
21:58.24bjohnsonmachinehd: they both have static ips or does one have dynamic?
21:58.26freatmachinehd: it isn't that bad at all actually. are both on static IPs?
21:58.27eKo1modulus_: i don't think google does communist countries
21:58.28freathehe
21:58.47machinehdboth are static, but ones is ipmapped to a private
21:59.06modulus_they're not communist
21:59.08freatjust make sure to forward all 4569 traffic to the one that is behind nat
21:59.20modulus_they're just KJ-ist
21:59.28bjohnsonif both have a static internet ip, then you don't need to register .. use the host option in the device section
21:59.37bjohnsonand forward as per freat
22:00.00freatthen you just need to construct your dial command correctly to include the user:password
22:00.14bjohnsonmachinehd: then set the username and secret on each side and try dialing from one to the other
22:00.19bjohnsonhehe
22:00.48machinehdgreat, thanks guys :) I'll try it right now
22:00.53freatbasically, when one * box connects to the other, the usual setup is to send the extension that was dialed to the other box.
22:00.57*** join/#asterisk cicide (chris@photon.netgeeks.net)
22:01.18freatthe other box then decides with it's dialplan, and the context that the dialing box is in, what to do with that extension
22:01.54machinehdso then which box logs the call for billing?
22:01.58machinehdboth?
22:02.10cicideHey folks, I'm looking for someone familiar with the CFAS group code.  I've got a strange problem with a CFAS group from SBC
22:02.19*** join/#asterisk Gimmemylanta (~Gimmemyla@adsl-64-223.swiftdsl.com.au)
22:02.50Derkommissaris there any free PPC sip phones that support small codecs like ,,, g723 ?
22:03.25freatmachinehd: I don't do billing, but you can set it for individual IAX accounts
22:03.35eKo1free sip phone <--- get a softphone
22:03.58file[laptop]Derkommissar: G723 and G729 cost, therefore you can't get 'em for free
22:04.01freatso the receiving box could do the billing for the incoming IAX from the first one. In that situation all incoming from that one IAX device would be billed to the same account
22:04.06ManxPowerHere is the licensing priceing info for G723.1 direct from the patent holder's web site: http://www.dspg.com/technology/LicensePricing.html
22:04.23modulus_jbot .dk?
22:04.24jboti guess .dk is the TLD for Denmark
22:04.36Derkommissarim talking about the software not the codec
22:04.47Derkommissaranyhow anything small enough to work over GPRS :-)
22:04.50*** join/#asterisk [1]NormAst (HydraIRC@Ottawa-HSE-ppp4122952.sympatico.ca)
22:05.05modulus_what time is it in denmark right now?
22:05.17modulus_6am?
22:05.18file[laptop]Derkommissar: the software uses the codec...
22:05.20freatGPRS? Have you tested your latency?
22:05.27Derkommissaryes
22:05.30eKo1modulus_: should be about 12:00 AM or 1:00 AM
22:05.34file[laptop]and the codec costs money
22:05.36freatI have GPRS on two phones and it sucks
22:05.51freatI get decent speed, but the latency is 1-3 seconds usually
22:05.55modulus_i want to call our client in denmark using voipjet
22:06.07ChulJincall your client in denmark using voipjet.
22:06.09Derkommissarlatancy is about 300 ms
22:06.12eKo1modulus_: you should have done that this morning
22:06.15Derkommissarthats better than satelite
22:06.20freatthat's really good
22:06.20Derkommissarwhere you get 500 ;)
22:06.24modulus_yeah i wasn't ready then
22:06.30modulus_i just finished his fbsd install
22:06.49ChulJinI'm sure he wouldn't mind being awaked for good news.
22:07.00*** join/#asterisk MasterYoda (~mnicholso@dhcp-155.digium.com)
22:07.12MasterYodais supervised '#' transfer in cvs head?
22:07.16modulus_i wonder if he speaks engrish
22:07.19eKo1installed fbsd is not good news
22:07.23cicideAnyone here that is familiar with the NFAS group code?
22:07.27*** part/#asterisk cjk (~cjk@80.92.75.85)
22:07.51*** join/#asterisk Godsey (lanny@2001:470:1f01:ffff:0:0:0:1f5)
22:07.53modulus_eko1, installed fbsd is good news to me
22:08.23*** join/#asterisk dsfr (~dsfr@zeus.booksys.com)
22:08.24eKo1installed fbsd is neither good news nor bad news. it's just news
22:08.38TrIpLeFFFbad news for zaptel
22:08.40TrIpLeFFFif bsd
22:08.58modulus_eko1, compared to rh it's good news
22:09.13TrIpLeFFFits there anyway to trace why i get socket errors on RH9 ?
22:09.38eKo1modulus_: depends
22:09.50samueltcsomeone may suggest me a voip hard phone? i've got a budgetone, work fine...
22:10.11eKo1cisco cp 7970g
22:10.27eKo1it doesn't get any better than that
22:10.37samueltchow much?$?$
22:10.44Nuggetwell, it would get better if you could put a sip firmware on the 7970.
22:11.28freatwe like the Polycom IP500s
22:11.30eKo1samueltc: about $700
22:11.34samueltcI want something SIP (iax) compatible, with no codec fee
22:11.38freatgreat speakerphone, SIP
22:11.39samueltceKo1: hehehehe too much
22:12.01freatyou can get them for under $200 each
22:12.05*** join/#asterisk ennuyeux72 (~ennuyeux7@62.53.79.208)
22:12.08samueltci want about 80 units
22:12.10eKo1samueltc: ebay dud
22:12.15eKo1s/dud/dude
22:12.32samueltcshould I go with a digital channel back?
22:12.40freatat 80 units the IP500s could be about $170-180 (we have gotten them for $169)
22:12.48progcariboubjohnson: I did what you told me but no luck ...
22:12.56samueltcfreat: ok
22:13.04*** join/#asterisk bonbon-home (~happy@81-86-0-190.dsl.pipex.com)
22:14.10*** join/#asterisk venix (~venix@Z-pc1-198-S1.gw1.tor1.sprint-canada.net)
22:15.47progcariboubjohnson: sip show peers doesn't show any peers ...
22:15.49*** join/#asterisk ZX81 (matt@222-153-51-36.jetstream.xtra.co.nz)
22:15.59samueltcchannel bank
22:16.13ZX81morning everyone
22:17.19progcaribougood afternoon
22:19.02*** join/#asterisk zoa (zoa@82.103.76.147)
22:19.05*** join/#asterisk charlesIII (~charlesII@65.171.196.23)
22:19.12zoaand asterisk made it to slashdot again
22:19.14charlesIIIhi
22:19.20ZX81:-)
22:19.22charlesIIIanyone used astcc?
22:19.25ZX81yah
22:19.29ZX81getting on to it
22:19.39ZX815 news articles waiting to be added
22:19.40ZX81:-)
22:19.43bjohnsonprogcaribou: if you changed sip.conf did you reload?
22:20.11ZX81anyone really know what the status is on HDLC on digium's T100P?
22:20.49progcaribouyes i did
22:21.36zoagoing to bed now
22:21.38zoacheers
22:21.44ZX81sweet
22:21.46ZX81night zoa
22:21.47eKo1i really don't understand how one can patent codecs. might as well start patenting the quicksort algorithm
22:21.48progcaribouciao zoa
22:22.00ZX81eKo1: now there's an idea
22:22.05ZX81I wonder if someone has
22:22.08ZX81or bubblesort
22:22.10ZX81:-)
22:22.11eKo1no, no one has
22:22.15ZX81hmmmmmm
22:22.18ZX81:-)
22:22.23*** join/#asterisk cjk (~cjk@80.92.75.85)
22:22.25eKo1i don't pay anyone to use quicksort
22:22.46progcariboubjohnson: it seems the spa3k doesn't go thru asterisk, but directly to the pstn
22:22.58ZX81:-)
22:22.58cjkhi, does anyone know a way to replicate the traffic on my linux router to another interface connected to a dedicated box for monitoring
22:23.08ST-3does making modem calls over g711ulaw work?
22:23.16eKo1might as well patent irc too
22:23.18progcaribouI did what the howto said for line 1
22:23.54*** join/#asterisk mogorman (~mogorman@216.207.244.182)
22:24.03mogormanmorning
22:24.18bjohnsonprogcaribou: did you set the asterisk ip as the line 1 proxy?
22:24.39progcaribouyes
22:24.42bjohnsonprogcaribou: I follwed that howto for the fxs from factory settings and it worked
22:24.52bjohnsonhave to go now
22:24.56progcaribouok
22:24.59progcaribouthanks again
22:29.45redder86anyone get this in their polycom logs: ?  Error: soCoreAudioPropSet: Profile for TX codec (3) not found
22:31.18*** join/#asterisk JimVanM (~JimVanM@HSE-MTL-ppp77210.qc.sympatico.ca)
22:31.59*** join/#asterisk JonR800 (jr@pcp05013027pcs.plyntv01.mi.comcast.net)
22:35.18Derkommissarhas anyone here used x-pro for win ce ?
22:35.54tzafrir_homeeKo1: the only reason you cannot patent quicksort and IRC is because they were invented over 20 years ago and well-documented.
22:36.14mishehubah.
22:36.21antiArgh, anyone have trouble reaching broadvoice? I call, sit on hold for 5 minutes then get hung up on.
22:37.02modulus_you're calling the company?
22:37.36eKo1tzafrir_home: what does time have to do with anything
22:37.45nestArwhere can i download the bootrom image and other files to setup a bootserver for my Polycom phones
22:37.48Yoda-BZH`ZzZune bonne nuit je vous souhaite ! / A good night I wish you
22:38.29*** join/#asterisk Saric (~konversat@199.243.57.65)
22:38.39develis there any way to have asterisk report what context it's in for each step?  i'm trying to trace some dialplan issues...
22:38.55*** join/#asterisk heison (~heison@ns.somanetworks.com)
22:39.13tzafrir_homeprior art. Too well known. The fact that something is trivial doesn't exactly count because even the most trivial claim can be obfuscated as something really innovative
22:39.25heison~seen JerJer
22:39.26jbotjerjer is currently on #asterisk (5d 17h 26m 51s).  Has said a total of 118 messages.  Is idling for 2h 38m 30s
22:40.02eKo1tzafrir_home: what about sip and iax. i don't see a patent for those
22:40.15tzafrir_homedevel: NoNp(${CONTEXT}) or something
22:40.31develtzafrir_home, thanks, i'll check that out.
22:40.32tzafrir_homedevel: NoOp, that is
22:41.00eKo1maybe i'll patent sip. hehe...
22:41.41tzafrir_homeeKo1: I wouldn't bet on SIP. It is an existing standard. Others must have bitten you to t if there was a chance
22:42.09fearnorzx: what do you need about HDLC?
22:42.12fearnorit works
22:42.13ZX81:-)
22:42.16ZX81really?
22:42.18fearnorsure.
22:42.21fearnorsort of. :)
22:42.24ZX81hehe
22:42.35ZX81can you split it to HDLC and Voice?
22:42.36fearnorwhere are you stuck at.
22:42.40fearnorzx: sure
22:42.47ZX81not really stuck
22:42.52fearnorzapata.conf - nethdlc
22:42.53fearnoretc
22:42.56ZX81know how - just thought there were probs
22:42.57ZX81yeah
22:42.59fearnorread the wiki yet?
22:43.03ZX81heh
22:43.07ZX81I should think so
22:43.08ZX81:-)
22:44.48mikesanderis there a good codec comparrison page - i'm looking for bandwidth, quality and cost details
22:45.03*** join/#asterisk enzo123 (~enzo123@rdu26-62-161.nc.rr.com)
22:45.10enzo123hi all
22:45.22enzo123got a question...
22:45.28*** join/#asterisk terrapen (~cjs@fw-01.satx.bikeworld.net)
22:45.32kramthis is my first time using the net from within an aircraft
22:45.33terrapenfucking a.
22:45.38terrapenDoS on our upstream provider
22:45.43terrapenthis BLOWS
22:45.46fearnorkram: gotte jetconnect?
22:45.51fearnorjetconnect > *
22:45.51Mikeguys anyone can see in the list the email i just sent? i want to be sure it didnt get blocked it goes by the name Miguel Cavazos a few minutes ago
22:45.52eKo1tzafrir_home: so if i independently come up with a new protocol for voip, i can patent it right?
22:45.55terrapeni wish i had a dialup backup in my colo cage
22:46.07terrapenand some BGP action
22:46.16fearnorterra: get a real provider who can deal with ddos ;)
22:46.21fearnorterra: how big of ddos.
22:46.25terrapenwell, they are cheap :)
22:46.26ZX81kram: wow l337
22:46.29ZX81:-)
22:46.30terrapenseveral T3s worth
22:46.34kramyah
22:46.39tzafrir_homeeKo1: but you'll have to convince others to use it even though you have a patent on it
22:46.41terrapenfor what we pay for the bandwidth we get, this provider is a hell of a good deal
22:46.42kramthey'll make me shut it down soon i'm sure
22:46.43ZX81isn't it like $100000000000000000000 per minute
22:46.45terrapenwe will never leave them
22:46.46ZX81:-)
22:46.50enzo123if i have 3 boxes.. A---B---C  a it the provider box with a pri , B our server and C is a client * box.. the problem is that the call is being bridged to Box C after setup can we stop this so we have billing ?
22:47.05fearnorterra: i get ddos'd with >1gbps about once every two weeks ;(
22:47.05terrapenthank fucking god we don't use VoIP for our phonebank
22:47.20terrapenfearnor, what network is this?
22:47.21fearnorand handle/filter it ;)
22:47.24file[laptop]enzo123: IAX2 right? notransfer=yes in the iax.conf config for the peer... or user...
22:47.25ZX81slashdot
22:47.26ZX81lol
22:47.29fearnorterra: my network. ;)
22:47.30ZX81:-)
22:47.32terrapenwell, they are doing a good job
22:47.32fearnoras26627
22:47.35terrapenslashdot?
22:47.41enzo123file iax.. correct
22:47.46ZX81Asterisk slashdotted again
22:47.47ZX81~adn
22:47.48jbotit has been said that adn is the Asterisk Daily News - http://www.sineapps.com/news.php for HTML and http://www.sineapps.com/rssfeed.php for RSS
22:48.01implicithey file, where do you live again?
22:48.02implicitwhat state?
22:48.23*** join/#asterisk zotz (~zotz@24.244.133.136)
22:48.32file[laptop]I don't live in the US
22:48.51implicitoh
22:48.57*** join/#asterisk firestrm (F1r3570rm@S010600047577bccd.gv.shawcable.net)
22:48.58ZX81he lives on Mars
22:49.06ZX81:-)
22:49.13file[laptop]yup and we have awful ping times
22:49.41Sariccan skype connect to asterisk?
22:50.14*** join/#asterisk Pantanero (~Pantanero@a213-22-82-157.netcabo.pt)
22:50.59enzo123file so it will stay in bridged like this: jason-iax-  00004/00012  00018/00015  [Native Bridged to ID=00005]
22:51.00ZX81not really
22:51.07ZX81unless you have windows
22:51.20Saricok thanks zx81
22:51.27ZX81np
22:51.33develi have a problem that acts like a context issue.  two different incoming contexts, both include the same context, it doesn't work from one of them.  another pair of servers, same config, no issues.  ideas where to look next?
22:51.44file[laptop]enzo123: it won't transfer off your box...
22:52.13enzo123yeah that looks like it fixed it now it just shows them in native bridge
22:54.59*** join/#asterisk ebell_ (eric@CPE-65-30-220-56.wi.rr.com)
22:55.03TrIpLeFFFhey .. how much is 100kB for 2 hours
22:55.09*** join/#asterisk ArkyLady (ArkyLady@93-95.hspg-ubr2-blk1.cablelynx.com)
22:55.19TrIpLeFFFthats 100 * 60 *60 *2 ?
22:55.40eKo1man, the cli is just horrible with so many clients
22:55.52TrIpLeFFF360000 *2 = 720000K  or 720 MEG ?
22:55.57eKo1have to turn down verbosity
22:56.14TrIpLeFFFshit cinemanow.com .. that must cost alot more then they make.. 2.99 per movie.. strwamed at 100k ..
22:56.21*** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com)
22:56.27TrIpLeFFFthats . like 3.5 per gig
22:56.30terrapenlooks like the DoS has been contained
22:56.33kramcvs commit from the plane
22:56.33krami'm hard core
22:56.50ZX81no way
22:56.51eKo1the plane!?
22:56.55ZX81thats awesome
22:57.03eKo1i didn't know asterisk was that profitable
22:57.25ZX81:-)
22:57.29eKo1bastard
22:57.32ZX81Asterisk - 1
22:57.35ZX81didn't you know
22:57.36ZX81:-)
22:57.54bkw_kram youre shitting me
22:57.59bkw_you did one from the plane?
22:58.00bkw_via what?
22:58.20drumkillahe's gone now, i think ...
22:58.27drumkillabut it was gprs
22:58.32ZX81hehe
22:58.45eKo1he is gone because the transmission screwed the planes electronics so the plane is going down
22:59.00ZX81lol
22:59.08eKo1hope he has a parachute
22:59.13drumkillaSend 405 Method Not Allowed when message received outside of call (bug #3324, commited from within the plane to huntsville)
22:59.15ZX81indeed
22:59.24firestrmcellphones have no effect on aircraft electronics..
22:59.28ZX81that has to go on the news!
22:59.28eKo1then he'll cvs commit from the parachute
22:59.32ZX81lol
22:59.58eKo1firestrm: that depends on the cell phone
23:00.29firestrmeKo1. i have made 100's of calles from the cockpit, never even once had a glitch
23:01.08firestrmit just annoys the cell providers, ties up a bunch of channels
23:01.09*** join/#asterisk kevinl (~klindsay@staff-nat.netnation.com)
23:01.31devi0usfirestrm. what kind of plane?
23:01.32*** join/#asterisk jnorell (~jesse@slug.kci.net)
23:01.33eKo1let me lend you my 10 kW cell phone
23:01.42firestrmlol
23:01.51firestrmsure, lets go try it :)
23:01.53devi0usi've been working on trying to figure out how to wire a headset cable into an intercom box
23:01.57modulus_broadvoice seems to suck
23:02.04modulus_stupid SIP provider
23:02.12firestrmdevi0us a330
23:02.13*** join/#asterisk mrverizone (~chatzilla@pa-robinson1b-88.pit.adelphia.net)
23:02.27enzo123anyone hear of a chan driver for skype ?
23:02.34file[laptop]enzo123: no.
23:02.41devi0usoh. heh. that's a bit quieter than what i usually fly
23:02.51devi0usinside at least
23:02.52file[laptop]skype is, proprietary like man, omg!
23:03.21firestrmdevi0us, i started out on twin otter,, your ears ring for 4 hours after a flight
23:03.31mrverizoneHello
23:03.44*** join/#asterisk PoWeRKiLL (~PoWeRKiLL@l01m-34-60.d4.club-internet.fr)
23:03.46enzo123file didn't know that
23:03.48devi0usanyone having any issues making outbound calls with nufone or voipjet? i can only get through to 800 numbers, everything else hits congestion
23:03.50eKo1firestrm: didn't you have earplugs
23:03.55mrverizonequestion, g729 is theer any issures on the free bsd box wtih this codec
23:04.08devi0usfirestrm: i can believe it
23:04.14ManxPowermrverizone: You mean it actually runs on *BSD?
23:04.19mrverizoneyes
23:04.32devi0usfreebsd works fine
23:04.39firestrmeKo1, earplugs dont help when the noise is low frequency, it comes up you A$$ through the seat
23:05.00ZX81~adn
23:05.01jbotadn is probably the Asterisk Daily News - http://www.sineapps.com/news.php for HTML and http://www.sineapps.com/rssfeed.php for RSS
23:05.05ZX81hehe
23:05.14bkw_kram is crazy
23:05.14bkw_how l33t
23:05.16ZX81history is made with first plane comit
23:05.18ZX81reckon
23:05.20bkw_CVS commit from 23,000 ft
23:05.24ZX81:-)
23:05.25mrverizoneI am using a wireless phone, has to codec in it, G-711 and G-729, of course i would like to use the 729 codec. but i have heard taht it might not register
23:05.35bkw_quick someone post to /.
23:05.49enzo123deivous..  no problems with voipjet
23:05.53ZX81mrverizone: you need to get a g729 licence from digium
23:05.57file[laptop]bkw_: hug me hug me, hold me hold me
23:06.24mrverizonethanks you, but there is not know problems with this codec and asterisk and Free BSD
23:06.46mrverizoneokay will try, and let you know
23:06.52mrverizonethanks for the help
23:06.54*** join/#asterisk fearnor (~alex@66.250.55.66)
23:06.55firestrmwar flying can be real fun!!, however you need something slower that what i fly. otherwise you need to be really quick at 450 kts
23:07.01drumkillabkw_: that's a pretty good idea
23:07.30eKo1firestrm: have you flown any commercial airliners?
23:07.37charlesIIIso if I am configuring the trunks table in astcc... and using ZAP, under Peer/Trunk should this just be g1 for group 1 or zap/g1? neither seems to work for me.
23:07.42firestrmeKo1, thats my day job
23:08.01eKo1oh yeah, you said you fly an airbus right?
23:08.30firestrmeKo1, correct
23:08.51mrverizonehas any one on here use the cn720 motorola cell phone GSM and Sip, any one
23:08.52eKo1what airline?
23:09.03*** join/#asterisk t3t (~t3t@207.67.0.18)
23:09.20firestrmfile[laptop], with a little more notice i probbly could swing it.. i would have to come up with some convoluted way that we are related (family flys free)
23:09.43firestrmeKo1, (SC)air Canada
23:10.08eKo1has air canada purchased any of those a380s yet
23:10.11mrverizoneFirestrm they do not come to the USA do they
23:10.29file[laptop]c'mon, VON is 2 months away... ya know you wanna
23:10.52firestrmeKo1, the orders are in apparently, i begin conversion training in england in july
23:10.57modulus_broadvoice is horrible
23:11.04modulus_sometimes i receive calls sometimes i don't
23:11.06bkw_file call hell please
23:11.09bkw_paging file to hell please
23:11.12file[laptop]awwwwww
23:11.21firestrmmrverizone, scair canada flys everywhere in the world
23:11.21t3thowdy
23:11.23file[laptop]my phone, she is dead
23:11.28bkw_file you have 10 phones
23:11.29bkw_pick one
23:11.30mrverizoneI agree, we are having the same problem with broad voice
23:11.31file[laptop]one sec...
23:11.42terrapenwe got a telemarking call for lightbulbs today
23:11.47bkw_file you order your mac mini yet?
23:11.56terrapeni'm adding a "lighting" section to voicemail hell when i get * working
23:12.04t3tI have the same problem with livevoip
23:12.10terrapen"Press 5 if you selling lighting products"
23:12.12terrapenthen
23:12.19GodseyI'm driving up to get my mini mac tonight
23:12.30firestrmterrapen, thats exactly what i want to set up :)
23:12.38terrapen"Press 1 for neon lighting.  Press 2 for Blue LED lighting"
23:12.42filenooooooooooooooooooooooooooo
23:12.43Godseypay for it anyway
23:12.49t3tterrapen: Don't forget to give them a carrot... like "Meet us next week Tuesday at our vendor fair at the airport..."
23:12.50terrapen"Press 3 for gro-lights"
23:12.52Godseythey said I'll get it by the 25th :)
23:12.53terrapenheh
23:12.59*** part/#asterisk taoflier (~taoflier@dsl017-021-036.chi1.dsl.speakeasy.net)
23:13.05terrapeni will put incandescent and flourescent lighting at the end
23:13.13terrapenof several "pages" of menus
23:13.16enzo123anyone know iax.cc ?
23:13.30firestrmterrapen, dont forget coal oil lighting ;)
23:13.34terrapenhahahahahah
23:13.36terrapenyeah
23:13.47modulus_US-CERT Technical Cyber Security Alert TA05-012A -- Multiple Vulnerabilities in Microsoft Windows Icon and Cursor Processing
23:13.48modulus_haha
23:13.52modulus_windows icon?
23:13.52modulus_haha
23:14.07eKo1not the icons, the icon processing
23:14.07firestrmif you sell coal lighting product press 7, if you sell coal press 8
23:14.27terrapencandles, press 9
23:14.28t3tDoes someone know how to set the default incoming callerID for T400 fxo?  "asterisk" is confusing to the users :)
23:14.34eKo1probably something to do with gdi or gdi=
23:14.38eKo1eh, gdi+
23:14.40firestrmlol
23:14.59terrapena requirement for Voice Mail Hell is to record the entire call
23:15.07ckruetzeWhere do I report bugs with the bugtracker?
23:15.07terrapenand listen to what the caller says as he/she listens to the prompts
23:15.33firestrmterrapen, i want mine to start out with, for service in english press 1 for service in french please hang up
23:15.59terrapenno way
23:16.05terrapenPress 1 for Hmong
23:16.20firestrmrotfl
23:16.31terrapenPress 2 for Divehi
23:16.35terrapenPress 3 for Cherokee
23:16.49terrapenPress 4 for Icelandic
23:17.12terrapenPress 5 for Bantu
23:17.16*** part/#asterisk yogurt2ungue (~charlie@203-132-126-200.fibertel.com.ar)
23:17.25terrapenhave i missed anything
23:17.32*** part/#asterisk ckruetze (~ckruetze@62.214.134.0)
23:17.32ZX81~ping
23:17.33jbotpong
23:17.34terrapenPress 6 for Armenian
23:17.43*** join/#asterisk ZX81 (matt@222-153-51-36.jetstream.xtra.co.nz)
23:17.45enzo1237 for spanish .. !!
23:17.48firestrmlol, to speak to an operator please enter any 13 digit prime number using youe telephone keypad
23:18.05enzo123divided by 3.14
23:18.05*** join/#asterisk santiago (~santiago@63.245.86.97)
23:18.40devi0ushttp://www.123world.com/languages/
23:18.43devi0usjust do all of em
23:18.45devi0uswith english last
23:18.50*** join/#asterisk AgiNamu (~AgiNamu@200.12.43.74)
23:19.03enzo123or http://www.engrish.com/
23:19.05devi0usactually. that's a short list
23:19.06firestrmdont forget klingon and vulcan
23:19.07enzo123engrish
23:19.24devi0ushrm. wonder if there's a engrish voice for festival
23:19.28AgiNamulol
23:19.36devi0usthat would give credibility to an extension in hong kong
23:19.37enzo123shicken
23:19.37AgiNamu"IAX2 your data . net protocol need flat roof!"
23:19.42*** join/#asterisk BoRiS (~boris@S01060040ca1e5b54.wp.shawcable.net)
23:19.54fileBoRiS!!!!!!!!!!!!!!!!!!!!!!!!!!
23:19.57BoRiSfile!!!!!!!!!!!
23:20.03BoRiShey ya!!!!!!
23:20.06BoRiSis the conf down?
23:20.07filego to hell
23:20.12filenope...
23:20.21BoRiShmmm
23:20.26AgiNamuon my voip phone: Security and Notes:
23:20.26AgiNamu◆     Don’t use it in chemical plant、gas station or near the exploder place.
23:21.20firestrmim laughing so hard it hurts
23:22.24*** join/#asterisk KleinJonp (~chatzilla@dsl-213-023-223-206.arcor-ip.net)
23:22.30t3tThere has to be someone willing to share their insight about setting the default incoming callerID string for calls without it set...
23:22.34AgiNamui emailed someone for prices, and got this: delay of its news, If has left it is possible that it can send its data to me of I telephone to be able to communicate with its person and power to me to have a direct contact but. until soon.
23:23.10KleinJonpHi everyone, say, how do i make * accept all sip calls from specific IP, without user/passwd combination ?
23:23.31AgiNamuhost=theip
23:23.35AgiNamuand dont put a secret
23:23.49firestrmAgiNamu, translators gotta love it.. still rotfl
23:24.10AgiNamu:)
23:24.13modulus_i wonder what lang. they're translating from
23:24.13KleinJonpAgiNamu: should i still put a username ? or leave it out as well ?
23:24.20AgiNamuno, no username
23:24.34KleinJonpAgiNamu: thanks
23:25.07PoincareAnyone experience with hiring 'voices' to make IVR prompts?
23:25.35KleinJonpAgiNamu: how do i put multiple hosts ? just create many lines with host= ?
23:25.44firestrmsomone sent me this.. if programmers were klingon.... What is this talk of 'release'? Klingons do not make software 'releases'. Our software escapes, leaving a bloody trail of designers and quality assurance people in its wake!
23:25.48terrapeni can get you a good female voice if you want
23:26.09AgiNamuKlein, not sure.
23:26.17PoincareI would like to know what reasonable fees are...
23:26.26terrapena voice is nothing special
23:26.33terrapenfind someone on the street with a good voice
23:26.35terrapenor in your company
23:26.40Poincareterrapen: your voice speaks dutch?
23:26.44terrapenoh.
23:26.44terrapenno.
23:26.48Poincare:-D
23:26.51t3tbkw_: you still there?
23:26.52Godseyvoicemail hell at my house is about to be "you have 10 seconds to dial $RAND"
23:26.55*** join/#asterisk kludgebox (~bob@12.171.178.194)
23:27.01terrapenyou're telling me that you can't find a cute Dutch girl out on the street?
23:27.03terrapenwith a good voice?
23:27.12Godsey"You have strong Chi and passed the first test!  You have 5 seconds to dial $RAND"
23:27.16PoincareWell... Someone I asked wanted about 75 USD per sentence...
23:27.20terrapendress up in a nice suit and walk around and ask girls
23:27.23terrapenyou might even get laid!
23:27.26firestrmpoincare, i have a very good dutch voice avail..
23:27.38ta[i]ntedis there a term to describe the ratio between the actual number of available lines and the number of users of the system?
23:27.44firestrmPoincare, she also does danish
23:27.53GodseyPoincare: just prank call people and record the calls
23:27.59terrapenhahah
23:28.00Poincarefirestrm: dutch-holland or dutch-flemish?
23:28.00Godseytry getting them to say all the words you need
23:28.02terrapenget them to say them
23:28.04terrapenyeah
23:28.10niZonPoincare: use a celebrity sound board
23:28.28Poincarea sound board?
23:28.41Godseysales calls are easiest
23:28.42firestrmPoincare, im going to guess dutch-holland, its her second language, do i doubt it would be flemish
23:28.45terrapeni would do my voices with SBAITSO.EXE
23:28.52niZonPoincare: http://www.ebaumsworld.com/soundboards.shtml
23:28.53niZon:P
23:28.56terrapenSBSAY.EXE
23:29.19Godseydamn ebaumsworld is in our squidGuard block list :)
23:30.14Poincarefirestrm: from where is she?
23:30.15niZonhaha
23:30.24niZonGodsey: find a cgiproxy :P
23:30.25*** join/#asterisk philz (~philzama@borg.zamigo.net)
23:30.53firestrmPoincare, she is from denmark, but married dutch, lived in holland for many years before coming to canada.
23:30.54*** join/#asterisk bjohnson (~bjohnson@ip226-181.tor.istop.com)
23:31.37ZX81~seen wsuff
23:31.40jbotwsuff <~wsuff@69.161.191.145> was last seen on IRC in channel #asterisk, 11d 7h 22m 45s ago, saying: 'faithx: iaxcomm works in linux'.
23:31.44antiCan anyone recommend a good asterisk friendly voip provider with great (or unlimited) international rates for business customers?
23:32.04ZX81you won't get unlimited for business
23:32.11*** join/#asterisk KoLLi (~KoLLi@80.77.132.5)
23:32.13ZX81unless you're willing to pay a *lot*
23:32.20ZX81nuvox is good
23:32.22ZX81oops
23:32.24ZX81i mean
23:32.25ZX81nuphone
23:32.36terrapenhow do they know that you are using it for business?
23:32.36firestrmPoincare, the nice thing is that her daughter (my wife) has exactly the same voice (without the accent) so she can do a seamless english version too
23:33.01Poincareaaah... that's where you know her from :-D
23:33.05Godseymy wife used to do walmart radio commercials :)
23:33.20terrapenyou must be from South Carolina
23:33.21t3tZX81: hi, it's http://www.nufone.net/
23:33.40AgiNamuhey, here's a feature i might do: change Dial (well, IAX)so that it can also take a codec
23:33.49AgiNamuso you can specify the codec at call-time as well
23:34.01AgiNamuAND, you can also specify "current", and get a better chance of reducing transcoding
23:34.25AgiNamulike, suppose Client B calls in, and only supports G729. And they call out to ServerA, which supports both, but prefers ULAW.
23:34.44*** part/#asterisk MasterYoda (~mnicholso@dhcp-155.digium.com)
23:34.50AgiNamuthen you could switch the codec on the fly
23:34.52t3tAgiNamu: interesting, so instead of per-login-session, it's per-call.  Nice.
23:34.53philzAgiNamu, Nice
23:34.58AgiNamuis it a good idea?
23:35.34t3tWhat are the possible problems with it?
23:35.45AgiNamunot sure
23:35.51AgiNamuIt wouldnt change default functionaluty
23:35.56AgiNamuif you dont pass codec, then it'd ignore it
23:36.03AgiNamuand you could also pass codec "default" or "current"
23:36.07AgiNamuin addition to g729, ulaw, etc.
23:36.10t3tAddiotional code complexity, but that's never stopped anyone before :)
23:36.14AgiNamulol yea
23:36.22AgiNamuAny suggestions on the dial string?
23:36.29AgiNamuit is now like : IAX2/user:pass@server:port/extention
23:36.39AgiNamu(actually ,it can be extention and context and priority i think)
23:37.01*** mode/#asterisk [+r] by bkw_
23:37.30t3tYou'd probably need a unique delimiter before the codec so that order of the other optional arguments isn't affected
23:37.41AgiNamuyea
23:37.51AgiNamusomething like /-?
23:38.22t3tMaybe, but it's pretty close to the / before the extension
23:38.33t3tI'd look for something that's used elsewhere
23:38.42firestrmi wonder if i were to offer float plane fishing for help with asterisk if i would get any takers...
23:38.48Godseyvariable?
23:38.55Godseylike monitor file name?
23:38.55AgiNamuOr perhaps "Dial IAX2/bla/bla/bla|codec
23:39.09AgiNamugodse, what do you mean?
23:39.09*** part/#asterisk JimVanM (~JimVanM@HSE-MTL-ppp77210.qc.sympatico.ca)
23:39.38philzfirestrm, what do you need help with?
23:39.53Godseyexten => s,2,Monitor(wav,${FILENAME},m)
23:40.05Godseyset PreferedCodec()
23:40.06Godseybefore dial?
23:40.08AgiNamuAlso, what do you think of my freakishly hackish solution for packetloss by sending dup packets ?
23:40.14firestrmphilz, im trying to wrap my brain around the whole extensions.conf dial plan thing..
23:40.14Godseyand just make dial know how to use in internally?
23:40.18AgiNamuGodsey, yea, I was thinking of doing it like that
23:40.23AgiNamuyea, right
23:40.28AgiNamuthat might be best
23:40.37AgiNamuBut, you have to make the patches to IAX, sip, etc.
23:40.41AgiNamuSince THEY negotiate the codecs.
23:40.42AgiNamunot dial
23:40.45AgiNamu... right?
23:40.46philzfirestrm, yeah. im feelin ya. takes time
23:41.03AgiNamuWell, hey, we can bypass DIAL completely
23:41.09AgiNamuAnd just make IAX aware of a channel var
23:41.15AgiNamuWhat do you guys think of that?
23:41.20Godseyover my head :)
23:41.40AgiNamuOK, from usability
23:41.50AgiNamuDo you prefer having more complex dialstrings?
23:41.59philzfirestrm, have you done any of the tutorials yet?
23:42.01Godseyno
23:42.04AgiNamuor a channel variable?
23:42.06firestrmphilz, i have my iax.conf working ok,its connecting to FWD, but it doesnt know what to do with incoming calls and errors out
23:42.10AgiNamu(or something similar, perhaps a separate app)
23:42.12GodseyI like variables
23:42.59philzfirestrm, what does the error say?
23:43.18Godseythere is a pesky bug I'm working on
23:43.33GodseySIP/63.x-2b217d78 got tired of being parked
23:43.38Godseyand asterisk crashes
23:44.46firestrmphils, somthing like sent to invalid extension in context incoming but no invalid handler..
23:45.27firestrmphils, i obviously have something set up wrong, but don't know enough what to do about it..
23:45.41philzbtw, can anyone recomend a gui for asterisk? something that would be simple enough for public school type admins to use.
23:45.43bjohnsonAgiNamu: freat was looking for iax phone info.  i sent him the info you sent me
23:45.59antiin asterisk, when connecting to a sip provider and going through a nat box, is there any forwarding I have to do on the nat box or does having nat=yes in asterisk suffice?
23:46.08sektor195is there a howto some place I can read on howto setup call queues
23:46.09sektor195?
23:46.15*** join/#asterisk carlos-d-man (~carlos@201.128.216.12)
23:46.50ZX81~docs
23:46.51jbotDocumentation 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
23:47.17firestrmphilz, i was looking into this one http://www.ifrance.com/belikewater/code/actos.html
23:47.33firestrmphilz, it looks fairly usable..
23:48.02GodseyI have some asterisk core files :)
23:48.42carlos-d-manCan asterisk take advantage of a regular 52k v92 telephone fax modem?
23:49.01GodseyCore was generated by `/usr/sbin/asterisk -gfvvvv'.
23:49.01GodseyProgram terminated with signal 11, Segmentation fault.
23:49.01Godsey#0  0x0805ac4e in ?? ()
23:49.07AgiNamuthanks bjohnson
23:49.23Godseylooks like I need to compile w/ debugging symbols right?
23:49.43AgiNamuthat'd be cool: an error reporting utility, ala Windows
23:49.48AgiNamuthat helps SO much
23:50.57Godseyhumm
23:50.59GodseyELF 32-bit LSB executable, Intel 80386, version 1 (SYSV), for GNU/Linux 2.0.0, dynamically linked (uses shared libs), not stripped
23:51.03modulus_any kind of reporting would be cool ala Windows
23:51.08GodseyI just must not know how to use gdb :)
23:51.19file[laptop]bt
23:51.23GodseyI did backtrace
23:51.31AgiNamucarlos, if you mean use it as an FXO, probably not
23:51.50GodseyI don't understand what the offsets and in ?? () mean tho
23:52.01GodseyI was expecting more detail than ?? ()
23:52.05file[laptop]you pooched it
23:52.08AgiNamumeanst it cant find the symbols
23:52.20file[laptop]I just HAD to use that word
23:52.31AgiNamutoo many blueberries.
23:52.36Godseydo I need to do file /usr/sbin/asterisk too?
23:53.00Godseyoh there we go
23:53.24Godsey#0  0x0805ac4e in ast_hangup (chan=0x2b233db8) at channel.c:679
23:53.24Godsey679             if (clone->pvt->writetrans)
23:53.53Godseyso gdb
23:53.57Godseyfile /usr/sbin/asterisk
23:54.01Godseycore /core.12342
23:54.03Godseybt
23:54.04Godseyright?
23:54.12AgiNamuoh yea
23:54.23file[laptop]funky
23:54.23AgiNamuI thouight you loaded asterisk and the core from  the command line
23:54.34file[laptop]that crash is interesting
23:54.36GodseyI have not used gdb since 96
23:54.53file[laptop]almost like the pointer to the pvt isn't there...
23:56.26Godseyhttp://pastebin.ca/4117
23:56.44carlos-d-manX100Pis there a FXO that may plug into asterisk or something?
23:57.31file[laptop]Godsey: very odd...
23:57.48file[laptop]Godsey: can you replicate the problem? I may have a fix
23:57.53GodseyI think so
23:57.59Godseycustomer sits in a hold queue
23:58.05Godseyagent dials #700 to park
23:59.13file[laptop]very odd

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